For following the ALSA PCM core change, a new PCM copy ops is added
toe ASoC component framework: snd_soc_component_driver receives the
copy ops, and snd_soc_pcm_component_copy() helper is provided.
This also fixes a long-standing potential bug where the ASoC driver
covers only copy_user PCM callback and misses the copy from kernel
pointers (such as OSS PCM layer), too.
As of this patch, the old copy_user is still kept, but it'll be
dropped later after all drivers are converted.
Reviewed-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230815190136.8987-19-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add two more helpers for copying memory between iov_iter and iomem,
which will be used by the new PCM copy ops in a few drivers.
The existing helpers became wrappers of those now.
Note that copy_from/to_iter() returns the copied bytes, hence the
error condition is adjusted accordingly.
Link: https://lore.kernel.org/r/20230815190136.8987-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
iov_iter is a universal interface to copy the data chunk from/to
user-space and kernel in a unified manner. This API can fit for ALSA
PCM copy ops, too; we had to split to copy_user and copy_kernel in the
past, and those can be unified to a single ops with iov_iter.
This patch adds a new PCM copy ops that passes iov_iter for copying
both kernel and user-space in the same way. This patch touches only
the ALSA PCM core part, and the actual users will be replaced in the
following patches.
The expansion of iov_iter is done in the PCM core right before calling
each copy callback. It's a bit suboptimal, but I took this now as
it's the most straightforward replacement. The more conversion to
iov_iter in the caller side is a TODO for future.
As of now, the old copy_user and copy_kernel ops are still kept.
Once after all users are converted, we'll drop the old copy_user and
copy_kernel ops, too.
Link: https://lore.kernel.org/r/20230815190136.8987-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert the RT5682S to use GPIO descriptors and drop the
legacy GPIO headers.
We remove the global GPIO number from the platform data,
but it is still possible to create board files using GPIO
descriptor tables, if desired.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20230817-descriptors-asoc-rt-v2-5-02fa2ca3e5b0@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Convert the RT5682 to use GPIO descriptors and drop the
legacy GPIO headers.
We remove the global GPIO number from the platform data,
but it is still possible to create board files using GPIO
descriptor tables, if desired.
Make sure to make sure SDW devices can associate with
an LDO1 EN descriptor too, if they so desire by putting
the lookup into the common code.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20230817-descriptors-asoc-rt-v2-4-02fa2ca3e5b0@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Convert the RT5668 to use GPIO descriptors and drop the
legacy GPIO headers.
We remove the global GPIO number from the platform data,
but it is still possible to create board files using GPIO
descriptor tables, if desired.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20230817-descriptors-asoc-rt-v2-3-02fa2ca3e5b0@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The RT5665 driver has some stub support for GPIO descriptors
going back to the initial driver commit, where there are
two GPIO descriptors for the LDO and headphone detection
defined in the device state. Well, let's make use of the
descriptor properly.
We remove the global GPIO number from the platform data,
but it is still possible to create board files using GPIO
descriptor tables, if desired.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Link: https://lore.kernel.org/r/20230817-descriptors-asoc-rt-v2-2-02fa2ca3e5b0@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Now all users of snd_device_intialize() are gone, let's drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Tested-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20230816160252.23396-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Embedding the struct device to snd_compr object may result in UAF when
the delayed kobj release is used. Like other devices, let's detach
the struct device from the snd_compr by allocating dynamically via
snd_device_alloc().
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Tested-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20230816160252.23396-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch detaches the struct device from the snd_rawmidi object by
allocating via snd_device_alloc(), just like done for other devices.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Tested-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20230816160252.23396-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like control and PCM devices, it's better to avoid the embedded struct
device for hwdep (although it's more or less well working), too.
Change it to allocate via snd_device_alloc(), and free the memory at
the common snd_hwdep_free().
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Tested-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20230816160252.23396-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we use the embedded struct device for each PCM substreams in
struct snd_pcm. This may result in UAF when the delayed kobj release
is used; each corresponding struct device is still accessed at the
(delayed) device release, while the snd_pcm object may be already
gone.
As a workaround, detach the struct device from the snd_pcm object by
allocating via the new snd_device_alloc() helper.
A caveat is that we store the PCM substream pointer to drvdata since
the device resume and others require the access to it.
This patch is based on the fix Curtis posted initially. In this
patch, the changes are split and use the new helper function instead.
Link: https://lore.kernel.org/r/20230801171928.1460120-1-cujomalainey@chromium.org
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Tested-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20230816160252.23396-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Embedding the ctl_dev in the snd_card object may result in UAF when
the delayed kobj release is used; at the delayed kobj release, it
still accesses the struct device itself while the card memory (that
embeds the struct device) may be already gone.
As a workaround, detach the struct device from the card object by
allocating via the new snd_device_alloc() helper. The rest are just
replacing ctl_dev access to the pointer.
This is based on the fix Curtis posted initially. In this patch, the
changes are split and use the new helper function instead.
Link: https://lore.kernel.org/r/20230801171928.1460120-1-cujomalainey@chromium.org
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Tested-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20230816160252.23396-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new helper, snd_device_alloc(), for allocating a struct
device that is bound with the sound class. It's a replacement of
snd_device_initialize().
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Tested-by: Curtis Malainey <cujomalainey@chromium.org>
Link: https://lore.kernel.org/r/20230816160252.23396-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The existing soc-dapm code may add a prefix to control names, which in
some cases is useful but in others leads to long and confusing kcontrol
names such as "gain 2.1 Main Playback Volume".
This patch suggests an added flag to prevent the widget name prefix
from being added. That flag will be set in the topology file on a
per-widget basis.
The flag no_wname_in_kcontrol_name is added to struct snd_soc_dapm_widget,
and the logic in dapm_create_or_share_kcontrol() is changed to not to
add widget name if the flag is set.
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Jyri Sarha <jyri.sarha@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230814232325.86397-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_driver has .ops for call back functions (A), but it also
has other call back functions (B). It is duplicated and confusable.
struct snd_soc_dai_driver {
...
^ int (*probe)(...);
| int (*remove)(...);
(B) int (*compress_new)(...);
| int (*pcm_new)(...);
v ...
(A) const struct snd_soc_dai_ops *ops;
...
}
This patch merges (B) into (A).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v8dpb0w6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
This patchset first fixes a number of errors made in the hda-mlink
support, then adds Lunar Lake definitions. The main contribution is
the hda-dai changes where the HDaudio DMA is now used for SSP, DMIC
and SoundWire. In previous hardware the GPDMA (aka DesignWare) was
used and controlled by the audio firmware. The volume of code is
minimized with the abstraction added in previous kernel cycles.
Due to cross-dependencies between ASoC and SoundWire trees, the full
support for jack detection will be deferred to the next kernel
cycle. There's not much point to ask for a sync of the two trees to
support one patch for each tree - we are at -rc5 already.
We need to retrieve the current value to deal with the HDAudio
WAKEEN/WAKESTS setup.
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>
Link: https://lore.kernel.org/r/20230807210959.506849-18-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The switch component was never completed and sat half empty for over 3
years. It was recently deleted. For modern components this would
require not change in the kernel but since this was a legacy allocation
from the enum days of IPC3 we should mark the respective enum as
deprecated.
The splitter component was never even got a source file in the firmware.
Therefore also delete it since this is not needed.
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230731213242.434594-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Move existing function in common library to make sure the code can be
reused by other SoC vendors.
No functionality change outside of the move and added prefix.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20230731213242.434594-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Here's an initial batch of updates for ASoC for this release cycle.
We've got a bunch of new drivers in here, a bit of core work from
Morimoto-san and quite a lot of janitorial work. There's several
updates that pull in changes from other subsystems in order to build
on them:
- An adaptor to allow use of IIO DACs and ADCs in ASoC which pulls in
some IIO changes.
- Create a library function for intlog10() and use it in the NAU8825
driver.
- Include the ASoC tests, including the topology tests, in the default
KUnit full test coverage. This also involves enabling UML builds of
ALSA since that's the default KUnit test environment which pulls in
the addition of some stubs to the driver.
- More factoring out from Morimoto-san.
- Convert a lot of drivers to use the more modern maple tree register
cache.
- Support for AMD machines with MAX98388 and NAU8821, Cirrus Logic
CS35L36, Intel AVS machines with ES8336 and RT5663 and NXP i.MX93.
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Merge tag 'asoc-v6.6-early' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v6.6
Here's an initial batch of updates for ASoC for this release cycle.
We've got a bunch of new drivers in here, a bit of core work from
Morimoto-san and quite a lot of janitorial work. There's several
updates that pull in changes from other subsystems in order to build
on them:
- An adaptor to allow use of IIO DACs and ADCs in ASoC which pulls in
some IIO changes.
- Create a library function for intlog10() and use it in the NAU8825
driver.
- Include the ASoC tests, including the topology tests, in the default
KUnit full test coverage. This also involves enabling UML builds of
ALSA since that's the default KUnit test environment which pulls in
the addition of some stubs to the driver.
- More factoring out from Morimoto-san.
- Convert a lot of drivers to use the more modern maple tree register
cache.
- Support for AMD machines with MAX98388 and NAU8821, Cirrus Logic
CS35L36, Intel AVS machines with ES8336 and RT5663 and NXP i.MX93.
To enable the speaker output in external boost mode, 2 registers must
be set, one after another. The longer the time between the writes of
the two registers, the more likely, and more loudly a pop may occur.
To minimize this, an mbox command can be used to allow the firmware
to perform this action, minimizing any delay between write, thus
minimizing any pop or click as a result. The old method will remain
when running without firmware.
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230721151816.2080453-2-sbinding@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move issuing of a CS35L56_MBOX_CMD_SHUTDOWN command and then waiting for
the DSP to reach CS35L56_HALO_STATE_SHUTDOWN in the register appropriate
for the hardware revision into a common function.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230721132120.5523-10-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the waits for CS35L56_CONTROL_PORT_READY_US into a common
function, and also allow a wider range of allowed wait times.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230721132120.5523-9-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Part of the initialization code in cs35l56_init() can be re-used
by the HDA driver so move it into a new function in the shared
library.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230721132120.5523-8-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the code that initialized the struct cs_dsp members
into the shared library so that the HDA driver can use it.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230721132120.5523-7-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The majority of runtime_suspend and runtime_resume handling
doesn't have anything specific to the ASoC driver, so can be
shared by the HDA driver. Move this code into the shared
library.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230721132120.5523-6-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the cs35l56 utility functions into the shared file so they are
available for use in HDA.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230721132120.5523-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The ASoC and HDA drivers have structures that contain some of the same
information - instead of maintaining two locations for this data the
drivers should share a common data structure as this will enable common
utility functions to be created.
The first step is to move the location of these members in the ASoC
driver.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/20230721132120.5523-2-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a new helper to add multiple vmaster followers in a shot. The
same function was open-coded in various places, and this helper
replaces them.
Link: https://lore.kernel.org/r/20230721071643.3631-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A commonly seen pattern is to run snd_ctl_find_id() for a mixer
control element with a given string. Let's provide a standard helper
for achieving that for simplifying the code.
Link: https://lore.kernel.org/r/20230720082108.31346-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few ALSA control API helpers like snd_ctl_rename(), snd_ctl_remove()
and snd_ctl_find_*() suppose the callers taking card->controls_rwsem.
But it's error-prone and fragile. This patch set tries to change
those API functions to take the card->controls>rwsem internally by
themselves, so that the drivers don't need to take care of lockings.
After applying this patch set, only a couple of places still touch
card->controls_rwsem (which are OK-ish as they need for traversing the
control linked list).
Link: https://lore.kernel.org/r/20230718141304.1032-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'tags/ctl-lock-fixes-6.6' into for-next
ALSA: Make control API taking controls_rwsem consistently
A few ALSA control API helpers like snd_ctl_rename(), snd_ctl_remove()
and snd_ctl_find_*() suppose the callers taking card->controls_rwsem.
But it's error-prone and fragile. This patch set tries to change
those API functions to take the card->controls>rwsem internally by
themselves, so that the drivers don't need to take care of lockings.
After applying this patch set, only a couple of places still touch
card->controls_rwsem (which are OK-ish as they need for traversing the
control linked list).
Link: https://lore.kernel.org/r/20230718141304.1032-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For reducing the unnecessary use of controls_rwsem in the drivers,
this patch adds a new variant for snd_ctl_find_*() helpers:
snd_ctl_find_id_locked() and snd_ctl_find_numid_locked() look for a
kctl element inside the card->controls_rwsem -- that is, doing the
very same as what snd_ctl_find_id() and snd_ctl_find_numid() did until
now. snd_ctl_find_id() and snd_ctl_find_numid() remain same,
i.e. still unlocked version, but they will be switched to locked
version once after all callers are replaced.
The patch also replaces the calls of snd_ctl_find_id() and
snd_ctl_find_numid() in a few places; all of those are places where we
know that the functions are called properly with controls_rwsem held.
All others are without rwsem (although they should have been).
After this patch, we'll turn on the locking in snd_ctl_find_id() and
snd_ctl_find_numid() to be more race-free.
Link: https://lore.kernel.org/r/20230718141304.1032-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The soft (firmware) registers for volume/mute/posture are not reset by
a chip soft-reset, so use a regmap patch to set them to defaults.
cs35l56_reread_firmware_registers() has been removed. Its intent was to
use whatever the firmware set as a default. But the driver now patches the
defaults to the registers.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230718144625.39634-1-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of using local macro to match PCI device, use global one. As
Apollolake is Broxton-P successor that made it to the market, be precise
and use APL shortcut. IS_CFL() macro is dropped as it is unused.
Acked-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230717114511.484999-9-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some HDA controllers require additional handling, so there are macros to
match them, however those are spread across multiple files. Add them all
in one place, so they can be reused.
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230717114511.484999-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Given that the filter is already set to neutral for PCM voices, the
only observable effect is that the Z1/Z2/FXBUS registers don't have a
stray bit set for negative numbers anymore. The bit is below the ones
significant for output, but it would mess with 32-bit sample
recombination, which we intend to add.
kX-project does that, but I had to figure out myself why.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230715160802.326872-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
85;95;0c
This uses IRQs to track spontaneous changes to the word clock source
register.
FWIW, that this can happen in the first place is the reason why it is
futile to lock the clock source mixer setting while the device is open -
we can't consistently control the rate anyway. Though arguably, we
should reset any open streams when that happens, as they become
corrupted anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230715160738.326832-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If CPU/Codec driver keeps its DAI node, we can directly identify actual
DAI by using snd_soc_get_dai_via_args().
This means we can use multi Component.
This patch enables multi Component support on Audio Graph Card/Card2.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a5w4o949.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To use multi Component support, we need to check dai_args whether
Card could get DAI from args (CPU/Codec needs set dai_args on DAI driver).
If it could, we need to allocate dai_args for dlc.
This patch adds snd_soc_copy_dai_args() for it.
This is helper function for multi Component support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87bkgko94e.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_is_matching_component() checks "of_node" or "dai_args".
Thus coping "of_node" only is not enough to use CPU as Platform.
This patch adds snd_soc_dlc_use_cpu_as_platform() and help it.
This is helper function for multi Component support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87cz10o94k.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To enable multi Component, Card driver need to get DAI via dai_args
to identify it. This patch adds snd_soc_get_dai_via_args() for it.
This is helper function for multi Component support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87edlgo94p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC Card is using dlc (snd_soc_dai_link_component) to find
target DAI / Component to be used.
Current dlc has below 3 items to identify DAI / Component
(a) name for Component
(b) of_node for Component
(c) dai_name for DAI
(a) or (b) is used to identify target Component, and (c) is used
to identify DAI.
One of the biggest issue on it today is dlc needs "name matching"
for "dai_name" (c).
It was not a big deal when we were using platform_device, because we
could specify nessesary "dai_name" via its platform_data.
But we need to find DAI name pointer from whole registered datas and/or
each related driver somehow in case of DT, because we can't specify it.
Therefore, Card driver parses DT and assumes the DAI, and find its name
pointer. How to assume is based on each Component and/or Card.
Next biggest issue is Component node (a)/(b).
Basically, Component is registered when CPU/Codec driver was
probed() (X). Here, 1 Component is possible to have some DAIs.
int xxx_probe(struct platform_device *pdev)
{
...
(X) ret = devm_snd_soc_register_component(pdev->dev,
&component_driver,
&dai_driver, dai_driver_num);
...
}
The image of each data will be like below.
One note here is "driver" is included for later explanation.
+-driver------+
|+-component-+|
|| dai0||
|| dai1||
|| ...||
|+-----------+|
+-------------+
The point here is 1 driver has 1 Component, because basically driver
calles snd_soc_register_component() (= X) once.
Here is the very basic CPU/Codec connection image.
HW image SW image
+-- Board ------------+ +-card--------------------------+
|+-----+ +------+| |+-driver------+ +-driver------+|
|| CPU | <--> |CodecA|| ||+-component-+| |+-component-+||
|+-----+ +------+| ||| dai|<=>|dai |||
+---------------------+ ||+-----------+| |+-----------+||
|+-------------+ +-------------+|
+-------------------------------+
It will be very complex if it has multi DAIs.
Here is intuitive easy to understandable HW / SW example.
HW image SW image
+-- Board ---------------+ +-card--------------------------+
|+--------+ +------+| |+-driver------+ +-driver------+|
|| CPU ch0| <--> |CodecA|| ||+-component-+| |+-component-+||
|| | +------+| ||| ch0 dai|<=>|dai |||
|| | +------+| ||| || |+-----------+||
|| ch1| <--> |CodecB|| ||| || +-------------+|
|+--------+ +------+| ||| || +-driver------+|
+------------------------+ ||| || |+-component-+||
||| ch1 dai|<=>|dai |||
||+-----------+| |+-----------+||
|+-------------+ +-------------+|
+-------------------------------+
It will be handled as multi interface as "one Card".
card0,0: CPU-ch0 - CodecA
card0,1: CPU-ch1 - CodecB
^
But, here is the HW image example which will be more complex
+-- Basic Board ---------+
|+--------+ +------+|
|| CPU ch0| <--> |CodecA||
|| ch1| <-+ +------+|
|+--------+ | |
+-------------|----------+
+-- expansion board -----+
| | +------+|
| +->|CodecB||
| +------+|
+------------------------+
We intuitively think we want to handle these as "2 Sound Cards".
card0,0: CPU-ch0 - CodecA
card1,0: CPU-ch1 - CodecB
^
But below image which we can register today doesn't allow it,
because the same Component will be connected to both Card0/1,
but it will be rejected by (Z).
+-driver------+
|+-component-+|
+-card0-------------------------+
||| || +-driver------+|
||| || |+-component-+||
||| ch0 dai|<=>|dai |||
||| || |+-----------+||
||| || +-------------+|
+-------------------------------+
|| ||
+-card1-------------------------+
||| || +-driver------+|
||| || |+-component-+||
||| ch1 dai|<=>|dai |||
||| || |+-----------+||
||| || +-------------+|
+-------------------------------+
|+-----------+|
+-------------+
static int soc_probe_component()
{
...
if (component->card) {
(Z) if (component->card != card) {
dev_err(component->dev, ...);
return -ENODEV;
}
return 0;
}
...
}
So, how about to call snd_soc_register_component() (= X) multiple times
on probe() to avoid buplicated component->card limitation, to be like
below ?
+-driver------+
+-card0-------------------------+
|| | +-driver------+|
||+-component-+| |+-component-+||
||| ch0 dai|<=>|dai |||
||+-----------+| |+-----------+||
|| | +-------------+|
+-------------------------------+
| |
+-card1-------------------------+
|| | +-driver------+|
||+-component-+| |+-component-+||
||| ch1 dai|<=>|dai |||
||+-----------+| |+-----------+||
|| | +-------------+|
+-------------------------------+
+-------------+
Yes, looks good. But unfortunately it doesn't help us for now.
Let's see soc_component_to_node() and snd_soc_is_matching_component()
static struct device_node
*soc_component_to_node(struct snd_soc_component *component)
{
...
(A) of_node = component->dev->of_node;
...
}
static int snd_soc_is_matching_component(...)
{
...
(B) if (dlc->of_node && component_of_node != dlc->of_node)
...
}
dlc checkes "of_node" to identify target component (B),
but this "of_node" came from component->dev (A) which is added
by snd_soc_register_component() (X) on probe().
This means we can have different "component->card", but have same
"component->dev" in this case.
Even though we calls snd_soc_register_component() (= X) multiple times,
all Components have same driver's dev, thus it is impossible to
identified the Component.
And if it was impossible to identify Component, it is impossible to
identify DAI on current implementation.
So, how to handle above complex HW image today is 2 patterns.
One is handles it as "1 big sound card".
The SW image is like below.
SW image
+-card--------------------------+
|+-driver------+ +-driver------+|
||+-component-+| |+-component-+||
||| ch0 dai|<=>|dai |||
||| || |+-----------+||
||| || +-------------+|
||| || +-driver------+|
||| || |+-component-+||
||| ch1 dai|<->|dai |||
||+-----------+| |+-----------+||
|+-------------+ +-------------+|
+-------------------------------+
But the problem is not intuitive.
We want to handle it as "2 Cards".
2nd pattern is like below.
SW image
+-card0-------------------------+
|+-driver------+ +-driver------+|
||+-component-+| |+-component-+||
||| ch0 dai|<=>|dai |||
||+-----------+| |+-----------+||
|+-------------+ +-------------+|
+-------------------------------+
+-card1-------------------------+
|+-driver------+ +-driver------+|
||+-component-+| |+-component-+||
||| ch1 dai|<=>|dai |||
||+-----------+| |+-----------+||
|+-------------+ +-------------+|
+-------------------------------+
It handles as "2 Cards", but CPU part needs to be probed as 2 drivers.
It is also not intuitive.
To solve this issue, we need to have multi Component support.
In current implementation, we need to identify Component first
to identify DAI, and it is using name matching to identify DAI.
But how about to be enable to directly identify DAI by unique way
instead of name matching ? In such case, we can directly identify DAI,
then it can identify Component from DAI.
For example Simple-Card / Audio-Graph-Card case, it is specifying DAI
via its node.
Simple-Card
sound-dai = <&cpu-sound>;
Audio-Graph-Card
dais = <&cpu-sound>;
If each CPU/Codec driver keeps this property when probing,
we can identify DAI directly from Card.
Being able to identify DAI directly means being able to identify its
Component as well even though Component has same dev (= B).
This patch adds new "dai_node" for it.
To keeping compatibility, it checks "dai_node" first if it has,
otherwise, use existing method (name matching).
Link: https://lore.kernel.org/r/87fskz5yrr.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fs5wo94v.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is specifying and checking DAI name.
But where it came from and how to check was ambiguous.
This patch adds snd_soc_dai_name_get() / snd_soc_dlc_dai_is_match()
and makes it clear.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h6qco952.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SND_SOC_DAPM_* helpers family are used to build widgets array in a
static way.
Convert them to return a compound literal in order to use them in both
static and dynamic way.
With this conversion, the different SND_SOC_DAPM_* parameters can be
computed by the code and the widget can be built based on this parameter
computation.
static int create_widget(char *input_name)
{
struct snd_soc_dapm_widget widget;
char name*;
...
name = input_name;
if (!name)
name = "default";
widget = SND_SOC_DAPM_INPUT(name);
...
}
Signed-off-by: Herve Codina <herve.codina@bootlin.com>
Suggested-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Christophe Leroy <christophe.leroy@csgroup.eu>
Link: https://lore.kernel.org/r/20230623085830.749991-12-herve.codina@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Yet more preliminary work for the upcoming USB gadget support.
Now export the helpers to convert between legacy MIDI1 and UMP data
for handling the MIDI 1.0 USB interface. The header file is moved to
include/sound.
The API functions are slightly changed, so that they can be used
without the direct access to snd_ump object. The allocation is done
in ump.c itself as it's a simple kcalloc().
Link: https://lore.kernel.org/r/20230623075530.10976-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a small patch set to change the UMP core for the upcoming
gadget driver support. Basically exporting a couple of helper
functions and adding a flag to suppress the internal UMP handling.
No functional changes by those alone.
Link: https://lore.kernel.org/r/20230621110241.4751-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is another preliminary patch for USB MIDI 2.0 gadget driver.
Export the currently local snd_ump_receive_ump_val(). It can be used
by the gadget driver for processing the UMP data.
Link: https://lore.kernel.org/r/20230621110241.4751-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is another preliminary patch for USB MIDI 2.0 gadget driver.
Add a new flag, no_process_stream, to snd_ump for suppressing the UMP
Stream message handling in UMP core.
Link: https://lore.kernel.org/r/20230621110241.4751-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for MIDI 2.0 USB gadget driver.
Export a new helper to allow changing the current MIDI protocol from
the outside.
Link: https://lore.kernel.org/r/20230621110241.4751-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the driver core allows for struct class to be in read-only
memory, making all 'class' structures to be declared at build time
placing them into read-only memory, instead of having to be dynamically
allocated at load time.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Ivan Orlov <ivan.orlov0322@gmail.com>
Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Cc: Geoff Levand <geoff@infradead.org>
Cc: Thierry Reding <treding@nvidia.com>
Cc: "Uwe Kleine-König" <u.kleine-koenig@pengutronix.de>
Cc: alsa-devel@alsa-project.org
Suggested-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Ivan Orlov <ivan.orlov0322@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Link: https://lore.kernel.org/r/20230620175633.641141-2-gregkh@linuxfoundation.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current snd_soc_of_get_dai_name() doesn't accept index
for #sound-dai-cells. It is not useful for user.
This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87pm5qdgng.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc-core.c has snd_soc_{of_}get_dai_name() to get DAI name
for dlc (snd_soc_dai_link_component). It gets .dai_name, but we need
.of_node too. Therefor user need to arrange.
It will be more useful if it gets both .dai_name and .of_node.
This patch adds snd_soc_{of_}get_dlc() for it, and existing functions
uses it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r0q6dgnm.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
simple_dai_props has cpus/codecs/platforms. These pointer were used
for dai_link before, but are allocated today since
commit 050c7950fd ("ASoC: simple-card-utils: alloc dai_link
information for CPU/Codec/Platform").
We don't need to keep it anymore. This patch removes these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87bkhhxpc6.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current Audio Graph Card/Card2 implements asoc_simple_parse_dai()
on each driver, but these are same function.
This patch share it as asoc_graph_parse_dai().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o7lihpvy.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC is assuming that trigger starting order is
Link -> Component -> DAI as default, and its reverse order for stopping.
But some Driver / Card want to reorder it for some reasons.
We have such flags, but is unbalance like below.
struct snd_soc_component_driver :: start_dma_last
struct snd_soc_dai_link :: stop_dma_first
We want to have more flexible, and more generic method.
This patch adds new snd_soc_trigger_order for start/stop at
component / DAI-link.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87r0qmfnzx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, ASoC supports dailinks with the following mappings:
1 cpu DAI to N codec DAIs
N cpu DAIs to N codec DAIs
But the mapping between N cpu DAIs and M codec DAIs is not supported.
The reason is that we didn't have a mechanism to map cpu and codec DAIs
This patch suggests a new snd_soc_dai_link_codec_ch_map struct in
struct snd_soc_dai_link{} which provides codec DAI to cpu DAI mapping
information used to implement N cpu DAIs to M codec DAIs
support.
When a dailink contains two or more cpu DAIs, we should set channel
number of cpus based on its channel mask. The new struct also provides
channel mask information for each codec and we can construct the cpu
channel mask by combining all codec channel masks which map to the cpu.
The N:M mapping is however restricted to the N <= M case due to physical
restrictions on a time-multiplexed bus such as I2S/TDM, AC97, SoundWire
and HDaudio.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20230607031242.1032060-2-yung-chuan.liao@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
As the updated MIDI 2.0 spec has been published freshly, this is a
catch up to add the support for new specs, especially UMP v1.1
features, on Linux kernel.
The new UMP v1.1 introduced the concept of Function Blocks (FB), which
is a kind of superset of USB MIDI 2.0 Group Terminal Blocks (GTB).
The patch set adds the support for FB as the primary information
source while keeping the parse of GTB as fallback. Also UMP v1.1
supports the groupless messages, the protocol switch, static FBs, and
other new fundamental features, and those are supported as well.
Link: https://www.midi.org/midi-articles/details-about-midi-2-0-midi-ci-profiles-and-property-exchange
Link: https://lore.kernel.org/r/20230612081054.17200-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UMP v1.1 supports the protocol switch via a UMP Stream message. When
it's received, we need to take care of the midi_version field in the
corresponding sequencer client, too.
This patch introduces a new ops to notify the protocol change to
snd_seq_ump_ops for handling it.
Link: https://lore.kernel.org/r/20230612081054.17200-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements the handling of the dynamic update of FB info.
When the FB info update is received after the initial parsing, it
means the dynamic FB info update. We compare the result, and if the
actual update is detected, it's notified via a new ops,
notify_fb_change, to the sequencer client, and the corresponding
sequencer ports are updated accordingly.
Link: https://lore.kernel.org/r/20230612081054.17200-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UMP Utility and Stream messages are "groupless", i.e. an incoming
groupless packet should be sent only to the UMP EP port, and the event
with the groupless message is sent to UMP EP as is without the group
translation per port.
Also, the former reserved bit 0 for the client group filter is now
used for groupless events. When the bit 0 is set, the groupless
events are filtered out and skipped.
Link: https://lore.kernel.org/r/20230612081054.17200-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the basic support for UMP Endpoint and UMP Function
Block parsing, which are extended in the new UMP v1.1 spec.
The patch provides a new helper function to perform the query of the
UMP Endpoint information and builds up the UMP blocks based on UMP
Function Block information. For the communication over the UMP
Endpoint, it opens the rawmidi device once internally, inquiries the
UMP Endpoint and Function Block info by sending new UMP Stream
messages, and waits for the response for each query.
The new UMP spec allows to update the FB info and change its
associated groups or its activeness on the fly, too. For catching it,
the UMP core keeps watching the incoming UMP messages, and
snd_ump_receive() handles the incoming UMP Stream messages to refresh
the FB info.
Link: https://lore.kernel.org/r/20230612081054.17200-3-tiwai@suse.de
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>
We are using get_stream_cpu() to get CPU stream which cares
Codec2Codec. But it is static function for now, and we want to use it
from other files. This patch makes it global function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fs7cj9mf.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
- 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>
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.
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>
The #endif is placed obviously at a wrong position, which caused a
build error on the big endian machine.
Fixes: 0b5288f5fe ("ALSA: ump: Add legacy raw MIDI support")
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Link: https://lore.kernel.org/r/20230524135448.3ecad334@canb.auug.org.au
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support of selecting insertion detection polarity
- Default polarity (Low)
- Inverted polarity (High)
Correct the keywords of parsing `dlg,jack-det-rate`
bases on the new DT binding.
Signed-off-by: David Rau <David.Rau.opensource@dm.renesas.com>
Link: https://lore.kernel.org/r/20230523161821.4260-4-David.Rau.opensource@dm.renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is a (largish) patch set for adding the support of MIDI 2.0
functionality, mainly targeted for USB devices. MIDI 2.0 is a
complete overhaul of the 40-years old MIDI 1.0. Unlike MIDI 1.0 byte
stream, MIDI 2.0 uses packets in 32bit words for Universal MIDI Packet
(UMP) protocol. It supports both MIDI 1.0 commands for compatibility
and the extended MIDI 2.0 commands for higher resolutions and more
functions.
For supporting the UMP, the patch set extends the existing ALSA
rawmidi and sequencer interfaces, and adds the USB MIDI 2.0 support to
the standard USB-audio driver.
The rawmidi for UMP has a different device name (/dev/snd/umpC*D*) and
it reads/writes UMP packet data in 32bit CPU-native endianness. For
the old MIDI 1.0 applications, the legacy rawmidi interface is
provided, too.
As default, USB-audio driver will take the alternate setting for MIDI
2.0 interface, and the compatibility with MIDI 1.0 is provided via the
rawmidi common layer. However, user may let the driver falling back
to the old MIDI 1.0 interface by a module option, too.
A UMP-capable rawmidi device can create the corresponding ALSA
sequencer client(s) to support the UMP Endpoint and UMP Group
connections. As a nature of ALSA sequencer, arbitrary connections
between clients/ports are allowed, and the ALSA sequencer core
performs the automatic conversions for the connections between a new
UMP sequencer client and a legacy MIDI 1.0 sequencer client. It
allows the existing application to use MIDI 2.0 devices without
changes.
The MIDI-CI, which is another major extension in MIDI 2.0, isn't
covered by this patch set. It would be implemented rather in
user-space.
Roughly speaking, the first half of this patch set is for extending
the rawmidi and USB-audio, and the second half is for extending the
ALSA sequencer interface.
The patch set is based on 6.4-rc2 kernel, but all patches can be
cleanly applicable on 6.2 and 6.3 kernels, too (while 6.1 and older
kernels would need minor adjustment for uapi header changes).
The updates for alsa-lib and alsa-utils will follow shortly later.
The author thanks members of MIDI Association OS/API Working Group,
especially Andrew Mee, for great helps for the initial design and
debugging / testing the drivers.
Link: https://lore.kernel.org/r/20230523075358.9672-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch introduces a new ALSA sequencer client for the kernel UMP
object, snd-seq-ump-client. It's a UMP version of snd-seq-midi
driver, while this driver creates a sequencer client per UMP endpoint
which contains (fixed) 16 ports.
The UMP rawmidi device is opened in APPEND mode for output, so that
multiple sequencer clients can share the same UMP endpoint, as well as
the legacy UMP rawmidi devices that are opened in APPEND mode, too.
For input, on the other hand, the incoming data is processed on the
fly in the dedicated hook, hence it doesn't open a rawmidi device.
The UMP packet group is updated upon delivery depending on the target
sequencer port (which corresponds to the actual UMP group).
Each sequencer port sets a new port type bit,
SNDRV_SEQ_PORT_TYPE_MIDI_UMP, in addition to the other standard
types for MIDI.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-33-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Starting from this commit, we add the basic support of UMP (Universal
MIDI Packet) events on ALSA sequencer infrastructure. The biggest
change here is that, for transferring UMP packets that are up to 128
bits, we extend the data payload of ALSA sequencer event record when
the client is declared to support for the new UMP events.
A new event flag bit, SNDRV_SEQ_EVENT_UMP, is defined and it shall be
set for the UMP packet events that have the larger payload of 128
bits, defined as struct snd_seq_ump_event.
For controlling the UMP feature enablement in kernel, a new Kconfig,
CONFIG_SND_SEQ_UMP is introduced. The extended event for UMP is
available only when this Kconfig item is set. Similarly, the size of
the internal snd_seq_event_cell also increases (in 4 bytes) when the
Kconfig item is set. (But the size increase is effective only for
32bit architectures; 64bit archs already have padding there.)
Overall, when CONFIG_SND_SEQ_UMP isn't set, there is no change in the
event and cell, keeping the old sizes.
For applications that want to access the UMP packets, first of all, a
sequencer client has to declare the user-protocol to match with the
latest one via the new SNDRV_SEQ_IOCTL_USER_PVERSION; otherwise it's
treated as if a legacy client without UMP support.
Then the client can switch to the new UMP mode (MIDI 1.0 or MIDI 2.0)
with a new field, midi_version, in snd_seq_client_info. When switched
to UMP mode (midi_version = 1 or 2), the client can write the UMP
events with SNDRV_SEQ_EVENT_UMP flag. For reads, the alignment size
is changed from snd_seq_event (28 bytes) to snd_seq_ump_event (32
bytes). When a UMP sequencer event is delivered to a legacy sequencer
client, it's ignored or handled as an error.
Conceptually, ALSA sequencer client and port correspond to the UMP
Endpoint and Group, respectively; each client may have multiple ports
and each port has the fixed number (16) of channels, total up to 256
channels.
As of this commit, ALSA sequencer core just sends and receives the UMP
events as-is from/to clients. The automatic conversions between the
legacy events and the new UMP events will be implemented in a later
patch.
Along with this commit, bump the sequencer protocol version to 1.0.3.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-26-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a new variant of snd_seq_expand_var_event() for expanding the
data starting from the given byte offset. It'll be used by the new
UMP sequencer code later.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-19-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch extends the UMP core code to support the legacy MIDI 1.0
rawmidi devices. When the new kconfig CONFIG_SND_UMP_LEGACY_RAWMIDI
is set, the UMP core allows to attach an additional rawmidi device for
each UMP Endpoint. The rawmidi device contains 16 substreams where
each substream corresponds to a UMP Group belonging to the EP. The
device reads/writes the legacy MIDI 1.0 byte streams and translates
from/to UMP packets.
The legacy rawmidi devices are exclusive with the UMP rawmidi devices,
hence both of them can't be opened at the same time unless the UMP
rawmidi is opened in APPEND mode.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-15-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a code refactoring for abstracting the rawmidi access to the
UMP's own helpers. It's a preliminary work for the later code
refactoring of the UMP layer.
Until now, we access to the rawmidi substream directly from the
driver via rawmidi access helpers, but after this change, the driver
is supposed to access via the newly introduced snd_ump_ops and
receive/transmit via snd_ump_receive() and snd_ump_transmit() helpers.
As of this commit, those are merely wrappers for the rawmidi
substream, and no much function change is seen here.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-14-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UMP devices may have more interesting information than the traditional
rawmidi. Extend the rawmidi_global_ops to allow the optional proc
info output and show some more bits in the proc file for UMP.
Note that the "Groups" field shows the first and the last UMP Groups,
and both numbers are 1-based (i.e. the first group is 1).
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the support helpers for UMP (Universal MIDI Packet) in
ALSA core.
The basic design is that a rawmidi instance is assigned to each UMP
Endpoint. A UMP Endpoint provides a UMP stream, typically
bidirectional (but can be also uni-directional, too), which may hold
up to 16 UMP Groups, where each UMP (input/output) Group corresponds
to the traditional MIDI I/O Endpoint.
Additionally, the ALSA UMP abstraction provides the multiple UMP
Blocks that can be assigned to each UMP Endpoint. A UMP Block is a
metadata to hold the UMP Group clusters, and can represent the
functions assigned to each UMP Group. A typical implementation of UMP
Block is the Group Terminal Blocks of USB MIDI 2.0 specification.
For distinguishing from the legacy byte-stream MIDI device, a new
device "umpC*D*" will be created, instead of the standard (MIDI 1.0)
devices "midiC*D*". The UMP instance can be identified by the new
rawmidi info bit SNDRV_RAWMIDI_INFO_UMP, too.
A UMP rawmidi device reads/writes only in 4-bytes words alignment,
stored in CPU native endianness.
The transmit and receive functions take care of the input/out data
alignment, and may return zero or aligned size, and the params ioctl
may return -EINVAL when the given input/output buffer size isn't
aligned.
A few new UMP-specific ioctls are added for obtaining the new UMP
endpoint and block information.
As of this commit, no ALSA sequencer instance is attached to UMP
devices yet. They will be supported by later patches.
Along with those changes, the protocol version for rawmidi is bumped
to 2.0.3.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A new callback, ioctl, is added to snd_rawmidi_global_ops for allowing
the driver to deal with the own ioctls. This is another preparation
patch for the upcoming UMP support.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_rawmidi_kernel_open() is used only internally from ALSA sequencer,
so far, and parsing the card / device matching table at each open is
redundant, as each sequencer client already gets the rawmidi object
beforehand.
This patch optimizes the path by passing the rawmidi object directly
at snd_rawmidi_kernel_open(). This is also a preparation for the
upcoming UMP rawmidi I/O support.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230523075358.9672-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230519201711.4073845-6-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of separate voices, we now allocate non-interleaved channels,
which may in turn contain two interleaved voices each. The higher-level
code keeps only one pointer per channel. The channels are not allocated
in one block any more, as there is no reason to do that. As a
consequence of that, and because it is cleaner regardless, we now let
the allocator store these pointers at a specified location, rather than
returning only the first one and having the calling code deduce the
remaining ones.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140947.3725394-8-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The voice allocator clearly knows about the field (it resets it), so
it's more consistent (and leads to less duplicated code) to have the
constructor take it as a parameter.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140947.3725394-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Richard Fitzgerald <rf@opensource.cirrus.com>:
First two patches are bugfixes.
Third patch skips the overhead of rebooting the amp after applying
firmware files when we know that it isn't necessary.
If the device is in secure mode it's unnecessary to send a SHUTDOWN and
SYSTEM_RESET around the firmware download. It could only be patching
insecure tunings. A tuning patch doesn't need a SHUTDOWN and only needs
a REINIT afterwards. This will reduce the overhead of exiting system
suspend in secure mode.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/Message-Id: <20230518150250.1121006-4-rf@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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.
Link: https://github.com/thesofproject/sof/issues/7596
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
This allows us to drop the code that tries to preserve already allocated
voices upon repeated hw_param callback invocations. Getting it right for
multi-channel voices would otherwise get a bit hairy.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140947.3725394-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliminate the MIDI type, as there is no such thing - the MPU401 port
doesn't have anything to do with voices.
For clarity, differentiate between regular and extra voices.
Don't atomize the enum into bits in the table display.
Simplify/optimize the storage.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140947.3725394-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Audigy, the send amounts are merely targets, presumably to avoid
sound distortion due to sudden changes, which the EMU8K docu explicitly
warns about.
However, that "soft-start" would prevent bit-for-bit reproduction, so
we now force the current send amounts to their final values at PCM
playback init.
One might want to do that for the MIDI synthesizer as well, though it
seems mostly pointless due to the attack phase each note has anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518140339.3722279-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While this nicely denoises the code, the real intent is being able to
write many registers pseudo-atomically, which will come in handy later.
Idea stolen from kX-project.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230518093134.3697955-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cache causes a fixed delay regardless of stream parameters.
Consequently, all that "cache invalidate size" calculation stuff was
garbage (which can be traced right back to Creative's OSS driver).
This also removes the definitions of registers CD1..CDF, because they
are accessed only relative to CD0 anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This workaround fails to address the underlying problem, which is
actually wholly self-made. Subsequent patches will fix it.
This reverts commit 56385a12d9.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230517174256.3657060-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Shrink the {in,out}put_source arrays and their data type to what is
actually necessary.
To be still on the safe side, add some static asserts.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536508-11-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unlike the other models, this is actually a distinct card, rather than
an E-MU 1010 with different "dongles". It is stereo only, and supports
no ADAT (there is no trace of ADAT in the manual, switching the output
mode to ADAT has no effect, and switching the input mode to ADAT just
breaks input (presumably ... my only ADAT source is the card's output)).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536508-10-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define arrays of strings instead of snd_kcontrol_new.
While at it, move the E-MU source & destination enum defs next to their
hardware defs, which is a lot more logical and will come in handy in a
followup commit. And add some static asserts to verify that the array
sizes match.
This also applies the compactization from the previous commit to the
destination registers.
While reshuffling the arrays anyway, switch the order of the HAMOA_DAC
& HANA_SPDIF output destinations for the 1010 card, so they follow a
more regular pattern. This should have no functional impact.
The code is somewhat de-duplicated by the extraction of add_ctls().
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536508-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many registers are meaningless for stereo slaves and the extra voices.
This patch cleans up these unnecessary register writes.
snd_emu10k1_playback_{trigger,stop}_voice() is not called for stereo
slaves any more.
snd_emu10k1_playback_prepare_voice() is renamed to
snd_emu10k1_playback_unmute_voice(), as this better reflects its
remaining function. It's not called for the extra voices any more.
Accordingly, snd_emu10k1_playback_mute_voice() is factored out from
snd_emu10k1_playback_stop_voice(), and is called selectively as well.
This doesn't add conditionals which would avoid initializing
sub-registers, as that wouldn't pull its weight.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536451-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We initialize them at card init and don't touch them later, so there is
no need to reset them again at voice start.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536451-4-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We (rightfully) don't enable the envelope engine for PCM voices, so any
related setup is entirely pointless - the EMU8K documentation makes that
very clear, and the fact that the various open drivers all use different
values to no observable detriment pretty much confirms it.
The remaining initializations are regrouped for clarity.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230516093612.3536451-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The voice volume is a raw fractional multiplier that can't actually
represent 1.0. To still enable real pass-through, we now set the volume
to 0.5 (which results in no loss of precision, as the FX bus provides
fractional values) and scale up the samples in DSP code.
To maintain backwards compatibility with existing configuration files,
we rescale the values in the mixer controls. The range is extended
upwards from 0xffff to 0x1fffd, which actually introduces the
possibility of specifying an amplification.
There is still a minor incompatibility with user space, namely if
someone loaded custom DSP code. They'll just get half the volume, so
this doesn't seem like a big deal.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230514170323.3408834-8-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fractional multiplication with the maximal value 2^31-1 causes some tiny
distortion. Instead, we want to multiply with the full 2^31. The catch
is of course that this cannot be represented in the DSP's signed 32 bit
registers.
One way to deal with this is to encode 1.0 as a negative number and
special-case it. As a matter of fact, the SbLive! code path already
contained such code, though the controls never actually exercised it.
A more efficient approach is to use negative values, which actually
extend to -2^31. Accordingly, for all the volume adjustments we now use
the MAC1 instruction which negates the X operand.
The range of the controls in highres mode is extended downwards, so -1
is the new zero/mute. At maximal excursion, real zero is not mute any
more, but I don't think anyone will notice this behavior change. ;-)
That also required making the min/max/values in the control structs
signed. This technically changes the user space interface, but it seems
implausible that someone would notice - the numbers were actually
treated as if they were signed anyway (and in the actual mixer iface
they _are_). And without this change, the min value didn't even make
sense in the first place (and no-one noticed, because it was always 0).
Tested-by: Jonathan Dowland <jon@dow.land>
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230514170323.3408834-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The idea to encode the bitfield manipulation in the register address is
quite clever, but doing that by hand is ugly and error-prone. So derive
it automatically from the mask instead.
Macros cannot #define other macros, so we now declare enums instead.
This also adds macros for decoding the register definitions. These will
be used by later commits.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230514170323.3408798-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These tables are used for 'nocodec' and SoundWire mockups+RVP tests.
The LNL RVP has a single rt711-sdca SoundWire codec.
Co-developed-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Link: https://lore.kernel.org/r/20230512173305.65399-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Add a function to allow ASoC drivers to easily notify an ALSA control
change. This function will automatically add any component naming
prefix into the control name.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com
Link: https://lore.kernel.org/r/20230512122838.243002-3-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org
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
Same functionality as for DMIC/SSP with different ID.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com
Reviewed-by: Rander Wang <rander.wang@intel.com
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com
Link: https://lore.kernel.org/r/20230512174611.84372-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org
Evidently, the channel delay bug exists in all E-MU cards; it's in the
Hana FPGA program, and was never fixed.
Note that the implementation is somewhat lazy - to localize the code
paths, we actually waste a GPR and a DSP instruction by keeping two
delay registers for the same physical source.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230510173917.3073107-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit 5bbb1ab5bd ("control: use counting semaphore as write lock
for ELEM_WRITE operation"), this has been locking the controls including
their values, not just the list of controls.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230428095941.1706278-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC uses dummy Component, sharing snd_soc_dai_link_component
for it is better idea. This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com
Link: https://lore.kernel.org/r/87a5yy0zyk.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org
The bulk of the commits here are for the conversion of drivers to use
void remove callbacks but there's a reasonable amount of other stuff
going on, the pace of development with the SOF code continues to be high
and there's a bunch of new drivers too:
- More core cleanups from Morimto-san.
- Update drivers to have remove() callbacks returning void, mostly
mechanical with some substantial changes.
- Continued feature and simplification work on SOF, including addition
of a no-DSP mode for bringup, HDA MLink and extensions to the IPC4
protocol.
- Hibernation support for CS35L45.
- More DT binding conversions.
- Support for Cirrus Logic CS35L56, Freescale QMC, Maxim MAX98363,
nVidia systems with MAX9809x and RT5631, Realtek RT712, Renesas R-Car
Gen4, Rockchip RK3588 and TI TAS5733.
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Merge tag 'asoc-v6.4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v6.4
The bulk of the commits here are for the conversion of drivers to use
void remove callbacks but there's a reasonable amount of other stuff
going on, the pace of development with the SOF code continues to be high
and there's a bunch of new drivers too:
- More core cleanups from Morimto-san.
- Update drivers to have remove() callbacks returning void, mostly
mechanical with some substantial changes.
- Continued feature and simplification work on SOF, including addition
of a no-DSP mode for bringup, HDA MLink and extensions to the IPC4
protocol.
- Hibernation support for CS35L45.
- More DT binding conversions.
- Support for Cirrus Logic CS35L56, Freescale QMC, Maxim MAX98363,
nVidia systems with MAX9809x and RT5631, Realtek RT712, Renesas R-Car
Gen4, Rockchip RK3588 and TI TAS5733.
Firstly, fix the distribution between public and private headers.
Otherwise, some of the already public macros wouldn't actually work, and
the SNDRV_EMU10K1_IOCTL_DBG_READ result for Audigy would be useless.
Secondly, add condition code registers for Audigy. These are just
aliases for selected constant registers, and thus are generation-
specific. At least A_CC_REG_ZERO is actually correct ...
Finally, shuffle around some defines to more logical places while at it,
and fix up some more comments.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230422161021.1143903-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For documentation purposes and later use.
Some pre-existing but (mostly) unused definitions were renamed for
consistency.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230422161021.1143903-6-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Firstly, remove the FXWC_* defines - the comment on FXWC implies that
the relevant defines are the (A_)EXTOUT_* ones. It's unclear where this
came from - it was in the initial ALSA import, but neither the driver
from Creative nor kX-project have these defines.
Secondly, remove A_HR, which made plain no sense (was unused, and
clashed with FXRT). Amends commit cbb7d8f9b7 ("emu10k1: Update
registers defines for the Audigy 2/emu10k2.5").
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230422161021.1143903-5-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move comments to better locations, de-duplicate, fix/remove incorrect/
outdated ones, add new ones, and unify spacing somewhat.
While at it, also add testing credits for Jonathan Dowland (SB Live!
Platinum) and myself (E-MU 0404b).
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230422161021.1143903-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unlike the Alice2 chips used on 1st generation E-MU cards, the
Tina/Tina2 chips used on the 2nd gen cards have only six GPIN pins,
which means that we need to use a smaller mask. Failure to do so would
falsify the read data if the FPGA tried to raise an IRQ right at that
moment. This wasn't a problem so far, as we didn't actually enable FPGA
IRQs, but that's going to change soon.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230422132430.1057490-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the register definition clearly states, this is a 16-bit register,
yet we did all accesses as 32-bit. The writes in particular would have
the potential to clear the TIMER register (depending on how the bus/card
actually handles the too long writes).
This commit also introduces a separate define A_GPIO which aliases
A_IOCFG, which better reflects the distinct usage on E-MU cards.
This is done in the same commit to keep the churn down, as we're
touching all involved lines anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230421141006.1005539-2-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Assert the validity of the registers and values, as them being out of
range would indicate an error in the driver. Consequently, don't bother
returning error codes; they were ignored everywhere anyway.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230421141006.1005539-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Detach it better from the main PCM driver, which it really doesn't have
much in common with.
In particular, this moves the interrupt handler implementation into
p16v.c, and makes it access the substream runtime status more directly,
so it doesn't need to abuse structs snd_emu10k1_pcm and
snd_emu10k1_voice any more.
We don't need private pcm runtime data at all, as the only thing it was
used for (except the back-link to the substream) was the `running` flag.
So store that directly in runtime->private_data.
This somewhat radical strip-down shows that this driver contains some
complexity that was never actually utilized. I suppose the right way to
fully utilize the hardware in a simple way would be introducing more
substreams. This wouldn't require any of the removed code.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230421141006.1005452-7-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The use of the variable was removed in commit 2b637da5a1 ("clean up
card features"). That commit also broke user space (the ioctl
structure), at which point the defines became meaningless, so I don't
think purging them is a problem.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230421141006.1005452-3-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
extin_mask and extout_mask are used only by the SbLive! microcode, so
they have no effect on Audigy.
Eliminate fxbus_mask entirely, as it wasn't actually used for anything.
As a drive-by, remove the pointless pad1 field from struct
snd_emu10k1_fx8010 - it is not visible to user space, so it has no
binary compatibility constraints.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Link: https://lore.kernel.org/r/20230421141006.1005509-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-silencer supports two modes: "thresholded" to fill up "just
enough", and "top-up" to fill up "as much as possible". The two modes
used rather distinct code paths, which this patch unifies. The only
remaining distinction is how much we actually want to fill.
This fixes a bug in thresholded mode, where we failed to use new_hw_ptr,
resulting in under-fill.
Top-up mode is now more well-behaved and much easier to understand in
corner cases.
This also updates comments in the proximity of silencing-related data
structures.
Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen@gmx.de>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230420113324.877164-1-oswald.buddenhagen@gmx.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename the mixer source defines from CS35L56_INPUT_SRC_SWIRE_RXn
to CS35L56_INPUT_SRC_SWIRE_DP1_CHANNELn to match the latest
datasheet.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230418144309.1100721-5-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The mixer source index value for SDW2RX1 is different between
A1 and B0 silicon. As the driver doesn't provide a DAI for SDW2
just remove it as a mixer source option.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230418144309.1100721-4-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Reduce SDW1 to 4 channels and remove the controls for SDW1
TX5 and TX6.
The TX5 and TX6 channels have been removed from B0 silicon.
There is no need to support them on A1 silicon.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230418144309.1100721-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC need to use card->mutex with _INIT or _RUNTIME,
but there is no helper function for it.
This patch adds its helper function and use it.
Because people might misunderstand that _init() is mutex initialization,
this patch renames _INIT to _ROOT and adds new
snd_soc_card_mutex_lock_root() for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a5zlx3tw.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc-pcm.c has snd_soc_dpcm_mutex_lock/unlock(),
but other files can't use it because it is static function.
It requests snd_soc_pcm_runtime as parameter (A), but sometimes we
want to use it by snd_soc_card (B).
(A) static inline void snd_soc_dpcm_mutex_lock(struct snd_soc_pcm_runtime *rtd)
{
mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass);
} ^^^^^^^^^
(B) mutex_lock_nested(&card->pcm_mutex, card->pcm_subclass);
^^^^
We want to use it with both "rtd" and "card" for dapm lock/unlock.
To enable it, this patch uses _Generic macro.
This patch makes snd_soc_dpcm_mutex_{un}lock() global function, and use it on
each files.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87bkk1x3ud.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc.h has snd_soc_dapm_mutex_lock/unlock() definition and
many drivers are using it, but soc-dapm.c is not.
1st reason is snd_soc_dapm_mutex_lock/unlock() requests
snd_soc_dapm_context pointer as parameter (A), but sometimes soc-dapm.c
needs to use snd_soc_card (B).
(A) static inline void snd_soc_dapm_mutex_lock(struct snd_soc_dapm_context *dapm)
{
mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
} ^^^^^^^^^^
(B) mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
^^^^
2nd reason is it want to use SND_SOC_DAPM_CLASS_INIT for mutex_lock_nested(),
but helper is using _RUNTIME (A).
The conclusion is we want to use "dapm vs card" and "_RUNTIME vs _INIT"
for dapm lock/unlock. To enable this selfish request, this patch uses
_Generic macro. We can use snd_soc_dapm_mutex_lock/unlock() for both
dapm and card case.
snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_mutex_unlock(dapm);
snd_soc_dapm_mutex_lock(card); snd_soc_dapm_mutex_unlock(card);
Current soc-dapm.c is using both mutex_lock() and mutex_lock_nested().
This patch handles mutex_lock() as mutex_lock_nested(..., 0),
in other words, handles below as same.
mutex_lock(&card->dapm_mutex);
mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
Because people might misunderstand that _init() is mutex initialization,
this patch renames _INIT to _ROOT and adds new
snd_soc_dapm_mutex_lock_root() for it.
This patch also moves snd_soc_dapm_subclass definition from soc-dapm.h
to soc.h to keep related code together.
Because very complex soc.h vs soc-dapm.h relationship,
it is difficult/impossible to define these helper into soc-dapm.h.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87cz4hx3v0.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Check during cs35l56_system_resume() whether the firmware patch must
be applied again.
The FIRMWARE_MISSING flag in the PROTECTION_STATUS register indicates
whether the firmware has been patched.
In non-secure mode the FIRMWARE_MISSING flag is cleared at the end of
dsp_work(). If it is set after system-resume we know that dsp_work()
must be run again.
In secure mode the pre-OS loader will have done the secure patching
and cleared the FIRMWARE_MISSING flag. So this flag does not tell us
whether firmware memory was lost. But the driver could only be
downloading non-secure tunings, which is always safe to do.
If the driver has control of RESET we will have asserted it during
suspend so the firmware patch will have been lost. The driver would only
have control of RESET in non-secure mode.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/168122674550.26.8545058503709956172@mailman-core.alsa-project.org
Signed-off-by: Mark Brown <broonie@kernel.org>
For SoundWire usages, we need to use the global eml_lock to
serialize/protect all accesses to shared registers. Due to the split
implementation across two subsystems, we need to pass a pointer
around.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-19-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For DMIC and SSP, the DSP will be responsible for programming the
blobs and link registers.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-18-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Small helpers to make DAI ops simpler.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-17-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Each SoundWire peripheral can be programmed from the manager side
either with a regular command FIFO, or with the HDaudio CORB/RIRB
DMA-based mechanism. The mapping between SoundWire peripheral and SDI
address is handled with the LSDIID register.
This mapping only works of course if each peripheral has a unique
address across all links. This has already been enforced in previous
Intel contributions allowing for an IDA-based solution for the device
number allocation.
The checks on the dev_num are handled at the SoundWire level, but the
locking is handled at the hda-mlink level.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-16-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This helper is an optimization where sync_go is only called when the
cmdsync field is actually set to a non-zero value.
Since this is also only used by SoundWire for now, only expose the
_unlocked version.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-15-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The multi-link synchronization uses the same concept and registers,
but moved to the HDAudio extended links.
Add helpers for sync_arm and sync_go which are the basic for the bus
reset, bank switch and clock stop.
Since SoundWire is the only user of those helpers, only expose the
_unlocked versions for now.
Note that SYNCGO is a write-only bit, so no error can be reported. We
still return 0 for compatibility with the SoundWire stream management
headers.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-14-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
These helpers configure the ratio between the base clock and the
hardware signal used for link synchronization.
The SYNCPRD is written before the first sublink is powered-up. The
SYNCPU bit is set, but it will only be cleared after the link is
powered-up, hence the implementation with a set/wait pattern.
These helpers are currently only needed by SoundWire support, where
the lock is taken at a higher level, so only the _unlocked versions
are exposed for now.
Note that the _wait_bit() implementation is similar to previous
helpers in drivers/soundwire, but with sleep duration and timeout
aligned with hardware recommendations. If desired, this helper could
be modified in a second step with e.g. readl_poll_timeout().
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-13-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When INTC is set, LCTL exposes INTEN and INTSTS fields.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-12-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This is needed for SoundWire integration.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-11-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The updated SoundWire Intel driver will need to rely on Extended
HDaudio links for power management, but it doesn't need to be aware of
all the HDaudio structures. Add convenience helpers to avoid polluting
SoundWire drivers too much with HDaudio information.
Since the SoundWire/Intel solution already takes the lock at a higher
level, the _unlocked PM helpers are used.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-10-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helpers to program SPA/CPA bits, using a mutex to access the
shared LCTL register if required.
All links are managed with the same LCTLx.SPA bits. However there are
quite a few implementation details to be aware of:
Legacy HDaudio multi-links are powered-up when exiting reset, which
requires the ref_count to be manually set to one when initializing the
link.
Alternate links for SoundWire/DMIC/SSP need to be explicitly
powered-up before accessing the SHIM/IP/Vendor-Specific SHIM space for
each sublink. DMIC/SSP/SoundWire are all different cases with a
different device/dai/hlink relationship.
SoundWire will handle power management with the auxiliary device
resume/suspend routine. The ref_count is not necessary in this case.
The DMIC/SSP will by contrast handle the power management from DAI
.startup and .shutdown callbacks.
The SSP has a 1:1 mapping between sublink and DAI, but it's
bidirectional so the ref_count will help avoid turning off the sublink
when one of the two directions is still in use.
The DMIC has a single link but two DAIs for data generated at
different sampling frequencies, again the ref_count will make sure the
two DAIs can be used concurrently.
And last the SoundWire Intel require power-up/down and bank switch to
be handled with a lock already taken, so the 'eml_lock' is made
optional with the _unlocked versions of the helpers.
Note that the _check_power_active() implementation is similar to
previous helpers in sound/hda/ext, with sleep duration and timeout
aligned with hardware recommendations. If desired, this helper could
be modified in a second step with .e.g. readl_poll_timeout()
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-9-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Extend hdac_ext_link to store information needed for ALT
links. Follow-up patches will include more functional patches for
power-up and down.
Note that this patch suggests the use of an 'eml_lock' to serialize
access to shared registers. SoundWire-specific sequence require the
lock to be taken at a higher level, as a result the helpers added in
follow-up patches will provide 'unlocked' versions when needed.
Also note that the low-level sequences with the 'hdaml_' prefix are
taken directly from the hardware specifications - naming conventions
included. The code will be split in two, with locking and linked-list
management handled separately to avoid mixing required hardware setup
and Linux-based resource management.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-7-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Some of the functions will be used for SoundWire enumeration and power
management, to avoid cycles in module dependencies and simplify
integration all the HDaudio multi-link needs to move to a dedicated
module.
Drop no longer needed headers at the same time.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-6-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add new definitions for the HDaudio Extended link support,
specifically new registers for SoundWire, Intel DMIC and INTEL SSP
interfaces.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20230404104127.5629-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_link has params/num_params, but it is unclear that
params for what. This patch clarify it is params for Codec2Codec.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87o7o5c2lk.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The DSPless mode of the ASoC/SOF driver can be used for hardware
verification and debug on platforms with HDaudio codecs. The DSP mode is
still needed on existing platforms for SSP, DMIC, SoundWire interfaces
managed by the GP-DMA.
This mode is also helpful to compare the legacy HDaudio driver with the
ASoC/SOF driver wrt. codec management and handling. In theory we use the
same code but differences are sometimes seen on jack detection and event
handling.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@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: Rander Wang <rander.wang@intel.com>
Link: https://lore.kernel.org/r/20230404092115.27949-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
For allowing the build without CONFIG_PM, add definitions of dummy
functions for snd_ac97_suspend() and snd_ac97_resume() without
CONFIG_PM, too.
Link: https://lore.kernel.org/r/20230330132847.12882-1-tiwai@suse.de
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Tasos Sahanidis <tasos@tasossah.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current ASoC supports snd_soc_add_pcm_runtime(), but user need to
call it one-by-one if it has multi dai_links.
This patch adds snd_soc_add_pcm_runtimes() which supports multi
dai_links.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87h6u76nhq.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
As PCM ack callback may handle the XRUN situation gracefully now,
change the indirect PCM helpers to give a proper error (-EPIPE).
Also, change the pointer callback helpers to deal with the XRUN error
properly, too.
This requires the PCM core change by the commit 8c721c53dd
("ALSA: usb-audio: Fix recursive locking at XRUN during syncing").
Link: https://lore.kernel.org/r/20230323065237.5062-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
On a platform when the DSP is in use, we cannot select individual links
to use or not use the DSP, it is either all or none. On some audio
endpoint, like HDMI/DP, it is preferred to not use any processing in DSP
to reduce the latency and to allow bytestream pass-through (DTS, DD,
etc)
IPC4 introduces a new type of end-to-end connection within the DSP which
is using the host DMA and link DMA in a single buffer, working
back-to-back, passing the received data without looking at it or trying
to understand the format, content.
This mode reduces the latency and allows non PCM streams to be sent from
userspace.
The feature is enabled per PCM bases, signalled in topology.
In the chained DMA mode, the firmware allocates buffers for the host
and link DMA, and takes care of copying data between host- and
link-DMA buffers in a low-latency thread. This is different to a
regular pipeline, no processing is allowed, and the connection between
host- and link DMA is handled with a dedicated IPC.
This patch exposes the macros needed to create the required IPC messages.
Signed-off-by: Jyri Sarha <jyri.sarha@intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20230321092654.7292-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The CS35L56 combines a high-performance mono audio amplifier, Class-H
tracking inductive boost converter, Halo Core(TM) DSP and a DC-DC boost
converter supporting Class-H tracking.
Supported control interfaces are I2C, SPI or SoundWire.
Supported audio interfaces are I2S/TDM or SoundWire.
Most chip functionality is controlled by on-board ROM firmware that is
always running. The driver must apply patch/tune to the firmware
before using the CS35L56.
Signed-off-by: Simon Trimmer <simont@opensource.cirrus.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20230320112245.115720-9-rf@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
substream->runtime will be attached when substream was opened
at snd_pcm_attach_substream(). When it uses DPCM,
FE substream->runtime is attached, but BE substream->runtime is not.
Thus, we are copying FE substream->runtime to BE.
But, we are copyig FE substream->runtime to FE dpcm->runtime first (A),
and copy it to BE dpcm->runtime (B), and copy it to
BE substream->runtime (C).
static int dpcm_fe_dai_open(...) {
...
(A) fe->dpcm[stream].runtime = fe_substream->runtime;
...
}
static int dpcm_be_connect(...) {
...
(B) be->dpcm[stream].runtime = fe->dpcm[stream].runtime;
...
}
int dpcm_be_dai_startup(...) {
...
(C) be_substream->runtime = be->dpcm[stream].runtime;
...
}
It is too roundabout and troublesome.
OTOH, it is directly copying fe_substream->runtime at dpcm_be_reparent()
without using be->dpcm[stream].runtime.
static void dpcm_be_reparent(...)
{
...
for_each_dpcm_fe(be, stream, dpcm) {
...
=> be_substream->runtime = fe_substream->runtime;
break;
}
}
This patch removes indirect copying.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87v8je64dh.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The create pipeline message can carry the target code_id which is set to
0 at the moment.
Add macros to set the core_id in the message extension.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20230307123556.31328-2-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Shared boost allows two amplifiers to share a single boost circuit by
communicating on the MDSYNC bus.
The passive amplifier does not control the boost and receives data from
the active amplifier.
Shared Boost is not supported in HDA Systems.
Based on David Rhodes shared boost patches.
Signed-off-by: Lucas Tanure <lucas.tanure@collabora.com>
Reviewed-by: David Rhodes <david.rhodes@cirrus.com>
Link: https://lore.kernel.org/r/20230223084324.9076-4-lucas.tanure@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A collection of various small fixes that have been gathered since
the last PR. The majority of changes are for ASoC, and there is
a small change in ASoC PCM core, but the rest are all for driver-
specific fixes / quirks / updates.
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Merge tag 'sound-fix-6.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of various small fixes that have been gathered since the
last PR.
The majority of changes are for ASoC, and there is a small change in
ASoC PCM core, but the rest are all for driver- specific fixes /
quirks / updates"
* tag 'sound-fix-6.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (32 commits)
ALSA: ice1712: Delete unreachable code in aureon_add_controls()
ALSA: ice1712: Do not left ice->gpio_mutex locked in aureon_add_controls()
ALSA: hda/realtek: Add quirk for HP EliteDesk 800 G6 Tower PC
ALSA: hda/realtek: Improve support for Dell Precision 3260
ASoC: mediatek: mt8195: add missing initialization
ASoC: mediatek: mt8188: add missing initialization
ASoC: amd: yc: Add DMI entries to support HP OMEN 16-n0xxx (8A43)
ASoC: zl38060 add gpiolib dependency
ASoC: sam9g20ek: Disable capture unless building with microphone input
ASoC: mt8192: Fix range for sidetone positive gain
ASoC: mt8192: Report an error if when an invalid sidetone gain is written
ASoC: mt8192: Fix event generation for controls
ASoC: mt8192: Remove spammy log messages
ASoC: mchp-pdmc: fix poc noise at capture startup
ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding
ASoC: soc-pcm: add option to start DMA after DAI
ASoC: mt8183: Fix event generation for I2S DAI operations
ASoC: mt8183: Remove spammy logging from I2S DAI driver
ASoC: mt6358: Remove undefined HPx Mux enumeration values
ASoC: mt6358: Validate Wake on Voice 2 writes
...
Almost all of this is driver specific fixes and new IDs that have come
in during the merge window. A good chunk of them are simple ones from
me which came about due to a bunch of Mediatek Chromebooks being enabled
in KernelCI, there's more where that came from.
We do have one small feature added to the PCM core by Claudiu Beznea in
order to allow the sequencing required to resolve a noise issue with the
Microchip PDMC driver.
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Merge tag 'asoc-fix-v6.3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.3
Almost all of this is driver specific fixes and new IDs that have come
in during the merge window. A good chunk of them are simple ones from
me which came about due to a bunch of Mediatek Chromebooks being enabled
in KernelCI, there's more where that came from.
We do have one small feature added to the PCM core by Claudiu Beznea in
order to allow the sequencing required to resolve a noise issue with the
Microchip PDMC driver.
Add option to start DMA component after DAI trigger. This is done
by filling the new struct snd_soc_component_driver::start_dma_last.
Signed-off-by: Claudiu Beznea <claudiu.beznea@microchip.com>
Link: https://lore.kernel.org/r/20230228110145.3770525-2-claudiu.beznea@microchip.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Here is the large set of driver core changes for 6.3-rc1.
There's a lot of changes this development cycle, most of the work falls
into two different categories:
- fw_devlink fixes and updates. This has gone through numerous review
cycles and lots of review and testing by lots of different devices.
Hopefully all should be good now, and Saravana will be keeping a
watch for any potential regression on odd embedded systems.
- driver core changes to work to make struct bus_type able to be moved
into read-only memory (i.e. const) The recent work with Rust has
pointed out a number of areas in the driver core where we are
passing around and working with structures that really do not have
to be dynamic at all, and they should be able to be read-only making
things safer overall. This is the contuation of that work (started
last release with kobject changes) in moving struct bus_type to be
constant. We didn't quite make it for this release, but the
remaining patches will be finished up for the release after this
one, but the groundwork has been laid for this effort.
Other than that we have in here:
- debugfs memory leak fixes in some subsystems
- error path cleanups and fixes for some never-able-to-be-hit
codepaths.
- cacheinfo rework and fixes
- Other tiny fixes, full details are in the shortlog
All of these have been in linux-next for a while with no reported
problems.
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Merge tag 'driver-core-6.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/driver-core
Pull driver core updates from Greg KH:
"Here is the large set of driver core changes for 6.3-rc1.
There's a lot of changes this development cycle, most of the work
falls into two different categories:
- fw_devlink fixes and updates. This has gone through numerous review
cycles and lots of review and testing by lots of different devices.
Hopefully all should be good now, and Saravana will be keeping a
watch for any potential regression on odd embedded systems.
- driver core changes to work to make struct bus_type able to be
moved into read-only memory (i.e. const) The recent work with Rust
has pointed out a number of areas in the driver core where we are
passing around and working with structures that really do not have
to be dynamic at all, and they should be able to be read-only
making things safer overall. This is the contuation of that work
(started last release with kobject changes) in moving struct
bus_type to be constant. We didn't quite make it for this release,
but the remaining patches will be finished up for the release after
this one, but the groundwork has been laid for this effort.
Other than that we have in here:
- debugfs memory leak fixes in some subsystems
- error path cleanups and fixes for some never-able-to-be-hit
codepaths.
- cacheinfo rework and fixes
- Other tiny fixes, full details are in the shortlog
All of these have been in linux-next for a while with no reported
problems"
[ Geert Uytterhoeven points out that that last sentence isn't true, and
that there's a pending report that has a fix that is queued up - Linus ]
* tag 'driver-core-6.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/driver-core: (124 commits)
debugfs: drop inline constant formatting for ERR_PTR(-ERROR)
OPP: fix error checking in opp_migrate_dentry()
debugfs: update comment of debugfs_rename()
i3c: fix device.h kernel-doc warnings
dma-mapping: no need to pass a bus_type into get_arch_dma_ops()
driver core: class: move EXPORT_SYMBOL_GPL() lines to the correct place
Revert "driver core: add error handling for devtmpfs_create_node()"
Revert "devtmpfs: add debug info to handle()"
Revert "devtmpfs: remove return value of devtmpfs_delete_node()"
driver core: cpu: don't hand-override the uevent bus_type callback.
devtmpfs: remove return value of devtmpfs_delete_node()
devtmpfs: add debug info to handle()
driver core: add error handling for devtmpfs_create_node()
driver core: bus: update my copyright notice
driver core: bus: add bus_get_dev_root() function
driver core: bus: constify bus_unregister()
driver core: bus: constify some internal functions
driver core: bus: constify bus_get_kset()
driver core: bus: constify bus_register/unregister_notifier()
driver core: remove private pointer from struct bus_type
...
The majority of works in this cycle are about ASoC spread over trees.
Most of them are for new devices and cleanups / refactoring works,
and not much significant changes are seen in the core side.
Below are some highlights:
ASoC:
- Continued refactoring to move into common helper functions
- Lots of DT schema conversons and stylistic nits
- Continued work on building out the new SOF IPC4 scheme
- Continued work for Intel AVS
- New drivers for Awinc AT88395, Infineon PEB2466, Iron Device
SMA1303, Mediatek MT8188, Realtek RT712, Renesas IDT821034,
Samsung/Tesla FSD SoC I2S, and TI TAS5720A-Q1
ALSA:
- A few cleanups to make the remove callbacks to void returns
- FireWire refactoring and enhancements
- PCM kselftest enhancements
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Merge tag 'sound-6.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"The majority of works in this cycle are about ASoC spread over trees.
Most of them are for new devices and cleanups / refactoring works, and
not much significant changes are seen in the core side.
Below are some highlights:
ASoC:
- Continued refactoring to move into common helper functions
- Lots of DT schema conversons and stylistic nits
- Continued work on building out the new SOF IPC4 scheme
- Continued work for Intel AVS
- New drivers for Awinc AT88395, Infineon PEB2466, Iron Device
SMA1303, Mediatek MT8188, Realtek RT712, Renesas IDT821034,
Samsung/Tesla FSD SoC I2S, and TI TAS5720A-Q1
ALSA:
- A few cleanups to make the remove callbacks to void returns
- FireWire refactoring and enhancements
- PCM kselftest enhancements"
* tag 'sound-6.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (398 commits)
ALSA: hda/hdmi: Register with vga_switcheroo on Dual GPU Macbooks
ASoC: soc-ac97: Return correct error codes
ASoC: soc-dapm.h: fixup warning struct snd_pcm_substream not declared
ASoC: cs35l45: Remove separate namespace for tables
ASoC: cs35l45: Remove separate tables module
ASoC: soc-ac97: Convert to agnostic GPIO API
ASoC: dt-bindings: renesas,rsnd.yaml: drop "dmas/dma-names" from "rcar_sound,ssi"
ALSA: hda: cs35l41: Enable Amp High Pass Filter
ALSA: hda: cs35l41: Ensure firmware/tuning pairs are always loaded
ALSA: hda: cs35l41: Correct error condition handling
ASoC: codecs: wcd934x: Use min macro for comparison and assignment
ASoC: Intel: Skylake: Fix struct definition
ASoC: tlv320adcx140: extend list of supported samplerates
ASoC: imx-pcm-rpmsg: Remove unused variable
SoC: rt5682s: Disable jack detection interrupt during suspend
ASoC: SOF: Intel: hda-dsp: Set streaming flag for d0i3
ASoC: SOF: Intel: Enable d0i3 work for ipc4
ASoC: SOF: ipc4: Wake up dsp core before sending ipc msg
ASoC: SOF: Intel: hda-dsp: use set_pm_gate according to ipc version
ASoC: SOF: Introduce a new set_pm_gate() IPC PM op
...
There's been quite a lot of activity this release, but not really
one big feature - lots of new devices, plus a lot of cleanup and
modernisation work spread throughout the subsystem:
- More factoring out of common operations into helper functions
by Morimoto-san.
- DT schema conversons and stylistic nits.
- Continued work on building out the new SOF IPC4 scheme.
- Support for Awinc AT88395, Infineon PEB2466, Iron Device
SMA1303, Mediatek MT8188, Realtek RT712, Renesas IDT821034,
Samsung/Tesla FSD SoC I2S, and TI TAS5720A-Q1.
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Merge tag 'asoc-v6.3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v6.3
There's been quite a lot of activity this release, but not really
one big feature - lots of new devices, plus a lot of cleanup and
modernisation work spread throughout the subsystem:
- More factoring out of common operations into helper functions
by Morimoto-san.
- DT schema conversons and stylistic nits.
- Continued work on building out the new SOF IPC4 scheme.
- Support for Awinc AT88395, Infineon PEB2466, Iron Device
SMA1303, Mediatek MT8188, Realtek RT712, Renesas IDT821034,
Samsung/Tesla FSD SoC I2S, and TI TAS5720A-Q1.
Add struct snd_pcm_substream forward declaration
Fixes: 078a85f280 ("ASoC: dapm: Only power up active channels from a DAI")
Signed-off-by: Lucas Tanure <lucas.tanure@collabora.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Link: https://lore.kernel.org/r/20230215132851.1626881-1-lucas.tanure@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This 2nd variables are all set as true in treewide. So I think
it can be removed for easy understanding.
Signed-off-by: Zhang Yiqun <zhangyiqun@phytium.com.cn>
Acked-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230209121723.14328-1-zhangyiqun@phytium.com.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All callers from other files ignore the return value of this function.
And it can only ever return a non-zero value if the parameter card is NULL.
This cannot happen in snd_card_free() as card was dereferenced just before
snd_card_free_when_closed() is called. So the error handling can be dropped
there.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Geoff Levand <geoff@infradead.org>
Acked-by: Thierry Reding <treding@nvidia.com>
Link: https://lore.kernel.org/r/20230207191907.467756-3-u.kleine-koenig@pengutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All callers from other files ignore the return value of this function.
And it can only ever return a non-zero value if the parameter card is NULL.
Move the check for card being NULL into snd_card_free_when_closed() to keep
the previous behaviour. Note this isn't necessary for
snd_card_disconnect_sync() because if card was NULL in there the dereference
of card for dev_err() would oops the kernel. Replace this by an oops
triggered by the dereference of card for spin_lock_irq().
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Geoff Levand <geoff@infradead.org>
Acked-by: Thierry Reding <treding@nvidia.com>
Link: https://lore.kernel.org/r/20230207191907.467756-2-u.kleine-koenig@pengutronix.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some motherboards have multiple HDA codecs connected to the serial bus.
The current code may create multiple mixer controls with the almost
identical identification.
The current code use id.device field from the control element structure
to store the codec address to avoid such clashes for multiple codecs.
Unfortunately, the user space do not handle this correctly. For mixer
controls, only name and index are used for the identifiers.
This patch fixes this problem to compose the index using the codec
address as an offset in case, when the control already exists. It is
really unlikely that one codec will create 10 similar controls.
This patch adds new kernel module parameter 'ctl_dev_id' to allow
select the old behaviour, too. The CONFIG_SND_HDA_CTL_DEV_ID Kconfig
option sets the default value.
BugLink: https://github.com/alsa-project/alsa-lib/issues/294
BugLink: https://github.com/alsa-project/alsa-lib/issues/205
Fixes: 54d1740315 ("[ALSA] hda-codec - Fix connection list parsing")
Fixes: 1afe206ab6 ("ALSA: hda - Try to find an empty control index when it's occupied")
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20230202092013.4066998-1-perex@perex.cz
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename Audio buffer and soundwire manager instance registers.
Remove scratch registers as these registers can be accessed
using ACP_SCRATCH_REG_0 register relative offset.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Link: https://lore.kernel.org/r/20230201165626.3169041-1-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Stefan Binding <sbinding@opensource.cirrus.com>:
The CS42L42 has a SoundWire interface for control and audio. This
chain of patches adds support for this.
Patches #1 .. #5 split out various changes to the existing code that
are needed for adding Soundwire. These are mostly around clocking and
supporting the separate probe and enumeration stages in SoundWire.
Patches #6 .. #8 actually adds the SoundWire handling.
The SOFT_RESET_REBOOT register is needed to recover CS42L42 state after
a Soundwire bus reset.
This is required to be set whenever there is severe/hard bus reset.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20230127165111.3010960-3-sbinding@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_dai has data for Playback/Capture, but it is very
random. Someone is array (A), someone is playback/capture (B),
and someone is tx/rx (C);
struct snd_soc_dai {
...
(A) unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1];
(B) struct snd_soc_dapm_widget *playback_widget;
(B) struct snd_soc_dapm_widget *capture_widget;
(B) void *playback_dma_data;
(B) void *capture_dma_data;
...
(C) unsigned int tx_mask;
(C) unsigned int rx_mask;
};
Because of it, the code was very complicated.
This patch creates new data structure to merge these into one,
and tidyup the code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87cz6vea1v.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC framework/driver checks whether card was instantiated every
where. Then, it should check card pointer too in such case.
This patch adds snd_soc_card_is_instantiated() for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/875ycnfoqp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_get_widget() requests SNDRV_PCM_STREAM_PLAYBACK/CAPTURE.
This patch adds helper for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/877cx3foqz.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC has tx/rx_mask, and is directly accessing to them,
but accessing to it via function is nice idea.
This patch adds snd_soc_dai_tdm_mask_set/get() for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/878rhjfor8.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC has snd_soc_dai_set/get_dma_data() which is assuming
struct snd_pcm_substream to get Playback/Capture direction.
But, many drivers want to use it not through snd_pcm_substream.
This patch adds more low level snd_soc_dai_dma_data_set/get() for it,
and previous functions will be macro for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87a61zfori.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ASoC has snd_soc_dai_get_widget() (= _get_) but doesn't
have _set_ function. This patch adds it.
This patch also cleanup unnecessary line break for _get_ function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/87bkmfforp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
struct snd_soc_dobj only needs pointer to the unload function, instead
however, there is pointer to all topology operations. Change code to use
the function pointer instead of pointer to structure containing all
operations.
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20230127231111.937721-12-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
The following series will enable multi-stream support for playback and capture
streams.
Currently only a single PCM can be connected to a DAI, with the multi-stream
support it is possible to connect multiple PCMs to a single DAI.
To achieve this we need to make sure that DAIs/AIF are only set up once since
other stream could be connected to it later.
We also need to introduce reference or use counting for widgets to make sure
that they are not going to be destroyed while other streams are still using
them.
With the multi-stream support we also need to extend our current locking scheme
which worked well for simple paths.
The uevent() callback in struct bus_type should not be modifying the
device that is passed into it, so mark it as a const * and propagate the
function signature changes out into all relevant subsystems that use
this callback.
Acked-by: Rafael J. Wysocki <rafael@kernel.org>
Acked-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20230111113018.459199-16-gregkh@linuxfoundation.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Add a generic way to create jack inputs for auxiliary jack detection
drivers (e.g. via i2c, spi), which are not part of any real codec.
The simple-card can be used as combining card driver to add the jacks,
no new one is required.
Create a jack (for input-events) for jack devices in the auxiliary
device list (aux_devs). A device which returns a valid value on
get_jack_type counts as jack device; set_jack is required
to add the jack to the device.
Signed-off-by: Astrid Rost <astrid.rost@axis.com>
Link: https://lore.kernel.org/r/20230123135913.2720991-3-astrid.rost@axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add function to return the jack type of snd_jack_types.
This allows a generic card driver to add a jack with the specified
type.
Signed-off-by: Astrid Rost <astrid.rost@axis.com>
Link: https://lore.kernel.org/r/20230123135913.2720991-2-astrid.rost@axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the list of pipelines in the PCM stream's pipeline info to trigger
the pipelines in the right order. Add a helper for triggering pipelines
in batch mode that will be used to trigger multiple pipelines at the
same time.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
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/20230127120031.10709-12-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Export the widget_in_list() function to be used by other modules.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
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/20230127120031.10709-3-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit fc7a6209d5 ("bus: Make remove callback return void")
forces bus_type::remove be void-returned, it doesn't make much sense
for any bus based driver implementing remove callbalk to return
non-void to its caller.
As such, change the remove function for ac97 based drivers to return
void.
Signed-off-by: Dawei Li <set_pte_at@outlook.com>
Link: https://lore.kernel.org/r/TYCP286MB2323A5AB1B2578EF4FA15DA7CAFB9@TYCP286MB2323.JPNP286.PROD.OUTLOOK.COM
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The s3c24xx SoC platform was completely removed, as were most of the
s3c64xx based board files, leaving only the DT based machines as well
as the MACH_WLF_CRAGG_6410 machine. All other board specific ASoC
driver can can now be recycled.
Reviewed-by: Krzysztof Kozlowski <krzysztof.kozlowski@linaro.org>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
When the new style KAE keep-alive implementation is used on compatible
Intel hardware, the clocks are maintained when codec is in D3. The
generic code in hda_cleanup_all_streams() can however interfere with
generation of audio samples in this mode, by setting the stream and
channel ids to zero.
To get full benefit of the keepalive, set the new
no_stream_clean_at_suspend quirk bit on affected Intel hardware. When
this bit is set, stream cleanup is skipped in hda_call_codec_suspend().
Special handling is needed for the case when system goes to suspend. The
stream id programming can be lost in this case. This will also cause
codec->cvt_setups to be out of sync. Handle this by implementing custom
suspend/resume handlers. If keep-alive is active for any converter, set
the quirk flags no_stream_clean_at_suspend and forced_resume. Upon
resume, keepalive programming is restored if needed.
Fixes: 15175a4f2b ("ALSA: hda/hdmi: add keep-alive support for ADL-P and DG2")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20221209101822.3893675-4-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more updates for v6.2 which can hopefully go into a later pull
request, the bulk of these are fixes, minor cleanups or new board quirks
- the one big bit that isn't is support for getting diagnostic data out
of the Intel AVS firmwares.
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Merge tag 'asoc-v6.2-2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v6.2
A few more updates for v6.2 which can hopefully go into a later pull
request, the bulk of these are fixes, minor cleanups or new board quirks
- the one big bit that isn't is support for getting diagnostic data out
of the Intel AVS firmwares.
This is a fairly sedate release for the core code, but there's been a
lot of driver work especially around the x86 platforms and device tree
updates:
- More cleanups of the DAPM code from Morimoto-san.
- Factoring out of mapping hw_params onto SoundWire configuration by
Charles Keepax.
- The ever ongoing overhauls of the Intel DSP code continue, including
support for loading libraries and probes with IPC4 on SOF.
- Support for more sample formats on JZ4740.
- Lots of device tree conversions and fixups.
- Support for Allwinner D1, a range of AMD and Intel systems, Mediatek
systems with multiple DMICs, Nuvoton NAU8318, NXP fsl_rpmsg and
i.MX93, Qualcomm AudioReach Enable, MFC and SAL, RealTek RT1318 and
Rockchip RK3588
There's more cross tree updates than usual, though all fairly minor:
- Some OMAP board file updates that were depedencies for removing their
providers in ASoC, as part of a wider effort removing the support for
the relevant OMAP platforms.
- A new I2C API required for updates to the new I2C probe API.
- A DRM update making use of a new API for fixing the capabilities
advertised via hdmi-codec.
Since this is being sent early I might send some more stuff if you've
not yet sent your pull request and there's more come in.
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Merge tag 'asoc-v6.2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v6.2
This is a fairly sedate release for the core code, but there's been a
lot of driver work especially around the x86 platforms and device tree
updates:
- More cleanups of the DAPM code from Morimoto-san.
- Factoring out of mapping hw_params onto SoundWire configuration by
Charles Keepax.
- The ever ongoing overhauls of the Intel DSP code continue, including
support for loading libraries and probes with IPC4 on SOF.
- Support for more sample formats on JZ4740.
- Lots of device tree conversions and fixups.
- Support for Allwinner D1, a range of AMD and Intel systems, Mediatek
systems with multiple DMICs, Nuvoton NAU8318, NXP fsl_rpmsg and
i.MX93, Qualcomm AudioReach Enable, MFC and SAL, RealTek RT1318 and
Rockchip RK3588
There's more cross tree updates than usual, though all fairly minor:
- Some OMAP board file updates that were depedencies for removing their
providers in ASoC, as part of a wider effort removing the support for
the relevant OMAP platforms.
- A new I2C API required for updates to the new I2C probe API.
- A DRM update making use of a new API for fixing the capabilities
advertised via hdmi-codec.
Since this is being sent early I might send some more stuff if you've
not yet sent your pull request and there's more come in.
Currently only PCM streams can enlist hdac_stream for their data
transfer. Add cstream field to hdac_ext_stream to expose possibility of
compress stream assignment in place of PCM one.
Limited to HOST-type only as there no other users on the horizon.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20221202152841.672536-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Mark Brown <broonie@kernel.org>:
The recently added pcm-test selftest has pointed out that systems with
the tda998x driver end up advertising that they support capture when in
reality as far as I can see the tda998x devices are transmit only. The
DAIs registered through hdmi-codec are bidirectional, meaning that for
I2S systems when combined with a typical bidrectional CPU DAI the
overall capability of the PCM is bidirectional. In most cases the I2S
links will clock OK but no useful audio will be returned which isn't so
bad but we should still not advertise the useless capability, and some
systems may notice problems for example due to pinmux management.
This is happening due to the hdmi-codec helpers not providing any
mechanism for indicating unidirectional audio so add one and use it in
the tda998x driver. It is likely other hdmi-codec users are also
affected but I don't have those systems to hand.
Mark Brown (2):
ASoC: hdmi-codec: Allow playback and capture to be disabled
drm: tda99x: Don't advertise non-existent capture support
drivers/gpu/drm/i2c/tda998x_drv.c | 2 ++
include/sound/hdmi-codec.h | 4 ++++
sound/soc/codecs/hdmi-codec.c | 30 +++++++++++++++++++++++++-----
3 files changed, 31 insertions(+), 5 deletions(-)
base-commit: f0c4d9fc9c
--
2.30.2
Currently the hdmi-codec driver always registers both playback and capture
capabilities but for most systems there's no actual capture capability,
usually HDMI is transmit only. Provide platform data which allows the users
to indicate what is supported so that we don't end up advertising things
to userspace that we can't actually support.
In order to avoid breaking existing users the flags in platform data are
a bit awkward and specify what should be disabled rather than doing the
perhaps more expected thing and defaulting to not supporting capture.
Reviewed-by: Russell King (Oracle) <rmk+kernel@armlinux.org.uk>
Link: https://lore.kernel.org/r/20221130184644.464820-2-broonie@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The vast majority of the current users of the SoundWire framework
have almost identical code for converting from hw_params to SoundWire
configuration. Whilst complex devices might require more, it is very
likely that most new devices will follow the same pattern. Save a
little code by factoring this out into a helper function.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20221123165432.594972-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Shifting signed 32-bit value by 31 bits is undefined, so changing
significant bit to unsigned. The UBSAN warning calltrace like below:
UBSAN: shift-out-of-bounds in sound/core/pcm_native.c:2676:21
left shift of 1 by 31 places cannot be represented in type 'int'
...
Call Trace:
<TASK>
dump_stack_lvl+0x8d/0xcf
ubsan_epilogue+0xa/0x44
__ubsan_handle_shift_out_of_bounds+0x1e7/0x208
snd_pcm_open_substream+0x9f0/0xa90
snd_pcm_oss_open.part.26+0x313/0x670
snd_pcm_oss_open+0x30/0x40
soundcore_open+0x18b/0x2e0
chrdev_open+0xe2/0x270
do_dentry_open+0x2f7/0x620
path_openat+0xd66/0xe70
do_filp_open+0xe3/0x170
do_sys_openat2+0x357/0x4a0
do_sys_open+0x87/0xd0
do_syscall_64+0x34/0x80
Fixes: 1da177e4c3 ("Linux-2.6.12-rc2")
Signed-off-by: Baisong Zhong <zhongbaisong@huawei.com>
Link: https://lore.kernel.org/r/20221121110044.3115686-1-zhongbaisong@huawei.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The addition of AMD_HS breaks Mediatek platforms by using an index
previously allocated to Mediatek. This is a backwards-compatibility
issue and needs to be fixed. All firmware released by AMD needs to be
re-generated and re-distributed.
Fixes: ed2562c64b ("ASoC: SOF: Adding amd HS functionality to the sof core")
Link: https://github.com/thesofproject/sof/issues/6615
Link: https://lore.kernel.org/alsa-devel/36a45c7a-820a-7675-d740-c0e83ae2c417@collabora.com/
Reported-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Reviewed-by: AngeloGioacchino Del Regno <angelogioacchino.delregno@collabora.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Basavaraj Hiregoudar <basavaraj.hiregoudar@amd.com>
Reviewed-by: V sujith kumar Reddy <Vsujithkumar.Reddy@amd.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20221117232120.112639-1-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A relatively large collection of fixes and new platform quirks here,
they're all fairly minor though - the widest possible impact is the fix
to the use of prefixes on regulator names which would have broken any
device that integrates regulators with DAPM and was used in a system
where it had a name prefix assigning to it.
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Merge tag 'asoc-fix-v6.2-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.1
A relatively large collection of fixes and new platform quirks here,
they're all fairly minor though - the widest possible impact is the fix
to the use of prefixes on regulator names which would have broken any
device that integrates regulators with DAPM and was used in a system
where it had a name prefix assigning to it.
There are some Dell SKUs that need to set the parameters of the
crossover filter (biquad). Each amplifier connects to one tweeter
speaker and one woofer speaker. We should control HPF/LPF to output the
proper frequency for the different speakers. If the codec driver got
the BQ parameters from the device property, it will apply these
parameters to the hardware.
No one is using asoc_simple_convert_fixup(), we don't need to
export its symbol. This patch removes it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/874jvj8ftp.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Originally in commit b2ebcf42a4 ("ASoC: SOF: free widgets in
sof_tear_down_pipelines() for static pipelines"), freeing of pipeline
components at suspend was only done with recent FW as there were known
limitations in older firmware versions.
Tests show that if static pipelines are used, i.e. all pipelines are
setup whenever firmware is powered up, the reverse action of freeing all
components at power down, leads to firmware failures with also SOF2.0
and SOF2.1 based firmware.
The problems can be specific to certain topologies with e.g. components
not prepared to be freed at suspend (as this did not happen with older
SOF kernels).
To avoid hitting these problems when kernel is upgraded and used with an
older firmware, bump the firmware requirement to SOF2.2 or newer. If an
older firmware is used, and pipeline is a static one, do not free the
components at suspend. This ensures the suspend flow remains backwards
compatible with older firmware versions. This limitation does not apply
if the product configuration is updated to dynamic pipelines.
The limitation is not linked to firmware ABI, as the interface to free
pipeline components has been available already before ABI3.19. The
problem is in the implementation, so firmware version should be used to
decide whether it is safe to use the newer flow or not. This patch adds
a new SOF_FW_VER() macro to compare SOF firmware release versions.
Link: https://github.com/thesofproject/sof/issues/6475
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: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20221101114913.1292671-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow for waiting for DRSM bit for specified stream to be cleared from
HDAudio library level. Drivers may utilize this optional step during the
stream resume procedure.
Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20221027124702.1761002-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Implement suspend/resume() operations for component drivers. For most
scenarios, the PM flow is similar to standard streaming one, except for
the part where the position register are being saved and the lack of PCM
pages freeing. To reduce code duplication, all avs_dai_suspend_XXX() and
avs_dai_resume_XXX() functions reuse their non-PM equivalents.
Given that path binding/unbinding happens only in FE part of the stream,
the order of suspend() goes:
1. hw_free() all FE DAIs, paths are unbound here
2. hw_free() all BE DAIs
Consequently, for resume() its:
1. hw_params() all BE DAIs
2. hw_params() all FE DAIs, paths are bound here
3. prepare() all BE DAIs
4. prepare() all FE DAIs
As component->suspend/resume() do not provide substream pointer, store
it ourselves so that the PM flow has all the necessary information to
proceed.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20221027124702.1761002-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Quite a few fixes here, a lot driver specific, plus some new quirks.
There was a bit of a mess with the runtime PM handling due to some
confusion in the API there which resulted in a number of commits and
reverts but that should all be stable now.
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Merge tag 'asoc-fix-v6.1-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v6.1
Quite a few fixes here, a lot driver specific, plus some new quirks.
There was a bit of a mess with the runtime PM handling due to some
confusion in the API there which resulted in a number of commits and
reverts but that should all be stable now.
Merge series from Peter Ujfalusi <peter.ujfalusi@linux.intel.com>:
In IPC4 all DSP loadable executable is a 'library' containing modules. The main
or basefw is also a library which contains multiple modules.
IPC4 allows to use loadable libraries to extend the functionality of the booted
basefw.
This series adds support for loading external libraries in case they are needed
by the loaded topology file.
The libraries must be placed to a specific firmware directory (fw_lib_prefix),
which is:
intel/avs-lib|sof-ipc4-lib/ followed by the platform name and in case of
community key use a 'community' directory.
For example for upx-i11 (community key): intel/avs-lib/tgl/community is the
default path.
The name of the library should be the UUID of the module it contains since the
library loading is going to look for the file as <module_UUID>.bin
In case there is a need to bundle multiple modules into single library, symlinks
can be used to point to the file:
module_boundle.bin
<UUID1>.bin -> module_boundle.bin
<UUID2>.bin -> module_boundle.bin
<UUID3>.bin -> module_boundle.bin
But note that in this case all modules will be loaded to the DSP since only the
whole library can be loaded, not individual modules.
On Intel HDA platforms the library loading is done via DMA and an IPC
message is also need to be sent to initiate the downloading of the new
library.
Co-developed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Chao Song <chao.song@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20221020121238.18339-16-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
IPC4 based firmware supports dynamically loaded external libraries.
The libraries will be not stored alongside of the firmware or tplg files.
For intel platforms the default path will be:
intel/avs-lib|sof-ipc4-lib/<platform>/ if a community key is used on the
given machine then the libraries will be under 'community' directory, like
it is done for the firmware itself.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Chao Song <chao.song@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20221020121238.18339-12-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SOF stack now uses the sdev->basefw to work with the SOF firmware, the
information from plat_data can be dropped.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Chao Song <chao.song@intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20221020121238.18339-7-peter.ujfalusi@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The SPIB and DRMS capabilities are orthogonal to the DSP enablement
and can be used whether the stream is coupled or not.
The existing code partitioning makes limited sense, the capabilities
are parsed at the sound/hda level but helpers are located in
sound/hda/ext.
This patch moves all the SPIB/DRMS functionality to the sound/hda
layer. This reduces the complexity of the sound/hda/ext layer which is
now limited to handling the multi-link extensions and stream
coupling/decoupling helpers.
Note that this is an iso-functionality code move and rename, the
HDaudio legacy driver would need additional changes to make use of
these capabilities.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20221019162115.185917-11-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All the helpers dealing with multi-link configurations are located in
the hdac_ext_controller.c, except the two set/clear routines that
modify the LOSIDV registers.
For consistency, move the two helpers and add the 'bus' prefix. One
could argue that the 'ml' prefix might be more relevant but that would
be a larger code change.
No functionality change, just move and rename.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20221019162115.185917-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should only use 'link' in the context of multi-link
configurations. Streams are configured from a different register space
and are not dependent on link except for LOSIDV settings.
Not functionality change, just pure rename.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20221019162115.185917-7-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No functionality change, just prefix addition to clearly identify that
the helper only applies to the 'ext' part for Intel platforms.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20221019162115.185917-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have two helpers with confusing names and different purposes.
Rename bus_get_link() and bus_get_link_at() as bus_get_hlink_by_name()
and bus_get_hlink_by_addr() respectively.
No functionality change
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20221019162115.185917-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge series from Siarhei Volkau <lis8215@gmail.com>:
The patchset fixes:
- Line In path stays powered off during capturing or
bypass to mixer.
- incorrectly represented dB values in alsamixer, et al.
- incorrect represented Capture input selector in alsamixer
in Playback tab.
- wrong control selected as Capture Master
The "convert-xxx" properties only have an effect for DPCM DAI links.
A DAI link is only created as DPCM if the device tree requires it;
part of this involves checking for the use of "convert-xxx" properties.
When the convert-sample-format property was added, the checks got out
of sync. A DAI link that specified only convert-sample-format but did
not pass any of the other DPCM checks would not go into DPCM mode and
the convert-sample-format property would be silently ignored.
Fix this by adding a function to do the "convert-xxx" property checks,
instead of open-coding it in simple-card and audio-graph-card. And add
"convert-sample-format" to the check function so that DAI links using
it will be initialized correctly.
Fixes: 047a05366f ("ASoC: simple-card-utils: Fixup DAI sample format")
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0@gmail.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20221019012302.633830-1-aidanmacdonald.0x0@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current soc-pcm.c is coping fe hw_param to dpcm->hw_param (A),
fixup it (B), and copy it to be (C).
int dpcm_be_dai_hw_params(...)
{
...
for_each_dpcm_be(fe, stream, dpcm) {
...
/* copy params for each dpcm */
(A) memcpy(&dpcm->hw_params, &fe->dpcm[stream].hw_params, ...) ;
/* perform any hw_params fixups */
(B) ret = snd_soc_link_be_hw_params_fixup(be, &dpcm->hw_params);
...
/* copy the fixed-up hw params for BE dai */
(C) memcpy(&be->dpcm[stream].hw_params, &dpcm->hw_params, ...);
...
}
...
}
But here, (1) it is coping hw_params without caring stream (Playback/Capture),
(2) we can get same value from be. We don't need to have dpcm->hw_params.
This patch removes it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/87v8ogsl6h.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
