snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A call of pci_iounmap() call without CONFIG_PCI leads to a build error
on some architectures. We tried to address this and add a check of
IS_ENABLED(CONFIG_PCI), but this still doesn't seem enough for sh.
Ideally we should fix it globally, it's really a corner case, so let's
paper over it with a simpler ifdef.
Fixes: 1e73359a24 ("ALSA: hda/ca0132 - make pci_iounmap() call conditional")
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the T+A VID to the generic check in order to enable
native DSD support for T+A devices. This works with the new T+A USB
DAC model SD3100HV and will also work with future devices which
support the XMOS/Thesycon style DSD format.
Signed-off-by: Udo Eberhardt <udo.eberhardt@thesycon.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make sure that all children entries are registered by a single call of
snd_info_register(). OTOH, don't register if a parent isn't
registered yet.
This allows us to create the whole procfs tree in a shot at the last
stage of card registration phase in a later patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we covered all callers with NULL device pointer, let's catch the
remaining calls with NULL and warn explicitly.
Acked-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should pass a proper non-NULL device object to memory allocators
although it was accepted in the past. The card->dev points to the
most appropriate device object in such a case, so let's put it.
Acked-by: Christoph Hellwig <hch@lst.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should pass a proper non-NULL device object to memory allocators
although it was accepted in the past. The card->dev points to the
most appropriate device object in such a case, so let's put it.
Acked-by: Christoph Hellwig <hch@lst.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This driver use the gpio consumer interface.
Add the header as it's needed.
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in a dev_warn message. Fix this.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should pass a proper non-NULL device object to memory allocators
although it was accepted in the past. The card->dev points to the
most appropriate device object in such a case, so let's put it.
Acked-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should pass a proper non-NULL device object to memory allocators
although it was accepted in the past. The card->dev points to the
most appropriate device object in such a case, so let's put it.
Acked-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
soc_tplg_link_config() will find the physical dai link and call
soc_tplg_dai_link_load() to load the BE dai link. Currently remove_link()
is only used to remove the FE dai link which is created by the topology.
The BE dai link cannot however be unloaded in snd_soc_tplg_component
_remove(), which is problematic if anything needs to be released or
reinitialized.
This patch aligns the definitions of dynamic types with the existing
UAPI and adds a new remove_backend_link() routine to unload the the BE
dai link when snd_soc_tplg_component_remove() is invoked.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Originally snd_soc_init_platform was not cleaning up its pointers, this
was fixed to always reallocate dynamic memory but created a memory leak
when snd_soc_init_platform was called multiple times during the same
probe attempt and also threw away any changes made to the struct between
calls. In order to avoid reallocating memory that is still valid, the
behaviour will be changed to clear the dynamically set pointers on a
probe error and a unregister event and snd_soc_init_platform will go
back to its original behaviour of only allocating null pointers so it will
stop throwing away valid changes.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The of_find_device_by_node() takes a reference to the underlying device
structure, we should release that reference.
Signed-off-by: Wen Yang <yellowriver2010@hotmil.com>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, in some complex cases, more than one widgets have same
name and registed from differnt dapm context, and route add from
another context too. When snd_soc_dapm_add_route, the previous
registered widget will overwritten by the latest same name widget,
will cause unexpect error. For Asoc framework we cant avoid this
situation and we cant decide which widget that wanted with route.
At least we can give users a notice.
Signed-off-by: Zhiwei Jiang <qq282012236@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently all widgets attached to a DAI link will be powered
up when the DAI is active, however this may include routes
that are not actually in use if there are unused channels
available on the DAI.
The macros for creating AIF widgets already include an entry for
slot, it is proposed to change that to channel. The effective
difference here being respresenting the logical channel index
rather than the physical slot index. The CODECs currently
using the slot entry on the DAPM_AIF macros are using it in
a manner consistent with this, the CODECs not using it just
have the field set to zero.
A variable is added to snd_soc_dapm_widget to represent
this channel index and then for each AIF widget attached to
a DAI this is compared against the number of channels on
the stream. Enabling the links for those which will be in
use. This has the nice property that the CODECs which haven't
used the slot/channel entry in the macro will function exactly
as before due to all the AIF widgets having a channel of zero
and a stream by definition having at least one channel.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rmmod/modprobe tests expose a kernel oops when accessing the dai
driver pointer. This comes from the topology design which operates in
multiple passes. Each object removal happens at a specific iteration,
and the code checks for the iteration (order) number after the memory
containing the order was freed.
Fix this be clearing a reference to the dai driver and check its
validity to avoid dereferences.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 7620fe9161 ("ASoC: topology: fix memory leak in
soc_tplg_dapm_widget_create") fixed a memory leak issue, but
additional tests and KASAN reports show a use-after-free in soc-dapm.
The widgets are created with a kmemdup operating on a template. The
"name" string is also duplicated, but the "sname" string is not. As a
result, when the template is freed after widget creation, its sname
string is still used.
Fix by explicitly duplicating the "sname" string, and freeing it when
required.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver function for transferring/receiving
BT encoded data to/from BT firmware.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set the channel number on each AIF widget to allow unused channels not
to be powered up across AIFs.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound card need to judge that whether it is using
"TDM Split mode". To judge it and for other purpose, it has
rsnd_parse_connect_simple() and rsnd_parse_connect_graph(),
but these are using different judgement policy for
TDM Split mode.
It is pointless and confusable.
This patch add new rsnd_parse_tdm_split_mode() and use common
judgement policy for simple-card/audio-graph.
Without this patch, CTU will be judged as TDM Split mode
on audio-graph card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver has below function to check connection
rsnd_parse_connect_simple()
rsnd_parse_connect_graph()
But these have different parameters. This patch synchronize these
for cleanup.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound device has many IPs and many situations.
If platform/board uses MIXer, situation will be more complex.
To avoid duplicate DVC kctrl registration when MIXer was used,
it had original flags.
But it was issue when sound card was re-binded, because
no one can't cleanup this flags then.
To solve this issue, commit 9c698e8481 ("ASoC: rsnd: tidyup
registering method for rsnd_kctrl_new()") checks registered
card->controls, because if card was re-binded, these were cleanuped
automatically. This patch could solve re-binding issue.
But, it start to avoid MIX kctrl.
To solve these issues, we need below.
To avoid card re-binding issue: check registered card->controls
To avoid duplicate DVC registration: check registered rsnd_kctrl_cfg
To allow multiple MIX registration: check registered rsnd_kctrl_cfg
This patch do it.
Fixes: 9c698e8481 ("ASoC: rsnd: tidyup registering method for rsnd_kctrl_new()")
Reported-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-By: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We used to pass NULL to memory allocators for ISA devices due to
historical reasons. But we prefer rather a proper device object to be
assigned, so let's fix it by replacing snd_dma_isa_data() call with
card->dev reference, and kill snd_dma_isa_data() definition.
Reviewed-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DMA API generally relies on a struct device to work properly, and
only barely works without one for legacy reasons. Pass the easily
available struct device from the platform_device to remedy this.
Also use GFP_KERNEL instead of GFP_USER as the gfp_t for the memory
allocation, as we should treat this allocation as a normal kernel one.
Signed-off-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DMA API generally relies on a struct device to work properly, and
only barely works without one for legacy reasons. Pass the easily
available struct device from the platform_device to remedy this.
Signed-off-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current code, the codec registration may happen both at the
codec bind time and the end of the controller probe time. In a rare
occasion, they race with each other, leading to Oops due to the still
uninitialized card device.
This patch introduces a simple flag to prevent the codec registration
at the codec bind time as long as the controller probe is going on.
The controller probe invokes snd_card_register() that does the whole
registration task, and we don't need to register each piece
beforehand.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the open-codes in many places with a new common helper for
performing the same thing: referring to the primary headphone pin.
This eventually fixes the potentially missing headphone pin on some
weird devices, too.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When auto_mute = no or spec->suppress_auto_mute = 1, cfg->hp_pins will
lose value.
Add this patch to find hp_pins value.
I add fixed for ALC282 ALC225 ALC256 ALC294 and alc_default_init()
alc_default_shutup().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only three fixes: a fix for Realtek HD-audio looks lengthy, but it's
just a code shuffling, and the actual changes are fairly small. The
rest are a PCM core fix for a long-standing bug that was recently
scratched by syzkaller, and a trivial USB-audio quirk for DSD
support.
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Merge tag 'sound-5.0-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Only three fixes.
The fix for Realtek HD-audio looks lengthy, but it's just a code
shuffling, and the actual changes are fairly small.
The rest are a PCM core fix for a long-standing bug that was recently
scratched by syzkaller, and a trivial USB-audio quirk for DSD support"
* tag 'sound-5.0-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fixed hp_pin no value
ALSA: pcm: Fix tight loop of OSS capture stream
ALSA: usb-audio: Add Opus #3 to quirks for native DSD support
On the System76 Darter Pro (darp5), there is a headset microphone
input attached to 0x1a that does not have a jack detect. In order to
get it working, the pin configuration needs to be set correctly, and
the ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied.
This is similar to the MIC_NO_PRESENCE fixups for some Dell laptops,
except we have a separate microphone jack that is already configured
correctly.
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declaration of snd_pcm_drop() in sound/core/pcm_native.c is superfluous
since the function isn't called before being defined. Remove the
declaration.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Spreadtrum DMA engine uses the link-list mode to support audio playback
or capture, thus this patch adds audio DMA platform support for CPU DAI to
trigger DMA link-list transfer.
Signed-off-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
struct snd_soc_dapm_route has been modified to be a dynamic
object so that it can be used to save driver specific
data while parsing topology and clean up
driver-specific data during driver unloading.
This patch makes the following changes to accomplish the above:
1. Set the dobj member of snd_soc_dapm_route during the
SOC_TPLG_PASS_GRAPH pass of topology parsing.
2. Add the remove_route() routine that will be called while
removing all dynamic objects from the component driver.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
template.sname and template.name are only freed when an error occur.
They should be freed in the success return case, too.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
dtexts is two dimensional array, so we also need to free it after
freeing its fields.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when we unload and reload machine driver few times we end with
corrupted list and try to cleanup no longer existing objects. Fix this
by removing dobj from the list.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We already have passed dobj, there is no reason to access it through
containing structs.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the registration and free of beep input device was done
manually from the register and the disconnect callbacks of the
assigned codec object. This seems working in most cases, but this may
be a cause of some races at probe. Moreover, due to these manual
calls, the total code became unnecessarily lengthy.
This patch rewrites the beep registration code to follow the standard
sound device object style. This allows us reducing the code, in
addition to avoiding the nested device registration calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The init sequence for ALC294 headphone stuff is needed not only for
the boot up time but also for the resume from hibernation, where the
device is switched from the boot kernel without sound driver to the
suspended image. Since we record the PM event in the device
power_state field, we can now recognize the call pattern and apply the
sequence conditionally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we deal with single codec and suspend codec callbacks for
all S3, S4 and runtime PM handling. But it turned out that we want
distinguish the call patterns sometimes, e.g. for applying some init
sequence only at probing and restoring from hibernate.
This patch slightly modifies the common PM callbacks for HD-audio
codec and stores the currently processed PM event in power_state of
the codec's device.power field, which is currently unused. The codec
callback can take a look at this event value and judges which purpose
it's being called.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For some reason we test if the machine is passed as a parameter before
fixing up the codec name. This is unnecessary, generates false
positives in static analysis tools and done only in this machine
driver, remove and adjust indentation.
Reported-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix hp_pin always no value.
[More notes on the changes:
The hp_pin value that is referred in alc294_hp_init() is always zero
at the moment the function gets called, hence this is actually
useless as in the current code.
And, this kind of init sequence should be called from the codec init
callback, instead of the parser function. So, the first fix in this
patch to move the call call into its own init_hook.
OTOH, this function is needed to be called only once after the boot,
and it'd take too long for invoking at each resume (where the init
callback gets called). So we add a new flag and invoke this only
once as an additional fix.
The one case is still not covered, though: S4 resume. But this
change itself won't lead to any regression in that regard, so we
leave S4 issue as is for now and fix it later. -- tiwai ]
Fixes: bde1a74596 ("ALSA: hda/realtek - Fixed headphone issue for ALC700")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a codec driver to control ChromeOS EC codec.
Use EC Host command to enable/disable I2S recording and control other
configurations.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Reviewed-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
XMOS/Thesycon family of USB Audio Class firmware flags DSD altsetting
separate from the PCM ones. Thus the DSD altsetting can be auto-detected
based on the flag and doesn't need maintaining specific altsetting
whitelist.
In addition, static VID:PID-to-altsetting whitelisting causes problems
when firmware update changes the altsetting, or same VID:PID is reused
for another device that has different kind of firmware.
This patch removes existing explicit whitelist mappings for XMOS VID
(0x20b1) and Thesycon VID (0x152a).
Also corrects placement of Hegel HD12 and NuPrime DAC-10 to keep list
sorted based on VID.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds audio routing for both playback and capture.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds required dapm widgets for playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds basic controls found in wcd9335 codec.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
CLASS-H controller/Amplifier is common accorss Qualcomm WCD codec series.
This patchset adds basic CLASS-H controller apis for WCD codecs after
wcd9335 to use.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC,
It supports both I2S/I2C and SLIMbus audio interfaces.
On slimbus interface it supports two data lanes; 16 Tx ports
and 8 Rx ports. It has Seven DACs and nine dedicated interpolators,
Seven (six audio ADCs, and one VBAT ADC), Multibutton headset
control (MBHC), Active noise cancellation and Sidetone paths
and processing.
This patchset adds very basic support for playback and capture
via the 9 interpolators and ADC respectively.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper to override dailink platform name, if passed as parameter
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Duende Classic was produced by Solid State Logic in 2006, as a
first model of Duende DSP series. The following model, Duende Mini
was produced in 2008. They are designed to receive isochronous
packets for PCM frames via IEEE 1394 bus, perform signal processing by
downloaded program, then transfer isochronous packets for converted
PCM frames.
These two models includes the same embedded board, consists of several
ICs below:
- Texus Instruments Inc, TSB41AB3 for physical layer of IEEE 1394 bus
- WaveFront semiconductor, DICE II STD ASIC for link/protocol layer
- Altera MAX 3000A CPLD for programs
- Analog devices, SHARC ADSP-21363 for signal processing (4 chips)
This commit adds support for the two models to ALSA dice driver. Like
support for the other devices, packet streaming is just available.
Userspace applications should be developed if full features became
available; e.g. program uploader and parameter controller.
$ ./hinawa-config-rom-printer /dev/fw1
{ 'bus-info': { 'adj': False,
'bmc': False,
'chip_ID': 349771402425,
'cmc': True,
'cyc_clk_acc': 255,
'generation': 1,
'imc': True,
'isc': True,
'link_spd': 2,
'max_ROM': 1,
'max_rec': 512,
'name': '1394',
'node_vendor_ID': 20674,
'pmc': False},
'root-directory': [ ['VENDOR', 20674],
['DESCRIPTOR', 'Solid State Logic'],
['MODEL', 112],
['DESCRIPTOR', 'Duende board'],
[ 'NODE_CAPABILITIES',
{ 'addressing': {'64': True, 'fix': True, 'prv': True},
'misc': {'int': False, 'ms': False, 'spt': True},
'state': { 'atn': False,
'ded': False,
'drq': True,
'elo': False,
'init': False,
'lst': True,
'off': False},
'testing': {'bas': False, 'ext': False}}],
[ 'UNIT',
[ ['SPECIFIER_ID', 20674],
['VERSION', 1],
['MODEL', 112],
['DESCRIPTOR', 'Duende board']]]]}
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In function rt5651_i2c_probe(), local variable "ret" could
be uninitialized if function regmap_read() returns -EINVAL.
However, this value is used in if statement. This is
potentially unsafe.
Signed-off-by: Yizhuo <yzhai003@ucr.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rationale behind the current calculation is somewhat obscure [1]
and can yield slightly wrong dividers in certain cases, which the
machine drivers for some boards (like the HiFiBerry DAC+ Pro)
seemingly try to circumvent, by updating the rate fraction so as to
suit this calculation.
The updated calculation should correctly yield the smallest bit clock
rate that would fit the frame.
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2019-January/144219.html
Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some boards, such as the HiFiBerry DAC+ Pro, use a pair of external
oscillators, to generate 44.1 or 48kHz multiples and are forced to
resort to hacks [1] in order to support 24-bit data without ending up
with fractional dividers. This patch allows the machine driver to use
32-bit frames for 24-bit data to avoid such issues.
Although the datasheet (p. 15) seems to suggest that only a handful
of ratios are supported, it's not very explicit about it, so we allow
the full range of values supported by the underlying register in the
callback, to avoid needlessly rejecting potentially usable
configurations.
[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143442.html
Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can use for_each_link_codecs() without waiting
for_each_rtd_codec_dai() on soc_bind_dai_link().
Let's use for_each macro.
Fixes: 50acc7e49 ("ASoC: core: Fix multi-CODEC setups")
Fixes: 10dff9b0d ("ASoC: soc-core: use for_each_link_codecs() for dai_link codecs")
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the trigger=off is passed for a PCM OSS stream, it sets the
start_threshold of the given substream to the boundary size, so that
it won't be automatically started. This can be problematic for a
capture stream, unfortunately, as detected by syzkaller. The scenario
is like the following:
- In __snd_pcm_lib_xfer() that is invoked from snd_pcm_oss_read()
loop, we have a check whether the stream was already started or the
stream can be auto-started.
- The function at this check returns 0 with trigger=off since we
explicitly disable the auto-start.
- The loop continues and repeats calling __snd_pcm_lib_xfer() tightly,
which may lead to an RCU stall.
This patch fixes the bug by simply allowing the wait for non-started
stream in the case of OSS capture. For native usages, it's supposed
to be done by the caller side (which is user-space), hence it returns
zero like before.
(In theory, __snd_pcm_lib_xfer() could wait even for the native API
usage cases, too; but I'd like to stay in a safer side for not
breaking the existing stuff for now.)
Reported-by: syzbot+fbe0496f92a0ce7b786c@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds quirk VID/PID IDs for the Opus #3 DAP (made by 'The Bit')
in order to enable Native DSD support.
[ NOTE: this could be handled in the generic way with fp->dvd_raw if
we add 0x10cb to the vendor whitelist, but since 0x10cb shows a
different vendor name (Erantech), put to the individual entry at
this time -- tiwai ]
Signed-off-by: Olek Poplavsky <woodenbits@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For devices implemented as a MFD it is common to only have a single node
in devicetree representing the whole device. As such when looking up
components in soc_find_components we should match against both the devices
of_node and the devices parent's of_node, as is already done in the rest
of the ASoC core.
This causes regressions for some DAI links at the moment as
soc_find_component was recently added as a check in soc_init_dai_link.
Fixes: 8780cf1142 ("ASoC: soc-core: defer card probe until all component is added to list")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
An open-coded error path in __snd_pcm_lib_xfer() can be replaced with
the simple goto to the common error path. This also makes the error
handling more consistent, i.e. when some samples have been already
processed, return that size instead of the error code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The > should be >= or otherwise we potentially read one element beyond
the end of the ff->tx_midi_substreams[] array.
Fixes: 73f5537fb2 ("ALSA: fireface: support tx MIDI functionality of Fireface UCX")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't populate the const arrays on the stack but instead make
it static. Makes the object code smaller, for example:
Before:
text data bss dec hex filename
14107 8832 224 23163 5a7b bytcht_es8316.o
After:
text data bss dec hex filename
14015 8896 224 23135 5a5f bytcht_es8316.o
(gcc version 8.2.0 x86_64)
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A significant amount of fixes at this time, mostly for covering the
recent ASoC issues.
- Fixes for the missing ASoC driver initialization with non-deferred
probes; these triggered other problems in chain, which resulted in
yet more fix commits
- DaVinci runtime PM fix; the diff looks large but it's just a code
shuffling
- Various fixes for ASoC Intel drivers: a regression in HD-A HDMI,
Kconfig dependency, machine driver adjustments, PLL fix.
- Other ASoC driver-specific stuff including the trivial fixes
caught by static analysis
- Usual HD-audio quirks
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Merge tag 'sound-5.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A significant amount of fixes at this time, mostly for covering the
recent ASoC issues.
- Fixes for the missing ASoC driver initialization with non-deferred
probes; these triggered other problems in chain, which resulted in
yet more fix commits
- DaVinci runtime PM fix; the diff looks large but it's just a code
shuffling
- Various fixes for ASoC Intel drivers: a regression in HD-A HDMI,
Kconfig dependency, machine driver adjustments, PLL fix.
- Other ASoC driver-specific stuff including the trivial fixes caught
by static analysis
- Usual HD-audio quirks"
* tag 'sound-5.0-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (30 commits)
ALSA: hda - Add mute LED support for HP ProBook 470 G5
ASoC: amd: Fix potential NULL pointer dereference
ASoC: imx-audmux: change snprintf to scnprintf for possible overflow
ASoC: rt5514-spi: Fix potential NULL pointer dereference
ASoC: dapm: change snprintf to scnprintf for possible overflow
ASoC: rt5682: Fix PLL source register definitions
ASoC: core: Don't defer probe on optional, NULL components
ASoC: core: Make snd_soc_find_component() more robust
ASoC: soc-core: fix init platform memory handling
ASoC: intel: skl: Fix display power regression
ALSA: hda/realtek - Fix typo for ALC225 model
ASoC: soc-core: Hold client_mutex around soc_init_dai_link()
ASoC: Intel: Boards: move the codec PLL configuration to _init
ASoC: soc-core: defer card probe until all component is added to list
ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages
ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode
ASoC: ti: davinci-mcasp: Move context save/restore to runtime_pm callbacks
ASoC: Variable "val" in function rt274_i2c_probe() could be uninitialized
ASoC: rt5682: Fix recording no sound issue
ASoC: Intel: atom: Make PCI dependency explicit
...
This patch changes the parent pointer assignment of snd_info_entry
object to be always non-NULL. More specifically,check the parent
argument in snd_info_create_module_entry() & co, and assign
snd_proc_root if NULL is passed there.
This assures that the proc object is always freed when the root is
freed, so avoid possible memory leaks. For example, some error paths
(e.g. snd_info_register() error at snd_minor_info_init()) may leave
snd_info_entry object although the proc file itself is freed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The proc files are recursively freed by calling with the root
snd_info_entry object, so we don't have to keep each object for
releasing one by one. Move the release of the PCM stream proc root at
the beginning, so that we can remove the redundant code and resource.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In Fireface series, registration of higher 4 bytes of destination
address for asynchronous transaction of MIDI messages is done by
a write transaction to model-specific register.
On the other hand, registration of lower 4 bytes of the address is
selectable from 4 options. A register for this registration includes
the other purpose options such as input attenuation. Thus this
driver expects userspace applications to configure the register.
Actual behaviour for the asynchronous transaction is different
depending on protocols. In former protocol, destination offset
of each transaction is the same as the registered address even if
it is block request. In latter models, destination offset of each
transaction is the offset of previous transaction plus 4 byte
and the transaction is quadlet request.
This commit cleanups comments about the above mechanism.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the previous code refactoring, the PCM stream locking code
became nothing but the PCM group lock with self_group object. Use the
existing helper function for simplifying the code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the hackish down_write_nonfifo() that was introduced as a
workaround of rwsem deadlock.
It used to be a problem for non-atomic PCM streams that take the rwsem
for the locking and hit the high lock contention. Since the current
PCM locking refactoring, we'll no longer hit it as the hot code-paths
don't take global locks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have currently two global locks, a rwlock and a rwsem, that are
used for managing linking the PCM streams. Due to these global locks,
once when a linked stream is used, the lock granularity suffers a
lot.
This patch attempts to eliminate the former global lock for atomic
ops. The latter rwsem needs remaining because of the loosy way of the
loop calls in snd_pcm_action_nonatomic(), as well as for avoiding the
deadlock at linking. However, these are used far rarely, actually
only by two actions (prepare and reset), where both are no timing
critical ones. So this can be still seen as a good improvement.
The basic strategy to eliminate the rwlock is to assure group->lock at
adding or removing a stream to / from the group. Since we already
takes the group lock whenever taking the all substream locks under the
group, this shouldn't be a big problem. The reference to group
pointer in snd_pcm_substream object is protected by the stream lock
itself.
However, there are still pitfalls: a race window at re-locking and the
lifecycle of group object. The former is a small race window for
dereferencing the substream group object opened while snd_pcm_action()
performs re-locking to avoid ABBA deadlocks. This includes the unlink
of group during that window, too. And the latter is the kfree
performed after all streams are removed from the group while it's
still dereferenced.
For addressing these corner cases, two new tricks are introduced:
- After re-locking, the group assigned to the stream is checked again;
if the group is changed, we retry the whole procedure.
- Introduce a refcount to snd_pcm_group object, so that it's freed
only when it's empty and really no one refers to it.
(Some readers might wonder why not RCU for the latter. RCU in this
case would cost more than refcounting, unfortunately. We take the
group lock sooner or later, hence the performance improvement by RCU
would be negligible. Meanwhile, because we need to deal with
schedulable context depending on the pcm->nonatomic flag, it'll become
dynamic RCU/SRCU switch, and the grace period may become too long.)
Along with these changes, there are a significant amount of code
refactoring. The complex group re-lock & ref code is factored out to
snd_pcm_stream_group_ref() function, for example.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert 10dff9b0d (ASoC: soc-core: use for_each_link_codecs() for
dai_link codecs) for now as Sylwester Nawrocki reports that it causes
oopses on at least Odroid boards.
Reported-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In latter model of Fireface series, asynchronous transaction includes
a prefix byte to indicate the way to decode included MIDI bytes.
Upper 4 bits of the prefix byte indicates port number, and the rest 4
bits indicate the way to decode rest of bytes for MIDI messages.
Basically the rest bits indicates the number of bytes for MIDI message.
However, if the last byte of each MIDi message is included, the rest
bits are 0xf. For example:
message: f0 00 00 66 14 20 00 00 f7
offset: content (big endian, port 0)
'0030: 0x02f00000
'0030: 0x03006614
'0030: 0x03200000
'0030: 0x0ff70000
This commit supports encoding scheme for the above and allows
applications to transfer MIDI messages via ALSA rawmidi interface.
An unused member (running_status) is reused to keep state of
transmission of system exclusive messages.
For your information, this is a dump of config rom.
$ sudo ./hinawa-config-rom-printer /dev/fw1
{ 'bus-info': { 'bmc': False,
'chip_ID': 13225063715,
'cmc': False,
'cyc_clk_acc': 0,
'imc': False,
'isc': True,
'max_rec': 512,
'name': '1394',
'node_vendor_ID': 2613},
'root-directory': [ [ 'NODE_CAPABILITIES',
{ 'addressing': {'64': True, 'fix': True, 'prv': False},
'misc': {'int': False, 'ms': False, 'spt': True},
'state': { 'atn': False,
'ded': False,
'drq': True,
'elo': False,
'init': False,
'lst': True,
'off': False},
'testing': {'bas': False, 'ext': False}}],
['VENDOR', 2613],
['DESCRIPTOR', 'RME!'],
['EUI_64', 2873037108442403],
[ 'UNIT',
[ ['SPECIFIER_ID', 2613],
['VERSION', 4],
['MODEL', 1054720],
['DESCRIPTOR', 'Fireface UCX']]]]}
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Between former and latter models, content of asynchronous transaction
for MIDI messages from driver to device is different.
This commit is a preparation to support latter models. A protocol-specific
operation is added to encode MIDI messages to the transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Between former and latter models, destination address to receive
asynchronous transactions for MIDI messages is different.
This commit adds model-dependent parameter for the addresses.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface UCX transfers asynchronous transactions for MIDI messages.
One transaction includes quadlet data therefore it can transfer 3
message bytes as maximum. Base address of the destination is
configured by two settings; a register for higher 8 byte of the
address, and a bitflag to option register indicates lower 8byte.
The register for higher address is 0x'ffff'0000'0034. Unfortunately,
firmware v24 includes a bug to ignore registered value for the
destination address and transfers to 0x0001xxxxxxxx always. This
driver doesn't work well if the bug exists, therefore users should
install the latest firmware (v27).
The bitflag is a part of value to be written to option register
(0x'ffff'0000'0014).
lower addr: bitflag (little endian)
'0000'0000: 0x00002000
'0000'0080: 0x00004000
'0000'0100: 0x00008000
'0000'0180: 0x00010000
This register includes more options but they are not relevant to
packet streaming or MIDI functionality. This driver don't touch it.
Furthermore, the transaction is sent to address offset incremented
by 4 byte to the offset in previous time. When it reaches base address
plus 0x7c, next offset is the base address.
Content of the transaction includes a prefix byte. Upper 4 bits of
the byte indicates port number, and the rest 4 bits indicate the way
to decode rest of bytes for MIDI message.
Except for system exclusive messages, the rest bits are the same as
status bits of the message without channel bits. For system exclusive
messages, the rest bits are encoded according to included message bytes.
For example:
message: f0 7e 7f 09 01 f7
offset: content (little endian, port 0)
'0000: 0x04f07e7f
'0004: 0x070901f7
message: f0 00 00 66 14 20 00 00 00 f7
offset: content (little endian, port 1)
'0014: 0x14f00000
'0018: 0x14661420
'001c: 0x14000000
'0020: 0x15f70000
message: f0 00 00 66 14 20 00 00 f7
offset: content (little endian, port 0)
'0078: 0x04f00000
'007c: 0x04661420
'0000: 0x070000f7
This commit supports decoding scheme for the above and allows
applications to receive MIDI messages via ALSA rawmidi interface.
The lower 8 bytes of destination address is fixed to 0x'0000'0000,
thus this driver expects userspace applications to configure option
register with bitflag 0x00002000 in advance.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In Fireface series, drivers can register destination address for
asynchronous transaction which transfers MIDI messages from device.
In former models, all of the transactions arrive at the registered
address without any offset. In latter models, each of the transaction
arrives at the registered address with sequential offset within 0x00
to 0x7f. This seems to be for discontinuity detection.
This commit adds model-dependent member for the address range.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a series of Fireface, devices transfer asynchronous transaction with
MIDI messages. In the transaction, content is different depending on
models. ALSA fireface driver has protocol-dependent handler to pick up
MIDI messages from the content.
In latter models of the series, the transaction is transferred to range
of address sequentially. This seems to check continuity of transferred
messages.
This commit changes prototype of the handler to receive offset of
address for received transactions.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD/DA ASRC function control two ASRC clock sources separately.
Whether AD/DA filter select which clock source, we enable AD/DA ASRC
function for all cases.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AZX_DCAPS_PM_RUNTIME flag is added to indicate support for runtime PM.
azx_has_pm_runtime() is used to check if above is enabled and thus
forbid runtime PM calls if needed.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch moves clock enable/disable from system resume/suspend to
runtime resume/suspend respectively. Along with this hda controller
chip init or stop is also moved. System resume/suspend can invoke
runtime callbacks and do necessary setup.
chip->running can be used to check for probe completion and device
access during runtime_resume or runtime_suspend can be avoided if
probe is not yet finished. This helps to avoid kernel panic during
boot where runtime PM callbacks can happen from system PM.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Explicit clock enable is not required during probe, as this would be
managed by runtime PM calls. Clock can be enabled/disabled in runtime
resume/suspend. This way it is easier to balance clock enable/disable
counts.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moved devm_clk_get() API calls to a separate function and the same
can be called early in the probe. This is done before runtime PM
for the device is enabled. The runtime resume/suspend callbacks can
later enable/disable clocks respectively(the support would be added
in subsequent patches). Clock handles should be available by the
time runtime suspend/resume calls can happen.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables runtime power management(runtime PM) support for
hda. pm_runtime_enable() and pm_runtime_disable() are added during
device probe and remove respectively. The runtime PM callbacks will
be forbidden if hda controller does not have support for runtime PM.
pm_runtime_get_sync() and pm_runtime_put() are added for hda register
access. The callbacks for above will be added in subsequent patches.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current simple-card is using asoc_simple_card_canonicalize_dailink().
Its naming is "dailink", but is for "platform".
We already have asoc_simple_card_canonicalize_cpu() for "cpu",
let's follow same naming rule.
It never return error, so, void function is better idea.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can use for_each_link_codecs() without waiting
for_each_rtd_codec_dai() on soc_bind_dai_link().
Let's use for_each macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to cleanup component when soc_probe_component() was
failed, or when soc_remove_component() was called.
But they are cleanuping component on each way.
(And soc_probe_component() doesn't call snd_soc_dapm_free(),
but it should).
Same code in many places makes code un-understandable.
This patch adds new soc_cleanup_component() and call it from
snd_probe_component() and snd_remove_component().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Deep nested codec is not readable.
Let's reduce if/else nest.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to cleanup card resources when snd_soc_instantiate_card() was
failed, or when snd_soc_unbind_card() was called.
But they are cleanuping card resources on each way.
Same code in many places makes code un-understandable.
This patch reuses soc_cleanup_card_resources() for cleanuping code
resource. Then, it makes avoiding cleanup order.
It will be called from snd_soc_instantiate_card() and
snd_soc_unbind_card().
Then, original soc_cleanup_card_resources() included
snd_soc_flush_all_delayed_work(), but it is now separated.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
soc-core is calling flush_delayed_work() many times for same purpose.
Same code in many places makes code un-understandable.
This patch adds new snd_soc_flush_all_delayed_work() for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_dai_link is starting to use snd_soc_dai_link_component
(= modern) style for Platform, but it is still assuming single Platform
so far. We will need to have multi Platform support in the not far
future.
Currently only simple card is using it as sound card driver,
and other drivers are converted to it from legacy style by
snd_soc_init_platform().
To avoid future problem of multi Platform support, let's add
num_platforms before it is too late.
In the same time, to make it same naming mothed, "platform" should
be "platforms". This patch fixup it too.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_pcm_group_for_each_entry() loop found in snd_pcm_unlink() is
only for taking the first list entry. Use list_first_entry() to make
clearer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make a common helper to re-assign the PCM link using list_move() instead
of open code with manual list_del() and list_add_tail(). This assures
the consistency and we can get rid of snd_pcm_group.count field -- its
purpose is only to check whether the list is singular, and we can know
it by list_is_singular() call now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are multiple open codes that initialize the same object.
Create a common helper function instead.
Also, use kzalloc() to be safer at creating a group object, and move
the initialization out of the critical section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_card_unref() call in snd_pcm_link() looks suspicious through a
quick glance, but it's a correct usage; this is needed just because
the file descriptor check in is_pcm_file() calls the helper
snd_lookup_minor_data() that keeps the card refcount.
Despite of the correctness, the code still looks confusing.
Basically, keeping the card ref for the whole code isn't needed
as fdget() blocks the release of the opened file. Hence it's more
understandable if snd_card_unref() is moved into is_pcm_file(), then
the caller doesn't have to take care after the call.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support speaker and mic mute LEDs on HP ProBook 470 G5.
BugLink: https://bugs.launchpad.net/bugs/1811254
Signed-off-by: Anthony Wong <anthony.wong@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface UFX was shipped by RME GmbH in 2012. This model supports later
protocol for management of isochronous communication and synchronization
of sampling transmission frequency.
This commit adds support for the model. At present, it's not clear how
to encode MIDI messages and decide destination address for asynchronous
transaction, thus this commit adds support for isochronous communication
for PCM frames only.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A procedure to retrieve clock configuration is used by two callers.
Each of caller has duplicated code to parse bits.
This commit adds refactoring to remove the duplicated code.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds refactoring for dump of sync status by adding
tables for check bits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a member for a callback function to get clock status
to former protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a member for a callback function to switch frame
fetching mode to former protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a member for a callback function to dump status and
move existing code to former protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a series of Fireface, latter protocol has no way for drivers to
retrieve current clock configuration. On the other hand, this driver
has proc node for it.
This commit removes a proc node to dump both clock configuration
and synchronization status in one proc node.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit moves codes for Fireface 400 to a file of former protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In a series of Fireface, later model supports different protocol
from former models.
This commit is a preparation to support both of protocols.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The unused variable was forgotten to be removed and now we get a
compiler warning:
sound/pci/hda/hda_codec.c: In function 'hda_codec_runtime_suspend':
sound/pci/hda/hda_codec.c:2926:18: warning: unused variable 'pcm'
Fixes: 17bc4815de ("ALSA: pci: Remove superfluous snd_pcm_suspend*() calls")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current ALSA SoC is assuming 1 CPU 1 Platform (= DMA) style system.
Because of this background, it is directly using
xxx_name / xxx_of_node / xxx_dai_name on dai_link.
Let's call it as legacy style here.
More complex style system like multi CPU multi Platform (= DMA) will
coming. To supporting it, we can use snd_soc_dai_link_component on
dai_link. Let's call it as modern style here.
But current ALSA SoC can't support it so far. Thus, we need to have
multi CPU / multi Codec / multi Platform style in the future on ALSA SoC.
Currently we already have multi Codec support. Platform is starting to
use modern style on dai_link, but still style only. Multi Platform is
not yet implemented. And we still don't have multi CPU support on ALSA
SoC, and not have modern style either.
Currently, if driver is using legacy style Codec/Platform, it will be
converted to modern style on soc-core. This means, we are using glue code
for legacy vs modern style so far on ALSA SoC.
We can fully switch to modern style on all drivers if ALSA SoC supported
modern style for CPU, and then, legacy style code will be removed from
ALSA SoC.
Untile then, we need to keep both legacy/modern style and its glue code.
This patch adds such future plan and background on soc.h
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
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Merge tag 'asoc-fix-v5.0-rc2' into asoc-5.1
ASoC: Fixes for v5.0
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
When stopping audio, ASoC will first stop DMA then CPU DAI.
Sometimes there is a delay between DMA stop and CPU DAI stop, which
triggers an underrun error. Now, because of the delay introduced
by dev_err another underrun error will occur causing a vicious circle
making impossible to stop CPU DAI.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Deferred probes shouldn't cause error messages in the boot log, so
change the dev_err() to the more harmless dev_info().
Signed-off-by: Stefan Agner <stefan@agner.ch>
Signed-off-by: Mark Brown <broonie@kernel.org>
Probe deferral is to be expected during normal operation, so avoid
printing an error when it is encountered.
Removing the goto would not be strictly necessary. However, if
code gets added later, the cleanup in the EPROBE_DEFER case likely
would get missed.
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Probe deferral is to be expected during normal operation, so avoid
printing an error when it is encountered.
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Probe deferral is to be expected during normal operation, so avoid
printing an error when it is encountered.
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Not finding the codec/SSI instance can be due to probe deferral.
Do not print error messages in those cases.
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make sure to properly put the of node in case finding the codec
fails.
Fixes: 81e8e49261 ("ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000")
Signed-off-by: Stefan Agner <stefan@agner.ch>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull the PCM suspend improvement / cleanup.
This moves the most of snd_pcm_suspend*() calls into PCM's own device
PM ops. There should be no change from the functionality POV.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hdmi-codec oopses the kernel when it is unbound from a successfully
bound audio subsystem, and is then rebound:
Unable to handle kernel NULL pointer dereference at virtual address 0000001c
pgd = ee3f0000
[0000001c] *pgd=3cc59831
Internal error: Oops: 817 [#1] PREEMPT ARM
Modules linked in: ext2 snd_soc_spdif_tx vmeta dove_thermal snd_soc_kirkwood ofpart marvell_cesa m25p80 orion_wdt mtd spi_nor des_generic gpio_ir_recv snd_soc_kirkwood_spdif bmm_dmabuf auth_rpcgss nfsd autofs4 etnaviv thermal_sys hwmon gpu_sched tda9950
CPU: 0 PID: 1005 Comm: bash Not tainted 4.20.0+ #1762
Hardware name: Marvell Dove (Cubox)
PC is at hdmi_dai_probe+0x68/0x80
LR is at find_held_lock+0x20/0x94
pc : [<c04c7de0>] lr : [<c0063bf4>] psr: 600f0013
sp : ee15bd28 ip : eebd8b1c fp : c093b488
r10: ee048000 r9 : eebdab18 r8 : ee048600
r7 : 00000001 r6 : 00000000 r5 : 00000000 r4 : ee82c100
r3 : 00000006 r2 : 00000001 r1 : c067e38c r0 : ee82c100
Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none[ 297.318599] Control: 10c5387d Table: 2e3f0019 DAC: 00000051
Process bash (pid: 1005, stack limit = 0xee15a248)
...
[<c04c7de0>] (hdmi_dai_probe) from [<c04b7060>] (soc_probe_dai.part.9+0x34/0x70)
[<c04b7060>] (soc_probe_dai.part.9) from [<c04b81a8>] (snd_soc_instantiate_card+0x734/0xc9c)
[<c04b81a8>] (snd_soc_instantiate_card) from [<c04b8b6c>] (snd_soc_add_component+0x29c/0x378)
[<c04b8b6c>] (snd_soc_add_component) from [<c04b8c8c>] (snd_soc_register_component+0x44/0x54)
[<c04b8c8c>] (snd_soc_register_component) from [<c04c64b4>] (devm_snd_soc_register_component+0x48/0x84)
[<c04c64b4>] (devm_snd_soc_register_component) from [<c04c7be8>] (hdmi_codec_probe+0x150/0x260)
[<c04c7be8>] (hdmi_codec_probe) from [<c0373124>] (platform_drv_probe+0x48/0x98)
This happens because hdmi_dai_probe() attempts to access the HDMI
codec private data, but this has not been assigned by hdmi_dai_probe()
before it calls devm_snd_soc_register_component(). Move the call to
dev_set_drvdata() before devm_snd_soc_register_component() to avoid
this oops.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
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Merge tag 'asoc-fix-v5.0-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
Add a default pdata which can fit most HW design. So we don't need to
add a lot of DMI checking in this driver.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Check return value from call to devm_kzalloc() in order to prevent a
potential NULL pointer dereference.
Also, notice that it makes no sense to allocate any resources if
res = platform_get_resource(pdev, IORESOURCE_MEM, 0); fails,
so move the call to devm_kzalloc() below the mentioned code.
Lastly, improve the use of sizeof in the call to devm_kzalloc() by
changing it from sizeof(struct i2s_dev_data) to sizeof(*adata)
This issue was detected with the help of Coccinelle.
Fixes: ac289c7ec0 ("ASoC: amd: add ACP3x PCM platform driver")
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change snprintf to scnprintf. There are generally two cases where using
snprintf causes problems.
1) Uses of size += snprintf(buf, SIZE - size, fmt, ...)
In this case, if snprintf would have written more characters than what the
buffer size (SIZE) is, then size will end up larger than SIZE. In later
uses of snprintf, SIZE - size will result in a negative number, leading
to problems. Note that size might already be too large by using
size = snprintf before the code reaches a case of size += snprintf.
2) If size is ultimately used as a length parameter for a copy back to user
space, then it will potentially allow for a buffer overflow and information
disclosure when size is greater than SIZE. When the size is used to index
the buffer directly, we can have memory corruption. This also means when
size = snprintf... is used, it may also cause problems since size may become
large. Copying to userspace is mitigated by the HARDENED_USERCOPY kernel
configuration.
The solution to these issues is to use scnprintf which returns the number of
characters actually written to the buffer, so the size variable will never
exceed SIZE.
Signed-off-by: Silvio Cesare <silvio.cesare@gmail.com>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <fabio.estevam@nxp.com>
Cc: Dan Carpenter <dan.carpenter@oracle.com>
Cc: Kees Cook <keescook@chromium.org>
Cc: Will Deacon <will.deacon@arm.com>
Cc: Greg KH <greg@kroah.com>
Signed-off-by: Willy Tarreau <w@1wt.eu>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a potential NULL pointer dereference in case devm_kzalloc()
fails and returns NULL.
Fix this by adding a NULL check on rt5514_dsp.
This issue was detected with the help of Coccinelle.
Fixes: 6eebf35b0e ("ASoC: rt5514: add rt5514 SPI driver")
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change snprintf to scnprintf. There are generally two cases where using
snprintf causes problems.
1) Uses of size += snprintf(buf, SIZE - size, fmt, ...)
In this case, if snprintf would have written more characters than what the
buffer size (SIZE) is, then size will end up larger than SIZE. In later
uses of snprintf, SIZE - size will result in a negative number, leading
to problems. Note that size might already be too large by using
size = snprintf before the code reaches a case of size += snprintf.
2) If size is ultimately used as a length parameter for a copy back to user
space, then it will potentially allow for a buffer overflow and information
disclosure when size is greater than SIZE. When the size is used to index
the buffer directly, we can have memory corruption. This also means when
size = snprintf... is used, it may also cause problems since size may become
large. Copying to userspace is mitigated by the HARDENED_USERCOPY kernel
configuration.
The solution to these issues is to use scnprintf which returns the number of
characters actually written to the buffer, so the size variable will never
exceed SIZE.
Signed-off-by: Silvio Cesare <silvio.cesare@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Dan Carpenter <dan.carpenter@oracle.com>
Cc: Kees Cook <keescook@chromium.org>
Cc: Will Deacon <will.deacon@arm.com>
Cc: Greg KH <greg@kroah.com>
Signed-off-by: Willy Tarreau <w@1wt.eu>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix typo which causes headphone no sound while using BCLK
as PLL source.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
cpu and platform are optional components in DAI links. For example
codec-codec links usually have no platform set.
Call snd_soc_find_component only if the name or of_node of
a cpu or platform is set. Otherwise it will return NULL and
soc_init_dai_link bails out immediately with -EPROBE_DEFER,
meaning registering a card with NULL cpu or platform in DAI links
can never succeed.
Fixes: 8780cf1142 ("ASoC: soc-core: defer card probe until all component is added to list")
Signed-off-by: Matthias Reichl <hias@horus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_suspend() is no longer called from outside, so let's make it
local static. Also drop a superfluous NULL check there.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of snd_pcm_suspend_all() & co became superfluous since we
call it in the PCM PM ops. Let's remove them.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATIIXP driver supports the full PCM resume and saves/restores the
running PCM pointer. This used to be done in the suspend and resume
callbacks together with snd_pcm_suspend() call. But since we moved
the snd_pcm_supsend*() call in PCM device PM ops, this should be moved
to a more appropriate place, i.e. the trigger callback.
Along with the movement of the PCM suspend/resume code, remove the
superfluous snd_pcm_suspend_all() call, too.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Until now we rely on each driver calling snd_pcm_suspend*() explicitly
at its own PM handling. However, this can be done far more easily by
setting the PM ops to each actual snd_pcm device object.
This patch adds the device_type object for PCM stream and assigns to
each PCM stream object. The type contains only the PM ops for system
suspend; we don't need to deal with the resume in general.
The suspend hook simply calls snd_pcm_suspend_all() for the given PCM
streams. This implies that the PM order is correctly put, i.e. PCM is
suspended before the main (or codec) driver, which should be true in
general. If a special ordering is needed, you'd need to adjust the
device PM order manually later.
This patch introduces a new flag, snd_pcm.no_device_suspend, too.
With this flag set, the PCM device object won't invoke
snd_pcm_suspend_all() by itself. This is needed for ASoC who wants to
manage the PM call orders in its serialized way, and the flag is set
in soc_new_pcm() as default.
For the non-ASoC world, we can get rid of the manual snd_pcm_suspend
calls. This will be done in the later patches.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are some use cases where you're checking for a lot of things on a
card and it makes sense that you might end up trying to call
snd_soc_find_component() without either a name or an of_node. Currently
in that case we try to dereference the name and crash but it's more
useful to allow the caller to just treat that as a case where we don't
find anything, that error handling will already exist.
Inspired by a patch from Ajit Pandey fixing some callers.
Fixes: 8780cf1142 ("ASoC: soc-core: defer card probe until all component is added to list")
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_init_platform initializes pointers to snd_soc_dai_link which is
statically allocated and it does this by devm_kzalloc. In the event of
an EPROBE_DEFER the memory will be freed and the pointers are left
dangling. snd_soc_init_platform sees the dangling pointers and assumes
they are pointing to initialized memory and does not reallocate them on
the second probe attempt which results in a use after free bug since
devm has freed the memory from the first probe attempt.
Since the intention for snd_soc_dai_link->platform is that it can be set
statically by the machine driver we need to respect the pointer in the
event we did not set it but still catch dangling pointers. The solution
is to add a flag to track whether the pointer was dynamically allocated
or not.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
AES channel status carries various audio parameters. If channel status is
detected, current patch extracts sample rate and bit depth parameters of
the incoming stream during capture.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix below build error:
ERROR: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-msm8916-digital.ko] undefined!
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes unused variables which also fixes below warnings:
msm8916-wcd-digital.c:245:30: warning: 'rx2_mix2_inp1_chain_enum'
defined but not used [-Wunused-const-variable=]
static const struct soc_enum rx2_mix2_inp1_chain_enum = SOC_ENUM_SINGLE(
^~~~~~~~~~~~~~~~~~~~~~~~
msm8916-wcd-digital.c:234:30: warning: 'rx_mix2_inp1_chain_enum'
defined but not used [-Wunused-const-variable=]
static const struct soc_enum rx_mix2_inp1_chain_enum = SOC_ENUM_SINGLE(
^~~~~~~~~~~~~~~~~~~~~~~
msm8916-wcd-digital.c:224:26: warning: 'adc2_mux_text'
defined but not used [-Wunused-const-variable=]
static const char *const adc2_mux_text[] = { "ZERO", "INP2", "INP3" };
^~~~~~~~~~~~~
msm8916-wcd-digital.c:223:26: warning: 'rx_mix2_text'
defined but not used [-Wunused-const-variable=]
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver cs4341 can be built with SPI and/or I2C, but it has to be one
of them at least. When I2C is set as a module we see the warning below:
sound/soc/codecs/cs4341.c:213:12: warning: ‘cs4341_probe’
defined but not used [-Wunused-function]
static int cs4341_probe(struct device *dev)
^~~~~~~~~~~~
Rework so that we use IS_ENABLED instead of defined. Also change so
SND_SOC_CS4341 depends on SND_SOC_I2C_AND_SPI to we dont' get a link
error when SND_SOC_CS4341=y, I2C=m and REGMAP_I2C=m is set.
Suggested-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Anders Roxell <anders.roxell@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Program codec stripe through AC_VERB_SET_STRIPE_CONTROL to use multiple
sdo lines if supported. Audio needs to be striped across number of sdo
lines for simultaneous playbacks of higher resolutions to work.
This needs to be implemented only for an Audio Output Converter and only
if the stripe bit(AC_WCAP_STRIPE) of Audio Widget Capabilities parameter
is 1.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Platforms having multiple SORs and hdmi/dp sinks require higher
bandwidth to support simultaneous playbacks of higher resolution.
If hda controller supports multiple SDO lines, STRIPE can be used
to indicate how many of the SDO lines the stream should be striped
across.
During stream start stripe control bits are programmed to use given
number of sdo lines and the same is cleared during stream stop.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Controllers and codecs can support striping of audio out across
multiple SDO lines. The number of supported SDO lines can be
specific to chip. GCAP register can be read to know the maximum
supported SDO lines.
snd_hdac_get_stream_stripe_ctl() is exposed to program stripe bits
on controller and codec side.
stripe value: 0 for 1SDO, 1 for 2SDO, 2 for 4SDO lines, etc.,
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Mohan Kumar D <mkumard@nvidia.com>
Reviewed-by: Ravindra Lokhande <rlokhande@nvidia.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the refactoring of HD-audio display power management, the
display power status is managed per domain. Meanwhile the ASoC
hdac_hdmi driver still keeps and relies (incorrectly) on the
refcounting together with ASoC skl driver, and this leads to the
display state always on.
This patch is an attempt to address the regression by simplifying the
PM code of ASoC skl and hdac_hdmi drivers. Basically, since the
refactoring, we don't have to manage the display power at HD-audio
controller suspend / resume but only at HD-audio HDMI codec suspend /
resume. So the patch drops the superfluous snd_hdac_display_power()
calls in skl driver.
Meanwhile, in hdac_hdmi side, we rewrite the PM call just to re-use
the runtime PM callbacks like other drivers do. Now the logic is
simple: turn off at suspend and turn on at resume.
The patch also fixes the possibly missing display-power off at skl
driver removal as well as some error paths at probe.
Fixes: 029d92c289 ("ALSA: hda: Refactor display power management")
Reported-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix typo for model alc255-dell1 to alc225-dell1.
Enable headset mode support for new WYSE NB platform.
Fixes: a26d96c780 ("ALSA: hda/realtek - Comprehensive model list for ALC259 & co")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've always had a weird situation around dma_zalloc_coherent. To
safely support mapping the allocations to userspace major architectures
like x86 and arm have always zeroed allocations from dma_alloc_coherent,
but a couple other architectures were missing that zeroing either always
or in corner cases. Then later we grew anothe dma_zalloc_coherent
interface to explicitly request zeroing, but that just added __GFP_ZERO
to the allocation flags, which for some allocators that didn't end
up using the page allocator ended up being a no-op and still not
zeroing the allocations.
So for this merge window I fixed up all remaining architectures to zero
the memory in dma_alloc_coherent, and made dma_zalloc_coherent a no-op
wrapper around dma_alloc_coherent, which fixes all of the above issues.
dma_zalloc_coherent is now pointless and can go away, and Luis helped
me writing a cocchinelle script and patch series to kill it, which I
think we should apply now just after -rc1 to finally settle these
issue.
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Merge tag 'remove-dma_zalloc_coherent-5.0' of git://git.infradead.org/users/hch/dma-mapping
Pull dma_zalloc_coherent() removal from Christoph Hellwig:
"We've always had a weird situation around dma_zalloc_coherent. To
safely support mapping the allocations to userspace major
architectures like x86 and arm have always zeroed allocations from
dma_alloc_coherent, but a couple other architectures were missing that
zeroing either always or in corner cases.
Then later we grew anothe dma_zalloc_coherent interface to explicitly
request zeroing, but that just added __GFP_ZERO to the allocation
flags, which for some allocators that didn't end up using the page
allocator ended up being a no-op and still not zeroing the
allocations.
So for this merge window I fixed up all remaining architectures to
zero the memory in dma_alloc_coherent, and made dma_zalloc_coherent a
no-op wrapper around dma_alloc_coherent, which fixes all of the above
issues.
dma_zalloc_coherent is now pointless and can go away, and Luis helped
me writing a cocchinelle script and patch series to kill it, which I
think we should apply now just after -rc1 to finally settle these
issue"
* tag 'remove-dma_zalloc_coherent-5.0' of git://git.infradead.org/users/hch/dma-mapping:
dma-mapping: remove dma_zalloc_coherent()
cross-tree: phase out dma_zalloc_coherent() on headers
cross-tree: phase out dma_zalloc_coherent()
soc_init_dai_link() calls soc_find_component() which needs
to be within client_mutex lock. Add client_mutex lock around
soc_init_dai_link() in snd_soc_register_card() to avoid
lockdep warning.
Fixes: 8780cf1142 ("ASoC: soc-core: defer card probe until all component is added to list")
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_byt_cht_es8316_mc_remove() use the platform drvdata as a type
of 'struct byt_cht_es8316_private', but snd_byt_cht_es8316_mc_probe()
set it to 'struct snd_soc_card', as suggested by Dan Carpenter, fix
the usage in snd_byt_cht_es8316_mc_remove().
Fixes: 0d3e91da07 ("ASoC: Intel: bytcht_es8316: Add external speaker mux support")
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
move the codec PLL to rt5682_codec_init, because codec only need to config the clock source/PLL once.
As the result, remove the platform_clock_controls since no need to control clock anymore.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_DAPM_MICBIAS is deprecated, replace it with SND_SOC_DAPM_SUPPLY.
MICBIAS voltage wasn't supplied to the microphone with the older
SND_SOC_DAPM_MICBIAS widget, hence the microphone wouldn't work.
This patch fixes the problem.
Signed-off-by: b-ak <anur.bhargav@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
By making MCLK parent of DAI clocks, when querying the rate of the
clock the rate returned is now given from the parent clock so
gives the MCLK rate rather than 0 as previously returned. This is
a bit misleading, and actually there's no major reason why we can't
at least return the DAI WCLK rate, as set in HW, so that's what we
now do.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For platforms using the Common Clock Framework to control the
codec's DAI clocks, MCLK should be enabled prior to DAI clocks
being turned on. For some platforms the codec is already
provided with an MCLK reference and can therefore control MCLK
itself as it needs to.
To improve functionality MCLK is now added as a parent to the
DAI clocks, if MCLK was provided, so that if they are enabled MCLK
will automatically be enabled as a prerequisite by the CCF.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAI component probe is not called if it is not present
in component list during sound card registration.
Check if component is available in component list for
platform and cpu dai before soundcard registration.
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/usb/mixer.c: In function 'parse_audio_feature_unit':
sound/usb/mixer.c:1838:28: warning:
variable 'first_ch_bits' set but not used [-Wunused-but-set-variable]
It never used since 2.6
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To enable SIE(Stream Interrupt Enable) in snd_hdac_stream_start(), we
should set both mask and value to be "1 << azx_dev->index" for register
update, the mask was 0, here fix it.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
E.g. for azx_int_enable(), we should set both mask and value to be
"AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN"(the mask was 0) to enable
controller CIE and GIE.
We have similar issues on setting AZX_GCTL_RESET and AZX_GCTL_UNSOL,
here try to correct all of them.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable Headset Mic VREF for headset mode of ALC225.
This will be controlled by coef bits of headset mode functions.
[ Fixed a compile warning and code simplification -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Forgot to add unplug function to unplug state of headset mode
for ALC225.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fix typo after a recent commit causing headphones to have no sound
Fixes: ad43d528a7 (ALSA: usb-audio: Define registers for CM6206)
Signed-off-by: Amadeusz Sławiński <amade@asmblr.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In function rt274_jack_detect(), local variable "buf" could
be uninitialized if function regmap_read() returns -EINVAL.
However, it will be used to calculate "hp" and "mic" and
make their value unpredictable while those value are used
in the caller. This is potentially unsafe.
Signed-off-by: Yizhuo <yzhai003@ucr.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
On capture through some of dmic we observe a glitch at the
start of record. This is because we start capturing even before
dmic is ready to send out data.
The optional delay will be applied after enabling the mic.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We already need to zero out memory for dma_alloc_coherent(), as such
using dma_zalloc_coherent() is superflous. Phase it out.
This change was generated with the following Coccinelle SmPL patch:
@ replace_dma_zalloc_coherent @
expression dev, size, data, handle, flags;
@@
-dma_zalloc_coherent(dev, size, handle, flags)
+dma_alloc_coherent(dev, size, handle, flags)
Suggested-by: Christoph Hellwig <hch@lst.de>
Signed-off-by: Luis Chamberlain <mcgrof@kernel.org>
[hch: re-ran the script on the latest tree]
Signed-off-by: Christoph Hellwig <hch@lst.de>
The "chip->dsp_spos_instance" can be NULL on some of the ealier error
paths in snd_cs46xx_create().
Reported-by: "Yavuz, Tuba" <tuba@ece.ufl.edu>
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a DMI quirk for the Point of View TAB-P1006W-232 (v1.0) tablet, this
tablet is special in a number of ways:
1) It uses the 2nd GPIO resource in the ACPI tables for jack-detect rather
then using the rt5651 codec's builtin jack-detect functionality
2) It uses the 3th GPIO resource in the ACPI tables to control the
external amplifier rather then the usual first non GpioInt resource and
the GPIO is active-low.
3) It is a BYTCR device, without a CHAN package and it uses SSP0-AIF1
rather then the default SSP0-AIF2.
4) Its internal mic is a digital mic (the first x86 rt5651 device that
I'm aware of which does this), combined with having its headset-mic
connected to IN2.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some board designs hook the jack-detect up to an external GPIO, rather
then to one of the codec pins, add support for this.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add quirks module parameter to allow manually specifying quirks
from the kernel commandline (or modprobe.conf).
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 37c7401e8c ("ASoC: Intel: bytcr_rt5651: Fix DMIC map
headsetmic mapping"), changed the headsetmic mapping from IN3P to IN2P,
this was based on the observation that all bytcr_rt5651 devices I have
access to (7 devices) where all using IN3P for the headsetmic. This was
an attempt to unifify / simplify the mapping, but it was wrong.
None of those devices was actually using a digital internal mic. Now I've
access to a Point of View TAB-P1006W-232 (v1.0) tabler, which does use a
DMIC and it does have its headsetmic connected to IN2P, showing that the
original mapping was correct, so this commit reverts the change changing
the mapping back to IN2P.
Fixes: 37c7401e8c ("ASoC: Intel: bytcr_rt5651: Fix DMIC map ... mapping")
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some board designs hook the jack-detect up to an external GPIO,
rather then to one of the codec pins, add support for this.
Figuring out which GPIO to use is pretty much board specific so I've
chosen to let the machine driver pass the gpio_desc as data argument to
snd_soc_component_set_jack() rather then add support for getting the
GPIO to the codec driver. This keeps the codec code nice and clean.
Note that using an external GPIO for this conflicts with button-press
support, so this commit disables button-press support when an
external GPIO is used.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some BYT platforms have a RT5651 codec while using an ACPI node with
a HID of 10EC5640 to describe the coded. Add the 10EC5640 HID to the
acpi_device_id list, so that the rt5651 will bind to the codec on these
devices.
Like the rt5645 and rt5670 drivers which also have the 10EC5640 ACPI HID
in their acpi_device_id list for similar reasons, the rt5651 driver checks
the codecs device-id register so that it will only bind if the codec
actually is a rt5651 and it will ignore actual rt5640 codecs.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Point of View TAB-P1006W-232 (v1.0) tablet uses 10EC5640 as
ACPI HID, but it has a rt5651 codec add a quirk for this.
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current BSDSR/BSDISR are using temporary/generic settings, but it can't
handle all SRCx/SoC. It needs to handle correctry.
Otherwise, sampling rate converted sound channel will be broken if it
was TDM. One note is that it needs to overwrite settings on E3 case.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: chaoliang qin <chaoliang.qin.jg@renesas.com>
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch introduces "sclk-strength" property to allow SCLK pad drive
strength to be changed via device tree.
When running playback test on LS1028ARDB, Tx Frame sync error interrupt
will occur sometimes. Some noises also exist. After changing SCLK pad
drive strength to the maximum value, the issues are gone.
Signed-off-by: Alison Wang <alison.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_lib_malloc_pages() may fail, so let's check its status and
return its error code upstream.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Added SPDIF driver build related changes.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Added SPDIF audio driver. This provides playback and capture of
AES audio over SPDIF interface.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During the bootup of the kernel, the DAPM bias level is in the OFF
state. As soon as the DAPM framework kicks in it pushes the codec
into STANDBY state.
The probe function doesn't prepare the clock, and STANDBY state
does a clk_disable_unprepare() without checking the previous state.
This leads to an OOPS.
Not transitioning from an OFF state to the STANDBY state fixes the
problem.
Signed-off-by: b-ak <anur.bhargav@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Add Digital Audio Interface driver that convers PDM bitstream to PCM
format.
Features:
- Fixed filtering characteristics for audio application.
- Full or partial set of channels operation with individual enable control.
- Programmable PDM clock generator.
- Programmable decimation rate.
- 16-bit signed output result.
- Overall stopband attenuation more than 80dB.
- Overall passband ripple less than 0.2dB.
Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila@nxp.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit f84a6273dd ("ASoC: pxa: remove raumfeld machine driver")
removed the Raumfeld ASoC machine driver but forgot to kill one line
in the Makefile.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt298.c:992:6-8: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:995:6-9: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:317:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:320:5-8: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:348:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt298.c:351:5-8: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The buf in rl6347a_hw_read is __be32.
Cc: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The eq parameters binary is stored in __be. However, it is unsigned short
in rt5645_eq_param_s{} which will cause incorrect type assignment. So add
struct rt5645_eq_param_s_be16{} to store the eq binary and convert it to
unsigned short in rt5645->eq_param.
Cc: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Sparse:
da7219.c:841:57: warning: dubious: x & !y
Cc: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Sparse.
da7219.c:440:44: warning: cast to restricted __le16
da7219.c:461:13: warning: incorrect type in assignment (different base types)
da7219.c:461:13: expected unsigned short [unsigned] [usertype] val
da7219.c:461:13: got restricted __le16 [usertype] <noident>
da7219.c:1451:16: warning: incorrect type in assignment (different base types)
da7219.c:1451:16: expected unsigned short [unsigned] [usertype] offset
da7219.c:1451:16: got restricted __le16 [usertype] <noident>
da7219-aad.c:150:37: warning: incorrect type in assignment (different base types)
da7219-aad.c:150:37: expected unsigned short [unsigned] [usertype] tonegen_freq_hptest
da7219-aad.c:150:37: got restricted __le16 [usertype] <noident>
da7219-aad.c:157:37: warning: incorrect type in assignment (different base types)
da7219-aad.c:157:37: expected unsigned short [unsigned] [usertype] tonegen_freq_hptest
da7219-aad.c:157:37: got restricted __le16 [usertype] <noident>
Cc: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
mt6351.c:1418:5-8: Unneeded variable: "ret". Return "0" on line 1437
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/tscs42xx.c:392:5-31: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
nau8824.c:810:6-12: ERROR: Assignment of bool to non-0/1 constant
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt5651.c:750:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5651.c:754:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5651.c:2192:1-16: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/max98927.c:508:2-20: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98927.c:889:3-28: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98927.c:891:3-28: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98927.c:893:2-27: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt5640.c:980:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5640.c:984:2-17: WARNING: Assignment of bool to 0/1
sound/soc/codecs/rt5640.c:2825:1-16: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt286.c:927:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt286.c:930:5-8: WARNING: Comparison to bool
sound/soc/codecs/rt286.c:299:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt286.c:302:5-8: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/rt274.c:958:6-8: WARNING: Comparison to bool
sound/soc/codecs/rt274.c:961:6-9: WARNING: Comparison to bool
sound/soc/codecs/rt274.c:384:5-7: WARNING: Comparison to bool
sound/soc/codecs/rt274.c:387:5-8: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/cs4271.c:226:2-16: WARNING: Assignment of bool to 0/1
sound/soc/codecs/cs4271.c:229:2-16: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Reported by Coccinelle:
sound/soc/codecs/max98373.c:411:2-20: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98373.c:922:2-27: WARNING: Assignment of bool to 0/1
sound/soc/codecs/max98373.c:924:2-27: WARNING: Assignment of bool to 0/1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some mux/mixer are not used. Remove them from the driver.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5645_if3_adc_in_mux, rt5645_inr_mux, and rt5645_inl_mux are not used.
Remove them from the driver.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Missing or spurious parameter descriptions. Fix warnings with W=1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix warnings with W=1
If these variables are useful this driver should be modified to expose
them.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix warnings with W=1
If these variables are useful then this driver should be modified to
expose them.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No reason why this is global, fix warnings with W=1
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AK4497 is a 32-bit 2ch DAC and has the same register
map as AK4458 with few exceptions:
* AK4497 has one more register at the end of register space
DFS_READ which is a read only register that allows users
to read FS Auto Detection mode. We currently do not use
this register so we use the same regmap structure as for ak4458.
* Because AK4458 is an 8ch DAC there are some fields that are
only used by AK4458 and marked as reserved for AK4497, so for
this reason we need to have a distinct set of controls, widgets
and routes.
Datasheet for AK4497 is at:
https://www.akm.com/akm/en/file/ev-board-manual/AK4497EQ.pdf
Datasheet for AK4458 is at:
https://www.akm.com/akm/en/file/datasheet/AK4458VN.pdf
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_ctl_add() could fail, so let's check its return value and return its
error code upstream upon failure.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix checks if snd_card_register() fails, and if so logs the error
via dev_err() consistent with other patches.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_i2c_sendbytes could fail. The fix checks its return value: if it
fails, issues an error message and returns with its error code.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ctl_add() could fail, so let's check its status and issue an error
message if it indeed fails.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's no reason for us to do that while we initialize dac_mute to
1. Also oxygen_init() has been clearing the OXYGEN_SPDIF_OUT_ENABLE
bit anyway.
Signed-off-by: Tom Yan <tom.ty89@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell has new platform for ALC274.
This will support to enable headset mode.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In `create_composite_quirk`, the terminating condition of for loops is
`quirk->ifnum < 0`. So any composite quirks should end with `struct
snd_usb_audio_quirk` object with ifnum < 0.
for (quirk = quirk_comp->data; quirk->ifnum >= 0; ++quirk) {
.....
}
the data field of Bower's & Wilkins PX headphones usb device device quirks
do not end with {.ifnum = -1}, wihch may result in out-of-bound read.
This Patch fix the bug by adding an ending quirk object.
Fixes: 240a8af929 ("ALSA: usb-audio: Add a quirck for B&W PX headphones")
Signed-off-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places where we access the data without checking the
actual object size from the USB audio descriptor. This may result in
OOB access, as recently reported.
This patch addresses these missing checks. Most of added codes are
simple bLength checks in the caller side. For the input and output
terminal parsers, we put the length check in the parser functions.
For the input terminal, a new argument is added to distinguish between
UAC1 and the rest, as they treat different objects.
Reported-by: Mathias Payer <mathias.payer@nebelwelt.net>
Reported-by: Hui Peng <benquike@163.com>
Tested-by: Hui Peng <benquike@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've had some sanity checks of the mixer unit descriptors but they
are too loose and some corner cases are overlooked. Add more strict
checks in uac_mixer_unit_get_channels() for avoiding possible OOB
accesses by malformed descriptors.
This also changes the semantics of uac_mixer_unit_get_channels()
slightly. Now it returns zero for the cases where the descriptor
lacks of bmControls instead of -EINVAL. Then the caller side skips
the mixer creation for such unit while it keeps parsing it.
This corresponds to the case like Maya44.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The parser for the processing unit reads bNrInPins field before the
bLength sanity check, which may lead to an out-of-bound access when a
malformed descriptor is given. Fix it by assignment after the bLength
check.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All McASP pin can be configured as GPIO.
Add gpiochip support for McASP and only enable it when the
gpio-controller is present in the DT node.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
McASP can loose it's context when runtime_pm is disabled.
Save and restore the context when suspending and resuming the device.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the platform drivers are selected by the DAI drivers (including
McASP) there is no longer a need to check whether the modules are built-in
or module.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add quirks to select the correct input map, jack-detect options
and channel map to make sound work on the ASUS MeMO Pad 7 (ME176C).
Note: Although sound works out of the box, jack detection currently
requires overriding the ACPI DSDT table. This is necessary because
the rt5640 ACPI device (10EC5640) has the wrong GPIO listed as
interrupt (one of the Bluetooth GPIOs).
The correct GPIO is GPO2 0x0004 (listed as the first GPIO in the
Intel(R) Audio Machine Driver - AMCR0F28 device).
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some devices detected as BYT-T by the PMIC-type based detection
have only a single IRQ listed in the 80860F28 ACPI device. This
causes -ENXIO later when attempting to get the IRQ at index 5.
It turns out these devices behave more like BYT-CR devices,
and using the IRQ at index 0 makes sound work correctly.
This patch adds a fallback for these devices to is_byt_cr():
If there is no IRQ resource at index 5, treating the device
as BYT-T is guaranteed to fail later, so we can safely treat
these devices as BYT-CR without breaking any working device.
Link: http://mailman.alsa-project.org/pipermail/alsa-devel/2018-December/143176.html
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
is_byt_cr() and its usage can be simplified by returning the bool
directly, instead of through a pointer. This works because the
return value is just treated as bytcr = false and is not used
otherwise.
This patch also removes the extra check of
IS_ENABLED(CONFIG_IOSF_MBI) in favor of checking
iosf_mbi_available() directly. The header already takes care
of returning false if the config option is not enabled.
No functional change.
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some BYTCR devices use an ES8316 codec, add an ACPI match table entry
for this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Depending on the input-map and on if 1 or 2 speakers are connected,
userspace needs to use a different UCM profile.
Since we already deal with quirks in the kernel driver and set the
input-map from the kernel, add a quirk for devices with a single / mono
speaker and set the card's long_name based on the input and speaker
quirks, so that userspace can use the long_name to pick the right UCM
profile.
This change, including how the long_name is build-up mirrors how we do
this in the bytcr_rt5640 and bytcr_rt5651 machine drivers.
Note since all devices I have access to use a mono speaker setup I've
chosen to default the speaker setting to mono.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After adding jack-detect support we have 3 microphone input switches:
"Microphone 1", "Microphone 2" and "Headset Mic". But the ES8316 has only
2 microphone inputs.
In the app-note explaining how to use the codec and on the 3 boards I
have one input is used for an internal microphone and one for the headset
microphone. On the 2 CHT boards I have the internal mic is on on MIC1 and
the headset mic is on MIC2, on the BYTCR board I have it is the other way
around.
This commit replaces the 2 "Microphone 1" and "Microphone 2" input switches
with a single "Internal Mic" switch and adds support for selecting either
possible input mapping.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ES8316 only has a single (amplified) output. The ES8316 appnote showing
the intended usage uses a jack-receptacle which physically disconnects the
speakers from the output when a jack is plugged in.
But all 3 devices using the es8316 which I have (2 Cherry Trail devices and
one Bay Trail CR device), use an analog mux to disconnect the speakers,
driven by a GPIO.
This commit adds support for this, modelling this as a separate speaker
widget / dapm pin-switch which sets the mux to drive the speakers when
selected.
The intend is for userspace to use the recently added jack-detect support
and then automatically select either the Headphone or Speaker output based
on that.
Note this commit includes a workaround for an ACPI table bug which is
present on 2 of the 3 devices I have, see the added comment in the code.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Hookup the jack-detect support added to the codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for having the codec connected to SSP0 instead of SSP2. This
is controlled through a new quirk parameter, similar to how this is done
in the bytcr_rt5640 and bytcr_rt5651 machine drivers.
Bay Trail CR (cost reduced) SoCs do not have an SSP2, so we default to SSP0
there.
Note the SPP0 quirk gets BIT(16) because bits 0-15 are reserved for non
boolean quirks like the input-map added in a later commit in this series.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some minor refactoring:
1) Group the code setting the card dev and prive pointers together with
registering the card
2) Properly put the comment about registering the card at the place where
we actually register the card and add a new comment for getting the clk
3) Add a struct device *dev helper variable (this will be used more in
follow up commits)
4) Reword error message to have the same "foo failed: %d" wording as others
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For lack of a better (non-random) way of sorting includes more and more
files in the kernel are moving over to sorting the includes alphabetically.
Move the bytcht_es8316 driver over to this sorting before we add a
bunch of more includes.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Export the DAC functionality to mix left + right together and then output
the same (mixed) signal on both outputs.
Various (x86) tablets with an ES8316 codec use a single speaker
connected between the headhpone LOUT and ROUT pins, expecting the output
to be in a mono differential mode. Presumably this is done to use the
power of both the left and right outputs to allow the speaker to be
louder.
The ES8316 codec does not have a differential output mode, but we can
emulate this by making both channels output the same through the mono mix
switch, combined with setting the Playback Polarity control to "R Invert",
which applias a 180 degrees phase inversion to the right channel.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adding jack-detect support may seem weird for a codec with only
a single output, but it is necessary. The ES8316 appnote showing
the intended usage uses a jack-receptacle which physically disconnects
the speakers from the output when a jack is plugged in.
But all 3 devices using the es8316 which I have (2 Cherry Trail
devices and one Bay Trail CR device), use an analog mux to disconnect
the speakers, driven by a GPIO. In order to enable/disable the speakers
at the right time, we need jack-detect.
The same goes for the microphone where we must correctly set the mux
for the single ADC to either the internal or the headset microphone.
All devices I have support the es8316's builtin jack-detect functionality.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Inside function rt274_i2c_probe(), if regmap_read() function
returns -EINVAL, then local variable "val" leaves uninitialized
but used in if statement. This is potentially unsafe.
Signed-off-by: Yizhuo <yzhai003@ucr.edu>
Signed-off-by: Mark Brown <broonie@kernel.org>
Nobody has actually used the type (VERIFY_READ vs VERIFY_WRITE) argument
of the user address range verification function since we got rid of the
old racy i386-only code to walk page tables by hand.
It existed because the original 80386 would not honor the write protect
bit when in kernel mode, so you had to do COW by hand before doing any
user access. But we haven't supported that in a long time, and these
days the 'type' argument is a purely historical artifact.
A discussion about extending 'user_access_begin()' to do the range
checking resulted this patch, because there is no way we're going to
move the old VERIFY_xyz interface to that model. And it's best done at
the end of the merge window when I've done most of my merges, so let's
just get this done once and for all.
This patch was mostly done with a sed-script, with manual fix-ups for
the cases that weren't of the trivial 'access_ok(VERIFY_xyz' form.
There were a couple of notable cases:
- csky still had the old "verify_area()" name as an alias.
- the iter_iov code had magical hardcoded knowledge of the actual
values of VERIFY_{READ,WRITE} (not that they mattered, since nothing
really used it)
- microblaze used the type argument for a debug printout
but other than those oddities this should be a total no-op patch.
I tried to fix up all architectures, did fairly extensive grepping for
access_ok() uses, and the changes are trivial, but I may have missed
something. Any missed conversion should be trivially fixable, though.
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Among a few HD-audio fixes, the only significant one is the
regression fix on some machines like Dell XPS due to the default
binding changes. We ended up reverting the whole since the fix for
ASoC HD-audio driver won't be available immediately.
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Merge tag 'sound-fix-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Among a few HD-audio fixes, the only significant one is the regression
fix on some machines like Dell XPS due to the default binding changes.
We ended up reverting the whole since the fix for ASoC HD-audio driver
won't be available immediately"
* tag 'sound-fix-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Revert DSP detection on legacy HD-audio driver
ALSA: hda/tegra: clear pending irq handlers
ALSA: hda/realtek: Enable the headset mic auto detection for ASUS laptops
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, if platform_get_irq_byname() fails, the returned error
turns into a huge value, once it is being store into a variable
of type unsigned int, hence never actually reporting any error
and causing unexpected behavior when using the values stored
in aud_drv_data->s2mm_irq and aud_drv_data->mm2s_irq.
Fix this by changing the type of variables s2mm_irq and mm2s_irq in
structure xlnx_pcm_drv_data from unsigned int to int.
Addresses-Coverity-ID: 1476096 ("Unsigned compared against 0")
Fixes: 796175a94a7f ("ASoC: xlnx: add pcm formatter platform driver")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds reset and precharge in shutdown of PCM device.
ACODEC goes to silence if we change Fs to 44.1kHz from 48kHz. This
workaround seems to work but I don't know this workaround is correct
sequence or not for ACODEC.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for audio CODEC core of rk3328.
Rockchip does not publish detail specification of this core
but driver source code is opened on their GitHub repository.
https://github.com/rockchip-linux/kernel
So I ported this code to linux-next and added some trivial fixes.
Signed-off-by: Katsuhiro Suzuki <katsuhiro@katsuster.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is using asoc_simple_card_xxx() for
function / data naming. Because of this long prefix, it is easy to be
80 character over.
Let's reduce prefix from asoc_simple_card_xxx() to simple_xxx().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
This patch cleanups the code by using asoc_simple_card_for_each_link()
which judges normal link / DPCM link.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
This patch adds/modifies counting and parsing function for
"normal sound" and "DPCM sound", and call it from link loop.
This is prepare for cleanup DAI link loop method.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
To preparing cleanup code, this patch adds link_info which handles
number of DAIs/Links/Codec Conf, and CPU/Codec turn.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card is now supporting normal sound and DPCM sound.
For DPCM sound, original sound card (= simple-scu-card) had been
supported 1 CPU : 1 Codec connection which uses hw_params_fixup()
for convert-rate/channel.
But, merged simple-card is completely forgeting about it.
This patch re-support it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card is now supporting normal sound and DPCM sound.
For DPCM sound, original sound card (= simple-scu-card) had been
supported 1 CPU : 1 Codec connection which uses hw_params_fixup()
for convert-rate/channel.
But, merged simple-card is completely forgeting about it.
To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(),
it need to judge whether it is DPCM by checking convert-rate/channel.
For this purpose, this patch adds asoc_simple_card_get_conversion()
as preparation
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is using asoc_graph_card_xxx() for
function / data naming. Because of this long prefix, it is easy to be
80 character over.
Let's reduce prefix from asoc_graph_card_xxx() to graph_xxx().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
This patch cleanups the code by using asoc_graph_card_for_each_link()
which judges normal link / DPCM link.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
This patch adds/modifies counting and parsing function for
"normal sound" and "DPCM sound", and call it from link loop.
This is prepare for cleanup DAI link loop method.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is parsing DAI link for both "normal sound" and
"DPCM sound". On this driver, it needs to count and parse
DAIs/Links/Codec Conf from each links.
Then, counting/parsing link loop are very similar, but using different
implementation. Because of this background, the link loop code is very
mysterious. Mystery code will be trouble in the future.
To preparing cleanup code, this patch adds link_info which handles
number of DAIs/Links/Codec Conf, and CPU/Codec turn.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card is now supporting normal sound and DPCM sound.
For DPCM sound, original sound card (= audio-graph-scu) had been
supported 1 CPU : 1 Codec connection which uses hw_params_fixup()
for convert-rate/channel.
But, merged audio-graph-card is completely forgeting about it.
This patch re-support it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio formatter PL IP supports DMA of two streams -
mm2s and s2mm for playback and capture respectively. Apart from
DMA, IP also does conversions like PCM to AES and viceversa.
This patch adds DMA component driver for the IP.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is already merged into simple-card.
simple-scu-card is no longer needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is already merged into audio-graph-card.
audio-graph-scu-card is no longer needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card is now supporting normal sound and DPCM sound.
For DPCM sound, original sound card (= audio-graph-scu) had been
supported 1 CPU : 1 Codec connection which uses hw_params_fixup()
for convert-rate/channel.
But, merged audio-graph-card is completely forgeting about it.
To re-support 1 CPU : 1 Codec DPCM for hw_params_fixup(),
it need to judge whether it is DPCM by checking convert-rate/channel.
For this purpose, this patch adds asoc_graph_card_get_conversion()
as preparation
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We will get DAI ID from "reg" property if it has on DT, otherwise get
it by counting port/endpoint.
But in below case, we need to get DAI ID = 0 via port reg = <0>, but
current implementation returns ID = 1, because it can't judge ID = 0 was
from "non reg" or "reg = <0>".
Thus, it will count port/endpoint number as "non reg" case.
of_graph_parse_endpoint() implementation itself is not a problem,
but because asoc_simple_card_get_dai_id() need to count port/endpoint
number when "non reg" case, it need to know ID = 0 was from
"non reg" or "reg = <0>".
This patch fix this issue.
port {
reg = <0>;
xxxx: endpoint@0 {
};
=> xxxx: endpoint@1 {
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix Sparse warnings with two machine drivers which weren't updated
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Detected with Coccinelle
skl-messages.c:419:5-32: WARNING: Comparison to bool
skl-pcm.c:1426:6-33: WARNING: Comparison to bool
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Detected with Coccinelle
sound/soc/intel/skylake/skl-topology.c:3106:16-20: WARNING: casting
value returned by memory allocation function to (char *) is useless.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
MCLK input is needed when accessing any register after enabling SYSCLK.
This also fixes imbalance of clk_enable / clk_disable when transitioning
between ON -> STANDBY -> ON bias levels.
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Save 2x unsigned int of .rodata.
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For platforms that use the audio-graph-card driver, the codec is
not selected by SoC-platform driver. Make it available.
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds Cirrus Logic CS4341.
This is a very simple, playback only, stereo DAC.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ADC mixer setting needs to restore to default value
after calibration.
Signed-off-by: Shuming Fan <shumingf@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After 'commit 5d32a66541 ("PCI/ACPI: Allow ACPI to be built without
CONFIG_PCI set")' dependencies on CONFIG_PCI that previously were
satisfied implicitly through dependencies on CONFIG_ACPI have to be
specified directly. This code relies on IOSF_MBI and IOSF_MBI depends
on PCI. For this reason, add a direct dependency on CONFIG_PCI to the
IOSF_MBI driver.
Fixes: 5d32a66541 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set")
Signed-off-by: Sinan Kaya <okaya@kernel.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The problem is seen in the q6asm_dai_compr_set_params() function:
ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys,
(prtd->pcm_size / prtd->periods),
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
prtd->periods);
In this code prtd->pcm_size is the buffer_size and prtd->periods comes
from params->buffer.fragments. If we allow the number of fragments to
be zero then it results in a divide by zero bug. One possible fix would
be to use prtd->pcm_count directly instead of using the division to
re-calculate it. But I decided that it doesn't really make sense to
allow zero fragments.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can't return directly if snd_dma_alloc_pages() fails; we first need
to free prtd->audio_client and prtd.
Fixes: 22930c79ac ("ASoC: qdsp6: q6asm-dai: Add support to compress offload")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
