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>
The q6asm_audio_client_alloc() doesn't return NULL, it returns error
pointers.
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>
The q6asm_fe_dais[] array has MAX_SESSIONS (8) elements so the >
comparison should be >= or we access one element beyond the end of the
array.
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>
We accidentally call mutex_unlock(&pcm512x->mutex); twice in a row.
I re-wrote the error handling to use "goto unlock;" instead of returning
directly. Hopefully, it makes the code a little simpler.
Fixes: 3500f1c589 ("ASoC: pcm512x: Implement the digital_mute interface")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviwed-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Changed License header from C to C++ style comment block.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some reason this field was set to zero when all other drivers use
.dynamic = 1 for front-ends. This change was tested on Dell XPS13 and
has no impact with the existing legacy driver. The SOF driver also works
with this change which enables it to override the fixed topology.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The intent was to print the address as a hexadecimal but there is an
extra "u" in the "0x%08ulx" format specification so it is displayed as
decimal.
Fixes: aef3b06ac6 ("[ALSA] SH7760 ASoC support")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Headset codec is connected over PRIMARY_MI2S interface. Call
set_jack for codec associated with Primary Mi2s interface.
Also, set_jack to NULL when jack is freed.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This essentially reverts the commits
c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present
and Skylake driver selected")
and
d82b51c855 ("ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+
driver selection")
for the path of legacy HD-audio controller (snd-hda-intel).
The automatic DSP detection and skip of binding with the legacy driver
caused regressions on several machines like Dell XPS13. They give the
PCI class 0x40380 indicating the availability of DSP while they don't
work with ASoC SKL driver (yet).
As the support of ASoC driver for such devices isn't available, it's
better to revert the whole DSP-detection-and-skip behavior of the
legacy driver, so that we can get the old good driver working on such
devices.
The pci_binding option for ASoC SKL driver is still kept so that it
can work without blacklisting.
Fixes: c337104b1a ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected")
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Reported-by: Hans de Goede <hdegoede@redhat.com>
Reported-by: Azat Khuzhin <dohardgopro@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even after disabling interrupts on the module, it could be possible
that irq handlers are still running. System hang is seen during
suspend path. It was found that, there were pending writes on the
HDA bus and clock was disabled by that time.
Above mentioned issue is fixed by clearing any pending irq handlers
before disabling clocks and returning from hda suspend.
Suggested-by: Mohan Kumar <mkumard@nvidia.com>
Suggested-by: Dara Ramesh <dramesh@nvidia.com>
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The headset mic of ASUS laptops like UX533FD, UX433FN and UX333FA, whose
CODEC is Realtek ALC294 has jack auto detection feature. This patch
enables the feature.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Merge tag 'for-linus-4.21-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/xen/tip
Pull xen updates from Juergen Gross:
"Xen features and fixes:
- a series to enable KVM guests to be booted by qemu via the Xen PVH
boot entry for speeding up KVM guest tests
- a series for a common driver to be used by Xen PV frontends (right
now drm and sound)
- two other fixes in Xen related code"
* tag 'for-linus-4.21-rc1-tag' of git://git.kernel.org/pub/scm/linux/kernel/git/xen/tip:
ALSA: xen-front: Use Xen common shared buffer implementation
drm/xen-front: Use Xen common shared buffer implementation
xen: Introduce shared buffer helpers for page directory...
xen/pciback: Check dev_data before using it
kprobes/x86/xen: blacklist non-attachable xen interrupt functions
KVM: x86: Allow Qemu/KVM to use PVH entry point
xen/pvh: Add memory map pointer to hvm_start_info struct
xen/pvh: Move Xen code for getting mem map via hcall out of common file
xen/pvh: Move Xen specific PVH VM initialization out of common file
xen/pvh: Create a new file for Xen specific PVH code
xen/pvh: Move PVH entry code out of Xen specific tree
xen/pvh: Split CONFIG_XEN_PVH into CONFIG_PVH and CONFIG_XEN_PVH
Pull sparc updates from David Miller:
- Automatic system call table generation, from Firoz Khan.
- Clean up accesses to the OF device names by using full_name instead
of path_component_name.
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/sparc-next:
ALSA: sparc: Use of_node_name_eq for node name comparisons
sbus: Use of_node_name_eq for node name comparisons
sparc: generate uapi header and system call table files
sparc: add system call table generation support
sparc: add __NR_syscalls along with NR_syscalls
sparc: move __IGNORE* entries to non uapi header
sparc: Use DT node full_name instead of name for resources
sparc: Remove unused leon_trans_init
sparc: Use device_type helpers to access the node type
sparc: Use of_node_name_eq for node name comparisons
sparc: Convert to using %pOFn instead of device_node.name
sparc: prom: use property "name" directly to construct node names
of: Drop full path from full_name for PDT systems
sparc: Convert to using %pOF instead of full_name
fs/openpromfs: Use of_node_name_eq for node name comparisons
fs/openpromfs: use full_name instead of path_component_name
There are no intensive changes in both ALSA and ASoC core parts while
rather most of changes are a bunch of driver fixes and updates.
A large diff pattern appears in ASoC TI part which now merges both
OMAP and DaVinci stuff, but the rest spreads allover the places.
Note that this pull request includes also some updates for LED trigger
and platform drivers for mute LEDs, appearing in the diffstat as well.
Some highlights:
ASoC:
- Preparatory work for merging the audio-graph and audio-graph-scu
cards
- A merge of TI OMAP and DaVinci directories, as both product lines
get merged together. Also including a few architecture changes as
well.
- Major cleanups of the Maxim MAX9867 driver
- Small fixes for tablets & co with Intel BYT/CHT chips
- Lots of rsnd updates as usual
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers
HD-audio:
- Introduce audio-mute LED trigger for replacing the former hackish
dynamic binding
- Huawei WMI hotkey and mute LED support
- Refactoring of PM code and display power controls
- Headset button support in the generic jack code
- A few updates for Tegra
- Fixups for HP EliteBook and ASUS UX391UA
- Lots of updates for Intel ASoC HD-audio, including the improved DSP
detection and the fallback binding from ASoC SST to legacy HD-audio
controller drivers
Others:
- Updates for FireWire TASCAM and Fireface devices, some other fixes
- A few potential Spectre v1 fixes that are all trivial
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Merge tag 'sound-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There are no intensive changes in both ALSA and ASoC core parts while
rather most of changes are a bunch of driver fixes and updates. A
large diff pattern appears in ASoC TI part which now merges both OMAP
and DaVinci stuff, but the rest spreads allover the places.
Note that this pull request includes also some updates for LED trigger
and platform drivers for mute LEDs, appearing in the diffstat as well.
Some highlights:
ASoC:
- Preparatory work for merging the audio-graph and audio-graph-scu
cards
- A merge of TI OMAP and DaVinci directories, as both product lines
get merged together. Also including a few architecture changes as
well.
- Major cleanups of the Maxim MAX9867 driver
- Small fixes for tablets & co with Intel BYT/CHT chips
- Lots of rsnd updates as usual
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx
I2S controllers
HD-audio:
- Introduce audio-mute LED trigger for replacing the former hackish
dynamic binding
- Huawei WMI hotkey and mute LED support
- Refactoring of PM code and display power controls
- Headset button support in the generic jack code
- A few updates for Tegra
- Fixups for HP EliteBook and ASUS UX391UA
- Lots of updates for Intel ASoC HD-audio, including the improved DSP
detection and the fallback binding from ASoC SST to legacy HD-audio
controller drivers
Others:
- Updates for FireWire TASCAM and Fireface devices, some other fixes
- A few potential Spectre v1 fixes that are all trivial"
* tag 'sound-4.21-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection
ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected
ALSA: HDA: export process_unsol_events()
ALSA: hda/realtek: Enable audio jacks of ASUS UX391UA with ALC294
ALSA: bebob: fix model-id of unit for Apogee Ensemble
ALSA: emu10k1: Fix potential Spectre v1 vulnerabilities
ALSA: rme9652: Fix potential Spectre v1 vulnerability
ASoC: ti: Kconfig: Remove the deprecated options
ARM: davinci_all_defconfig: Update the audio options
ARM: omap1_defconfig: Do not select ASoC by default
ARM: omap2plus_defconfig: Update the audio options
ARM: davinci: dm365-evm: Update for the new ASoC Kcofnig options
ARM: OMAP2: Update for new MCBSP Kconfig option
ARM: OMAP1: Makefile: Update for new MCBSP Kconfig option
MAINTAINERS: Add entry for sound/soc/ti and update the OMAP audio support
ASoC: ti: Merge davinci and omap directories
ALSA: hda: add mute LED support for HP EliteBook 840 G4
ALSA: fireface: code refactoring to handle model-specific registers
ALSA: fireface: add support for packet streaming on Fireface 800
ALSA: fireface: allocate isochronous resources in mode-specific implementation
...
For HDaudio and Skylake drivers, add module parameter "pci_binding"
When pci_binding == 0 (AUTO), the PCI class/subclass info is used to
select drivers based on the presence of the DSP.
pci_binding == 1 (LEGACY) forces the use of the HDAudio legacy driver,
even if the DSP is present.
pci_binding == 2 (ASOC) forces the use of the ASOC driver. The
information on the DSP presence is bypassed.
The value for the module parameter needs to be identical for both
drivers. This parameter is intended as a back-up solution if the
automatic detection fails or when the DSP usage fails. Such cases
should be reported on the alsa-devel mailing list for analysis.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the SST/Skylake driver supports per platform selectors, we
can add logic to automatically select the right driver.
If the Skylake driver is selected for a specific platform, and the DSP
is detected at run-time based on the PCI class/subclass/prog-if
information, the legacy HDaudio driver aborts the probe. This will
result in a single driver probing and remove the need for modprobe
blacklists.
Follow-up patches will add a module parameter to bypass the logic if
this automatic detection fails, or if the Skylake driver is unable to
actually support the platform (firmware authentication, missing
topology file, hardware issue, etc).
The same mechanism will be used to conflicts generated by the same PCI
ID being registered by both legacy HDAuudio and SOF drivers for Intel
platforms. In other words SOF will not require changes to the HDaudio
legacy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SOF implementation does not rely on the hdac_bus library, however
for HDMI and HDaudio codec support it does need to deal with
unsolicited events. Instead of re-inventing the wheel, export this
symbol to reuse this part of the library directly.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By default, there is no sound on Asus UX391UA on Linux.
This patch adds sound support on Asus UX391UA. Tested working by three
different users.
The problem has also been described at
https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1784485
Signed-off-by: Wandrille RONCE <w@ndrille.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ipcm->substream is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/pci/emu10k1/emufx.c:1031 snd_emu10k1_ipcm_poke() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap)
sound/pci/emu10k1/emufx.c:1075 snd_emu10k1_ipcm_peek() warn: potential spectre issue 'emu->fx8010.pcm' [r] (local cap)
Fix this by sanitizing ipcm->substream before using it to index emu->fx8010.pcm
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
info->channel is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/pci/rme9652/hdsp.c:4100 snd_hdsp_channel_info() warn: potential spectre issue 'hdsp->channel_map' [r] (local cap)
Fix this by sanitizing info->channel before using it to index hdsp->channel_map
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
Also, notice that I refactored the code a bit in order to get rid of the
following checkpatch warning:
ERROR: do not use assignment in if condition
FILE: sound/pci/rme9652/hdsp.c:4103:
if ((mapped_channel = hdsp->channel_map[info->channel]) < 0)
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use page directory based shared buffer implementation
now available as common code for Xen frontend drivers.
Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Boris Ostrovsky <boris.ostrovsky@oracle.com>
Not much work on the core this time around but we've seen quite a bit of
driver work, including on the generic DT drivers. There's also a large
part of the diff from a merge of the DaVinci and OMAP directories, along
with some active development there:
- Preparatory work from Morimoto-san for merging the audio-graph and
audio-graph-scu cards.
- A merge of the TI OMAP and DaVinci directories, the OMAP product line
has been merged into the DaVinci product line so there is now a lot
of IP sharing which meant that the split directories just got in the
way. This has pulled in a few architecture changes as well.
- A big cleanup of the Maxim MAX9867 driver from Ladislav Michl.
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers.
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Merge tag 'asoc-v4.21' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.21
Not much work on the core this time around but we've seen quite a bit of
driver work, including on the generic DT drivers. There's also a large
part of the diff from a merge of the DaVinci and OMAP directories, along
with some active development there:
- Preparatory work from Morimoto-san for merging the audio-graph and
audio-graph-scu cards.
- A merge of the TI OMAP and DaVinci directories, the OMAP product line
has been merged into the DaVinci product line so there is now a lot
of IP sharing which meant that the split directories just got in the
way. This has pulled in a few architecture changes as well.
- A big cleanup of the Maxim MAX9867 driver from Ladislav Michl.
- Support for Asahi Kaesi AKM4118, AMD ACP3x, Intel platforms with
RT5660, Meson AXG S/PDIF inputs, several Qualcomm IPs and Xilinx I2S
controllers.
We no longer have these options used anywhere.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Create new directory to contain all Texas Instruments specific DAI,
platform and machine drivers instead of scattering them under davinci and
omap directories.
There is already inter dependency between the two directories becasue of
McASP (on dra7x it is serviced by sDMA, not EDMA).
With the upcoming AM654 we will need to introduce new platform driver for
UDMA and it does not fit under davinci, nor under omap.
With the move I have restructured the Kconfig to be more usable in the era
of simple-sound-card:
CPU DAIs can be selected individually and they will select the platform
driver they can be served with.
To avoid breakage, I have moved over deprecated Kconfig options so
defconfig builds will work without regression.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
For sound/soc/{omap => ti}:
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Tested with 4.19.9.
v2: Changed from CXT_FIXUP_MUTE_LED_GPIO to CXT_FIXUP_HP_DOCK because
that's what the existing fixups for EliteBooks use.
Signed-off-by: Mantas Mikulėnas <grawity@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As a result of investigation for Fireface 800, 'struct snd_ff_spec.regs'
is just for higher address to receive tx asynchronous packets of MIDI
messages, thus it can be simplified.
This commit simplifies it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to multiplex PCM frames into isochronous
packets and demultiplex PCM frames from isochronous packets for ALSA PCM
applications.
Fireface 800 voluntarily maintains resources for tx isochronous
communication. It performs reservation of isochronous channel and
allocation/update of bandwidth in some cases below:
- at a first request to allocation after bus resets
- at requests to allocation when further bandwidth is required
When request is grant and the unit is prepared, read data from
0x0000801c0008 represents isochronous channel for tx stream, then
the unit can handle requests to start communication. If driver
send the request without checking the register, the unit takes
panic to continue bus resets. The unit starts transmission of
tx packets after receiving several rx packets from driver.
I note that the unit can process tx/rx packets and generate/record
sound regardless of HOST LED.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The way to maintain isochronous resources on bus is different between
Fireface 400/800.
This commit is a preparation. This commit moves a function to allocate resource to
model-dependent implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface 400/800 use three modes against the number of data channels in
data block for both tx/rx packets.
This commit adds refactoring for it. Some enumerators are added to
represent each of mode and a function is added to calculate the mode
from sampling frequency code (sfc).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both of Fireface 400/800 have the same register to switch frame fetching
mode regardless of difference of available number of PCM frames in
rx isochronous packet.
This commit moves a helper function from model-dependent implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to my memo at hand and saved records, writing 0x00000001 to
SND_FF_REG_FETCH_PCM_FRAMES disables fetching PCM frames in corresponding
channel, however current implement uses reversed logic. This results in
muted volume in device side during playback.
This commit corrects the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 76fdb3a9e1 ('ALSA: fireface: add support for Fireface 400')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An initial commit to add tracepoints for packets without CIP headers
uses different print formats for added tracepoints. However this is not
convenient for users/developers to prepare debug tools.
This commit uses the same format for the two tracepoints.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: b164d2fd6e ('ALSA: firewire_lib: add tracepoints for packets without CIP headers')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An initial commit to add tracepoints for packets without CIP headers
introduces a wrong assignment to 'data_blocks' value of
'out_packet_without_header' tracepoint.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: b164d2fd6e ('ALSA: firewire_lib: add tracepoints for packets without CIP headers')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-1/6 engine of ALSA firewire stack, a packet handler has a
second argument for 'the number of bytes in payload of isochronous
packet'. However, an incoming packet handler without CIP header uses the
value as 'the number of quadlets in the payload'. This brings userspace
applications to receive the number of PCM frames as four times against
real time.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 3b196c394d ('ALSA: firewire-lib: add no-header packet processing')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add support to Display_port_rx mixers required to
select path between ASM stream and AFE ports.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support of AFE DAI for Display_port_rx port.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for Display_Port_Rx
port in AFE.
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds MP3 playback support in q6asm dais, adding other codec
support should be pretty trivial.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to mp3 format in ASM module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Default copy function uses kmalloc to allocate buffers, lets check
if the runtime buffers are setup before making this allocations.
This can be useful if the buffers are dma buffers.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current SKYLAKE kconfig is a all-you-can-eat selection that will
support all known plaforms. This is however not necessarily a good
thing: most platforms for SKL and KBL don't support the DSP, but a
number of CNL/WHL ones do. Selecting this driver in all cases isn't
really smart and will require users to muck with blacklists.
Partition the configs to allow distributions to select on which
platform this driver is used. Keep the existing SND_SOC_INTEL_SKYLAKE
config to select everything for backwards compatibility. This patch does
not provide new functionality, only finer-grained choices in supported
platforms.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is handling "prefix" by many ways.
But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch supports it.
It will be overwrote if lower node has it.
sound {
simple-audio-card,prefix = "xxx"; // initial
simple-audio-card,dai-link {
prefix = "xxx"; // overwrite
cpu {
...
};
codec {
prefix = "xxx"; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is handling "convert_rate/channel"
by many ways. But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
simple-audio-card,convert_channels = <xxx>; // initial
simple-audio-card,dai-link {
convert_channels = <xxx>; // overwrite
cpu {
convert_channels = <xxx>; // overwrite
};
codec {
convert_channels = <xxx>; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is handling "mclk-fs" by many way.
But, it is not useful and readable.
We want to do is that allow having mclk-fs everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
simple-audio-card,mclk-fs = <xxx>; // for initial
simple-audio-card,dai-link {
mclk-fs = <xxx>; // overwrite
cpu {
mclk-fs = <xxx>; // overwrite
};
codec {
mclk-fs = <xxx>; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card and simple-scu-card are very similar driver,
but the former is supporting normal sound card,
the latter is supporting DPCM sound card.
We couldn't use normal sound and DPCM sound in same time by
one sound card. This patch merges both sound card into
simple-card. Now we can use both feature on same driver.
simple-card is now supporting .compatible = "simple-scu-audio-card".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is handling "prefix" by many ways.
But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch supports it.
It will be overwrote if lower node has it.
sound {
prefix = "xxx"; // initial
};
codec {
audio-graph-card,prefix = "xxx"; // overwrite
ports {
prefix = "xxx"; // overwrite
port {
prefix = "xxx"; // overwrite
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is handling "convert_rate/channel"
by many ways. But, it is not useful and readable.
We want to do is that allow having it everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
convert-channels = <xxx>; // initial
};
codec {
audio-graph-card,convert-channels = <xxx>; // overwrite
ports {
convert_channels = <xxx>; // overwrite
port {
convert_channels = <xxx>; // overwrite
endpoint {
convert_channels = <xxx>; // overwrite
};
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current audio-graph-card is handling "mclk-fs" by many way.
But, it is not useful and readable.
We want to do is that allow having mclk-fs everywere.
This patch support it.
It will be overwrote if lower node has it.
sound {
mclk-fs = <xxx>; // initial
};
codec {
ports {
mclk-fs = <xxx>; // overwrite
port {
mclk-fs = <xxx>; // overwrite
endpoint {
mclk-fs = <xxx>; // overwrite
};
};
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
audio-graph-card and audio-graph-scu-card are very similar driver,
but the former is supporting normal sound card,
the latter is supporting DPCM sound card.
We couldn't use normal sound and DPCM sound in same sound card by
audio-graph-card.
This patch merges both sound card into it.
Now we can use both feature on same driver.
audio-grap-card is now supporting .compatible = "audio-graph-scu-card".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit b6f3fc005a ("ASoC: simple-card-utils: fixup
asoc_simple_card_get_dai_id() counting") fixuped getting DAI ID method.
It will get DAI ID from OF graph "port", but, we want to consider about
"endpoint", too.
And, we also want to keep compatibility.
This patch fixup it as
if (driver has specified DAI ID)
use it as DAI ID
else if (OF graph endpoint has reg)
use it as DAI ID
else if (OF graph port has reg)
use it as DAI ID
else
use endpoint count as DAI ID
Fixes: commit b6f3fc005a ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting")
Reported-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove no_pcm check to invoke pcm_new() for backend dai-links
too. This fixes crash in hdmi codec driver during hdmi_codec_startup()
while accessing chmap_info struct. chmap_info struct memory is
allocated in pcm_new() of hdmi codec driver which is not invoked
in case of DPCM when hdmi codec driver is part of backend dai-link.
Below is the crash stack:
[ 61.635493] Unable to handle kernel NULL pointer dereference at virtual address 00000018
..
[ 61.666696] CM = 0, WnR = 1
[ 61.669778] user pgtable: 4k pages, 39-bit VAs, pgd = ffffffc0d6633000
[ 61.676526] [0000000000000018] *pgd=0000000153fc8003, *pud=0000000153fc8003, *pmd=0000000000000000
[ 61.685793] Internal error: Oops: 96000046 [#1] PREEMPT SMP
[ 61.722955] CPU: 7 PID: 2238 Comm: aplay Not tainted 4.14.72 #21
..
[ 61.740269] PC is at hdmi_codec_startup+0x124/0x164
[ 61.745308] LR is at hdmi_codec_startup+0xe4/0x164
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Clicks and pops of various volumes can be produced while the device is
opened, closed, put into and taken out of standby, or reconfigured.
Fix this, by implementing the digital_mute interface, so that the
output is muted during such operations.
Signed-off-by: Dimitris Papavasiliou <dpapavas@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Even if this spdif input driver is only supposed to be used on 64bits
platform, there is possible problem with 32bits and do_div, as reported
by the kbuild robot. Just fix it.
Fixes: 5ce5658375 ("ASoC: meson: add axg spdif input")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add error logs to make probe debug easier.
Also remove hard-coded dependency on NHLT. NHLT literally stands for
NonHdaudioLinkTable and is only required for SSP/DMIC interfaces.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
bus->ppcap is now tested upfront, there is no need to re-check if the
hardware is exposed as needed. Remove tests and remove indentation.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Check immediately if required HDaudio capabilities can't be found (no
PPCAP or no streams exposed in GCAP), and move all DMA inits after the
error tests.
PPCAP and GCAP are not reliable indicators of DSP presence, but if
they don't exist then the driver will not work.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing PPCAP and GCAP fields cannot be used reliably to
determine if the DSP is enabled by the BIOS. Instead rely on the
class/subclass information to find out if this driver can run or
not. The values in the code don't seem to be documented in publicly
available documents but are part of recommendations made to BIOS
writers and have been verified to be accurate on a number of
platforms.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It's with CNP, supposed to be equivalent with CNL entry.
Keep the existing declaration style for now, at a later point we may
transition and use PCI_DEVICE_DATA().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
I2S IP instance can work in transmitter/playback or receiver/capture mode
exclusively. The patch registers corresponding instance as ASoC component
with audio framework.
Signed-off-by: Maruthi Srinivas Bayyavarapu <maruthi.srinivas.bayyavarapu@xilinx.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The new Dell IoT platform uses kabylake + alc3277 codec, and alc3277
shares the driver with the codec rt5660, here we generate a new
machine driver based on kbl_da7219_max98357a.
The audio design on this IoT platform is as below:
- Intel kabylake platform
- connect the codec ALC3277 via SSP0
- line-out and line-in with Micbias jacks
- line-out mute control and jack detection of line-out and line-in
- two HDMI ports with audio capability
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the spdif input decoder of the axg SoC family
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
add IEC958_SUBFRAME_LE to the list of format accepted by the fifo frontend.
As opposed to what was initially noted in the toddr dai driver, the spdifin
does not place the msb at bit 28, it just output a whole spdif subframe.
Placing the msb at bit 28 in the toddr driver just filters out the parity,
user, channel status and validity bits. It is better to just provide the
whole spdif subframe to the userspace and let the iec958 plugin deal with
it.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 4fb7f4df49 ("ASoC: simple-card: use cpu/codec pointer on
simple_dai_props") updated {cpu,codec}_dai to be pointers in struct
simple_dai_props but didn't update these locations to dereference the
pointers.
This patch fixup it for non DT simple-card use case.
Signed-off-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
eukrea-tlv320.c machine driver runs on non-DT platforms
and include <asm/mach-types.h> header file in order to be able
to use some machine_is_eukrea_xxx() macros.
Building it for ARM64 causes the following build error:
sound/soc/fsl/eukrea-tlv320.c:28:10: fatal error: asm/mach-types.h: No such file or directory
Avoid this error by not allowing to build the SND_SOC_EUKREA_TLV320
driver when ARM64 is selected.
This is needed in preparation for the i.MX8M support.
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Acked-by: Shawn Guo <shawnguo@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some of Huawei laptops come with a LED in the micmute key. This patch
enables the use of micmute LED for these devices:
1. Matebook X (19e5:3200), (19e5:3201)
2. Matebook X Pro (19e5:3204)
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ayman Bagabas <ayman.bagabas@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch solves bug 200501 'Only 2 of 4 speakers playing sound.'
It enables the front speakers on Huawei Matebook X Pro laptops.
These laptops come with Dolby Atmos sound system and these pins
configuration enables the front speakers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=200501
Reviewed-by: Andy Shevchenko <andy.shevchenko@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ayman Bagabas <ayman.bagabas@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
stream is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/core/pcm.c:140 snd_pcm_control_ioctl() warn: potential spectre issue 'pcm->streams' [r] (local cap)
Fix this by sanitizing stream before using it to index pcm->streams
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
info.mode and info.port are indirectly controlled by user-space,
hence leading to a potential exploitation of the Spectre variant 1
vulnerability.
These issues were detected with the help of Smatch:
sound/synth/emux/emux_hwdep.c:72 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[i]->ctrls' [w] (local cap)
sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs' [w] (local cap)
sound/synth/emux/emux_hwdep.c:75 snd_emux_hwdep_misc_mode() warn: potential spectre issue 'emu->portptrs[info.port]->ctrls' [w] (local cap)
Fix this by sanitizing both info.mode and info.port before using them
to index emu->portptrs[i]->ctrls, emu->portptrs[info.port]->ctrls and
emu->portptrs.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://marc.info/?l=linux-kernel&m=152449131114778&w=2
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current rsnd is using RSND_REG_xxx for register naming,
and using RSND_REG_##f style macro for read/write.
The biggest reason why it uses this style is that
we can avoid non-existing register access.
But, its demerit is sequential register access code will
be very ugly.
Current rsnd driver is well tested, so, let's remove RSND_REG_
from rsnd_reg, and cleanup sequential register access code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fireface 800 is a flagship model of RME GmbH for audio and music units
on IEEE 1394 bus, shipped 2004. This model consists of four chips:
- TI TSB81BA3D for physical layer on cable environment of EEE 1394 bus
- TI TSB82AA2 for link layer for 1394 OHCI bus bridge to PCI bus
- Xilinx Spartan-3 FPGA XC3S400
- Xilinx High-Performance CPLD XC9572XL
This commit adds support Fireface 800. In this time, the support is
restricted to its MIDI functionality, thus this commit adds some
condition statements to avoid touching streaming functionality.
Unlike Fireface 400, Fireface 800 has no functionality to suppress
asynchronous transactions for MIDI messages except for unregister of
listen address in controller side, thus the feature is available as is.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Content of asynchronous transaction for MIDI messages differs between
Fireface 400 and 800.
This commit adds a model-specific handler for the transaction and adds
arrangement.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface 400 and 800 have the same mechanism to decide address to which
asynchronous transactions are sent for MIDI messages, however they use
different registers for controllers to notify higher 4 byte of the
address.
This commit adds a model-specific parameter to represent the address.
Additionally, it corrects some comments. I note that these two models have
a difference to enable/disable the transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating packet dumps from Fireface 400/800, a register
to receive asynchronous transactions for MIDI messages is the same. For
Fireface 800, minor register is used.
This commit declares macros for the transactions and obsoletes
model-specific parameters.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unlike Fireface 400, Fireface 800 have two pair of optical interface
for ADAT signal and S/PDIF signal. ADAT signals for the interface
are handled for sampling clock source separately.
This commit modifies a parser for clock configuration to distinguish
these two ADAT signals.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating packet dumps from Fireface 400/800, bits on
status registers for clock synchronization are the same.
This commit moves a parser for a register of clock configuration to
obsolete model-specific operations.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating packet dumps from Fireface 400/800, bits on
status registers for clock synchronization are the same.
This commit moves a parser for the registers to obsolete model-specific
operations.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating packet dumps from Fireface 400/800, status
registers for clock synchronization is common.
This commit moves some macros for them to header file.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current simple-scu-card didn't care about codec_conf
for multi DPCM case. This patch cares it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and
non DPCM case, it uses CPU-Codec dai links.
Now, we want to merge simple-card and simple-scu-card.
These sound cards are using silimar but not same logic on each functions.
Then, of course we want to share same logic.
To compromise, this patch uses cpu/codec pointer on simple-scu-card.
It is same logic with simple-card, thus easy merging.
This is prepare for merging simple card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-scu-card.c is supporting "convert-rate/channels" which is
used for DPCM.
But, sound card might have multi codecs, and each codec might need
each convert-rate/channels.
This patch supports each codec's convert-rate/channles support.
top node convert-rate/channels will overwrite settings if exist.
It can't support each codec's convert-rate/channels if sound card had
multi codecs without this patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links.
If sound card is caring only DPCM, link count = dai count,
but, if non DPCM case, link count != dai count.
Now, we want to merge simple-card and simple-scu-card,
then, we need to care both link / dai count more carefly
This patch cares it, and prepare for merging simple card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card is supporting dai-link support, but simple-scu-card
doesn't have it.
This patch support it. This is prepare for merging simple-card
and simple-scu-card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When building without CONFIG_PCI, we can (depending on the architecture)
get a link failure:
ERROR: "pci_iounmap" [sound/pci/hda/snd-hda-codec-ca0132.ko] undefined!
Adding a compile-time check for PCI gets it to work correctly on
32-bit ARM.
Fixes: d99501b857 ("ALSA: hda/ca0132 - Call pci_iounmap() instead of iounmap()")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've excluded the display_power_control flag for Intel HSW and BDW
codecs as the HD-audio controllers of the corresponding platforms take
care of the display power as well. But the recent refactoring
separates the controller and the codec power accounting, so it's fine
to call the display PM even for HSW/BDW codecs. This is less
confusing since we can avoid this well-hidden condition.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The display power is in unbalance at removing the driver since it
misses the snd_hdac_display_power(OFF) call.
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the recent refactoring, snd_hdac_display_power() doesn't return
any error, hence it can be defined to return void.
This makes many error checks redundant and allows us to reduce them
gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an error occurs in azx_probe_continue(), we should release the
display power. However, the current code ignores it and releases the
display power only for HSW/BDW cases. Fix it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hdac_display_power() can be called even for a HDA controller
without DRM binding. The same is true for other helpers,
snd_hdac_i915_set_bclk() and snd_hdac_set_codec_wakeup().
So all superfluous AZX_DCAPS_I915_POWERWELL checks in hda_intel.c can
be dropped, and the definition of AZX_DCAPS_I915_POWERWELL itself can
be removed as well. This simplifies the code a lot.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current HD-audio code manages the DRM audio power via too complex
redirections, and this seems even still unbalanced in a corner case as
Intel DRM CI has been intermittently reporting. This patch is a big
surgery for addressing the complexity and the possible unbalance.
Basically the patch changes the display PM in the following ways:
- Both HD-audio controller and codec drivers call a single helper,
snd_hdac_display_power(). (Formerly, the display power control from
a codec was done indirectly via link_power bus ops.)
- snd_hdac_display_power() receives the codec address index. For
turning on/off from the controller, pass HDA_CODEC_IDX_CONTROLLER.
- snd_hdac_display_power() doesn't manage refcounts any longer, but
keeps the power status in bitmap. If any of controller or codecs is
turned on, the function updates the DRM power state via get_power()
or put_power().
Also this refactor allows us more cleanup:
- The link_power bus ops is dropped, so there is no longer indirect
management, as mentioned in the above.
- hdac_device link_power_control flag is moved to hda_codec
display_power_control flag, as it's only for HDA legacy.
Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=106525
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current simple-scu-card driver is parsing codec position for DPCM
and consider DAI format. But, current operation is doing totally pointless,
because it should be called for each CPU/Codec pair.
Let's tidyup asoc_simple_card_parse_daifmt() timing.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In DPCM case, it uses CPU-dummy / dummy-Codec dai links, and
non DPCM case, it uses CPU-Codec dai links.
Now, we want to merge simple-card and simple-scu-card.
These sound cards are using silimar but not same logic on each functions.
Then, of course we want to share same logic.
To compromise, this patch uses cpu/codec pointer on simple-card.
It is same logic with simple-scu-card, thus easy merging.
This is prepare for merging audio card
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When CONFIG_OF is disabled, of_graph_parse_endpoint() does not
initialize 'info', and gcc can see that:
sound/soc/generic/simple-card-utils.c: In function 'asoc_simple_card_parse_graph_dai':
sound/soc/generic/simple-card-utils.c:284:13: error: 'info.port' may be used uninitialized in this function [-Werror=maybe-uninitialized]
It's probably best to check the return code anyway, and that also
takes care of the warning.
Fixes: b6f3fc005a ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Calling into the codec driver adds a dependency on that being reachable
from the module:
ERROR: "rt5663_sel_asrc_clk_src" [sound/soc/qcom/snd-soc-sdm845.ko]
undefined!
Add the corresponding select statement, as it is done in the other user
(Intel).
Fixes: f7485875a687 ("ASoC: sdm845: Add configuration for headset codec")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
From the da7219 spec, the button A, B, C and D are remapped to
0, 1, 2 and 3 respectively where button A is KEY_PLAYPAUSE,
B is KEY_VOLUMEUP, C is KEY_VOLUMEDOWN and D is KEY_VOICECOMMAND.
Signed-off-by: Zhuohao Lee <zhuohao@chromium.org>
Signed-off-by: Max Chang <changmax@chromium.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a quirk for the Point of View Mobii TAB-P1005W-232 v2.0 tablet, this
BYTCR device uses IN1 for its MIC and JD2 for jack-detect, rather then the
default IN3 and JD1.
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 a quirk for the Prowise PT301 tablet, this BYTCR tablet has no CHAN
package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which
is the default for BYTCR devices.
Also it uses IN1 for its MIC and JD2 for jack-detect, rather then the
default IN3 and JD1.
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 ASUS UX433FN and UX333FA with ALC294 cannot detect the headset MIC
and output through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS UX533FD with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone until
ALC294_FIXUP_ASUS_SPK and ALC294_FIXUP_ASUS_HEADSET_MIC quirk applied.
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The known ALC256_FIXUP_ASUS_MIC fixup can fix the headphone jack
sensing and enable use of the internal microphone on this laptop
X542UN. However, it's ALC294 so create a new fixup named
ALC294_FIXUP_ASUS_MIC to avoid confusion.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make unified suspend / resume helpers and call them from both the
runtime- and the system-PM callbacks for simplifying code.
There are slight changes of call orders, but there shouldn't be any
functional difference after refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In an initial commit, 'SYNC_STATUS' register is referred to get
clock configuration, however this is wrong, according to my local
note at hand for reverse-engineering about packet dump. It should
be 'CLOCK_CONFIG' register. Actually, ff400_dump_clock_config()
is correctly programmed.
This commit fixes the bug.
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 76fdb3a9e1 ('ALSA: fireface: add support for Fireface 400')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Users reported a mute LED regression on Lenovo X1 Carbon, the root
cause is we applied the fixup of ALC285_FIXUP_LENOVO_HEADPHONE_NOISE
to this machine, then the machine can't apply the fixup of
ALC269_FIXUP_THINKPAD_ACPI anymore. To fix it, we chain two fixup
together.
Fixes: c4cfcf6f42 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Driver rewritten, assign copyright notice and change module author
as original one remains silent and I want to be notified about bugs.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set DAI format and sysclk for headset codec.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set TDM time slots and DAI format for speaker codec.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sound capture and line bypass currently do not work as well as
some mixer controls. Fix that by building proper audio paths and
adjusting volume controls to match datasheet.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop "Common NI Values Table" and calculate LRCLK divider, then
add allowed rate constraints based on master clock frequency.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Implement set_bias_level to drive shutdown bit, so device is
put to sleep when unused.
Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch will enable headset button for new Chrome platform.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Extend some structs to add the support for jack button changes.
Now snd_hda_jack_add_kctl() receives two more arguments: the jack type
and the jack keymaps. Both are optional, and when zero are passed,
the function behaves just like before.
For reporting button state changes, you'd need to update
jack->button_state bits accordingly, typically in the jack callback.
Then the value OR'ed with button_state and the jack plug state is
passed to snd_jack_report().
Note that currently the code assumes only the one-shot button events,
i.e. it tries to send the button release soon after sending the button
event. If a driver really supports the button release handling by
itself, we may need to introduce some flag to control this behavior in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For allowing the callee to evaluate the associated jack information
and the unsolicited event data, add the new fields to
hda_jack_callback. They can be used, for example, to retrieve the
headset button state in the callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If it plugged headphone or headset into the jack, then
do the reboot, it will have a chance to cause headphone no sound.
It just need to run the headphone mode procedure after boot time.
The issue will be fixed.
It also suitable for ALC234 ALC274 and ALC294.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Realtek codec ALC3277 is 100% compatible with the codec RT5660
in I2S mode. And on the Dell IoT platform, the codec is ALC3277,
and the HID of the codec in the BIOS is 10EC3277, so adding this
ID to the ACPI match table.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Get the reset GPIO through the GPIO consumer API. This allows specifying the
DT property as "reset-gpios" without breaking existing DT users.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Get the reset GPIO through the GPIO consumer API. This allows specifying the
DT property as "reset-gpios" without breaking existing DT users.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Convert string compares of DT node names to use of_node_name_eq helper
instead. This removes direct access to the node name pointer.
For the FSL ASoC card, the full node names appear to be "ssi", "esai",
and "sai", so there's not any reason to use strstr and of_node_name_eq
can be used instead.
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <fabio.estevam@nxp.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
Signed-off-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
AMD platform device acp_audio_dma can only be created by parent PCI
device driver (drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c). Pass struct
device of the parent to snd_pcm_lib_preallocate_pages() so
dma_alloc_coherent() can use correct dma_ops. Otherwise, it will
use default dma_ops which is nommu_dma_ops on x86_64 even when
IOMMU is enabled and set to non passthrough mode.
Though platform device inherits some dma related fields during its
creation in mfd_add_device(), we can't simply pass its struct device
to snd_pcm_lib_preallocate_pages() because dma_ops is not among the
inherited fields. Even it were, drivers/iommu/amd_iommu.c would
ignore it because get_device_id() doesn't handle platform device.
This change shouldn't give us any trouble even struct device of the
parent becomes null or represents some non PCI device in the future,
because get_dma_ops() correctly handles null struct device or uses
the default dma_ops if struct device doesn't have it set.
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We shouldn't assume CPU physical address we get from page_to_phys()
is same as DMA address we get from dma_alloc_coherent(). On x86_64,
we won't run into any problem with the assumption when dma_ops is
nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled.
And it's most likely different from CPU physical address when AMD
IOMMU is not in passthrough mode.
Signed-off-by: Yu Zhao <yuzhao@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai