As an equivalent codec with CX20724,
CX8200 is also subject to the reboot bug.
Late 2017 and 2018 LG Gram and some HP Spectre laptops are known victims
to this issue, causing extremely loud noises upon reboot.
Now that we know that this bug is subject to multiple codecs,
fix the comment as well.
Signed-off-by: Park Ju Hyung <qkrwngud825@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hw_pll_init(), hw_dac_stop(), hw_dac_start() and hw_adc_init()
are never called in atomic context.
They call mdelay() to busily wait, which is not necessary.
mdelay() can be replaced with msleep().
This is found by a static analysis tool named DCNS written by myself.
Signed-off-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hw_pll_init(), hw_reset_dac() and hw_card_init() are never
called in atomic context.
They calls mdelay() to busily wait, which is not necessary.
mdelay() can be replaced with msleep().
This is found by a static analysis tool named DCNS written by myself.
Signed-off-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_usb_select_mode_quirk(), snd_usb_set_interface_quirk() and
snd_usb_ctl_msg_quirk() are never called in atomic context.
They call mdelay() to busily wait, which is not necessary.
mdelay() can be replaced with msleep() and usleep_range().
This is found by a static analysis tool named DCNS written by myself.
Signed-off-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The data types defined in SB CSP driver code are all in little-endian,
hence the proper type like __le32 should be used.
Spotted by sparse, a warning like:
sound/isa/sb/sb16_csp.c:330:14: warning: cast to restricted __le32
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DMA address table in atiixp modem driver is in little-endian,
hence we should define it with __le32 properly.
Spotted by sparse, a warning like:
sound/pci/atiixp_modem.c:360:28: warning: incorrect type in assignment (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DMA address table in atiixp driver is in little-endian, hence we should define it with __le32 properly.
Spotted by sparse, a warning like:
sound/pci/atiixp.c:393:28: warning: incorrect type in assignment (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The RISC data in bt87x is in little-endian, hence we should define it
with __le32 properly.
Spotted by sparse, a warning like:
sound/pci/bt87x.c:240:17: warning: incorrect type in assignment (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many data fields defined in echoaudio drivers are in little-endian,
hence they should be defined with __le16 or __le32. This makes it
easier to catch the forgotten conversions.
Spotted by sparse, a warning like:
sound/pci/echoaudio/echoaudio_dsp.c:990:36: warning: incorrect type in assignment (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASSP data passed to maestro3 driver is in little-endian format,
hence the data pointer should be with __le16.
Spotted by sparse, warnings like:
sound/pci/maestro3.c:2128:35: warning: cast to restricted __le16
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The BD address tables in intel8x0m driver are in little-endian, hence
they should be represented as __le32 instead u32.
Spotted by sparse, warnings like:
sound/pci/intel8x0m.c:406:40: warning: incorrect type in assignment (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The BD address tables in intel8x0 driver are in little-endian, hence
they should be represented as __le32 instead u32.
Spotted by sparse, warnings like:
sound/pci/intel8x0.c:688:40: warning: incorrect type in assignment (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The BDL entries in lola driver are little-endian while we code them as
u32. This leads to sparse warnings like:
sound/pci/lola/lola.c:105:40: warning: incorrect type in assignment (different base types)
sound/pci/lola/lola.c:105:40: expected unsigned int [unsigned] [usertype] <noident>
sound/pci/lola/lola.c:105:40: got restricted __le32 [usertype] <noident>
This patch fixes the declarations to the proper __le32 type.
Also, there was a typo in the original code, where __user was used
that was intended as __iomem. This was caused also by sparse:
sound/pci/lola/lola_mixer.c:132:27: warning: incorrect type in assignment (different address spaces)
Fixed in this patch as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The miXart driver deals with big-endian values as raw data, while it
declares most of variables as u32. This leads to sparse warnings like
sound/pci/mixart/mixart.c:1203:23: warning: cast to restricted __be32
Fix them by properly defining the structs and add the explicit cast to
macros.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SG descriptor of Riptide contains the little-endian values, hence
we need to define with __le32 properly. This fixes sparse warnings
like:
sound/pci/riptide/riptide.c:1112:40: warning: cast to restricted __le32
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The BDL pointer used in snd_hdac_dsp_prepare() should be declared as
__le32, as warned by sparse:
sound/hda/hdac_stream.c:655:47: warning: incorrect type in argument 4 (different base types)
sound/hda/hdac_stream.c:655:47: expected restricted __le32 [usertype] **bdlp
sound/hda/hdac_stream.c:655:47: got unsigned int [usertype] **<noident>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The TLB entries in Trident driver are represented in little-endian,
hence they should be declared as __le32.
This patch fixes the sparse warnings like:
sound/pci/trident/trident_memory.c:226:17: warning: incorrect type in assignment (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bank values are all little-endians, so they should be defined with
__le32. This fixes lots of sparse warnings like:
sound/pci/ymfpci/ymfpci_main.c:315:23: warning: cast to restricted __le32
sound/pci/ymfpci/ymfpci_main.c:342:32: warning: incorrect type in assignment (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The open codes with the bit shift in xen_snd_front_alsa.c give sparse
warnings as the PCM format type is with __bitwise.
There is already a standard macro to get the format bits, so let's use
it instead.
This fixes sparse warnings like:
sound/xen/xen_snd_front_alsa.c:191:47: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM format type is with __bitwise, and it can't be converted from
integer implicitly. Instead of an ugly cast, declare the function
argument of snd_sb_csp_autoload() with the proper snd_pcm_format_t
type.
This fixes the sparse warnings like:
sound/isa/sb/sb16_csp.c:743:22: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM format type in snd_pcm_format_t can't be treated as integer
implicitly since it's with __bitwise. We have already a helper
function to get the bit index of the given type, and use it in each
place instead.
This fixes sparse warnings like:
sound/isa/sb/sb16_main.c:61:44: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM format type is with __bitwise, and it can't be converted from
integer implicitly. Instead of an ugly cast, declare the function
argument of snd_wss_get_format() with the proper snd_pcm_format_t
type.
This fixes the sparse warnings like:
sound/isa/wss/wss_lib.c:551:14: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
asihpi driver treats -1 as an own invalid PCM format, but this needs
a proper cast with __force prefix since PCM format type is __bitwise.
Define a constant with the proper type and use it allover.
This fixes sparse warnings like:
sound/pci/asihpi/asihpi.c:315:9: warning: incorrect type in initializer (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM format type is with __bitwise, and it can't be converted from
integer implicitly. Instead of an ugly cast, declare the function
argument of vortex_alsafmt_aspfmt() with the proper snd_pcm_format_t
type.
This fixes the sparse warning like:
sound/pci/au88x0/au88x0_core.c:2778:14: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM format type is with __bitwise, and it can't be converted from
integer implicitly. Instead of an ugly cast, declare the function
argument of snd_ad1816a_get_format() with the proper snd_pcm_format_t
type.
This fixes the sparse warning like:
sound/isa/ad1816a/ad1816a_lib.c:93:14: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM format type is with __bitwise, hence it needs the explicit
cast with __force. It's ugly, but there is a reason for that cost...
This fixes the sparse warning:
sound/core/oss/pcm_oss.c:1854:55: warning: incorrect type in argument 1 (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM format type is with __bitwise, hence it needs to be explicitly
declared as snd_pcm_format_t, as warned by sparse:
sound/pci/riptide/riptide.c:1028:34: warning: incorrect type in argument 1 (different base types)
sound/pci/riptide/riptide.c:1028:34: expected restricted snd_pcm_format_t [usertype] format
sound/pci/riptide/riptide.c:1028:34: got unsigned char [unsigned] format
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM format type is defined with __bitwise, hence it can't be
passed as integer but needs an explicit cast. In this patch, instead
of the messy cast flood, define the format argument of
snd_hdac_calc_stream_format() to be the proper snd_pcm_format_t type.
This fixes sparse warnings like:
sound/hda/hdac_device.c:760:38: warning: incorrect type in argument 1 (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The virmidi output trigger tries to parse the all available bytes and
process sequencer events as much as possible. In a normal situation,
this is supposed to be relatively short, but a program may give a huge
buffer and it'll take a long time in a single spin lock, which may
eventually lead to a soft lockup.
This patch simply adds a workaround, a cond_resched() call in the loop
if applicable. A better solution would be to move the event processor
into a work, but let's put a duct-tape quickly at first.
Reported-and-tested-by: Dae R. Jeong <threeearcat@gmail.com>
Reported-by: syzbot+619d9f40141d826b097e@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The conversion from PCM format type to bits needs an explicit cast,
and it'll be uglier. Since we have a standard macro for that, let's
use it instead.
This patch fixes the sparse warning:
sound/soc/soc-generic-dmaengine-pcm.c:200:63: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The PCM format type is with __bitwise, so we should use the dedicated
snd_pcm_format_t instead of int.
This fixes the sparse warning like:
sound/soc/codecs/pcm186x.c:268:44: warning: incorrect type in initializer (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the new helper function snd_mask_set_format() for avoiding the
ugly cast with __force prefix.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
As sparse warns, the PCM format type can't be dealt as integer as
found in Intel SST driver codes.
Fix them in the following two ways:
- The open code with snd_mask_set() and params->masks reference is
replaced with params_set_format()
- The rest codes with snd_mask_set(fmt, SNDRV_PCM_FORMAT_XXX) are
replaced with the new helper, snd_mask_set_format().
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
fmt in snd_soc_dai_link_event() contains the format bit position, not
the format bit itself. Hence it can be a simple integer instead of
the explicit u64.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
It seems that __user prefix was forgotten to be added to
dmaengine_copy_user callback while we refactored the user-copy PCM
core.
This patch adds the missing prefix, remove the superfluous cast, and
add the needed cast (__force is needed for downgrading from user
pointer to kernel pointer), too.
Spotted by a sparse warning like:
sound/soc/soc-generic-dmaengine-pcm.c:397:27: warning: incorrect type in initializer (incompatible argument 4 (different address spaces))
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
zx-tdm driver sets the DAI driver definitions with the format bits
wrongly set with SNDRV_PCM_FORMAT_*, instead of SNDRV_PCM_FMTBIT_*.
This patch corrects the definitions.
Spotted by a sparse warning:
sound/soc/zte/zx-tdm.c:363:35: warning: restricted snd_pcm_format_t degrades to integer
Fixes: 870e0ddc43 ("ASoC: zx-tdm: add zte's tdm controller driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop amlogic prefix in front of the generic DT properties and change
property "name" to "model".
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use IRQ_RETVAL instead of the open coded ternary operation.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes
sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe':
sound/soc/amd/acp-da7219-max98357a.c:367:3: warning: 'ret' may
be used uninitialized in this function [-Wmaybe-uninitialized]
dev_err(&pdev->dev, "Failed to register regulator: %d\n",
ret);
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DA7219 for our platform need to be configured for 1.8V.
Hence, we add a volatge regulator with supplies
of 1.8V in the machine driver.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The meddlesome gcc warns about the possible shortname string in
trident driver code:
sound/pci/trident/trident.c: In function ‘snd_trident_probe’:
sound/pci/trident/trident.c:126:2: warning: ‘strcat’ accessing 17 or more bytes at offsets 36 and 20 may overlap 1 byte at offset 36 [-Wrestrict]
strcat(card->shortname, card->driver);
It happens since gcc calculates the possible string size from
card->driver, but this can't be true since we did set the string just
before that, and they are much shorter.
For shutting it up, use the exactly same string set to card->driver
for strcat() to card->shortname, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cast between user-space and kernel-space needs an explicit __force
prefix, but it's missing in many places in emu10k1 driver code.
Spotted by sparse as a warning like:
sound/pci/emu10k1/emufx.c:529:33: warning: cast removes address space of expression
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The user-copy callbacks in korg1212 driver contain the explicit cast
from a user pointer to a kernel pointer, but they missed __force
prefix. It's mandatory for converting between them.
Spotted by sparse, a warning like:
sound/pci/korg1212/korg1212.c:1329:33: warning: cast removes address space of expression
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_usb_feature_unit_ctl properly in mixer.h. Otherwise it's
error-prone.
This fixes the sparse warning:
sound/usb/mixer.c:1464:25: warning: symbol 'snd_usb_feature_unit_ctl' was not declared. Should it be static?
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the declarations of common variables into opl3_voice.h instead of
declaring at each file multiple times, which was error-prone.
This fixes sparse warnings like:
sound/drivers/opl3/opl3_synth.c:51:6: warning: symbol 'snd_opl3_regmap' was not declared. Should it be static?
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The knew->iface field is in snd_ctl_elem_iface_t, which is with
__bitwise, hence it can't be converted implicitly from integer.
Give an explicit cast for the invalid type.
Spotted by sparse:
sound/pci/hda/hda_codec.c:3280:25: warning: restricted snd_ctl_elem_iface_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use NULL for initializing the snd_kcontrol_new.tlv field, instead of
0, as warned by sparse:
sound/pci/hda/patch_ca0132.c:5519:22: warning: Using plain integer as NULL pointer
Also, the driver does the same initialization twice, once for
knew.tlv.c and another for knew.tlv.p while both point to the same
address (these are union). Drop the latter superfluous one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The io-mapped buffers used in msnd drivers need __iomem annotations.
This fixes sparse warnings like:
sound/isa/msnd/msnd_pinnacle.c:172:45: warning: incorrect type in initializer (different address spaces)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AU0828_DEVICE() macro in quirks-table.h uses USB_DEVICE_VENDOR_SPEC()
for expanding idVendor and idProduct fields. However, the latter
macro adds also match_flags and bInterfaceClass, which are different
from the values AU0828_DEVICE() macro sets after that.
For fixing them, just expand idVendor and idProduct fields manually in
AU0828_DEVICE().
This fixes sparse warnings like:
sound/usb/quirks-table.h:2892:1: warning: Initializer entry defined twice
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD playback support for the Encore mDSD USB DAC by
specifying the vendor and product ID's
Signed-off-by: Jeff Crukley <jcrukley@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One place in cs5535audio_build_dma_packets() does an extra conversion
via cpu_to_le32(); namely jmpprd_addr is passed to setup_prd() ops,
which writes the value via cs_writel(). That is, the callback does
the conversion by itself, and we don't need to convert beforehand.
This patch fixes that bogus conversion.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The endian conversions used in vxp_dma_read() and vxp_dma_write() are
superfluous and even wrong on big-endian machines, as inw() and outw()
already do conversions. Kill them.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The endian conversions used in vx2_dma_read() and vx2_dma_write() are
superfluous and even wrong on big-endian machines, as inl() and outl()
already do conversions. Kill them.
Spotted by sparse, a warning like:
sound/pci/vx222/vx222_ops.c:278:30: warning: incorrect type in argument 1 (different base types)
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sanity checks in ALSA sequencer and OSS sequencer emulation codes
return falsely -ENXIO from poll callback. They should be EPOLLERR
instead.
This was caught thanks to the recent change to the return value.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
include DAPM Mux and output widgets into the list.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This looks like a copy/paste issue, but clearly there is an inversion
that is obvious when checking the arguments.
Detected with Sparse - now that we have fewer warnings this one was
easy to find.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Simplify code and add relevant casts to make Sparse warnings go away
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sst_dma_new and sst_dma_free are not used in any other file and don't
have a prototype. Move to static functions and remove
EXPORT_SYMBOL_GPL statement.
Reported by sparse warnings.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make sure definitions are consistent with usage.
Detected with Sparse.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make all Sparse warnings go away by using le16/32_to_cpu.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On Qualcomm platforms, specifically with SLIMbus interfaced codecs,
the codec slim channel numbers are passed to DSP while configuring
the slim audio path. Having get_channel_map() would allow dais to
share such information across multiple dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch fixes the issue of the delay volume applied.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When CONFIG_SND_PCM_IEC958 is disabled, we get a link error for the
new driver:
sound/soc/meson/axg-spdifout.o: In function `axg_spdifout_hw_params':
axg-spdifout.c:(.text+0x650): undefined reference to `snd_pcm_create_iec958_consumer_hw_params'
The other users use 'select', so we should do the same here.
Fixes: 53eb4b7aaa ("ASoC: meson: add axg spdif output")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently HD-audio i915 audio binding doesn't support any delayed
binding, and supposes that the i915 driver registers the component
immediately. This has been OK, so far, but the work-in-progress
change in i915 may introduce the asynchronous binding, which
effectively delays the component registration.
For addressing it, implement a completion to be synced with the master
binding. The timeout is set to 10 seconds which should be long enough
and hopefully be not too annoying if anyone boots up a debugging
session with i915 KMS turned off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Thesycon provides solutions to XMOS chips, and has its own device
vendor id.
In this patch, we use generic method to detect DSD capability of
Thesycon-based UAC2 implementations in order to support a wide range
of current and future devices.
The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC
hence enable native DSD playback up to DSD512 format.
Signed-off-by: Yue Wang <yuleopen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_dma_alloc_pages_fallback() tries to allocate pages again when the
allocation fails with reduced size. But the first try actually
*increases* the size to power-of-two, which may give back a larger
chunk than the requested size. This confuses the callers, e.g. sgbuf
assumes that the size is equal or less, and it may result in a bad
loop due to the underflow and eventually lead to Oops.
The code of this function seems incorrectly assuming the usage of
get_order(). We need to decrease at first, then align to
power-of-two.
Reported-and-tested-by: he, bo <bo.he@intel.com>
Reported-by: zhang jun <jun.zhang@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A timer object for the classes SNDRV_TIMER_CLASS_CARD and
SNDRV_TIMER_CLASS_PCM has to be associated with a card object, but we
have no check at creation time. Such a timer object with NULL card
causes various unexpected problems, e.g. NULL dereference at reading
the sound timer proc file.
So as preventive measure while the creating the sound timer object is
created the card information availability is checked for the mentioned
entries and returned error if its NULL.
Signed-off-by: Srikanth K H <srikanth.h@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SSP DAI now handles the clocking setup itself, all it needs is the
master clock frequency. Remove the code from Zylonite and Magician
platforms.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the axg sound card to handle the specifities of the axg audio
sub system.
This card is required to:
* setup the dpcm links specific to the AXG (with a cpu sound dai)
* handle the 4 lanes masks of the tdm interfaces
* add the loopback link when a tdm pad interface has a playback
stream
* handle multi-codec links
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Amlogic's axg card driver can't use snd_soc_of_parse_tdm_slot()
directly because it needs to handle 4 mask for each direction.
Yet the parsing of each mask is the same, so export
snd_soc_of_get_slot_mask() to reuse the the existing code.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg TDM input driver which take the TDM signal of 4 input
lanes and push the decoded audio samples to TODDR fifo
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg tdm output driver which pulls data from FRDDR fifo
and produce the TDM signals for 4 output lanes.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg TDM interface driver. This driver manages the format
and clocks provided on the pads.
On this SoC, each stream direction provides 4 serial lanes. This makes
a maximum of 8 channels in i2s modes and 128 channels in DSP modes.
While each lanes operate on the same slot number (same bit clock), they
may have different TDM masks. This requires to provide a function to let
the card set the 4 masks, in lieu of the usual set_tdm_slots() callback
of the dai driver.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg TDM core driver. On this SoC, tdm is bit more
complex than usual, mainly because the different TDM input decoders can
be attached to any of TDM pad interface, including the output pads.
For the this, TDM on this SoC is modeled like this:
- TDM interface provides the DAIs the codecs will be attached to.
The main responsibility of this driver is to manage the pad format
and the TDM clock rates.
- TDM Formatters: These are the entities which are actually dealing with
the TDM signal. TDMOUT produce a TDM signal from the audio sample
provided by FRDDR using the clocks provided the TDM interface. TDMIN
feeds TODDR with audio sample using the clocks and TDM signal provided
by the TDM Interface.
- TDM Streams: This provides the link between 1 DAI stream of the TDM
interface and one (or more) TDM formatters.
This driver provides the TDM formatter and TDM stream operations.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
While the two error labels "err" and "err_clk_put" goto the same place
it is rather confusing that the earlier one is certainly used later
again.
Signed-off-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set the coherent_dma_mask for the PS3 ehci, ohci, and snd devices.
Silences WARN_ON_ONCE messages emitted by the dma_alloc_attrs() routine.
Reported-by: Fredrik Noring <noring@nocrew.org>
Signed-off-by: Geoff Levand <geoff@infradead.org>
Acked-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Acked-by: Alan Stern <stern@rowland.harvard.edu>
Signed-off-by: Michael Ellerman <mpe@ellerman.id.au>
Pull the generic drm_audio_component support, which will be used later
for AMD/ATI and other HD-audio HDMI codec drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for the "Connect Tablet 9" tablet, this tablet has a
mono-speaker. Otherwise it works fine with the defaults.
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 quirk table entries for the following tablets:
ITWorks TW701
Ployer Momo7w
Trekstor win7
Yours 8"
These all use the default settings, except that they only have a single
speaker and thus need the mono-speaker quirk.
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>
During my initial round of bytcr_rt5651 long-name patches I did not include
a difference for mono vs stereo speaker setups in the longname because it
seems that all 5651 devices with only a single speaker do some mixing of
left + right on the PCB.
However further testing has shown that while this works great when only
playing audio on the left or right channel, the output becomes garbled
when using both channels at once. Something which does not happen when
using the Stereo DAC MIXL / MIXR switches to mix the channels together
inside the codec and then only outputting on a single channel.
So we need to have separate UCM profiles and thus separate long-names
for devices with a mono speaker vs stereo speakers. Just as we already
have for the bytcr_rt5640 case.
This commit adds a new BYT_RT5651_MONO_SPEAKER quirk and adds "stereo-spk"
or "mono-spk" to the long-name based on this and enables this mapping on
devices with a mono speaker.
Changing the long-name like this is ok for now, since I'm still working
on the UCM profiles, so they are not in upstream alsa-lib yet.
This brings the long-name naming scheme fully in sync with the bytcr_rt5640
case, which is good from a consistency pov.
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>
During the recent cleanup series 3 of the 6 input mappings where removed
from the bytcr_rt5651 machine driver because testing showed that none of
them were used.
However some devices do actually have their internal mic on IN2 (and
only IN2, not IN1 and IN2), this did not show during previous tests
due to a bug in the userspace UCM input device switching code.
This commit re-adds the IN2 mapping for devices with the internal mic.
on IN2 and the headser mic on IN3 and enables this mapping on devices
with their internal mic on IN2.
This commit also changes the default internal mic input to IN2, because
all my 7 test devices have their mic there.
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>
With the default over current detect limit of 1500uA headsets on often
get detected as headphones on the VIOS LTH17 and even when detected as
headset the OVCD current triggers often while plugged in, resulting in
false-positive button press detection.
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 boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other
boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the
GpioIo one for the ext-amp-enable-gpio.
So far we've been assuming that the GpioIo one always comes first, this
commit adds code to detect which one comes first and to add the right
gpio-mapping.
This fixes sound not working on the Vios LTH17 laptop.
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 mixer control for the IN3 Boost volume, IN3 is used for the headset
mic on most devices, so this is necessary to control the headset mic
volume.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the compressed streams in DSP firmwares are
identified essentially by looking at a fixed location inside
the firmware. This is fragile and also limits things to a
single compressed stream.
Here a new form of firmware parameter is added, the HOST_BUFFER
which identifies a compressed stream from meta-data in the
firmware file. This is more robust and allows for the possiblity
of using multiple streams per core in the future. Currently the
implementation is still limited to a single stream and will
use the first HOST_BUFFER parameter encountered. If there aren't
any HOST_BUFFER parameters it will fall back to the legacy way
of finding the host buffer.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Newer voice control firmwares can capture multiple audio channels.
Allow up to 8 channels for future-proofing.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when creating ALSA control names for the DSP the length of any
prefix applied to the CODEC is not taken into account. Whilst this is
mostly harmless it does result in ALSA doing the truncation of the
control names and printing a warning. It is better to have the driver do
the truncation so it can truncate from the start of parameter name
itself to give a greater chance of the result maintain a unique name.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 6396bb2215 ("treewide: kzalloc() -> kcalloc()") was
overlooked when doing some refactoring to the algorithm list
handling, which lead to twice as much buffer being allocated
as required for reading the algorithm list. A kcalloc is no
longer appropriate since the allocation size is now in bytes
not registers, as such change back to kzalloc.
Fixes: 7f7cca08ab ("ASoC: wm_adsp: Simplify handling of alg offset and length")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
All controls derived from the loaded firmware should be created prior
to returning from the preloader's put function, such that they are
immediately available to user-space.
Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the audio device is externally clocked and set to a rate that does
not match the external clock, the clock will never be valid and we cannot
set the rate successfully. To fix this, allow a rate change even if
the clock is initially invalid, and validate again after the rate is
changed.
This fixes problems with MOTU UltraLite AVB hardware over USB.
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the spdif output serializer of the axg SoC family
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the capture memory interface of Amlogic's axg SoCs.
TDM, SPDIF or PDM input devices place audio samples inside this FIFO.
The FIFO content is then pushed to DDR
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the playback memory interface of Amlogic's axg SoCs.
This device pulls data from DDR to an internal FIFO.
This FIFO is then used to feed TDM and SPDIF Output devices.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Amlogic's axg SoCs have two types of fifos which are the memory
interfaces of the audio subsystem. FRDDR provides the playback
interface while TODDR provides the capture interface.
The way these fifos operate is very similar. Only a few settings
are specific to each.
They implement the same pcm driver here and the specifics of each
will be dealt with the related DAI driver.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add documentation for power management of HDAC HDMI codec device for
various scenarios such as S0/S3, probe and playback use case.
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Sanyog Kale <sanyog.r.kale@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
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 aims at achieving dynamic behaviour of audio card when
the dependent components disappear and reappear.
With this patch the card is removed if any of the dependent component
is removed and card is added back if the dependent component comes back.
All this is done using component framework and matching based on
component name.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The PCI SSID 1558:95e1 needs the same quirk for other Clevo P950
models, too. Otherwise no sound comes out of speakers.
Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=1101143
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_mmap_vmalloc() was supposed to be implemented with
somewhat special for vmalloc handling, but in the end, this turned to
just the default handler, i.e. NULL. As the situation has never
changed over decades, let's rip it off.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The size of in-kernel rawmidi buffers may be big up to 1MB, and it can
be specified freely by user-space; which implies that user-space may
trigger kmalloc() errors frequently.
This patch replaces the buffer allocation via kvmalloc() for dealing
with bigger buffers gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unify a few open codes with helper functions to improve the
readability. Minor behavior changes (rather fixes) are:
- runtime->drain clearance is done within lock
- active_sensing is updated before resizing buffer in
SNDRV_RAWMIDI_IOCTL_PARAMS ioctl.
Other than that, simply code cleanups.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply the standard idiom: rewrite the multiple unlocks in error paths
in the goto-error-and-single-unlock way.
Just a code refactoring, and no functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just minor coding style fixes like removal of superfluous white space,
adding missing blank lines, etc. No actual code changes at all.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the final step for more generic support of DRM audio
component. The generic audio component code is now moved to its own
file, and the symbols are renamed from snd_hac_i915_* to
snd_hdac_acomp_*, respectively. The generic code is enabled via the
new kconfig, CONFIG_SND_HDA_COMPONENT, while CONFIG_SND_HDA_I915 is
kept as the super-class.
Along with the split, three new callbacks are added to audio_ops:
pin2port is for providing the conversion between the pin number and
the widget id, and master_bind/master_unbin are called at binding /
unbinding the master component, respectively. All these are optional,
but used in i915 implementation and also other later implementations.
A note about the new snd_hdac_acomp_init() function: there is a slight
difference between this and the old snd_hdac_i915_init(). The latter
(still) synchronizes with the master component binding, i.e. it
assures that the relevant DRM component gets bound when it returns, or
gives a negative error. Meanwhile the new function doesn't
synchronize but just leaves as is. It's the responsibility by the
caller's side to synchronize, or the caller may accept the
asynchronous binding on the fly.
v1->v2: Fix missing NULL check in master_bind/unbind
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HD-audio i915 binding code contains a single pointer, hdac_acomp,
for allowing the access to audio component from the master bind/unbind
callbacks. This was needed because the callbacks pass only the device
pointer and we can't guarantee the object type assigned to the drvdata
(which is free for each controller driver implementation).
And this implementation will be a problem if we support multiple
components for different DRM drivers, not only i915.
As a solution, allocate the audio component object via devres and
associate it with the given device, so that the component callbacks
can refer to it via devres_find().
The removal of the object is still done half-manually via
devres_destroy() to make the code consistent (although it may work
without the explicit call).
Also, the snd_hda_i915_register_notifier() had the reference to
hdac_acomp as well. In this patch, the corresponding code is removed
by passing hdac_bus object to the function, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For allowing other drivers to use the DRM audio component, rename the
i915_audio_component_* with drm_audio_component_*, and split the
generic part into drm_audio_component.h. The i915 specific stuff
remains in struct i915_audio_component, which contains
drm_audio_component as the base.
The license of drm_audio_component.h is kept to MIT as same as the the
original i915_component.h.
This is a preliminary change for further development, and no
functional changes by this patch itself, merely code-split and
renames.
v1->v2: Use SPDX for drm_audio_component.h, fix remaining i915
argument in drm_audio_component.h
Reviewed-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SNDRV_RAWMIDI_IOCTL_PARAMS ioctl may resize the buffers and the
current code is racy. For example, the sequencer client may write to
buffer while it being resized.
As a simple workaround, let's switch to the resized buffer inside the
stream runtime lock.
Reported-by: syzbot+52f83f0ea8df16932f7f@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make use of the swap macro and remove unnecessary variable *tmp*. This
makes the code easier to read and maintain.
This code was detected with the help of Coccinelle.
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make use of the swap macro and remove unnecessary variable *tmp*. This
makes the code easier to read and maintain.
This code was detected with the help of Coccinelle.
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On modern laptop, there are more and more platforms
have two GPUs, and each of them maybe have audio codec
for HDMP/DP output. For some dGPU which is no output,
audio codec usually is disabled.
In currect HDA audio driver, it will set all codec as
VGA_SWITCHEROO_DIS, the audio which is binded to UMA
will be suspended if user use debugfs to contorl power
In HDA driver side, it is difficult to know which GPU
the audio has binded to. So set the bound gpu pci dev
to vga_switcheroo.
if the audio client is not the third registration, audio
id will set in vga_switcheroo enable function. if the
audio client is the last registration when vga_switcheroo
_ready() get true, we should get audio client id from bound
GPU directly.
Signed-off-by: Jim Qu <Jim.Qu@amd.com>
Reviewed-by: Lukas Wunner <lukas@wunner.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds some required quirk when uses headset or headphone on
Panasonic CF-SZ6.
Signed-off-by: YOKOTA Hiroshi <yokota.hgml@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audio mute led does not work on HP ProBook 455 G5,
this can be fixed by using CXT_FIXUP_MUTE_LED_GPIO to support it.
BugLink: https://bugs.launchpad.net/bugs/1781763
Reported-by: James Buren
Signed-off-by: Po-Hsu Lin <po-hsu.lin@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Except PCI_CLASS_DISPLAY_VGA, some PCI class is sometimes
PCI_CLASS_DISPLAY_3D or PCI_CLASS_DISPLAY_OTHER.
Signed-off-by: Jim Qu <Jim.Qu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch refactors the processing units min/max calculation logic
for the mixer controls and fixes an issue where the Mode Select
checking of the Up/Down mixers doesn't differentiate between the
UAC1 and UAC2 Control Selector (0x02) and the UAC3 one which is
different (0x01).
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Audio Control interface descriptor subtypes do not match
across all the UAC versions. That makes reusability of the
"virtual type" (Mixer, Processors, Selectors, etc) terminals
difficult. It also makes the mixer get the default names for
the virtual terminals wrong due to the overlap.
This patch proposes an unified approach by always using the most
comprehensive spec version to define them all (in this case UAC3).
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the Processig Units defined in
the UAC3 spec. The main difference with the previous specs
is the lack of on/off switches in the controls for these
units and the addiction of the new Multi Function Processing
Unit.
The current version of the UAC3 spec doesn't define any
useful controls for the new Multi Function Processing Unit
so no control will get created once this unit is parsed.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current support for UAC2 Processing Units does the parsing
as one control per bit in the bitmap. However, the UAC2 spec
defines the controls as bit pairs where b01 means read-only
and b11 means read/write control.
This patch fixes that and uses the helper functions for checking
controls readability/writability when the control is defined as
bit pairs (UAC2 and UAC3).
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add support for Selector Units and Clock Selector Units
defined in the new UAC3 spec.
Selector Units play a really important role in the new UAC3 spec as
Processing Units do not define an on/off switch control anymore.
This forces topology designers to add bypass paths in the topology
to enable/dissable the Processing Units.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Having interrupts enabled for ACP<->SYSMEM DMA transfer, we are in
for an interrupt storm.
For both playback and capture interrupts should be enabled for
I2S<->ACP DMA.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Earlier, ch1 was used to define ACP-SYSMEM transfer and ch2 for
ACP-I2S transfer. With recent patches ch1 is used to define channel
order number 1 and ch2 as channel order number 2. Thus,
Playback:
ch1:SYSMEM->ACP
ch2:ACP->I2S
Capture:
ch1:I2S->ACP
ch1:ACP->SYSMEM
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 943fa02282 ("ASoC: hdmi-codec: Use different name for playback
streams") broke hdmi-codec's routing between it's output "TX" widget
and the S/PDIF or I2S streams by renaming the streams.
Whether an error occurs or not is dependent on whether there is another
widget called "Playback" registered by some other component - if there
is, that widget will be (incorrectly) bound to the HDMI codec's "TX"
output widget. If we end up connecting "TX" incorrectly, it can result
in components not being started, causing no audio output.
Since the I2S and S/PDIF streams now have different names, we can't
use a static route at component level to describe the relationship, so
arrange to dynamically create the route when the DAI driver is probed.
Fixes: 943fa02282 ("ASoC: hdmi-codec: Use different name for playback streams")
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pointer 'ins' is being assigned but is never used hence it is
redundant and can be removed.
Cleans up clang warning:
warning: variable 'ins' set but not used [-Wunused-but-set-variable]
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pointer 'codec' is being assigned but is never used hence it is
redundant and can be removed.
Cleans up clang warning:
warning: variable 'codec' set but not used [-Wunused-but-set-variable]
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pointer runtime is being assigned but is never used hence it is
redundant and can be removed.
Cleans up clang warning:
warning: variable 'runtime' set but not used [-Wunused-but-set-variable]
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pointer private_data is being assigned but is never used hence it is
redundant and can be removed.
Cleans up clang warning:
warning: variable 'private_data' set but not used [-Wunused-but-set-variable]
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pointer chip is being assigned but is never used hence it is
redundant and can be removed.
Cleans up clang warning:
warning: variable 'chip' set but not used [-Wunused-but-set-variable]
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Variable opl3 is being assigned but is never used hence it is
redundant and can be removed.
Cleans up several clang warnings:
warning: variable 'opl3' set but not used [-Wunused-but-set-variable]
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The data->port_data[] array has AFE_MAX_PORTS elements so the check
should be >= instead of > or we write one element beyond the end of the
array.
Fixes: e3a33673e8 ("ASoC: qdsp6: q6routing: Add q6routing driver")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The main thing is that the data->priv[] array has AFE_PORT_MAX elements
so the > condition should be >=. But we may as well check for negative
values as well just to be safe.
Fixes: 24c4cbcfac ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the component does not match the configuration table provided
by the card, let soc-core check the component node for a name prefix
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The PCI subsystem in question for this quirk rule has been
identified as a Gigabyte GA-Z170X-Gaming 7 motherboard. Set the
device name appropriately.
Signed-off-by: Alastair Bridgewater <alastair.bridgewater@gmail.com>
Reviewed-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These motherboards have Sound Core3D and apparently "support"
Recon3Di. Added to the quirk list as QUIRK_R3DI.
Issue report, PCI Subsystem ID, and testing by a contributor on
IRC who wished to remain anonymous.
Signed-off-by: Alastair Bridgewater <alastair.bridgewater@gmail.com>
Reviewed-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As done for format and channels, add the possibility to merge
the backend rates on the frontend rates.
This useful if the backend does not support all rates supported by the
frontend, or if several backends (cpu and codecs) with different
capabilities are connected to the same frontend.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The goal of this patch is to simplify a bit dpcm runtime stream merge
by removing several local variables.
ATM, merge functions return the BE 'filter' values which should then be
filtered against the FE stream values. This create a lot of local
variable and unnecessary init of min and max.
Instead of this, we can pass the FE stream values directly and let the
BE filtering functions perform the merge 'in-place'
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Without this commit the following intervals [x y), (x y) were be
replaced to (y-1 y) by snd_interval_refine_last(). This was also done
if y-1 is part of the previous interval.
With this changes it will be replaced with [y-1 y) in case of y-1 is
part of the previous interval. A similar behavior will be used for
snd_interval_refine_first().
This commit adapts the changes for alsa-lib of commit
9bb985c ("pcm: snd_interval_refine_first/last: exclude value only if
also excluded before")
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable reporting of button presses now that the codec driver recently has
gotten support 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>
Disable jack-detection and thus the codec IRQ over suspend/resume.
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>
Enable button press detection for headsets by using the ovcd IRQ to get
notified of button presses.
This is modelled after (almost exactly copied from) the button press code
for the rt5640 which has identical ovcd hardware.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow the machine driver to disable jack-detect over a suspend/resume by
calling snd_soc_component_set_jack(NULL).
Note this renames rt5651_set_jack, where all the jack-enable work was done
to rt5651_enable_jack_detect. This function can now no longer fail as it
does not request the IRQ anymore. It can still be passed an invalid jack
source, but that should never happen, so this is now logged and treated as
no jack source.
Cc: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On removal we must free the IRQ *before* cancelling the jack-detect work,
so that the jack-detect work cannot be rescheduled by the IRQ.
Before this commit we were cancelling the jack-detect work from the
driver remove callback, while relying on devm to free the IRQ, which
happens after the remove callback.
This is the wrong order. This commit uses a devm-action to register
a devm callback which cancels the work, before requesting the IRQ
(devm tears things down in reverse order). This also allows us to
remove the now empty remove driver callback.
Cc: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rt5651 does not have a built-in speaker amplifier, so it is often
used together with an external amplifier. On Cherry Trail boards this
external amplifier's enable pin is driven through a GPIO, which is
given as the first GPIO in the ACPI resources of the codec fwnode.
This commit adds support to the bytcr_rt5651 for this GPIO, fixing
the speaker not working on CHT devices with a rt5651 codec.
Cc: Carlo Caione <carlo@endlessm.com>
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>
Move the getting of the codec_dev, to add device-props to it, out of
byt_rt5651_add_codec_device_props() and into its caller,
snd_byt_rt5651_mc_probe().
This is a preparation patch for adding support for an external amplifier
enable GPIO, which requires further accesses to the codec_dev.
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>
Remove is_valleyview helper, this is not necessary, we can simply call
x86_match_cpu() directly instead.
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 Lenovo Miix2 8 tablet, this tablet uses a digital
mic on DMIC1 and has a mono-speaker. The jack-detect uses the default
settings..
Reported-and-tested-by: russianneuromancer@ya.ru
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>
'cmd_pause' DMA channel capability means that respective DMA engine
supports both pausing and resuming given DMA channel. However, in some
cases it is important to know if DMA channel can be paused without the
need to resume it. This is a typical requirement for proper residue
reading on transfer timeout in UART drivers. There are also some DMA
engines with limited hardware, which doesn't really support resuming.
Reporting pause and resume capabilities separately allows UART drivers to
properly check for the really required capabilities and operate in DMA
mode also in systems with limited DMA hardware. On the other hand drivers,
which rely on full channel suspend/resume support, should now check for
both 'pause' and 'resume' features.
Existing clients of dma_get_slave_caps() have been checked and the only
driver which rely on proper channel resuming is soc-generic-dmaengine-pcm
driver, which has been updated to check the newly added capability.
Existing 'cmd_pause' now only indicates that DMA engine support pausing
given DMA channel.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
The playback DAI is connected to the DSP and the DSP might be sourcing
signals from the playback stream. Add a DAPM route between the two to make
sure that the playback DAI is powered up, when the DSP is active.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Alexandru Ardelean <alexandru.ardelean@analog.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For the moment, we can't enable CONFIG_SND_PXA_SOC_SSP unless we are
building for ARM PXA or MMP:
WARNING: unmet direct dependencies detected for PXA_SSP
Depends on [n]: PLAT_PXA [=n]
Selected by [y]:
- SND_PXA_SOC_SSP [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y]
This adds an explicit dependency for it.
Fixes: 0a94cf3457 ("ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The format specifier "%p" can leak kernel addresses.
Use "%pK" instead.
Signed-off-by: Benjamin Gaignard <benjamin.gaignard@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO.
This needs to be configurable for modern hardware like DSPs where no
pointer update in milliseconds can indicate terminal DSP errors.
Add a substream variable to set the wait time in ms. This allows userspace
and drivers to recover more quickly from terminal DSP errors.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have two new lenovo desktop models which need to apply the fixup of
ALC294_FIXUP_LENOVO_MIC_LOCATION, and they have the same pin cfg as
the machine with subsystem id:0x17aa3136, now use the pincfg table
to apply the fixup for them.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.
Notice that such constant is used in a context that expects an
expression of type u64 (64 bits, unsigned) and the following
expression is currently being evaluated using 32-bit arithmetic:
256 * fs * 2 * mclk_src_scaling[i].param
Addresses-Coverity-ID: 1432039 ("Unintentional integer overflow")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow setting a clock called 'extclk' in the device of the ssp-dai
device. If specified, this clock will be set to the mclk rate from the
DAI's .set_sysclk() callback. The DAI will also configure itself to
use that external clock.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
To comply with the style of all kernel messages, add newline
to the end of every message.
Fixes: 70fb10529f ("ASoC: rsnd: add MIX (Mixer) support")
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suffix ULL to constant 64 in order to give the compiler complete
information about the proper arithmetic to use.
Notice that such constant is used in a context that expects an
expression of type u64 (64 bits, unsigned) and the following
expression is currently being evaluated using 32-bit arithmetic:
rate[index] * txclk_df * 64
Addresses-Coverity-ID: 1222129 ("Unintentional integer overflow")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
num_channels for slim dais are aready set int set_channel_map,
do not overwrite them in hw_params.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch add routings mixer controls for slim rx ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to SLIMbus TX dais in AFE module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Existing code already has support for SLIMbus TX and RX, only thing
that was missing from TX side was mapping between virtual to actual
DSP port ids.
This patch adds those mappings.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The XRUN trigger from the driver should be done via
snd_pcm_stop_xrun(). It simplifies the locking as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The XRUN trigger from the driver should be done via
snd_pcm_stop_xrun(). It fixes the missing stream locking as a gratis,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The XRUN trigger from the driver should be done via
snd_pcm_stop_xrun(). It fixes the missing stream locking as a gratis,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace open-codes with the standard snd_pcm_stop_xrun() helper.
It simplifies codes a lot.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.
Notice that such constant is used in a context that expects an
expression of type u64 (64 bits, unsigned) and the following
expression is currently being evaluated using 32-bit arithmetic:
256 * fs * 2 * mclk_src_scaling[i].param
Addresses-Coverity-ID: 1339616 ("Unintentional integer overflow")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
asm ports are open as part of prepare, so for use cases like
"aplay sample.wav" were sample.wav is not present. This would
call port close eventhough port was never opened. DSP would
return errors for such use cases.
Avoid doing this by checking the port state.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
afe ports are open as part of prepare, so for use cases like
"aplay sample.wav" were sample.wav is not present. This would
call port close eventhough port was never opened. DSP would
return errors for such use cases.
Avoid doing this by checking the port state.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Basically the xrun injection routine can simply call the standard
helper snd_pcm_stop_xrun(), but with one exception: it may be called
even when the stream is closed.
Make snd_pcm_stop_xrun() more robust and check the NULL runtime state,
and simplify xrun injection code by calling it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM xrun injection triggers directly snd_pcm_stop() without the
standard xrun handler, hence it's not recorded on the event buffer.
Ditto for snd_pcm_stop_xrun() call and SNDRV_PCM_IOCTL_XRUN ioctl.
They are inconvenient from the debugging POV.
Let's make them to trigger XRUN via the standard helper more
consistently.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Machine drivers statically define a number of DAI links that currently
cannot be changed or removed by topology. This means PCMs and platform
components cannot be changed by topology at runtime AND machine drivers
are tightly coupled to topology.
This patch allows topology to override the machine driver DAI link config
in order to reuse machine drivers with different topologies and platform
components. The patch supports :-
1) create new FE PCMs with a topology defined PCM ID.
2) destroy existing static FE PCMs
3) change the platform component driver.
4) assign any new HW params fixups.
5) assign a new card name prefix to differentiate this topology to userspace.
The patch requires no changes to the machine drivers, but does add some
platform component flags that the platform component driver can assign
before loading topologies.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The generated clock (gclk) driver is able to set aclk as its parent and
change its rate alone, if needed. This means that our driver no longer
needs to configure aclk and we can let gclk select and configure its
clock source.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the es7154 which is basically an es7134 with an
embedded power amplifier and lower maximum sample rate
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop AVDD in favor of PVDD to match the names used in the datasheet
and only claim PVDD on the es7154. The es7134 and es7144 don't have
a separate supply for the digital I/O.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the I2S channel names are fixed, and DMA data flow order is
consistent (ch1 then ch2), we can simplify channel start order:
start the upstream channel and then the downstream channel for both
playback and capture cases.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The pointer() callback gets its value by reading the I2S BYTE_COUNT
register. This is a 64-bit runnning transaction counter. If a
transaction was aborted in the middle of a sample buffer, the counter will
stop counting on a number divisible by the buffer size. Since we actually
use it as a pointer into an aligned buffer, however, we do want to ensure
that it always starts at a number divisible by the buffer size when
starting a transaction, hence we reset it whenever starting a transaction.
To accomplish this, it wasn't necessary to zero bytescount at the
termination of each transaction, so remove this unnecessary code.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On capture, audio data is first copied from I2S to ACP memory, and then
from ACP to SYSRAM. The I2S_TO_ACP_DMA interrupt fires on every sample
transferred from I2S to ACP memory. That is it fires ~48000 times per
second when capturing @ 48 kHz. Since we don't do anything on this
interrupt anyway, disable it to save quite a few unnecessary interrupts.
The real "work" (calling snd_pcm_period_elapsed()) is done when transfer
from ACP to SYSRAM is complete.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On capture, audio data is first copied from I2S to ACP memory, and then
to SYSRAM. For each step the channel number increases, so the names in
the driver were wrong.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is always correct to subtract out the starting bytescount value. Even
in the case of 2^64 byte rollover (292 Million Years in the future
@ 48000 Hz) the math still works out.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 6b116dfb46 ("ASoC: AMD: make channel 1 dma as circular") made
both channels circular, so this comment and logic no longer applies. Always
stop ch2 (the channel closest to the output) before ch1. This ensures
that the downstream circular DMA channel does not continue to play/capture
repeated samples after the upstream circular DMA channel has already
stopped.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to enabling -Wimplicit-fallthrough, mark switch cases
where we are expecting to fall through.
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the everest es7241 which is a simple 2 channels
analog to digital converter.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We are currently using power saving mode for button detection.
However, it will impact the headset recording performance.
This patch will switch button detection to normal mode in capture
and switch to power saving mode in the end of capture.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Obtaining the runtime pm wakeref can fail, especially in a hotplug
scenario where i915.ko has been unloaded. If we do not catch the
failure, we end up with an unbalanced pm.
v2 additions by tiwai:
hdmi_present_sense() checks the return value and handle only a
negative error case and bails out only if it's really still suspended.
Also, snd_hda_power_down() is called at the error path so that the
refcount is balanced.
Along with it, the spec->pcm_lock is taken outside
hdmi_present_sense() in the caller side, so that it won't cause
deadlock at reentrace via runtime resume.
v3 fix by tiwai:
Missing linux/pm_runtime.h is included.
References: 222bde0388 ("ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug")
Signed-off-by: Chris Wilson <chris@chris-wilson.co.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB completion callback does not disable interrupts while acquiring
the lock. We want to remove the local_irq_disable() invocation from
__usb_hcd_giveback_urb() and therefore it is required for the callback
handler to disable the interrupts while acquiring the lock.
The callback may be invoked either in IRQ or BH context depending on the
USB host controller.
Use the _irqsave() variant of the locking primitives.
Signed-off-by: John Ogness <john.ogness@linutronix.de>
Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The USB completion callback does not disable interrupts while acquiring
the lock. We want to remove the local_irq_disable() invocation from
__usb_hcd_giveback_urb() and therefore it is required for the callback
handler to disable the interrupts while acquiring the lock.
The callback may be invoked either in IRQ or BH context depending on the
USB host controller.
Use the _irqsave() variant of the locking primitives.
Signed-off-by: John Ogness <john.ogness@linutronix.de>
Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Acked-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the VDD and AVDD power supplies to the DAPM graph as some board may
need to enable a regulator to turn them on.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Park Ju Hyung