This macro is unused since commit e369bd006f ("ASoC: wm8741: Allow master
clock switching").
Signed-off-by: Sergej Sawazki <ce3a@gmx.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_card_jack_new function can call snd_soc_jack_add_pins for
you, so pass directly the pins struct when you create the new jack.
Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove default and set I2S mode correctly both on codec and
cpu sides
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use ACPI ID 10EC3270 to load machine driver for cht-bsw-rt5645
and add reference to 3270 to use the rt5645 mode
Tested on Asus T100HA
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some platforms use AIF2, use routing information to set ASRC as needed
Suggested-by: Bard Liao <bardliao@realtek.com>
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This driver may be used on Baytrail CR platforms where SSP2 is
not available.
Add quirks and routing detection based on work done for RT5640.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALC3270 is a low-cost version of RT5645, add ACPI ID
to enable probe and use rt5645 codec driver
Tested on Asus T100HA
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the BIOS reports this codec as RT5640 but it's a rt5670. Use the
quirk mechanism to use the cht_bsw_rt5672 machine driver
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix classic issue of having multiple codecs listed in DSDT
but a single one actually enabled. The previous code did
not handle such errors and could also lead to uninitalized
configurations
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use machine driver initially defined for CherryTrail
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
add ACPI ID 10EC5648 found e.g on Asus X205TA and use
rt5645 driver
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the BIOS incorrectly reports this codec as 5640 but it is
really a rt5670
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
RT5651 is used on some Cherrytrail platforms, add the ACPI
ID in machine table.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=156191
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add machine entry for HP X2 Pavilion 10-p100.
This notebook contains rt5640 codec, but with ACPI ID "10EC3276".
Signed-off-by: Alexandrov Stansilav <neko@nya.ai>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add ACPI ID "10EC3276" for sound card found on notebook HP Pavilion X2 10-p000.
ACPI DSDT Table on this device describes this card as ALC3276, but it is in fact rt5640.
Signed-off-by: Alexandrov Stansilav <neko@nya.ai>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing code assumes a 19.2 MHz MCLK as the default
hardware configuration. This is valid for CherryTrail but
not for Baytrail.
Add explicit MCLK configuration to set the 19.2 clock on/off
depending on DAPM events.
This is a prerequisite step to enable devices with Baytrail
and RT5645 such as Asus X205TA
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current frame sync polarity definitions are inconsistent in the
Atom/DPCM driver, fix to align with regular ASoC definitions and
update code in platform and machine drivers for RT5640 and RT5651.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch corrects an omission in bytcr_rt5640 and bytcr_rt5651.
All existing machine drivers shall not use .pm_ops to avoid a double
suspend, as initially implemented by 3f2dcbeaeb
("ASoC: Intel: Remove soc pm handling to allow platform driver handle it").
Reported-by: Shrirang Bagul <shrirang.bagul@canonical.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Most of the devices are using stereo speakers so media_loop1 and
sprot_loop default mode should be stereo.
As per default all the routing UCM configuration doesn't enable Post
processing loops it is not impacting curent configurations.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit 8f5ebb1bee
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_card_name")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit 8f5ebb1bee.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit b6defcca0a
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_prefix")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit b6defcca0a.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit 1ef5bcd57b
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_simple_widgets")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit 1ef5bcd57b.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit 7364c8dc25
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_routing")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit 7364c8dc25.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_match_device could return NULL, and so can cause a NULL
pointer dereference later.
Signed-off-by: Shailendra Verma <shailendra.v@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Final block of feature work for 4.11:
- gen8 pd cleanup from Matthew Auld
- more cleanups for view/vma (Chris)
- dmc support on glk (Anusha Srivatsa)
- use core crc api (Tomue)
- track wedged requests using fence.error (Chris)
- lots of psr fixes (Nagaraju, Vathsala)
- dp mst support, acked for merging through drm-intel by Takashi
(Libin)
- huc loading support, including uapi for libva to use it (Anusha
Srivatsa)
* tag 'drm-intel-next-2017-01-23' of git://anongit.freedesktop.org/git/drm-intel: (111 commits)
drm/i915: Update DRIVER_DATE to 20170123
drm/i915: reinstate call to trace_i915_vma_bind
drm/i915: Assert that created vma has a whole number of pages
drm/i915: Assert the drm_mm_node is allocated when on the VM lists
drm/i915: Treat an error from i915_vma_instance() as unlikely
drm/i915: Reject vma creation larger than address space
drm/i915: Use common LRU inactive vma bumping for unpin_from_display
drm/i915: Do an unlocked wait before set-cache-level ioctl
drm/i915/huc: Assert that HuC vma is placed in GuC accessible range
drm/i915/huc: Avoid attempting to authenticate non-existent fw
drm/i915: Set adjustment to zero on Up/Down interrupts if freq is already max/min
drm/i915: Remove the double handling of 'flags from intel_mode_from_pipe_config()
drm/i915: Remove crtc->config usage from intel_modeset_readout_hw_state()
drm/i915: Release temporary load-detect state upon switching
drm/i915: Remove i915_gem_object_to_ggtt()
drm/i915: Remove i915_vma_create from VMA API
drm/i915: Add a check that the VMA instance we lookup matches the request
drm/i915: Rename some warts in the VMA API
drm/i915: Track pinned vma in intel_plane_state
drm/i915/get_params: Add HuC status to getparams
...
Declare snd_pcm_ops structures as const as they are either stored in the
ops field of a snd_pcm_substream structure or passed as an argument to
the function snd_pcm_set_ops. The function argument and the ops field
are of type const, so snd_pcm_ops structures having this property
can be made const too.
File size before: sound/pci/cs46xx/cs46xx_lib.o
text data bss dec hex filename
26047 5304 16 31367 7a87 sound/pci/cs46xx/cs46xx_lib.o
File size after: sound/pci/cs46xx/cs46xx_lib.o
text data bss dec hex filename
27335 4036 16 31387 7a9b sound/pci/cs46xx/cs46xx_lib.o
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the display resolution changes, the drm disables the
display pipes due to which audio rendering stops. At this
time, we need to ensure the existing audio pointers and
buffers are cleared out so that the playback can restarted
once the display pipe is enabled with a different N/CTS values
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Jerome Anand <jerome.anand@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hdmi audio driver based on the child platform device
created by gfx driver is implemented.
This audio driver is derived from legacy intel
hdmi audio driver.
The interfaces for interaction between gfx and audio
are updated and the driver implementation updated to
derive interrupts in its own address space based on
irq chip framework
The changes to calculate sub-period positions was triggered
by David Henningsson <david.henningsson@canonical.com> and is
accomodated in this patch
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Jerome Anand <jerome.anand@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Baytrail and Cherrytrail, HDaudio may be fused out or disabled
by the BIOS. This driver enables an alternate path to the i915
display registers and DMA.
Although there is no hardware path between i915 display and LPE/SST
audio clusters, this HDMI capability is referred to in the documentation
as "HDMI LPE Audio" so we keep the name for consistency. There is no
hardware path or control dependencies with the LPE/SST DSP functionality.
The hdmi-lpe-audio driver will be probed when the i915 driver creates
a child platform device.
Since this driver is neither SoC nor PCI, a new x86 folder is added
Additional indirections in the code will be cleaned up in the next series
to aid smoother DP integration
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Jerome Anand <jerome.anand@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CQ0093VC is no longer dependent on MFD_DAVINCI_VOICECODEC,
let's remove it. Otherwise, we can't compile it by COMPILE_TEST
on non-DAVINCE platform
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DMA channel(stream tag) used by the HDA link need to programmed in
codec so that codec receives packet from the link associated with the
same channel.
DMA channel is allocated in link BE dai hw_params, the same needs to be
set for the BE codec dai. Instead of using get/set dma_data(), use
dai_ops snd_soc_dai_set_tdm_slot() to set the stream tag.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of hdmi connect/disconnect or when stream need to be route to
multiple monitors, corresponding port and audio infoframe needs to be
reconfigured. Currently all the configuration are done in DAI ops which
results in silence playback.
So use dapm widget event handlers to program audio infoframe and enable
/disable port configuration when widget is power on/off.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently we are not disabling MEM_ENA on the error path, we should
really do this to unwind the state back to how it was. This patch adds a
clear of MEM_ENA on the error path, again there is no major issues
caused by this minor fix.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The booted and running flags should really only be set once all the
steps at that power level have been complete. Currently operations can
fail after the flags have been set, which would leave us in an
inconsistent state where the flags are set but the things expected to
reach that level have not happened. Whilst there isn't really any major
impact from this it is best to clean it up.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent refactoring overlooked some places which should be covered by
the pwr_lock, all code that affects or depends on the power status of
the DSP should be covered, this patch adds the missing coverage.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit c6644119a3 and
restores the ability to specify DMA channel names per DAI dma_data.
Unfortunately the functionality removed in the patch being reverted
cannot be entirely replaced by specifying DMA channel names in struct
snd_dmaengine_pcm_config as that does not cover devices with more than
2 DMA channels.
Together with patch "ASoC: Revert "samsung: Remove unneeded
initialization of chan_name"" this fixes broken sound on the s3c24xx
SoC platforms.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit cdaf9af1ea
which breaks I2S support on the non-DT Samsung SoC platforms,
since the default "tx", "rx" DMA channel names for playback
and capture streams or custom channel names in struct
snd_dmaengine_pcm_config are supported in the ASoC dmaengine
module only for devicetree booting case.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit adds a compatible string for everest,es8388. This is
an audio codec that is compatible with es8328.
Signed-off-by: Romain Perier <romain.perier@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils is getting clk by of_clk_get(), but didn't call
clk_free(). Now we can use devm_get_clk_from_child() for this purpose.
Let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 5f222a292 ("ASoC: rsnd: use for_each_rsnd_mod_xxx() ...")
modifies rsnd_dai_call() to use for_each_rsnd_mod_arrays().
Current rsnd is incrementing iterator in rsnd_mod_next(),
but the iterator will indicate +1 position in for_each loop in
this case. Incremental position should be inside for()
Reported-by: Hoan Nguyen An <na-hoan@jinso.co.jp>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add user interface to provide channel mapping.
In a first step this control is read only.
As TLV type, the control provides all configuration available for
HDMI sink(ELD), and provides current channel mapping selected by codec
based on ELD and number of channels specified by user on open.
When control is called before the number of the channel is specified
(i.e. hw_params is set), it returns all channels set to UNKNOWN.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During probe, DAIs can need to perform some actions that requests
the knowledge of the pcm runtime handle.
The callback is called during DAIs linking, after PCM device creation.
For instance this can be used to add relationship between a DAI pcm
control and the pcm device.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With codec read sometimes the pin_sense shows invalid monitor present
and eld_valid. Currently driver polls for few times to get the valid
eld data.
To avoid the latency, Instead of reading ELD from codec, read it
directly from the display driver using audio component framework.
and removed the unused direct codec helper functions.
Signed-off-by: Sandeep Tayal <sandeepx.tayal@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 66feeec9322132689d42723df2537d60f96f8e44
"RFC: ASoC: dapm: handle probe deferrals"
forgot a to update some two sites where the call
was used. The static codechecks quickly found them.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Fixes: 66feeec93221 ("RFC: ASoC: dapm: handle probe deferrals")
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The H3 SoC uses the same SPDIF block as found in earlier SoCs, but its
TXFIFO is mapped to another address.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When Ref capture is used during S0IX, only the DSP pipelines
are needed, thus remove the ignore_suspend for WoV streams so
that DMA can be suspended, but keep them for WoV endpoints.
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of enabling pin and cvt in pcm_open(), need to restore pin and
cvt state after system resume to restart the playback which is
paused/stopped before system suspend.
So enable pin and cvt in playback_prepare and call prepare when trigger
cmd is paused/started and resume to reconfigure pin and cvt.
Signed-off-by: Sachin Mokashi <sachinx.mokashi@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In system suspend, the firmware pipelines will be deleted and there
is no need to save the pipeline context. Driver will save the DPIB and
LPIB pointers in suspend.
In system resume, the firmware pipelines will be created again and the
RD/RW pointers in the Firmware buffer points to the base address. So
need to fetch the non-played data again to firmware buffer. LPIB
indicates the HW rendered position.
Instead of setting DPIB as resume point, set it to LPIB to restore from
the HW render position so that DMA would fetch the non-played data one
more time.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When pipe is pass-through, BE and FE modules are defined inside
a pipe, reset of pipe will be done in FE DAI prepare. So don't
reset in the BE prepare.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
arizona_spk_init uses snd_soc_dapm_new_control which since
commit 37e1df8c95 ("ASoC: dapm: handle probe deferrals") will
occasionally request a probe deferral. Which means we should propagate the
error out of our driver from it.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALC221 HP platform need to support Headphone Mic.
This patch will turn on headphone Mic supported.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As done previously for sun4i-codec, the DMA maxburst of 4
is not supported by every SoCs so the DMA controller engine
returns "unsupported value".
As a maxburst of 8 is supported by all variants, this patch
increases it to 8.
For more details, see commit from Chen-Yu Tsai:
commit 730e2dd0cb ("ASoC: sun4i-codec: Increase DMA max burst to 8")
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SAIF base oversample rates are either 512*fs or 384*fs. An additional
divider exists within the SAIF to generate sub-multiples of these two base
rates if MCLK is required by the codec.
* The sub-rates for the 512x base rate are: 256x, 128x, 64x, and 32x.
* The sub-rates for the 384x base rate are: 192x, 96x, and 48x.
Setting the base rate depending on the modulo operation with 32 and 48
give wrong results for some mclk.
If mclk=18.432MHz both modulo operations results in 0. As testing the
result with 32 is done first, a wrong base rate of 512*fs is set instead
of the correct 384*fs.
Fix this by setting the base rate depending on the calculated sub-rate.
Signed-off-by: Jörg Krause <joerg.krause@embedded.rocks>
Signed-off-by: Mark Brown <broonie@kernel.org>
If SAIF0 is used in master and SAIF1 in slave mode setting the SAIF1
register in mxs_saif_set_dai_fmt() does not have any effect on the
interface as the clk gate needs to be cleared before the register can be
written.
Signed-off-by: Jörg Krause <joerg.krause@embedded.rocks>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/rt5659.c:4236:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This starts to handle probe deferrals on regulators and clocks
on the ASoC DAPM.
I came to this patch after audio stopped working on Ux500 ages
ago and I finally looked into it to see what is wrong. I had
messages like this in the console since a while back:
ab8500-codec.0: ASoC: Failed to request audioclk: -517
ab8500-codec.0: ASoC: Failed to create DAPM control audioclk
ab8500-codec.0: Failed to create new controls -12
snd-soc-mop500.0: ASoC: failed to instantiate card -12
snd-soc-mop500.0: Error: snd_soc_register_card failed (-12)!
snd-soc-mop500: probe of snd-soc-mop500.0 failed with error -12
Apparently because the widget table for the codec looks like
this (sound/soc/codecs/ab8500-codec.c):
static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = {
/* Clocks */
SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"),
/* Regulators */
SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0, 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0, 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0, 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0, 0),
So when we call snd_soc_register_codec() and any of these widgets
get a deferred probe we do not get an -EPROBE_DEFER (-517) back as
we should and instead we just fail. Apparently the code assumes
that clocks and regulators must be available at this point and
not defer.
After this patch it rather looks like this:
ab8500-codec.0: Failed to create new controls -517
snd-soc-mop500.0: ASoC: failed to instantiate card -517
snd-soc-mop500.0: Error: snd_soc_register_card failed (-517)!
(...)
abx500-clk.0: registered clocks for ab850x
snd-soc-mop500.0: ab8500-codec-dai.0 <-> ux500-msp-i2s.1 mapping ok
snd-soc-mop500.0: ab8500-codec-dai.1 <-> ux500-msp-i2s.3 mapping ok
I'm pretty happy about the patch as it it, but I'm a bit
uncertain on how to proceed: there are a lot of users of the
external functions snd_soc_dapm_new_control() (111 sites)
and that will now return an occassional error pointer, which
is not handled in the calling sites.
I want an indication from the maintainers whether I should just
go in and augment all these call sites, or if deferred probe
is frowned upon when it leads to this much overhead.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel Broadwell machine driver will call API snd_soc_set_dmi_name() to
use DMI info to make the sound card long name.
For example, here are the changed long name for two Broadwell-based
machines:
Dell XPS-13(2015): DellInc.-XPS139343-01-0310JH
Intel WilsonBeach: Intel Corp.-BroadwellClientplatform-0.1-WilsonBeachSDS
They still share the same card name "broadwell-rt286".
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel DSP platform drivers are used by many different devices but are
difficult for userspace to differentiate. This patch adds an API to allow
the DMI name to be used in the sound card long name, thereby helping
userspace load the correct UCM configuration. Usually machine drivers
uses their own name as the sound card name (short name), and leave the
long name and driver name blank. This API will use the DMI info like
vendor, product and board to make up the card long name. If the machine
driver has already explicitly set the long name, this API will do nothing.
This patch also allows for further differentiation as many devices that
share the same DMI name i.e. Minnowboards, UP boards may be configured
with different codecs or firmwares. The API supports flavoring the DMI
name into the card longname to provide the extra differentiation required
for these devices.
For Use Case Manager (UCM) in the user space, changing card long name by
this API is backward compatible, since the card name does not change. For
a given sound card, even if there is no device-specific UCM configuration
file that uses the card long name, UCM will fall back to load the default
configuration file that uses the card name.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For selected only options the explicit dependencies do not make much sense
becase Kbuild ignores them anyway. Remove them explicitly.
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Select DW_DMAC_CORE like the rest of glue drivers do, e.g.
drivers/dma/dw/Kconfig.
While here group selectors under SND_SOC_INTEL_HASWELL and
SND_SOC_INTEL_BAYTRAIL.
Make platforms, which are using a common SST firmware driver, to be
dependent on DMADEVICES.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Rename SND_SST_MFLD_PLATFORM to SND_SST_ATOM_HIFI2_PLATFORM to make it clear
that is not only about Medfield platform.
The new name is derived from Intel Atom and HiFi2. HiFi2 is the DSP version,
it's public information for Intel *Field/*Trail parts, see
https://www.alsa-project.org/main/index.php/Firmware. By combining HiFi2 with
Atom we get a unique non-ambiguous description of the core+DSP hardware for
Intel Medfield through Intel Cherrytrail.
Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_INTEL_SKYLAKE selects SND_SOC_INTEL_SST already. Thus no need to
duplicate. Remove duplications.
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
MIPS allmodconfig results in this warning:
sound/mips/hal2.c: In function 'hal2_gain_get':
sound/mips/hal2.c:224:35: error: 'r' may be used uninitialized in this function [-Werror=maybe-uninitialized]
sound/mips/hal2.c:223:35: error: 'l' may be used uninitialized in this function [-Werror=maybe-uninitialized]
sound/mips/hal2.c: In function 'hal2_gain_put':
sound/mips/hal2.c:260:13: error: 'new' may be used uninitialized in this function [-Werror=maybe-uninitialized]
sound/mips/hal2.c:260:13: error: 'old' may be used uninitialized in this function [-Werror=maybe-uninitialized]
Returning an error for all unexpected cases shuts up the warning
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for new codec of ALC1220.
It's compatible with ALC882 & co.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio driver has a mechanism to fall back to the single cmd mode as
a last resort if the CORB/RIRB communication goes wrong even after
switching to the polling mode. The switching has worked in the past
well, but Enrico Mioso reported that his system crashes when this
happens.
Although the actual cause of the crash isn't still fully analyzed yet,
it'd be in anyway good to provide an option to turn off the fallback
mode. Now this patch extends the behavior of the existing single_cmd
option for that. Namely,
- The option is changed from bool to bint.
- As default, it is the mode allowing the fallback to single cmd.
- Once when either true/false value is given to the option, the driver
explicitly turns on/off the single cmd mode, but without the
fallback.
That is, if you want to disable the fallback, just pass single_cmd=0
option. Passing single_cmd=1 will keep working like before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some identifiers are referred just by one functions. In this case, they
can be put into the function definition. This brings two merits; readers
can easily follow codes related to the identifiers, developers are free
from name conflict.
This commit moves such identifiers to each function definition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ulseep_range() uses hrtimers and provides no advantage over msleep()
for larger delays. For this large delay msleep() is preferable.
Link: http://lkml.org/lkml/2017/1/11/377
Fixes: commit 2b26dd4c1f ("ASoC: rt5660: add rt5660 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop the const qualifier as it is being added by SOC_ENUM_DOUBLE_DECL()
already which is called by SOC_ENUM_SINGLE_DECL() here.
Fixes: commit 2b26dd4c1f ("ASoC: rt5660: add rt5660 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop the const qualifier as it is being added by SOC_ENUM_DOUBLE_DECL()
already which is called by SOC_ENUM_SINGLE_DECL() as well as the
double const by calls to SOC_VALUE_ENUM_SINGLE_DECL() via
SOC_VALUE_ENUM_DOUBLE_DECL).
Fixes: commit d3cb2de247 ("ASoC: rt5659: add rt5659 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Declar rt5659_i2c_driver, which is only being passed to
module_i2c_driver(rt5659_i2c_driver), static.
Fixes: commit d3cb2de247 ("ASoC: rt5659: add rt5659 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
ulseep_range() uses hrtimers and provides no advantage over msleep()
for larger delays. For this large delay msleep() is preferable.
Fixes: commit d3cb2de247 ("ASoC: rt5659: add rt5659 codec driver")
Link: http://lkml.org/lkml/2017/1/11/377
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
ulseep_range() uses hrtimers and provides no advantage over msleep()
for larger delays. Fix up the 70/80ms delays here passing the "min"
value to msleep(). This reduces the load on the hrtimer subsystem.
Link: http://lkml.org/lkml/2017/1/11/377
Fixes: commit 246693ba7b ("ASoC: rt5640: change widget sequence for depop")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make snd_rawmidi_substream.ops to be a const pointer to be safer and
allow more optimization. The patches to constify each rawmidi ops
will follow.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the DP MST audio support on i915 platform and
it will enable dyn_pcm_assign feature.
DP MST supports several device entry on the same port and each
device entry can map to one pcm stream. For example, on i915,
there are 3 pins, and each pin has 3 device entries. This means
there should be 3x3 pcms. However, there is only 3 pipe lines in
i915. This means 3 pcms are actived at most at the same moment.
We will create 5 pcms (pin number + dev entry num - 1) in this case.
For the details, please refer commit a76056f2e5
("ALSA: hda - hdmi dynamically bind PCM to pin when monitor hotplug")
Each device entry is a virtual pin. It is described by pin_nid and dev_id
in struct hdmi_spec_per_pin.
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Libin Yang <libin.yang@linux.intel.com>
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
Link: http://patchwork.freedesktop.org/patch/msgid/1484208294-8637-3-git-send-email-libin.yang@intel.com
As well as the usual smattering of driver specific fixes collected since
the merge window this has one particularly important fix to the core for
handling of aux_devs which was broken during the merge window by some of
the componentization refactoring.
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Merge tag 'asoc-fix-v4.10-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.10
As well as the usual smattering of driver specific fixes collected since
the merge window this has one particularly important fix to the core for
handling of aux_devs which was broken during the merge window by some of
the componentization refactoring.
gcc-7 caught what it considers a NULL pointer dereference:
sound/pci/hda/patch_ca0132.c: In function 'dspio_scp.constprop':
sound/pci/hda/patch_ca0132.c:1487:4: error: argument 1 null where non-null expected [-Werror=nonnull]
This is plausible from looking at the function, as we compare 'reply'
to NULL earlier in it. I have not tried to analyze if there are constraints
that make it impossible to hit the bug, but adding another NULL check in
the end kills the warning and makes the function more robust.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace mdelay to msleep to avoid busy loop on ak4642_lout_event().
Otherwise, sometimes playback doesn't work correctly when pulseaudio
was used.
Signed-off-by: Harunobu Kurokawa <harunobu.kurokawa.dn@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAI ID defines are back from the time when DAIs were referenced by a
numerical ID. These days a string is used instead and the defines are
unused. The last user of these defines was removed in commit f0fba2ad1b
("ASoC: multi-component - ASoC Multi-Component Support"). So remove the
defines as well.
This also means the mpc5200_psc_ac97.h file no longer has any content and
can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
When trying to use simple card with wm8962 the following probe
error happens:
wm8731 0-001a: simple-card: set_sysclk error
In simple-card.c the snd_soc_dai_set_sysclk() function is called with
clk_id as 0, which is an invalid clock for wm8731.
Adjust the clocks source definitions in wm8731.h so that the simple
card driver can work successfully
Signed-off-by: Jörg Krause <joerg.krause@embedded.rocks>
Signed-off-by: Mark Brown <broonie@kernel.org>
No one is using snd_soc_platform_trigger().
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Plantronics BT600 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1" and "cannot get freq at
ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and
avoids those error messages.
Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mono differential output for "Line Out" downmixes the stereo audio
from the mixer, instead of just taking the left channel.
Add a route from the "Right Mixer" to "Line Out Source Playback Route"
through the "Mono Differential" path, so DAPM doesn't shut down
everything if the left channel is muted.
Fixes: 0f909f98d7 ("ASoC: sun4i-codec: Add support for A31 Line Out
playback")
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No existing platform is using .bespoke_trigger.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No existing platform is using .delay.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sample size of 24 bits use in reality 32 bits for storage. We
can safelly enable this sample size and treat the data as
32 bits.
Tested in a x86_64 platform and in ARC AXS101 SDP platform.
Signed-off-by: Jose Abreu <joabreu@synopsys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Up until now PIO mode offered only playback support. With
this patch we add support for record mode. The PCM was
refactored so that we could reuse the existing infrastructure
without many changes.
We have support for 16 and 32 bits of sample size using
only 2 channels.
Tested in a x86_64 platform and in ARC AXS101 SDP platform.
Signed-off-by: Jose Abreu <joabreu@synopsys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The default capless power mode is low voltage mode. We should set
it to high voltage mode to get fair headphone performance.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The i2s clock pre-divider 1 is used for both i2s1 and sysclk.
The i2s1 is usually used for the main i2s and the pre-divider
will be set in hw_params function.
However, if i2s2 is used, the pre-divider is not set in the hw_params
function and the default value of i2s clock pre-divider 1 is too high
for sysclk and DMIC usage. Fix by overriding default divider value to 2.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When multiple front-ends are using the same back-end, putting state of a
front-end to STOP state upon receiving pause command will result in backend
stream getting released by DPCM framework unintentionally. In order to
avoid backend to be released when another active front-end stream is
present, put the stream state to PAUSED state instead of STOP state.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Check stream decoupled register value with requested value
before decoupling/coupling the stream.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Saved firmware ctx was not never released, so release Firmware
ctx in cleanup routine.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the optimized dsp_register_poll API to poll the DSP firmware
status register rather than open coding it.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Poll implementation is not quite accurate, especially for smaller
values of timeout or timeout values close to the actual timeout needed
Use jiffies to set the timeout value and time_before() to get the
accurate time. So update the dsp register poll implementation to
provide accurate timeout using jiffies.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of passing the topology manifest info directly to IPC library,
define the manifest info in topology and use this in IPC Library.
This will remove the dependency on topology interface definition with
IPC library.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As requirements to bring up audio paths are continuous getting tighter
and the DSP download to most ADSP devices happens over an external bus
it can become an important factor in the path bring up time. As such
sometimes it is a reasonable trade off to download the firmware ahead of
when it will be required and take a small hit on power consumption for
keeping the core powered up.
This "preloading" adds an additional control for each DSP core "DSPx
Preload Switch" that when set to true will power up the DSP core and
download the firmware currently selected in the "DSPx Firmware" control.
Whilst the core is preloaded the current firmware can not be changed and
the CODEC will be kept powered up and SYSCLK held on. Although future
improvements may allow the SYSCLK to be powered down as well because
the hardware only requires SYSCLK whilst the download is actually taking
place, but this is not covered in this series.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some identifiers are referred just by one functions. In this case, they
can be put into the function definition. This brings two merits; readers
can easily follow codes related to the identifiers, developers are free
from name conflict.
This commit moves such identifiers to each function definition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some identifiers are referred just by one functions. In this case, they
can be put into the function definition. This brings two merits; readers
can easily follow codes related to the identifiers, developers are free
from name conflict.
This commit moves such identifiers to each function definition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some identifiers are referred just by one functions. In this case, they
can be put into the function definition. This brings two merits; readers
can easily follow codes related to the identifiers, developers are free
from name conflict.
This commit moves such identifiers to each function definition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some identifiers are referred just by one functions. In this case, they
can be put into the function definition. This brings two merits; readers
can easily follow codes related to the identifiers, developers are free
from name conflict.
This commit moves such identifiers to each function definition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some identifiers are referred just by one functions. In this case, they
can be put into the function definition. This brings two merits; readers
can easily follow codes related to the identifiers, developers are free
from name conflict.
This commit moves such identifiers to each function definition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some identifiers are referred just by one functions. In this case, they
can be put into the function definition. This brings two merits; readers
can easily follow codes related to the identifiers, developers are free
from name conflict.
This commit moves such identifiers to each function definition.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are literally dozens of Insyde devices with a different
name but with the same audio routing. Use a generic quirk to
match on vendor name only to avoid recurring edits of the
same thing.
Signed-off-by: youling257 <youling257@gmail.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Frequency value of zero did not make sense, use same 24.576MHz
setting and only change the clock source in idle mode
Suggested-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAI ID defines are back from the time when DAIs were referenced by a
numerical ID. These days a string is used instead and the defines are
unused. The last user of these defines was removed in commit f0fba2ad1b
("ASoC: multi-component - ASoC Multi-Component Support"). So remove the
defines as well.
This also means the pxa2xx-ac97.h file no longer has any content and can be
removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Tested-by: Robert Jarzmik <robert.jarzmik@free.fr> (for mioa701_wm9713)
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 6b7e95d133. This commit
is based on a concern about value of the given parameter. It's expected
to be ORed value with some enumeration-constants, thus often it can not be
one of the enumeration-constants. I understood that this is out of
specification and causes implementation-dependent issues.
In C language specification, enumerated type can be interpreted as an
integer type, in which all of enumeration-constants in corresponding
enumerator-list can be stored. Implementations can select one of char,
signed int and unsigned int as its type, and this selection is
implementation-dependent.
In GCC, a signed integer is selected when at least one of
enumeration-constants has negative value, else an unsigned integer is
selected. This behaviour can be switched by -fshort-enums to short type.
Anyway, the type can be decided after scanning all of
enumeration-constants.
Totally, there's no rules to constrain the value of enumerated type to
be one of enumeration-constants. In short, in enumerated type, decision
of actual type for the type is the most important and
enumeration-constants are just used for the decision, thus it's permitted
to have an integer value in a range of enumeration-constants. In our case,
actual type for the type is currently deterministic to be either char or
unsigned int. Under GCC, it's unsigned int.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Testing EP_FLAG_RUNNING in snd_complete_urb() before running the completion
logic allows us to save a few cpu cycles by returning early, skipping the
pending urb in case the stream was stopped; the stop logic handles the urb
and sets the completion callbacks to NULL.
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream") was
incomplete causing another more severe kernel panic, so it got reverted.
This fixes both the original problem and its fallout kernel race/crash.
The original fix is to move the endpoint member NULL clearing logic inside
wait_clear_urbs() so the irq triggering the urb completion doesn't call
retire_capture/playback_urb() after the NULL clearing and generate a panic.
However this creates a new race between snd_usb_endpoint_start()'s call
to wait_clear_urbs() and the irq urb completion handler which again calls
retire_capture/playback_urb() leading to a new NULL dereference.
We keep the EP deactivation code in snd_usb_endpoint_start() because
removing it will break the EP reference counting (see [1] [2] for info),
however we don't need the "can_sleep" mechanism anymore because a new
function was introduced (snd_usb_endpoint_sync_pending_stop()) which
synchronizes pending stops and gets called inside the pcm prepare callback.
It also makes sense to remove can_sleep because it was also removed from
deactivate_urbs() signature in [3] so we benefit from more simplification.
[1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start")
[2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream")
[3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code")
Fixes: f8114f8583 ("Revert "ALSA: usb-audio: Fix race at stopping the stream"")
Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although the old quirk table showed ASUS X71SL with ALC663 codec being
compatible with asus-mode3 fixup, the bugzilla reporter explained that
asus-model8 fits better for the dual headphone controls. So be it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=191781
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fsl_ssi fifo watermark is by default set to 2 free spaces (i.e.
activate DMA on FIFO when only 2 spaces are left.) This means the
DMA must service the fifo within 2 audio samples, which is just not
enough time for many use cases with high data rate. In many
configurations the audio channel slips (causing l/r swap in stereo
configurations, or channel slipping in multi-channel configurations).
This patch gives more breathing room and allows the SSI to operate
reliably by changing the fifio refill watermark to 8.
There is no change in behavior for older chips (with an 8-deep fifo).
Only the newer chips with a 15-deep fifo get the new behavior. I
suspect a new fifo depth setting could be optimized on the older
chips too, but I have not tested.
Signed-off-by: Caleb Crome <caleb@crome.org>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The update of stream costs significantly, and we should avoid it
unless the stream really has started. Check pipe->running flag
instead of pipe->prepared.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pseudo DMA transfer codes in VX222 and VX-pocket driver have a
slight bug where they check the buffer boundary wrongly, and may
overflow. Also, the zero sample count might be handled badly for the
playback (although it shouldn't happen in theory). This patch
addresses these issues.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=141541
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"DAC L2 Power" and "DAC R2 Power" are used by both rt5639 and rt5640.
But it was defined in rt5640_specific_dapm_widgets[]. Move them to
rt5640_dapm_widgets will let both rt5639 and rt5640 can use it.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ASUS ROG Ranger VIII with ALC1150 codec requires the extra GPIO pin to
up for the front panel. Just use the existing fixup for setting up
the GPIO pins.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189411
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Imre Deak reported a deadlock of HD-audio driver at unbinding while
it's still in probing. Since we probe the codecs asynchronously in a
work, the codec driver probe may still be kicked off while the
controller itself is being unbound. And, azx_remove() tries to
process all pending tasks via cancel_work_sync() for fixing the other
races (see commit [0b8c82190c: ALSA: hda - Cancel probe work instead
of flush at remove]), now we may meet a bizarre deadlock:
Unbind snd_hda_intel via sysfs:
device_release_driver() ->
device_lock(snd_hda_intel) ->
azx_remove() ->
cancel_work_sync(azx_probe_work)
azx_probe_work():
codec driver probe() ->
__driver_attach() ->
device_lock(snd_hda_intel)
This deadlock is caused by the fact that both device_release_driver()
and driver_probe_device() take both the device and its parent locks at
the same time. The codec device sets the controller device as its
parent, and this lock is taken before the probe() callback is called,
while the controller remove() callback gets called also with the same
lock.
In this patch, as an ugly workaround, we unlock the controller device
temporarily during cancel_work_sync() call. The race against another
bind call should be still suppressed by the parent's device lock.
Reported-by: Imre Deak <imre.deak@intel.com>
Fixes: 0b8c82190c ("ALSA: hda - Cancel probe work instead of flush at remove")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC299 was similar as ALC225.
Add headset support for ALC299.
ALC3271 was for Dell rename.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As of kernel 4.10, ALSA dice driver is expected to be used in default
speed. In most cases, it's S400. While, IEEE 1394 specification describes
the other speed such as S800.
According to 'TCD30XX User Guide', its link layer controller supports
several transmission speed up to S800[0]. In Dice software interface,
transmission speed in output direction can be configured by asynchronous
transaction to 'TX_SPEED' offset in its address space. S800 may be
available.
This commit improves configuration of transmission unit before starting
packet streaming for this purpose. The value of 'max_speed' in 'fw_device'
data structure has available maximum speed decided in bus arbitration,
thus it's within capacity of the unit.
[0] TCD3xx User Guide - TCAT 1394 LLC, Revision 0.9.0-41360 (TC Applied Technologies, May 6 2015)
http://www.tctechnologies.tc/index.php/support/support-hardware/dice-iii-detailed-documentation
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 'amdtp_stream' structure is initialized by a call of
'amdtp_stream_init()'. Although a parameter of this function is for bit
flags of packet attributes, its type is enumerator.
This commit changes the type so that it's proper for a bit flags.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This module has a bug not to return error code in a case that data
structure for transmitted packets fails to be initialized.
This commit fixes the bug.
Fixes: 35efa5c489 ("ALSA: firewire-tascam: add streaming functionality")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA fireworks driver has a bug not to call an API to destroy
'cmp_connection' structure for input direction. Currently this causes no
issues because it just destroys 'mutex' structure, while it's better to
fix it for future work.
Fix: d23c2cc448 ("ALSA: fireworks/bebob/dice/oxfw: allow stream destructor after releasing runtime")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The clk_ref_div is not configured in the correct position of the
register. The patch fixes that clk_ref_div, Pre-Scalar, is assigned
the wrong value.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a module is not available in a pipeline, fail safely rather than
causing oops.
Signed-off-by: G Kranthi <gudishax.kranthikumar@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It has been seen that some newer SoCs have a different TX FIFO
address and we already have the difference with the A31 requiring
a reset. Add a quirks structure so that these can be managed
easily.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm components are now handled by the ALSA SoC SPDIF DIT driver
so can be removed.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the codec is powered on, it's registers are in reset state as the
power off will do a soft reset of the codec.
After the register sync we need to add delay to remove the pop-noise on
stream start.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The RESET register only have one self clearing bit and it should not be
cached. If it is cached, when we sync the registers back to the chip we
will initiate a software reset as well, which is not desirable.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds pointer to I2S device to clk_register_* functions.
This in the future allow clock framework to ensure proper runtime state
of the I2S device during all operations on the clocks provided by I2S
module.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds handling of parent operational clock to runtime PM
callbacks. This way it is ensured that when I2S module is in runtime
suspended state, all its parent clocks are disabled and unprepared.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch moves saving and restoring I2S registers to runtime PM
operations, what prepares the driver to operate with audio power domain.
When support for audio power domain is enabled and the domain is being
turned off, the I2S module will loose its context (registers), so runtime
callbacks have to handle it. System sleep suspend/resume operation are
implemented on top of runtime PM operations with generic
pm_runtime_force_suspend/resume helpers.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds calls to pm_runtime_get/put to ensure that any access to
I2S registers is done with proper (active) runtime PM state of I2S device.
Till now the driver enabled runtime PM, but didn't manage the state during
driver operation. The driver worked fine only because the runtime PM
callbacks managed device clock, which was enabled all the time because of
the additional enable call in the driver's probe function.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some unknown (maybe historical?) reasons support for secondary I2S DAI
was implemented by adding additional virtual platform device, which was
then probed again with the main I2S driver. This pattern is really hard
to follow and provides no benefits, so lets remove this hack and register
both DAIs during linear probe of Exynos I2S controller driver.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently binding of auxiliary devices doesn't work as in
soc_bind_aux_dev() function a bound component is not being added
to any list and in soc_probe_aux_devices() we are trying to walk
the component_dev_list list to probe auxiliary components but
at that time this list doesn't contain any auxiliary components
since they are being added to the card only in soc_probe_component().
This patch adds a list to the card where are stored bound but not
probed auxiliary devices, so that all aux devices can be probed.
Fixes: 1a653aa447 "ASoC: core: replace aux_comp_list to component_dev_list"
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Samsung Ativ Book 8 makes a loud click noise on boot, shutdown
and when the audio card enters or exits power saving states. All
these noises disappear applying ALC269_FIXUP_NO_SHUTUP.
In addition to that, fix the loud click noise that the laptop
makes when inserting or removing the headphone jack by automuting
via amp instead of pinctl.
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting shutup when the action is HDA_FIXUP_ACT_PRE_PROBE might
not have the desired effect since it could be overridden by
another more generic shutup function. Prevent this by setting
the more specific shutup function on HDA_FIXUP_ACT_PROBE.
Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DSD support for both little endian (DSD_U32_LE) and big endian
(DSD_U32_BE) version of the Amanero firmware.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Prepare to mark sensitive kernel structures for randomization by making
sure they're using designated initializers. These were identified during
allyesconfig builds of x86, arm, and arm64, with most initializer fixes
extracted from grsecurity.
Signed-off-by: Kees Cook <keescook@chromium.org>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When CONFIG_PM_SLEEP is disabled, SIMPLE_DEV_PM_OPS does not use
snd_cs5535audio_resume and snd_cs5535audio_suspend functions:
sound/pci/cs5535audio/cs5535audio_pm.c:77:12: warning: ‘snd_cs5535audio_resume’ defined but not used [-Wunused-function]
static int snd_cs5535audio_resume(struct device *dev)
^~~~~~~~~~~~~~~~~~~~~~
sound/pci/cs5535audio/cs5535audio_pm.c:58:12: warning: ‘snd_cs5535audio_suspend’ defined but not used [-Wunused-function]
static int snd_cs5535audio_suspend(struct device *dev)
^~~~~~~~~~~~~~~~~~~~~~~
Adding __maybe_unused to the declaration of these functions removes the
warnings.
Signed-off-by: Jérémy Lefaure <jeremy.lefaure@lse.epita.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ktime_set(S,N) was required for the timespec storage type and is still
useful for situations where a Seconds and Nanoseconds part of a time value
needs to be converted. For anything where the Seconds argument is 0, this
is pointless and can be replaced with a simple assignment.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Cc: Peter Zijlstra <peterz@infradead.org>
ktime is a union because the initial implementation stored the time in
scalar nanoseconds on 64 bit machine and in a endianess optimized timespec
variant for 32bit machines. The Y2038 cleanup removed the timespec variant
and switched everything to scalar nanoseconds. The union remained, but
become completely pointless.
Get rid of the union and just keep ktime_t as simple typedef of type s64.
The conversion was done with coccinelle and some manual mopping up.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Cc: Peter Zijlstra <peterz@infradead.org>
There is no point in having an extra type for extra confusion. u64 is
unambiguous.
Conversion was done with the following coccinelle script:
@rem@
@@
-typedef u64 cycle_t;
@fix@
typedef cycle_t;
@@
-cycle_t
+u64
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Cc: Peter Zijlstra <peterz@infradead.org>
Cc: John Stultz <john.stultz@linaro.org>
This was entirely automated, using the script by Al:
PATT='^[[:blank:]]*#[[:blank:]]*include[[:blank:]]*<asm/uaccess.h>'
sed -i -e "s!$PATT!#include <linux/uaccess.h>!" \
$(git grep -l "$PATT"|grep -v ^include/linux/uaccess.h)
to do the replacement at the end of the merge window.
Requested-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This reverts commit 16200948d8.
The commit was intended to cover the race condition, but it introduced
yet another regression for devices with the implicit feedback, leading
to a kernel panic due to NULL-dereference in an irq context.
As the race condition that was addressed by the commit is very rare
and the regression is much worse, let's revert the commit for rc1, and
fix the issue properly in a later patch.
Fixes: 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream")
Reported-by: Ioan-Adrian Ratiu <adi@adirat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Protect against corrupt firmware files by ensuring that the length we
get for the data in a region actually lies within the available firmware
file data buffer.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit df1a2776a7 ("ASoC: Intel: bytcr_rt5640: add MCLK support")
was merged but the corresponding clock framework patches have not,
after being bumped from audio to clock to x86 domains. The missing
clock-related patches result in a regression starting with 4.9 with
the audio card not being created.
Rather than reverting this commit and all following updates already
queued up for 4.10, handle run-time dependency on MCLK and fall back
to the previous bit-clock mode. This provides the same functionality
as in 4.8 for Baytrail devices. On Baytrail-CR most devices remain
silent with this fallback but additional patches are needed anyway.
As suggested by Mark Brown, the fallback is only allowed with -ENOENT,
all other run-time errors, including -EPROBE_DEFER, will stop the probe
with no sound card registered.
This patch should be applied to -stable as well as ASoC 4.10 fixes
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For TDM mode, the I2S data out line can be shared between mutliple
codecs. In this scenario, only the active codec should be using
the line, and all others should be high impedance. However,
currently in the driver this configuration isn't set when capture
is inactive, and the line remains driven.
This patch updates the AIF_OUT widget to set the DAI output pin of
the device as high impedance when not in use.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Platform only has pcm_new/pcm_free feature,
but it should be supported on Component level. This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/wm0010.c does not use any miscdevice so this patch
remove this unnecessary inclusion.
Signed-off-by: Corentin Labbe <clabbe.montjoie@gmail.com>
Acked-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/rt5677-spi.c does not use any miscdevice so this patch
remove this unnecessary inclusion.
Signed-off-by: Corentin Labbe <clabbe.montjoie@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/rt5514-spi.c does not use any miscdevice so this patch
remove this unnecessary inclusion.
Signed-off-by: Corentin Labbe <clabbe.montjoie@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Removed the unused function skl_get_format as the format is calculated
directly using the HDA core API.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds helper function to configure the host/link DMA when
the DMA is in decoupled mode.
Next patch adds the usage of this helper routines for configuring
DMA in Mixer event handler.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If system is suspended when PCM was paused/stopped, restart doesn't
configure DMA as it is we are in Pause state and results in IO error
eventually.
Configure host/link DMA before initializing DSP Gateway copier module
instead of DAI prepare(). So moved DMA configuration to mixer PRE_PMD
widget handler instead of DAI prepare.
This uses previously added new API to do the configuration and removes
old DAI prepare code.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To configure Host/Link DMA, additionally link index and format
are required based on the hw params. So added these parameters in
the pipe params and in hw_params the pipe params are updated.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This breaks devicetree compatibility, but in this case that is ok. All
affected units are either on my desk, or running an even older version
of the driver that is not compatible with the upstreamed version anyway
(and when these other units are eventually updated, they will get a
fresh dtb as well, so that is not a significant problem either).
All of that is of course assuming that noone else has managed to build
something that can use this driver, but that seems extremely improbable.
Signed-off-by: Peter Rosin <peda@axentia.se>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds a control to allow swapping HiFi DAC Left/Right channels.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver was checking for non-NULL address of struct's members:
- s3c_audio_pdata->type (union),
- s3c_audio_pdata->type.i2s (embedded struct).
This is pointless as these will be always non-NULL. The 's3c_audio_pdata'
is always initialized in static memory so it will be zeroed.
Additionally the 'type' member was an union with only one member.
It is safe to reorganize the structures to get rid of useless union and
checks for addresses to fix the coccinelle warning:
>> sound/soc/samsung/i2s.c:1270:2-4: ERROR: test of a variable/field address
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Reviewed-by: Bartlomiej Zolnierkiewicz <b.zolnierkie@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Both SND_SOC_SMARTQ and SND_SOC_SAMSUNG_TM2_WM5110
use gpio/consumer.h
This patch adds GPIOLIB || COMPILE_TEST to Kconfig entries
to fix runtime dependency.
See commit 638f958bae
("extcon: Allow compile test of GPIO consumers if !GPIOLIB")
for similar problem and explanations.
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Reported-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the following build errors on X86_32 !GPIOLIB
sound/soc/samsung/tm2_wm5110.c:220:3: error: implicit declaration
of function 'gpiod_set_value_cansleep' [-Werror=implicit-function-declaration]
sound/soc/samsung/tm2_wm5110.c:438:24: error: implicit declaration
of function 'devm_gpiod_get' [-Werror=implicit-function-declaration]
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
kcontrol->private_value is being kfree'd after kcontrol has been freed
(in previous call to snd_ctl_remove). Instead, fix this by kfreeing
the private_value before kcontrol.
CoverityScan CID#1388311 "Read from pointer after free"
Fixes: eea3dd4f12 ("ASoC: topology: Only free TLV for volume mixers of a widget")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In zx_i2s_hw_params(), 'format' is initialized and assigned bits based on
params_format, but never used. So remove it.
sound/soc/zte/zx296702-i2s.c: In function ‘zx_i2s_hw_params’:
sound/soc/zte/zx296702-i2s.c:228:21: warning: variable ‘format’ set but not used [-Wunused-but-set-variable]
unsigned long val, format;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In smdk_hw_params(), 'bfs' is initialized and assigned bits based on
params_width, but never used.
We could have removed the whole switch case but then driver might be
relying on checking bits, so I have kept the case for now.
sound/soc/samsung/smdk_wm8580.c: In function ‘smdk_hw_params’:
sound/soc/samsung/smdk_wm8580.c:35:6: warning: variable ‘bfs’ set but not used [-Wunused-but-set-variable]
int bfs, rfs, ret;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In sst_media_close(), 'ret_val' is initialized and assigned as return value
of stream ops close but never used. So remove it.
ound/soc/intel/atom/sst-mfld-platform-pcm.c: In function ‘sst_media_close’:
sound/soc/intel/atom/sst-mfld-platform-pcm.c:360:6: warning: variable ‘ret_val’ set but not used [-Wunused-but-set-variable]
int ret_val = 0, str_id;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In img_prl_out_hw_params(), 'format' is initialized but never used.
So remove it.
sound/soc/img/img-parallel-out.c: In function ‘img_prl_out_hw_params’:
sound/soc/img/img-parallel-out.c:126:19: warning: variable ‘format’ set but not used [-Wunused-but-set-variable]
snd_pcm_format_t format;
Cc: Damien.Horsley <Damien.Horsley@imgtec.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In pcm3168a_hw_params(), 'format' is initialized but never used.
sound/soc/codecs/pcm3168a.c: In function ‘pcm3168a_hw_params’:
sound/soc/codecs/pcm3168a.c:405:19: warning: variable ‘format’ set but not
used [-Wunused-but-set-variable]
snd_pcm_format_t format;
Cc: Damien.Horsley <Damien.Horsley@imgtec.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In max9867_dai_set_fmt(), 'ret' is initialized as return value of
regmap_raw_write() but never checked, so remove this and assignement.
sound/soc/codecs/max9867.c: In function ‘max9867_dai_set_fmt’:
sound/soc/codecs/max9867.c:312:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable]
int ret;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In adau17x1_pll_event(), 'ret' is initialized as return value of
regmap_raw_write() but never checked, so remove this and assignement.
sound/soc/codecs/adau17x1.c: In function ‘adau17x1_pll_event’:
sound/soc/codecs/adau17x1.c:68:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable]
int ret;
Cc: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In acp_dma_hw_params(), 'dma_buffer' is initialized, but not used. So
remove it.
sound/soc/amd/acp-pcm-dma.c: In function ‘acp_dma_hw_params’:
sound/soc/amd/acp-pcm-dma.c:673:25: warning: variable ‘dma_buffer’ set but not used [-Wunused-but-set-variable]
struct snd_dma_buffer *dma_buffer;
Cc: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In max98090_probe(), code checks for micbias being out of range. The
'micbias' variable in unsigned and checked against M98090_MBVSEL_2V2 which
is zero, so remove this check.
sound/soc/codecs/max98090.c: In function ‘max98090_probe’:
sound/soc/codecs/max98090.c:2459:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
} else if (micbias < M98090_MBVSEL_2V2 || micbias > M98090_MBVSEL_2V8) {
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In soc_tplg_pcm_elems_load, a variable 'err' is initialized but not
used.
It is assigned return values for pcm_new_ver() but never checked, so
remove it.
sound/soc/soc-topology.c: In function ‘soc_tplg_pcm_elems_load’:
sound/soc/soc-topology.c:1865:9: warning: variable ‘err’ set but not used [-Wunused-but-set-variable]
int i, err;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In process_fw_async_msg(), a variable 'msg_high' is initialized but
not used. So remove it.
sound/soc/intel/atom/sst/sst_ipc.c: In function ‘process_fw_async_msg’:
sound/soc/intel/atom/sst/sst_ipc.c:263:24: warning: variable ‘msg_high’ set but not used [-Wunused-but-set-variable]
union ipc_header_high msg_high;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In sst_free_stream(), a variable 'ops' is initialized but
not used. So remove it.
sound/soc/intel/atom/sst/sst_stream.c: In function ‘sst_free_stream’:
sound/soc/intel/atom/sst/sst_stream.c:397:24: warning: variable ‘ops’ set but not used [-Wunused-but-set-variable]
struct intel_sst_ops *ops;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In skl_tplg_mixer_dapm_post_pmd_event(), a variable 'ret' is initialized but
not used.
We don't check return of skl_delete_pipe, so remove the assignment as
well, so remove this variable.
sound/soc/intel/skylake/skl-topology.c: In function ‘skl_tplg_mixer_dapm_post_pmd_event’:
sound/soc/intel/skylake/skl-topology.c:976:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable]
int ret = 0;
^
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On an error, snd_ctl_add already free's kctrl, so calling snd_ctl_free_one
to free it again leads to a double free error. Fix this by removing
the extraneous snd_ctl_free_one call.
Issue found using static analysis with CoverityScan, CID 1372908
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No dramatic changes are found in this development cycle, but as usual,
many commits are applied in a wide range of drivers.
Most of big changes are in ASoC, where a few bits of framework work
and quite a lot of cleanups and improvements to existing code have
been done. The rest are usual stuff, a few HD-audio and USB-audio
quirks and fixes, as well as the drop of kthread usages in the whole
subsystem.
Below are some highlights:
ASoC:
- Support for stereo DAPM controls
- Some initial work on the of-graph sound card
- regmap conversions of the remaining AC'97 drivers
- A new version of the topology ABI; this should be backward compatible
- Updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona,
Intel skylake, atom-sst
- New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and
Realtek RT5665
USB-audio:
- Yet another race fix at disconnection
- Tolerated packet size calculation for some Android devices
- Quirks for Axe-Fx II, QuickCam, TEAC 501/503
HD-audio:
- Improvement of Dell pin fixup mapping
- Quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic
Misc:
- Replace all kthread usages with simple works
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Merge tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"No dramatic changes are found in this development cycle, but as usual,
many commits are applied in a wide range of drivers.
Most of big changes are in ASoC, where a few bits of framework work
and quite a lot of cleanups and improvements to existing code have
been done. The rest are usual stuff, a few HD-audio and USB-audio
quirks and fixes, as well as the drop of kthread usages in the whole
subsystem.
Below are some highlights:
ASoC:
- support for stereo DAPM controls
- some initial work on the of-graph sound card
- regmap conversions of the remaining AC'97 drivers
- a new version of the topology ABI; this should be backward
compatible
- updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona,
Intel skylake, atom-sst
- new drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and
Realtek RT5665
USB-audio:
- yet another race fix at disconnection
- tolerated packet size calculation for some Android devices
- quirks for Axe-Fx II, QuickCam, TEAC 501/503
HD-audio:
- improvement of Dell pin fixup mapping
- quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic
Misc:
- replace all kthread usages with simple works"
* tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (296 commits)
ALSA: hiface: Fix M2Tech hiFace driver sampling rate change
ALSA: usb-audio: Eliminate noise at the start of DSD playback.
ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC
ASoC: wm_adsp: wm_adsp_buf_alloc should use kfree in error path
ASoC: topology: avoid uninitialized kcontrol_type
ALSA: usb-audio: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks
ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
ASoC: zte: spdif: correct ZX_SPDIF_CLK_RAT define
ASoC: zte: spdif and i2s drivers are not zx296702 specific
ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting
ASoC: rsnd: enable/disable ADG when suspend/resume timing
ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params
ALSA: cs46xx: add a new line
ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture
ASoC: nau8825: disable sinc filter for high THD of ADC
ALSA: usb-audio: more tolerant packetsize
ALSA: usb-audio: avoid setting of sample rate multiple times on bus
ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting
ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO
ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq
...
We can no longer rely on the return value of
devm_snd_dmaengine_pcm_register(...) to check if the DMA
handle is declared in the DT.
Previously this check activated PIO mode but currently
dma_request_chan returns either a valid channel or -EPROBE_DEFER.
In order to activate PIO mode check instead if the interrupt
line is declared. This reflects better what is documented in
the DT bindings (see Documentation/devicetree/bindings/sound/
designware-i2s.txt).
Also, initialize use_pio variable which was never being set
causing PIO mode to never work.
Signed-off-by: Jose Abreu <joabreu@synopsys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Merge tag 'drm-for-v4.10' of git://people.freedesktop.org/~airlied/linux
Pull drm updates from Dave Airlie:
"This is the main pull request for drm for 4.10 kernel.
New drivers:
- ZTE VOU display driver (zxdrm)
- Amlogic Meson Graphic Controller GXBB/GXL/GXM SoCs (meson)
- MXSFB support (mxsfb)
Core:
- Format handling has been reworked
- Better atomic state debugging
- drm_mm leak debugging
- Atomic explicit fencing support
- fbdev helper ops
- Documentation updates
- MST fbcon fixes
Bridge:
- Silicon Image SiI8620 driver
Panel:
- Add support for new simple panels
i915:
- GVT Device model
- Better HDMI2.0 support on skylake
- More watermark fixes
- GPU idling rework for suspend/resume
- DP Audio workarounds
- Scheduler prep-work
- Opregion CADL handling
- GPU scheduler and priority boosting
amdgfx/radeon:
- Support for virtual devices
- New VM manager for non-contig VRAM buffers
- UVD powergating
- SI register header cleanup
- Cursor fixes
- Powermanagement fixes
nouveau:
- Powermangement reworks for better voltage/clock changes
- Atomic modesetting support
- Displayport Multistream (MST) support.
- GP102/104 hang and cursor fixes
- GP106 support
hisilicon:
- hibmc support (BMC chip for aarch64 servers)
armada:
- add tracing support for overlay change
- refactor plane support
- de-midlayer the driver
omapdrm:
- Timing code cleanups
rcar-du:
- R8A7792/R8A7796 support
- Misc fixes.
sunxi:
- A31 SoC display engine support
imx-drm:
- YUV format support
- Cleanup plane atomic update
mali-dp:
- Misc fixes
dw-hdmi:
- Add support for HDMI i2c master controller
tegra:
- IOMMU support fixes
- Error handling fixes
tda998x:
- Fix connector registration
- Improved robustness
- Fix infoframe/audio compliance
virtio:
- fix busid issues
- allocate more vbufs
qxl:
- misc fixes and cleanups.
vc4:
- Fragment shader threading
- ETC1 support
- VEC (tv-out) support
msm:
- A5XX GPU support
- Lots of atomic changes
tilcdc:
- Misc fixes and cleanups.
etnaviv:
- Fix dma-buf export path
- DRAW_INSTANCED support
- fix driver on i.MX6SX
exynos:
- HDMI refactoring
fsl-dcu:
- fbdev changes"
* tag 'drm-for-v4.10' of git://people.freedesktop.org/~airlied/linux: (1343 commits)
drm/nouveau/kms/nv50: fix atomic regression on original G80
drm/nouveau/bl: Do not register interface if Apple GMUX detected
drm/nouveau/bl: Assign different names to interfaces
drm/nouveau/bios/dp: fix handling of LevelEntryTableIndex on DP table 4.2
drm/nouveau/ltc: protect clearing of comptags with mutex
drm/nouveau/gr/gf100-: handle GPC/TPC/MPC trap
drm/nouveau/core: recognise GP106 chipset
drm/nouveau/ttm: wait for bo fence to signal before unmapping vmas
drm/nouveau/gr/gf100-: FECS intr handling is not relevant on proprietary ucode
drm/nouveau/gr/gf100-: properly ack all FECS error interrupts
drm/nouveau/fifo/gf100-: recover from host mmu faults
drm: Add fake controlD* symlinks for backwards compat
drm/vc4: Don't use drm_put_dev
drm/vc4: Document VEC DT binding
drm/vc4: Add support for the VEC (Video Encoder) IP
drm: Add TV connector states to drm_connector_state
drm: Turn DRM_MODE_SUBCONNECTOR_xx definitions into an enum
drm/vc4: Fix ->clock_select setting for the VEC encoder
drm/amdgpu/dce6: Set MASTER_UPDATE_MODE to 0 in resume_mc_access as well
drm/amdgpu: use pin rather than pin_restricted in a few cases
...
Commit 4bcc595ccd ("printk: reinstate KERN_CONT for printing
continuation lines") allows to define more message headers for a single
message. The motivation is that continuous lines might get mixed.
Therefore it make sense to define the right log level for every piece of
a cont line.
This patch allows to copy only the real message level. We should ignore
KERN_CONT because <filename:line> is added for each message. By other
words, we want to know where each piece of the line comes from.
[pmladek@suse.com: fix a check of the valid message level]
Link: http://lkml.kernel.org/r/20161111183444.GE2145@dhcp128.suse.cz
Link: http://lkml.kernel.org/r/1478695291-12169-5-git-send-email-pmladek@suse.com
Signed-off-by: Petr Mladek <pmladek@suse.com>
Cc: Joe Perches <joe@perches.com>
Cc: Sergey Senozhatsky <sergey.senozhatsky.work@gmail.com>
Cc: Steven Rostedt <rostedt@goodmis.org>
Cc: Jason Wessel <jason.wessel@windriver.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Chris Mason <clm@fb.com>
Cc: Josef Bacik <jbacik@fb.com>
Cc: David Sterba <dsterba@suse.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Sampling rate changes after first set one are not reflected to the
hardware, while driver and ALSA think the rate has been changed.
Fix the problem by properly stopping the interface at the beginning of
prepare call, allowing new rate to be set to the hardware. This keeps
the hardware in sync with the driver.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
[Problem]
In some USB DACs, a terrible pop noise comes to be heard
at the start of DSD playback (in the following situations).
- play first DSD track
- change from PCM track to DSD track
- change from DSD64 track to DSD128 track (and etc...)
- seek DSD track
- Fast-Forward/Rewind DSD track
[Cause]
At the start of playback, there is a little silence.
The silence bit pattern "0x69" is required on DSD mode,
but it is not like that.
[Solution]
This patch adds DSD silence pattern to the endpoint settings.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the following devices.
- TEAC NT-503
- TEAC UD-503
- TEAC UD-501
(1) Add quirks for native DSD support for TEAC devices.
(2) A specific vendor command is needed to switch between PCM/DOP and
DSD mode, same as Denon/Marantz devices.
Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's been a few bits of framework work this time around and quite a
lot of cleanups and improvements to existing code:
- Support for stereo DAPM controls from Chen-yu Tsai.
- Some initial work on the of-graph sound card from Morimoto-san, the
main bulk of this is currently in binding review.
- Lots of Renesas cleanups from Morimoto-san and sunxi work from
Chen-yu Tsai.
- regmap conversions of the remaining AC'97 drivers from Lars-Peter
Clausen.
- A new version of the topology ABI from Mengdong Lin.
- New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and Realtek
RT5665.
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Merge tag 'asoc-v4.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.10
There's been a few bits of framework work this time around and quite a
lot of cleanups and improvements to existing code:
- Support for stereo DAPM controls from Chen-yu Tsai.
- Some initial work on the of-graph sound card from Morimoto-san, the
main bulk of this is currently in binding review.
- Lots of Renesas cleanups from Morimoto-san and sunxi work from
Chen-yu Tsai.
- regmap conversions of the remaining AC'97 drivers from Lars-Peter
Clausen.
- A new version of the topology ABI from Mengdong Lin.
- New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and Realtek
RT5665.
buf was allocated by kzalloc() so it should be passed to kfree()
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When num_kcontrols is zero, widget->dobj.widget.kcontrol_type
gets set to an uninitialized local variable:
sound/soc/soc-topology.c: In function 'soc_tplg_dapm_widget_create':
sound/soc/soc-topology.c:1566:36: error: 'kcontrol_type' may be used uninitialized in this function [-Werror=maybe-uninitialized]
I could not figure out which of the valid types would be appropriate
here, so this sets it to '0', which is invalid but at least well-defined
here. There is probably a better way to address the issue.
Fixes: eea3dd4f12 ("ASoC: topology: Only free TLV for volume mixers of a widget")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Logitech QuickCam Communicate Deluxe/S7500 microphone fails with the
following warning.
[ 6.778995] usb 2-1.2.2.2: Warning! Unlikely big volume range (=3072),
cval->res is probably wrong.
[ 6.778996] usb 2-1.2.2.2: [5] FU [Mic Capture Volume] ch = 1, val =
4608/7680/1
Adding it to the list of devices in volume_control_quirks makes it work
properly, fixing related typo.
Signed-off-by: Con Kolivas <kernel@kolivas.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Axe-Fx II implicit feedback end point and the data sync endpoint
are in different interface descriptors. Add quirk to ensure a sync
endpoint is properly configured.
Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The macro ZX_SPDIF_CLK_RAT should be 2 instead of 4. With this
fix, we can get correct audio output on HDMI through SPDIF interface.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
ZTE ZX SPDIF and I2S drivers can work on not only ZX296702 but also
other ZTE ZX family SoCs like ZX296718, which is an arm64 platform.
Let's make a few renaming and tweak the Kconfig a bit to get the drivers
available for other ZTE ZX platforms.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Reviewed-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver setups BRGCKR/BRRA/BRRB when .probe timing.
But it breaks sound after Suspend/Resume. These should be setups
every start timing.
This patch is tested on R-Car Gen3 Salvator-X board
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Gaku Inami <gaku.inami.xw@bp.renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver enables ADG clock when .probe timing,
but it breaks sound after Suspend/Resume. These should be setups
every suspend/resume timing too.
This patch is tested on R-Car Gen3 Salvator-X board
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Gaku Inami <gaku.inami.xw@bp.renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ssi->usrcnt will be updated on snd_soc_dai_ops::trigger,
but snd_pcm_ops::hw_params will be called *before* it.
Thus, ssi->usrcnt is still 0 when 1st call.
rsnd_ssi_hw_params() needs to check its called count, this means
trigger should be if (ssi->usrcnt) instead of if (ssi->usrcnt > 1).
Reported-by: Nguyen Viet Dung <nv-dung@jinso.co.jp>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We accidentally deleted a newline so now the "nreallocated++;" statement
is hanging out way off to the right of the screen.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We accidentally introduced a dereference before the NULL check in
xmit_descs() as part of silencing a GCC warning.
Fixes: 16f46050e7 ("dbri: Fix compiler warning")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
This patch updates FE channel constraints & BE fixup to support
quad channel DMIC capture.
DMIC pin's BE fixup is configured based on channel input, i.e.
either stereo or quad.
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This bit will enable 4th order SINC filter.
=1, filter will enable; but it consumes higher power.
=0, the sinc filter is disable, and it should always keep 0 value to
get high THD.
Therefor, disable the filter when codec initiation for better
performance when recording.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
since commit 57e6dae108 ("ALSA: usb-audio: do not trust too-big
wMaxPacketSize values"), the expected packetsize is always limited
to nominal + 25%. It was discovered, that some devices (Android audio
accessory) have a much higher jitter in used packetsizes than 25%
which would result in BABBLE condition and dropping of packets.
A better solution is so assume the jitter to be the nominal packetsize:
-one nearly empty packet followed by a almost 150% sized one.
V2: changed to assume max frequency is +50 of nominal packetsize.
Signed-off-by: Andreas Pape <apape@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of userland applications call 'snd_pcm_hw_params()' and
'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()'
is called twice and the second 'snd_pcm_hw_prepare()' is called in
'SNDRV_PCM_STATE_PREPARED' state.
Some devices are not able to manage this and they will stop playback
if the sample rate will be configured several times over USB protocol.
V2: updated Changelog
Signed-off-by: Daniel Girnus <dgirnus@de.adit-jv.com>
Signed-off-by: Jens Lorenz <jlorenz@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The logic of "value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_XXXXXX;" is
unnecessary complex. By setting CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK
as the mask for regmap_update_bits() call, what the code does is exactly
the same as setting value = CS35L34_MCLK_RATE_XXXXXX.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HP Z1 Gen3 AiO with Conexant codec doesn't give an unsolicited event
to the headset mic pin upon the jack plugging, it reports only to the
headphone pin. It results in the missing mic switching. Let's fix up
by simply gating the jack event.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit [64047d7f49 ALSA: hda - ignore the assoc and seq when comparing
pin configurations] intented to ignore both seq and assoc at pin
comparing, but it only ignored seq. So that commit may still fail to
match pins on some machines.
Change the bitmask to also ignore assoc.
v2: Use macro to do bit masking.
Thanks to Hui Wang for the analysis.
Fixes: 64047d7f49 ("ALSA: hda - ignore the assoc and seq when comparing...")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the boot of the SST FW the firmware version is send back
to the driver. This patch is saving the FW version inside the
driver.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch is adding a sysfs entry in order to be able to get
access to SST FW version.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Using regmap_update_bits(..., mask, 1) with 'mask' following (1 << k)
and k greater than 0 is wrong. Indeed, _regmap_update_bits will perform
(mask & 1), which results in 0 if LSB of mask is 0. Thus the call
regmap_update_bits(..., mask, 1) is in reality equivalent to
regmap_update_bits(..., mask, 0).
In such a case, the correct use is regmap_update_bits(..., mask, mask).
This driver is performing such a mistake with the CS42L56_AIN*_REF_MASK
masks, which equal 0x10, 0x20, 0x40 and 0x80. Fix the driver to make it
consistent with the API. Please note that this change is untested,
as I do not have this piece of hardware. Testers are welcome!
Signed-off-by: Florian Vaussard <florian.vaussard@heig-vd.ch>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch will check the type of embedded controls for a widget, and
only free the TLV of volume mixers. Bytes controls don't have TLV.
Just free the private value which is used as struct soc_mixer_control
for volume mixers or soc_bytes_ext for bytes controls. No need to cast
to these types before freeing it.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
