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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 730e2dd0cbc7 ("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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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 2b26dd4c1fc5 ("ASoC: rt5660: add rt5660 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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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 2b26dd4c1fc5 ("ASoC: rt5660: add rt5660 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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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 d3cb2de2479b ("ASoC: rt5659: add rt5659 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Declar rt5659_i2c_driver, which is only being passed to
module_i2c_driver(rt5659_i2c_driver), static.
Fixes: commit d3cb2de2479b ("ASoC: rt5659: add rt5659 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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ulseep_range() uses hrtimers and provides no advantage over msleep()
for larger delays. For this large delay msleep() is preferable.
Fixes: commit d3cb2de2479b ("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>
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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 246693ba7b0b ("ASoC: rt5640: change widget sequence for depop")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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 a76056f2e57e
("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
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Add snd_hda_get_dev_select() and snd_hda_set_dev_select() functions
for DP MST audio support.
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/1484208294-8637-2-git-send-email-libin.yang@intel.com
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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.
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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>
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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>
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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 f0fba2ad1b6b
("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>
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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>
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Linux 4.10-rc1
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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>
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'asoc/fix/tlv320aic3x' and 'asoc/fix/topology' into asoc-linus
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'asoc/fix/dwc', 'asoc/fix/fsl-ssi' and 'asoc/fix/hdmi-codec' into asoc-linus
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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>
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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: 0f909f98d7cb ("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>
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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>
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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>
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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>
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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>
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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>
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