Previously the structure used bitfields, which do not guarantee bit
ordering.
This change makes sure the order is clearly defined. It also renames
and repurposes the field for general use.
Signed-off-by: Slawomir Blauciak <slawomir.blauciak@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We had a couple of misses with ABI changes, e.g. for Xtensa oops
information and the integration of sound trigger, before we set-up a
formal process to track evolutions.
With this patch, the SOF kernel patches are officially aligned with
the firmware 3.6 level. Changing this level has no impact on existing
users and is fully backwards-compatible.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add soundwire dai type and update ABI version.
Signed-off-by: Pan Xiuli <xiuli.pan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We missed these two definitions for GDB support and component
notifications, they are defined for the SOF firmware. Since they are
not used by the kernel so far, we can still add them without any ABI
change.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This tablet has an incorrect acpi identifier just like
Thinkpad10 tablet, which is why it is trying to load the RT5640 driver
instead of the RT5762 driver. The RT5640 driver, on the other hand, checks
the hardware ID, so no driver are loaded during boot. This fix resolves to
load the RT5672 driver on this tablet during boot. It also provides the
correct IO configuration, like the jack detect mode 3, for 1.8V pullup. I
would like to thank Pierre-Louis Bossart for helping with this patch.
Signed-off-by: Kovács Tamás <kepszlok@zohomail.eu>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allwinner DAC seems to have a delay in the Speaker audio routing. When
playing a sound for the first time, the sound gets chopped. On a second
play the sound is played correctly. After some time (~5s) the issue gets
back.
This commit seems to be fixing the same issue as bf14da7 but
for another codepath.
This is the DTS that was used to debug the problem.
&codec {
allwinner,pa-gpios = <&r_pio 0 11 GPIO_ACTIVE_HIGH>; /* PL11 */
allwinner,audio-routing =
"Speaker", "LINEOUT";
status = "okay";
}
Signed-off-by: Georgii Staroselskii <georgii.staroselskii@emlid.com>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
re-write hda_init_caps and remove the HDA reset, clean HDA
streams and clear interrupt steps in hda_dsp_probe so the
HDA init steps will not be called twice if the
CONFIG_SND_SOC_SOF_HDA is true.
Fixes: 8a300c8fb1 ("ASoC: SOF: Intel: Add HDA controller for Intel DSP")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Zhu Yingjiang <yingjiang.zhu@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently on all supported platforms the IPC IRQ thread first signals
the sender when an IPC response is received from the DSP, then unmasks
the IPC interrupt. Those actions are performed without holding any
locks, so the thread can be interrupted between them. IPC timeouts
have been observed in such scenarios: if the sender is woken up and it
proceeds with sending the next message without unmasking the IPC
interrupt, it can miss the next response. This patch takes a spin-lock
to prevent the IRQ thread from being preempted at that point. It also
makes sure, that the next IPC transmission by the host cannot take
place before the IRQ thread has finished updating all the required IPC
registers.
Fixes: 53e0c72d98 ("ASoC: SOF: Add support for IPC IO between DSP and Host")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The size for the bytes kcontrol should include the abi header, that is,
data->size + sizeof(*data), it is also aligned with get method after
this change.
Fixes: c3078f5397 ("ASoC: SOF: Add Sound Open Firmware KControl support")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the SOF hw_params() fail, typically with an IPC error thrown by the
firmware, the period_elapsed workqueue is not initialized, but we
still cancel it in hw_free(), which results in a kernel warning.
Move the initialization to the .open callback. Tested on Broadwell
(Samus) and IceLake.
Fixes: e2803e610a ("ASoC: SOF: PCM: add period_elapsed work to fix
race condition in interrupt context")
GitHub issue: https://github.com/thesofproject/linux/issues/932
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sof_pcm_hw_params() can only be called once to setup the FW hw_params.
So after calling sof_pcm_hw_params(), hw_params_upon_resume flag must
be cleared to avoid multiple invoking sof_pcm_hw_params() by prepare.
For example, after resume, there is an xrun happened, prepare() will
be called. As the hw_params_upon_resume flag is not cleared,
sof_pcm_hw_params() will be called and this will cause IPC timeout.
This patch fixes such issues.
Fixes: 868bd00f49 ("ASoC: SOF: Add PCM operations support")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In some configurations, it's a requirement to split the probe in two,
with a second part handled in a workqueue (e.g. for HDMI support
which depends on the DRM modules).
SOF already handles these configurations but the error flow is
incorrect. When an error occurs in the workqueue, the probe has
technically already completed. If we release the resources on errors,
this generates kernel oops/use-after-free when the resources are
released a second time on module removal.
GitHub issue: https://github.com/thesofproject/linux/issues/945
Fixes: c16211d622 ("ASoC: SOF: Add Sound Open Firmware driver core")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No need to call snd_soc_unregister_component in case of error
because the component device is resource-managed.
Fixes: c16211d622 ("ASoC: SOF: Add Sound Open Firmware driver core")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_sof_remove() disables the DSP and unmaps the DSP BAR.
Removing topology after disabling the DSP results in a
kernel panic while unloading the pipeline widget. This is
because pipeline widget unload attempts to power down
the core it is scheduled on by accessing the DSP registers.
So, the suggested fix here is to unregister the machine driver
first to remove the topology and then disable the DSP
to avoid the situation described above.
Note that the kernel panic only happens in cases where the
HDaudio link is not managed by the hdac library,
e.g. no codec or when HDMI is not supported.
When the hdac library is used, snd_sof_remove() calls
snd_hdac_ext_bus_device_remove() to remove the codec which
unregisters the component driver thereby also removing the
topology before the DSP is disabled.
Fixes: c16211d622 ("ASoC: SOF: Add Sound Open Firmware driver core")
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 53e947a0e1 ("ASoC: soc-core: merge card resources cleanup
method") merged cleanup method of snd_soc_instantiate_card() and
soc_cleanup_card_resources().
But, after this commit, if user uses unbind/bind to Component factor
drivers, Kernel might indicates refcount error at
soc_cleanup_card_resources().
The 1st reason is card->snd_card is still exist even though
snd_card_free() was called, but it is already cleaned.
We need to set NULL to it.
2nd is card->dapm and card create debugfs, but its dentry is still
exist even though it was removed. We need to set NULL to it.
Fixes: 53e947a0e1 ("ASoC: soc-core: merge card resources cleanup method")
Cc: stable@vger.kernel.org # for v5.1
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Removing link components results in topology unloading. So,
acquire the client_mutex before removing components in
soc_remove_link_components. This will prevent the lockdep warning
seen when dai links are removed during topology removal.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Revert commit 069d037aea ("ASoC: simple-card: Fix configuration of
DAI format"). During further review, it turns out that the actual issue
was caused by an incorrectly formatted device-tree node describing the
soundcard.
The following is incorrect because the simple-audio-card
'bitclock-master' and 'frame-master' properties should not reference the
actual codec phandle ...
sound {
compatible = "simple-audio-card";
...
=> simple-audio-card,bitclock-master = <&codec>;
=> simple-audio-card,frame-master = <&codec>;
...
simple-audio-card,cpu {
sound-dai = <&xxx>;
};
simple-audio-card,codec {
=> sound-dai = <&codec>;
};
};
Rather, these properties should reference the phandle to the
'simple-audio-card,codec' property as shown below ...
sound {
compatible = "simple-audio-card";
...
=> simple-audio-card,bitclock-master = <&codec>;
=> simple-audio-card,frame-master = <&codec>;
...
simple-audio-card,cpu {
sound-dai = <&xxx>;
};
=> codec: simple-audio-card,codec { /* simple-card wants here */
sound-dai = <&xxx>; /* not here */
};
};
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The use of BIT/GENMASK was incorrect, fix.
Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The RT5682 codec button mapping, initially copied from the DA7219 one,
needs to be corrected.
Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When configuring a codec to be both bit-clock and frame-master, it was
found that the codec was always configured as bit-clock and frame-slave.
Looking at the simple_dai_link_of() function there appears to be two
problems with the configuration of the DAI format, which are ...
1. The function asoc_simple_parse_daifmt() is called before the function
asoc_simple_parse_codec() and this means that the device-tree node
for the codec has not been parsed yet, which is needed by the
function asoc_simple_parse_daifmt() to determine who is the codec.
2. The phandle passed to asoc_simple_parse_daifmt() is the phandle to
the 'codec' node and not the phandle of the actual codec defined by
the 'sound-dai' property under the 'codec' node.
Fix the above by moving the call to asoc_simple_parse_daifmt() after the
the call to asoc_simple_parse_codec() and pass the phandle for the codec
to asoc_simple_parse_daifmt().
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The selection order of m/c in match table is corrected
to use common codec as last in the list.
Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a case when a we want to read a large number of bytes that
require a burst but is not a multiple of the word size (8). When this
happens rt5677_spi_reverse will run off the end of the buffer. The
solution is to tell spi_reverse the actual size of the destination and
stop if we reach it even if we have data left that we read.
Cc: Ben Zhang <benzh@chromium.org>
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_link_event() is updating snd_soc_dai :: active,
but it is unbalance.
It counts up if it has startup callback.
case SND_SOC_DAPM_PRE_PMU:
...
snd_soc_dapm_widget_for_each_source_path(w, path) {
...
if (source->driver->ops->startup) {
...
=> source->active++;
}
...
}
...
But, always counts down
case SND_SOC_DAPM_PRE_PMD:
...
snd_soc_dapm_widget_for_each_source_path(w, path) {
...
=> source->active--;
...
}
This patch always counts up when SND_SOC_DAPM_PRE_PMD.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add regcache_mark_dirty before regcache_sync for power
of codec may be lost at suspend, then all the register
need to be reconfigured.
Fixes: 0c516b4ff8 ("ASoC: cs42xx8: Add codec driver
support for CS42448/CS42888")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the output sample rate is [8kHz, 30kHz], the limitation
of the supported ratio range is [1/24, 8]. In the driver
we use (8kHz, 30kHz) instead of [8kHz, 30kHz].
So this patch is to fix this issue and the potential rounding
issue with divider.
Fixes: fff6e03c7b ("ASoC: fsl_asrc: add support for 8-30kHz
output sample rate")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_component_update_bits() may return 1 if operation
was successful and the value of the register changed.
Return a non-zero in ak4458_rstn_control for an error only.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If playback/capture is paused and system enters S3, after system returns
from suspend, BE dai needs to call prepare() callback when playback/capture
is released from pause if RESUME_INFO flag is not set.
Currently, the dpcm_be_dai_prepare() function will block calling prepare()
if the pcm is in SND_SOC_DPCM_STATE_PAUSED state. This will cause the
following test case fail if the pcm uses BE:
playback -> pause -> S3 suspend -> S3 resume -> pause release
The playback may exit abnormally when pause is released because the BE dai
prepare() is not called.
This patch allows dpcm_be_dai_prepare() to call dai prepare() callback in
SND_SOC_DPCM_STATE_PAUSED state.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AK4458 is probed successfully even if AK4458 is not present - this
is caused by probe function returning no error on i2c access failure.
Return an error on probe if i2c access has failed.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The cs4265_readable_register function stopped short of the maximum
register.
An example bug is taken from :
https://github.com/Audio-Injector/Ultra/issues/25
Where alsactl store fails with :
Cannot read control '2,0,0,C Data Buffer,0': Input/output error
This patch fixes the bug by setting the cs4265 to have readable
registers up to the maximum hardware register CS4265_MAX_REGISTER.
Signed-off-by: Matt Flax <flatmax@flatmax.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the erroneous addition of "SET_VALUE" to the GLB
IPC command string.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A race condition exists in handling firmware boot timeout.
If FW sends FW_READY just after boot timeout has expired in
driver, a kernel exception will result as FW_READY handler
will be run while the state is still being cleaned up in
snd_sof_run_firmware(). Avoid the race by setting
boot_complete also in the error case.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The nocodec option can be selected individually, leading to the following
issue:
sound/soc/sof/core.o: In function `snd_sof_device_probe':
core.c:(.text+0x4af): undefined reference to `sof_nocodec_setup'
Fix by selecting the SND_SOF_NOCODEC option as needed.
Reported-by: Hulk Robot <hulkci@huawei.com>
Reported-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent fix for the build fix caused a couple of unused variable
compiler warnings when CONFIG_SND_SOC_SOF_NOCODEC isn't set:
sound/soc/sof/core.c:263:6: warning: unused variable ‘ret’ [-Wunused-variable]
sound/soc/sof/core.c:262:28: warning: unused variable ‘machine’ [-Wunused-variable]
Fix them by adding another ifdef.
Fixes: ce38a75089 ("ASoC: SOF: core: fix undefined nocodec reference")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
- Set period minimum size. Ensure at least 5ms period
up to 48kHz/16 bits to prevent underrun/overrun.
- Remove MDMA constraints on period maximum size and
set period maximum to half the buffer maximum size.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some userspace apps, like pulseaudio, may call open, hw_params,
prepare to judge whether the pcm is ready or not. Current hdac_hdmi
will return -ENODEV if monitor is not connected, which will cause
the apps believe the pcm is not ready. Actually PCM for hdmi is ready,
even the monitor is not connected.
This patch removes the check of monitor presence in hw_params, just like
what the legacy HD-Audio driver does.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
mt6358_amic_disable() resets PGA to 0.
Save the gain settings from mixer control and restore them when using
the microphone.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Output volume settings from mixer controls would be lost.
Imagine that "Headphone Volume" has set to -10dB via amixer:
- in mtk_hp_enable()
- hp_store_gain() saves the volume setting -10dB from regmap_read()
to ana_gain[AUDIO_ANALOG_VOLUME_HPOUTL]
- headset_volume_ramp() ramps up from -10dB to -10dB
- in mtk_hp_disable()
- headset_volume_ramp() ramps down from -10dB to -40dB
Next time in mtk_hp_enable(), hp_store_gain() would save -40dB but
not -10dB. As a result, headset_volume_ramp() would ramp from -10dB to
-40dB (which is mute).
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Mt6358 ramps up from the smallest volume (i.e. -10dB) to target dB when
opening and ramps down from target dB to mute (i.e. -40dB) when closing.
If target is equal to -10dB when opening, headset_volume_ramp() simply
leaves current setting (which may not be -10dB) unchanged.
Execute the loop at least once to initialize the setting to the
starting point (i.e. from).
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing code mistakenly uses IS_ENABLED in C code instead of as
in conditional compilation, leading to the following error:
ld: sound/soc/sof/core.o: in function `sof_machine_check':
sound/soc/sof/core.c:279: undefined reference to `sof_nocodec_setup'
Fix by using #if !IS_ENABLED()
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SND_SOC_INTEL_COMMON Kconfig was removed months ago from SOF
Kconfig files but is still selected instead of the correct
SND_SOC_SOF_INTEL_COMMON kconfig which does select xtensa stuff,
leading to the following errors.
ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x120): undefined reference
to `sof_xtensa_arch_ops'
ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x180): undefined reference
to `sof_xtensa_arch_ops'
ld: sound/soc/sof/sof-acpi-dev.o:(.rodata+0x1e0): undefined reference
to `sof_xtensa_arch_ops'
Reported-by: kbuild test robot <lkp@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This fixes a compile warning below by properly handling the error code
from sof_get_ctrl_copy_params():
include/linux/kernel.h:843:43: warning: 'sparams.pl_size' may be used uninitialized in this function [-Wmaybe-uninitialized]
sound/soc/sof/ipc.c:639:34: note: 'sparams.pl_size' was declared here
The function returns an error before setting sparams.pl_size, so it'd
assign an uninitialized value at a later point.
Fixes: 53e0c72d98 ("ASoC: SOF: Add support for IPC IO between DSP and Host")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Paper over a trivial case leading to an uninitialized variable compile
warning:
sound/soc/rockchip/rockchip_pdm.c:179:3: warning: ‘clk_out’ may be used uninitialized in this function [-Wmaybe-uninitialized]
Fixes: 624e8e00ac ("ASoC: rockchip: pdm: fixup pdm fractional div")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
A trivial fix for the randconfig build error:
sound/soc/sof/ops.c:20:6: warning: ‘ret’ is used uninitialized in this function [-Wuninitialized]
Fixes: d1d95fcb63 ("ASoC: SOF: Add DSP HW abstraction operations")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A trivial fix for the randconfig build error:
sound/soc/codecs/da7219.c:2366:6: warning: unused variable ‘i’ [-Wunused-variable]
Fixes: d90ba6c8b5 ("ASoC: da7219: Expose BCLK and WCLK control through CCF")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix kconfig warning for unmet dependency for IOSF_MBI when
PCI is not set/enabled. Fixes this warning:
WARNING: unmet direct dependencies detected for IOSF_MBI
Depends on [n]: PCI [=n]
Selected by [y]:
- SND_SOC_SOF_ACPI [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_SOF_TOPLEVEL [=y] && (ACPI [=y] || COMPILE_TEST [=n]) && X86 [=y]
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: alsa-devel@alsa-project.org
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change trace level to debug to avoid spurious messages.
Return quietly when accessing iec958 control, while no
S/PDIF signal is available.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
- Set period minimum size. Ensure at least 5ms period
up to 48kHz/16 bits to prevent underrun/overrun.
- Remove MDMA constraints on period maximum size and
set period maximum to half the buffer maximum size.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The max98090 driver defines 3 DAPM muxes; one for the right line output
(LINMOD Mux), one for the left headphone mixer source (MIXHPLSEL Mux)
and one for the right headphone mixer source (MIXHPRSEL Mux). The same
bit is used for the mux as well as the DAPM enable, and although the mux
can be correctly configured, after playback has completed, the mux will
be reset during the disable phase. This is preventing the state of these
muxes from being saved and restored correctly on system reboot. Fix this
by marking these muxes as SND_SOC_NOPM.
Note this has been verified this on the Tegra124 Nyan Big which features
the MAX98090 codec.
Signed-off-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The current algorithm allows 3 types of transfers, 16bit, 32bit and
burst. According to Realtek, 16bit transfers have a special restriction
in that it is restricted to the memory region of
0x18020000 ~ 0x18021000. This region is the memory location of the I2C
registers. The current algorithm does not uphold this restriction and
therefore fails to complete writes.
Since this has been broken for some time it likely no one is using it.
Better to simply disable the 16 bit writes. This will allow users to
properly load firmware over SPI without data corruption.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org>
Reviewed-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
For Spreadtrum audio platform driver, it need allocate a larger DMA buffer
dynamically to copy audio data between userspace and kernel space, but that
will increase the risk of memory allocation failure especially the system
is under heavy load situation.
To make sure the audio can work in this scenario, we usually reserve one
region of memory to be used as a shared pool of DMA buffers for the
platform component. So add of_reserved_mem_device_init_by_idx() function
to initialize the shared pool of DMA buffers to be used by the platform
component.
Signed-off-by: Baolin Wang <baolin.wang@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
