CLASS-H controller/Amplifier is common accorss Qualcomm WCD codec series.
This patchset adds basic CLASS-H controller apis for WCD codecs after
wcd9335 to use.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC,
It supports both I2S/I2C and SLIMbus audio interfaces.
On slimbus interface it supports two data lanes; 16 Tx ports
and 8 Rx ports. It has Seven DACs and nine dedicated interpolators,
Seven (six audio ADCs, and one VBAT ADC), Multibutton headset
control (MBHC), Active noise cancellation and Sidetone paths
and processing.
This patchset adds very basic support for playback and capture
via the 9 interpolators and ADC respectively.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Mark Brown <broonie@kernel.org>
There is a potential execution path in which function
platform_get_resource() returns NULL. If this happens,
we will end up having a NULL pointer dereference.
Fix this by replacing devm_ioremap with devm_ioremap_resource,
which has the NULL check and the memory region request.
This code was detected with the help of Coccinelle.
Cc: stable@vger.kernel.org Fixes: 2bd8d1d5cf89 ("ASoC: sirf: Add audio usp interface driver") Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com> Signed-off-by: Mark Brown <broonie@kernel.org>
The recent fix moved the inline snd_sgbuf_aligned_pages() outside the
ifdef, and this triggered a build error on some architectures due to
the undefined PAGE_SIZE, as spotted by 0day bot.
Fix it by adding the missing header inclusion.
Fixes: 4cae99d9b530 ("ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally") Reported-by: kbuild test robot <lkp@intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: dmaengine: Use standard pcm_format_to_bits() macro
The conversion from PCM format type to bits needs an explicit cast,
and it'll be uglier. Since we have a standard macro for that, let's
use it instead.
This patch fixes the sparse warning:
sound/soc/soc-generic-dmaengine-pcm.c:200:63: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: doc: Replace open code with params_set_format()
The example code in dpcm.rst contains an open code calling
snd_mask_set(), and this can be better represented with
params_set_format() instead. This automatically fixes the sparse
warning about snd_pcm_format_t handling, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA: pcm: Add snd_mask_set_format() helper for standard usages
Many drivers calling snd_mask_set() need to do ugly cast with __force
for shutting up the sparse warnings. Actually almost all of them are
about setting the format, so it's far better to provide a common
helper snd_mask_set_format() to pass SNDRV_PCM_FORMAT_* directly
without the cast.
There are a few other calls of snd_mask_set(), but they are in the PCM
core code, so we leave them for now.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
fmt in snd_soc_dai_link_event() contains the format bit position, not
the format bit itself. Hence it can be a simple integer instead of
the explicit u64.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: dmaengine: Fix missing __user prefix in copy_user callback
It seems that __user prefix was forgotten to be added to
dmaengine_copy_user callback while we refactored the user-copy PCM
core.
This patch adds the missing prefix, remove the superfluous cast, and
add the needed cast (__force is needed for downgrading from user
pointer to kernel pointer), too.
Spotted by a sparse warning like:
sound/soc/soc-generic-dmaengine-pcm.c:397:27: warning: incorrect type in initializer (incompatible argument 4 (different address spaces))
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes
sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe':
sound/soc/amd/acp-da7219-max98357a.c:367:3: warning: 'ret' may
be used uninitialized in this function [-Wmaybe-uninitialized]
dev_err(&pdev->dev, "Failed to register regulator: %d\n",
ret);
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002
DA7219 for our platform need to be configured for 1.8V.
Hence, we add a volatge regulator with supplies
of 1.8V in the machine driver.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com> Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: Intel: common: make sst_dma functions static
sst_dma_new and sst_dma_free are not used in any other file and don't
have a prototype. Move to static functions and remove
EXPORT_SYMBOL_GPL statement.
Reported by sparse warnings.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: core: add support to snd_soc_dai_get_channel_map()
On Qualcomm platforms, specifically with SLIMbus interfaced codecs,
the codec slim channel numbers are passed to DSP while configuring
the slim audio path. Having get_channel_map() would allow dais to
share such information across multiple dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
When CONFIG_SND_PCM_IEC958 is disabled, we get a link error for the
new driver:
sound/soc/meson/axg-spdifout.o: In function `axg_spdifout_hw_params':
axg-spdifout.c:(.text+0x650): undefined reference to `snd_pcm_create_iec958_consumer_hw_params'
The other users use 'select', so we should do the same here.
Fixes: 53eb4b7aaa04 ("ASoC: meson: add axg spdif output") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Acked-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Add the axg sound card to handle the specifities of the axg audio
sub system.
This card is required to:
* setup the dpcm links specific to the AXG (with a cpu sound dai)
* handle the 4 lanes masks of the tdm interfaces
* add the loopback link when a tdm pad interface has a playback
stream
* handle multi-codec links
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Amlogic's axg card driver can't use snd_soc_of_parse_tdm_slot()
directly because it needs to handle 4 mask for each direction.
Yet the parsing of each mask is the same, so export
snd_soc_of_get_slot_mask() to reuse the the existing code.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg TDM interface driver. This driver manages the format
and clocks provided on the pads.
On this SoC, each stream direction provides 4 serial lanes. This makes
a maximum of 8 channels in i2s modes and 128 channels in DSP modes.
While each lanes operate on the same slot number (same bit clock), they
may have different TDM masks. This requires to provide a function to let
the card set the 4 masks, in lieu of the usual set_tdm_slots() callback
of the dai driver.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg TDM core driver. On this SoC, tdm is bit more
complex than usual, mainly because the different TDM input decoders can
be attached to any of TDM pad interface, including the output pads.
For the this, TDM on this SoC is modeled like this:
- TDM interface provides the DAIs the codecs will be attached to.
The main responsibility of this driver is to manage the pad format
and the TDM clock rates.
- TDM Formatters: These are the entities which are actually dealing with
the TDM signal. TDMOUT produce a TDM signal from the audio sample
provided by FRDDR using the clocks provided the TDM interface. TDMIN
feeds TODDR with audio sample using the clocks and TDM signal provided
by the TDM Interface.
- TDM Streams: This provides the link between 1 DAI stream of the TDM
interface and one (or more) TDM formatters.
This driver provides the TDM formatter and TDM stream operations.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Sun, 1 Jul 2018 09:30:23 +0000 (11:30 +0200)]
ASoC: Intel: bytcr_rt5640: Add quirk for the "Connect Tablet 9" tablet
Add a quirk for the "Connect Tablet 9" tablet, this tablet has a
mono-speaker. Otherwise it works fine with the defaults.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
These all use the default settings, except that they only have a single
speaker and thus need the mono-speaker quirk.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Wed, 18 Jul 2018 20:55:41 +0000 (22:55 +0200)]
ASoC: Intel: bytcr_rt5651: Add mono speaker quirk
During my initial round of bytcr_rt5651 long-name patches I did not include
a difference for mono vs stereo speaker setups in the longname because it
seems that all 5651 devices with only a single speaker do some mixing of
left + right on the PCB.
However further testing has shown that while this works great when only
playing audio on the left or right channel, the output becomes garbled
when using both channels at once. Something which does not happen when
using the Stereo DAC MIXL / MIXR switches to mix the channels together
inside the codec and then only outputting on a single channel.
So we need to have separate UCM profiles and thus separate long-names
for devices with a mono speaker vs stereo speakers. Just as we already
have for the bytcr_rt5640 case.
This commit adds a new BYT_RT5651_MONO_SPEAKER quirk and adds "stereo-spk"
or "mono-spk" to the long-name based on this and enables this mapping on
devices with a mono speaker.
Changing the long-name like this is ok for now, since I'm still working
on the UCM profiles, so they are not in upstream alsa-lib yet.
This brings the long-name naming scheme fully in sync with the bytcr_rt5640
case, which is good from a consistency pov.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Wed, 18 Jul 2018 20:55:40 +0000 (22:55 +0200)]
ASoC: Intel: bytcr_rt5651: Add IN2 input mapping
During the recent cleanup series 3 of the 6 input mappings where removed
from the bytcr_rt5651 machine driver because testing showed that none of
them were used.
However some devices do actually have their internal mic on IN2 (and
only IN2, not IN1 and IN2), this did not show during previous tests
due to a bug in the userspace UCM input device switching code.
This commit re-adds the IN2 mapping for devices with the internal mic.
on IN2 and the headser mic on IN3 and enables this mapping on devices
with their internal mic on IN2.
This commit also changes the default internal mic input to IN2, because
all my 7 test devices have their mic there.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Wed, 18 Jul 2018 20:55:39 +0000 (22:55 +0200)]
ASoC: Intel: bytcr_rt5651: Set OVCD limit for VIOS LTH17 to 2000uA
With the default over current detect limit of 1500uA headsets on often
get detected as headphones on the VIOS LTH17 and even when detected as
headset the OVCD current triggers often while plugged in, resulting in
false-positive button press detection.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Wed, 18 Jul 2018 20:55:38 +0000 (22:55 +0200)]
ASoC: Intel: bytcr_rt5651: Fix using the wrong GPIO for the ext-amp on some boards
Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other
boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the
GpioIo one for the ext-amp-enable-gpio.
So far we've been assuming that the GpioIo one always comes first, this
commit adds code to detect which one comes first and to add the right
gpio-mapping.
This fixes sound not working on the Vios LTH17 laptop.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the compressed streams in DSP firmwares are
identified essentially by looking at a fixed location inside
the firmware. This is fragile and also limits things to a
single compressed stream.
Here a new form of firmware parameter is added, the HOST_BUFFER
which identifies a compressed stream from meta-data in the
firmware file. This is more robust and allows for the possiblity
of using multiple streams per core in the future. Currently the
implementation is still limited to a single stream and will
use the first HOST_BUFFER parameter encountered. If there aren't
any HOST_BUFFER parameters it will fall back to the legacy way
of finding the host buffer.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: wm_adsp: Allow up to 8 channels for voice control
Newer voice control firmwares can capture multiple audio channels.
Allow up to 8 channels for future-proofing.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Thu, 19 Jul 2018 10:50:36 +0000 (11:50 +0100)]
ASoC: wm_adsp: Take prefix into account in control name length
Currently when creating ALSA control names for the DSP the length of any
prefix applied to the CODEC is not taken into account. Whilst this is
mostly harmless it does result in ALSA doing the truncation of the
control names and printing a warning. It is better to have the driver do
the truncation so it can truncate from the start of parameter name
itself to give a greater chance of the result maintain a unique name.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Charles Keepax [Thu, 19 Jul 2018 10:50:35 +0000 (11:50 +0100)]
ASoC: wm_adsp: Correct algorithm list allocation size
Commit 6396bb221514 ("treewide: kzalloc() -> kcalloc()") was
overlooked when doing some refactoring to the algorithm list
handling, which lead to twice as much buffer being allocated
as required for reading the algorithm list. A kcalloc is no
longer appropriate since the allocation size is now in bytes
not registers, as such change back to kzalloc.
Fixes: 7f7cca08abf4 ("ASoC: wm_adsp: Simplify handling of alg offset and length") Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Add the capture memory interface of Amlogic's axg SoCs.
TDM, SPDIF or PDM input devices place audio samples inside this FIFO.
The FIFO content is then pushed to DDR
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Add the playback memory interface of Amlogic's axg SoCs.
This device pulls data from DDR to an internal FIFO.
This FIFO is then used to feed TDM and SPDIF Output devices.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Amlogic's axg SoCs have two types of fifos which are the memory
interfaces of the audio subsystem. FRDDR provides the playback
interface while TODDR provides the capture interface.
The way these fifos operate is very similar. Only a few settings
are specific to each.
They implement the same pcm driver here and the specifics of each
will be dealt with the related DAI driver.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: hdac_hdmi: Add documentation for power management
Add documentation for power management of HDAC HDMI codec device for
various scenarios such as S0/S3, probe and playback use case.
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: core: add support to card re-bind using component framework
This patch aims at achieving dynamic behaviour of audio card when
the dependent components disappear and reappear.
With this patch the card is removed if any of the dependent component
is removed and card is added back if the dependent component comes back.
All this is done using component framework and matching based on
component name.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: AMD: For capture have interrupts on I2S->ACP channel
Having interrupts enabled for ACP<->SYSMEM DMA transfer, we are in
for an interrupt storm.
For both playback and capture interrupts should be enabled for
I2S<->ACP DMA.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: AMD: Send correct channel for configuring DMA descriptors
Earlier, ch1 was used to define ACP-SYSMEM transfer and ch2 for
ACP-I2S transfer. With recent patches ch1 is used to define channel
order number 1 and ch2 as channel order number 2. Thus,
Playback:
ch1:SYSMEM->ACP
ch2:ACP->I2S
Capture:
ch1:I2S->ACP
ch1:ACP->SYSMEM
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com> Signed-off-by: Mark Brown <broonie@kernel.org>
As done for format and channels, add the possibility to merge
the backend rates on the frontend rates.
This useful if the backend does not support all rates supported by the
frontend, or if several backends (cpu and codecs) with different
capabilities are connected to the same frontend.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
The goal of this patch is to simplify a bit dpcm runtime stream merge
by removing several local variables.
ATM, merge functions return the BE 'filter' values which should then be
filtered against the FE stream values. This create a lot of local
variable and unnecessary init of min and max.
Instead of this, we can pass the FE stream values directly and let the
BE filtering functions perform the merge 'in-place'
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Enable reporting of button presses now that the codec driver recently has
gotten support for this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Wed, 4 Jul 2018 22:59:34 +0000 (00:59 +0200)]
ASoC: Intel: bytcr_rt5651: Disable jack-detect over suspend/resume
Disable jack-detection and thus the codec IRQ over suspend/resume.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Wed, 4 Jul 2018 22:59:32 +0000 (00:59 +0200)]
ASoC: rt5651: Allow disabling jack-detect by calling set_jack(NULL)
Allow the machine driver to disable jack-detect over a suspend/resume by
calling snd_soc_component_set_jack(NULL).
Note this renames rt5651_set_jack, where all the jack-enable work was done
to rt5651_enable_jack_detect. This function can now no longer fail as it
does not request the IRQ anymore. It can still be passed an invalid jack
source, but that should never happen, so this is now logged and treated as
no jack source.
Cc: Carlo Caione <carlo@endlessm.com> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Wed, 4 Jul 2018 22:59:31 +0000 (00:59 +0200)]
ASoC: rt5651: Fix workqueue cancel vs irq free race on remove
On removal we must free the IRQ *before* cancelling the jack-detect work,
so that the jack-detect work cannot be rescheduled by the IRQ.
Before this commit we were cancelling the jack-detect work from the
driver remove callback, while relying on devm to free the IRQ, which
happens after the remove callback.
This is the wrong order. This commit uses a devm-action to register
a devm callback which cancels the work, before requesting the IRQ
(devm tears things down in reverse order). This also allows us to
remove the now empty remove driver callback.
Cc: Carlo Caione <carlo@endlessm.com> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Sun, 1 Jul 2018 18:36:31 +0000 (20:36 +0200)]
ASoC: Intel: bytcr_rt5651: Add support for externar amplifier enable GPIO
The rt5651 does not have a built-in speaker amplifier, so it is often
used together with an external amplifier. On Cherry Trail boards this
external amplifier's enable pin is driven through a GPIO, which is
given as the first GPIO in the ACPI resources of the codec fwnode.
This commit adds support to the bytcr_rt5651 for this GPIO, fixing
the speaker not working on CHT devices with a rt5651 codec.
Cc: Carlo Caione <carlo@endlessm.com> Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Sun, 1 Jul 2018 18:36:30 +0000 (20:36 +0200)]
ASoC: Intel: bytcr_rt5651: Move getting of codec_dev into probe()
Move the getting of the codec_dev, to add device-props to it, out of
byt_rt5651_add_codec_device_props() and into its caller,
snd_byt_rt5651_mc_probe().
This is a preparation patch for adding support for an external amplifier
enable GPIO, which requires further accesses to the codec_dev.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Remove is_valleyview helper, this is not necessary, we can simply call
x86_match_cpu() directly instead.
Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Hans de Goede [Sat, 7 Jul 2018 10:22:10 +0000 (12:22 +0200)]
ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix2 8 tablet
Add a quirk for the Lenovo Miix2 8 tablet, this tablet uses a digital
mic on DMIC1 and has a mono-speaker. The jack-detect uses the default
settings..
Reported-and-tested-by: russianneuromancer@ya.ru Signed-off-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
The playback DAI is connected to the DSP and the DSP might be sourcing
signals from the playback stream. Add a DAPM route between the two to make
sure that the playback DAI is powered up, when the DSP is active.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean@analog.com> Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: pxa: make SND_PXA_SOC_SSP depend on PLAT_PXA
For the moment, we can't enable CONFIG_SND_PXA_SOC_SSP unless we are
building for ARM PXA or MMP:
WARNING: unmet direct dependencies detected for PXA_SSP
Depends on [n]: PLAT_PXA [=n]
Selected by [y]:
- SND_PXA_SOC_SSP [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y]
This adds an explicit dependency for it.
Fixes: 0a94cf345740 ("ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org>
projects / mirror_ubuntu-eoan-kernel.git / log