The conversion from PCM format type to bits needs an explicit cast,
and it'll be uglier. Since we have a standard macro for that, let's
use it instead.
This patch fixes the sparse warning:
sound/soc/soc-generic-dmaengine-pcm.c:200:63: warning: restricted snd_pcm_format_t degrades to integer
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The PCM format type is with __bitwise, so we should use the dedicated
snd_pcm_format_t instead of int.
This fixes the sparse warning like:
sound/soc/codecs/pcm186x.c:268:44: warning: incorrect type in initializer (different base types)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the new helper function snd_mask_set_format() for avoiding the
ugly cast with __force prefix.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
As sparse warns, the PCM format type can't be dealt as integer as
found in Intel SST driver codes.
Fix them in the following two ways:
- The open code with snd_mask_set() and params->masks reference is
replaced with params_set_format()
- The rest codes with snd_mask_set(fmt, SNDRV_PCM_FORMAT_XXX) are
replaced with the new helper, snd_mask_set_format().
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
fmt in snd_soc_dai_link_event() contains the format bit position, not
the format bit itself. Hence it can be a simple integer instead of
the explicit u64.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
It seems that __user prefix was forgotten to be added to
dmaengine_copy_user callback while we refactored the user-copy PCM
core.
This patch adds the missing prefix, remove the superfluous cast, and
add the needed cast (__force is needed for downgrading from user
pointer to kernel pointer), too.
Spotted by a sparse warning like:
sound/soc/soc-generic-dmaengine-pcm.c:397:27: warning: incorrect type in initializer (incompatible argument 4 (different address spaces))
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
zx-tdm driver sets the DAI driver definitions with the format bits
wrongly set with SNDRV_PCM_FORMAT_*, instead of SNDRV_PCM_FMTBIT_*.
This patch corrects the definitions.
Spotted by a sparse warning:
sound/soc/zte/zx-tdm.c:363:35: warning: restricted snd_pcm_format_t degrades to integer
Fixes: 870e0ddc43 ("ASoC: zx-tdm: add zte's tdm controller driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop amlogic prefix in front of the generic DT properties and change
property "name" to "model".
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use IRQ_RETVAL instead of the open coded ternary operation.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes
sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe':
sound/soc/amd/acp-da7219-max98357a.c:367:3: warning: 'ret' may
be used uninitialized in this function [-Wmaybe-uninitialized]
dev_err(&pdev->dev, "Failed to register regulator: %d\n",
ret);
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DA7219 for our platform need to be configured for 1.8V.
Hence, we add a volatge regulator with supplies
of 1.8V in the machine driver.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
include DAPM Mux and output widgets into the list.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This looks like a copy/paste issue, but clearly there is an inversion
that is obvious when checking the arguments.
Detected with Sparse - now that we have fewer warnings this one was
easy to find.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Simplify code and add relevant casts to make Sparse warnings go away
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sst_dma_new and sst_dma_free are not used in any other file and don't
have a prototype. Move to static functions and remove
EXPORT_SYMBOL_GPL statement.
Reported by sparse warnings.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make sure definitions are consistent with usage.
Detected with Sparse.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make all Sparse warnings go away by using le16/32_to_cpu.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adopt the SPDX license identifier headers to ease license compliance
management.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On Qualcomm platforms, specifically with SLIMbus interfaced codecs,
the codec slim channel numbers are passed to DSP while configuring
the slim audio path. Having get_channel_map() would allow dais to
share such information across multiple dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
When CONFIG_SND_PCM_IEC958 is disabled, we get a link error for the
new driver:
sound/soc/meson/axg-spdifout.o: In function `axg_spdifout_hw_params':
axg-spdifout.c:(.text+0x650): undefined reference to `snd_pcm_create_iec958_consumer_hw_params'
The other users use 'select', so we should do the same here.
Fixes: 53eb4b7aaa ("ASoC: meson: add axg spdif output")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SSP DAI now handles the clocking setup itself, all it needs is the
master clock frequency. Remove the code from Zylonite and Magician
platforms.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the axg sound card to handle the specifities of the axg audio
sub system.
This card is required to:
* setup the dpcm links specific to the AXG (with a cpu sound dai)
* handle the 4 lanes masks of the tdm interfaces
* add the loopback link when a tdm pad interface has a playback
stream
* handle multi-codec links
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Amlogic's axg card driver can't use snd_soc_of_parse_tdm_slot()
directly because it needs to handle 4 mask for each direction.
Yet the parsing of each mask is the same, so export
snd_soc_of_get_slot_mask() to reuse the the existing code.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg TDM input driver which take the TDM signal of 4 input
lanes and push the decoded audio samples to TODDR fifo
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg tdm output driver which pulls data from FRDDR fifo
and produce the TDM signals for 4 output lanes.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg TDM interface driver. This driver manages the format
and clocks provided on the pads.
On this SoC, each stream direction provides 4 serial lanes. This makes
a maximum of 8 channels in i2s modes and 128 channels in DSP modes.
While each lanes operate on the same slot number (same bit clock), they
may have different TDM masks. This requires to provide a function to let
the card set the 4 masks, in lieu of the usual set_tdm_slots() callback
of the dai driver.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Amlogic's axg TDM core driver. On this SoC, tdm is bit more
complex than usual, mainly because the different TDM input decoders can
be attached to any of TDM pad interface, including the output pads.
For the this, TDM on this SoC is modeled like this:
- TDM interface provides the DAIs the codecs will be attached to.
The main responsibility of this driver is to manage the pad format
and the TDM clock rates.
- TDM Formatters: These are the entities which are actually dealing with
the TDM signal. TDMOUT produce a TDM signal from the audio sample
provided by FRDDR using the clocks provided the TDM interface. TDMIN
feeds TODDR with audio sample using the clocks and TDM signal provided
by the TDM Interface.
- TDM Streams: This provides the link between 1 DAI stream of the TDM
interface and one (or more) TDM formatters.
This driver provides the TDM formatter and TDM stream operations.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
While the two error labels "err" and "err_clk_put" goto the same place
it is rather confusing that the earlier one is certainly used later
again.
Signed-off-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a quirk for the "Connect Tablet 9" tablet, this tablet has a
mono-speaker. Otherwise it works fine with the defaults.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add quirk table entries for the following tablets:
ITWorks TW701
Ployer Momo7w
Trekstor win7
Yours 8"
These all use the default settings, except that they only have a single
speaker and thus need the mono-speaker quirk.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During my initial round of bytcr_rt5651 long-name patches I did not include
a difference for mono vs stereo speaker setups in the longname because it
seems that all 5651 devices with only a single speaker do some mixing of
left + right on the PCB.
However further testing has shown that while this works great when only
playing audio on the left or right channel, the output becomes garbled
when using both channels at once. Something which does not happen when
using the Stereo DAC MIXL / MIXR switches to mix the channels together
inside the codec and then only outputting on a single channel.
So we need to have separate UCM profiles and thus separate long-names
for devices with a mono speaker vs stereo speakers. Just as we already
have for the bytcr_rt5640 case.
This commit adds a new BYT_RT5651_MONO_SPEAKER quirk and adds "stereo-spk"
or "mono-spk" to the long-name based on this and enables this mapping on
devices with a mono speaker.
Changing the long-name like this is ok for now, since I'm still working
on the UCM profiles, so they are not in upstream alsa-lib yet.
This brings the long-name naming scheme fully in sync with the bytcr_rt5640
case, which is good from a consistency pov.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During the recent cleanup series 3 of the 6 input mappings where removed
from the bytcr_rt5651 machine driver because testing showed that none of
them were used.
However some devices do actually have their internal mic on IN2 (and
only IN2, not IN1 and IN2), this did not show during previous tests
due to a bug in the userspace UCM input device switching code.
This commit re-adds the IN2 mapping for devices with the internal mic.
on IN2 and the headser mic on IN3 and enables this mapping on devices
with their internal mic on IN2.
This commit also changes the default internal mic input to IN2, because
all my 7 test devices have their mic there.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With the default over current detect limit of 1500uA headsets on often
get detected as headphones on the VIOS LTH17 and even when detected as
headset the OVCD current triggers often while plugged in, resulting in
false-positive button press detection.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other
boards may have I2cSerialBusV2, GpioInt, GpioIo instead. We want the
GpioIo one for the ext-amp-enable-gpio.
So far we've been assuming that the GpioIo one always comes first, this
commit adds code to detect which one comes first and to add the right
gpio-mapping.
This fixes sound not working on the Vios LTH17 laptop.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a mixer control for the IN3 Boost volume, IN3 is used for the headset
mic on most devices, so this is necessary to control the headset mic
volume.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the compressed streams in DSP firmwares are
identified essentially by looking at a fixed location inside
the firmware. This is fragile and also limits things to a
single compressed stream.
Here a new form of firmware parameter is added, the HOST_BUFFER
which identifies a compressed stream from meta-data in the
firmware file. This is more robust and allows for the possiblity
of using multiple streams per core in the future. Currently the
implementation is still limited to a single stream and will
use the first HOST_BUFFER parameter encountered. If there aren't
any HOST_BUFFER parameters it will fall back to the legacy way
of finding the host buffer.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Newer voice control firmwares can capture multiple audio channels.
Allow up to 8 channels for future-proofing.
Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently when creating ALSA control names for the DSP the length of any
prefix applied to the CODEC is not taken into account. Whilst this is
mostly harmless it does result in ALSA doing the truncation of the
control names and printing a warning. It is better to have the driver do
the truncation so it can truncate from the start of parameter name
itself to give a greater chance of the result maintain a unique name.
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 6396bb2215 ("treewide: kzalloc() -> kcalloc()") was
overlooked when doing some refactoring to the algorithm list
handling, which lead to twice as much buffer being allocated
as required for reading the algorithm list. A kcalloc is no
longer appropriate since the allocation size is now in bytes
not registers, as such change back to kzalloc.
Fixes: 7f7cca08ab ("ASoC: wm_adsp: Simplify handling of alg offset and length")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the spdif output serializer of the axg SoC family
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the capture memory interface of Amlogic's axg SoCs.
TDM, SPDIF or PDM input devices place audio samples inside this FIFO.
The FIFO content is then pushed to DDR
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the playback memory interface of Amlogic's axg SoCs.
This device pulls data from DDR to an internal FIFO.
This FIFO is then used to feed TDM and SPDIF Output devices.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Amlogic's axg SoCs have two types of fifos which are the memory
interfaces of the audio subsystem. FRDDR provides the playback
interface while TODDR provides the capture interface.
The way these fifos operate is very similar. Only a few settings
are specific to each.
They implement the same pcm driver here and the specifics of each
will be dealt with the related DAI driver.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add documentation for power management of HDAC HDMI codec device for
various scenarios such as S0/S3, probe and playback use case.
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Sanyog Kale <sanyog.r.kale@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch aims at achieving dynamic behaviour of audio card when
the dependent components disappear and reappear.
With this patch the card is removed if any of the dependent component
is removed and card is added back if the dependent component comes back.
All this is done using component framework and matching based on
component name.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the final step for more generic support of DRM audio
component. The generic audio component code is now moved to its own
file, and the symbols are renamed from snd_hac_i915_* to
snd_hdac_acomp_*, respectively. The generic code is enabled via the
new kconfig, CONFIG_SND_HDA_COMPONENT, while CONFIG_SND_HDA_I915 is
kept as the super-class.
Along with the split, three new callbacks are added to audio_ops:
pin2port is for providing the conversion between the pin number and
the widget id, and master_bind/master_unbin are called at binding /
unbinding the master component, respectively. All these are optional,
but used in i915 implementation and also other later implementations.
A note about the new snd_hdac_acomp_init() function: there is a slight
difference between this and the old snd_hdac_i915_init(). The latter
(still) synchronizes with the master component binding, i.e. it
assures that the relevant DRM component gets bound when it returns, or
gives a negative error. Meanwhile the new function doesn't
synchronize but just leaves as is. It's the responsibility by the
caller's side to synchronize, or the caller may accept the
asynchronous binding on the fly.
v1->v2: Fix missing NULL check in master_bind/unbind
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HD-audio i915 binding code contains a single pointer, hdac_acomp,
for allowing the access to audio component from the master bind/unbind
callbacks. This was needed because the callbacks pass only the device
pointer and we can't guarantee the object type assigned to the drvdata
(which is free for each controller driver implementation).
And this implementation will be a problem if we support multiple
components for different DRM drivers, not only i915.
As a solution, allocate the audio component object via devres and
associate it with the given device, so that the component callbacks
can refer to it via devres_find().
The removal of the object is still done half-manually via
devres_destroy() to make the code consistent (although it may work
without the explicit call).
Also, the snd_hda_i915_register_notifier() had the reference to
hdac_acomp as well. In this patch, the corresponding code is removed
by passing hdac_bus object to the function, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For allowing other drivers to use the DRM audio component, rename the
i915_audio_component_* with drm_audio_component_*, and split the
generic part into drm_audio_component.h. The i915 specific stuff
remains in struct i915_audio_component, which contains
drm_audio_component as the base.
The license of drm_audio_component.h is kept to MIT as same as the the
original i915_component.h.
This is a preliminary change for further development, and no
functional changes by this patch itself, merely code-split and
renames.
v1->v2: Use SPDX for drm_audio_component.h, fix remaining i915
argument in drm_audio_component.h
Reviewed-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Having interrupts enabled for ACP<->SYSMEM DMA transfer, we are in
for an interrupt storm.
For both playback and capture interrupts should be enabled for
I2S<->ACP DMA.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Earlier, ch1 was used to define ACP-SYSMEM transfer and ch2 for
ACP-I2S transfer. With recent patches ch1 is used to define channel
order number 1 and ch2 as channel order number 2. Thus,
Playback:
ch1:SYSMEM->ACP
ch2:ACP->I2S
Capture:
ch1:I2S->ACP
ch1:ACP->SYSMEM
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the component does not match the configuration table provided
by the card, let soc-core check the component node for a name prefix
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
Disable jack-detection and thus the codec IRQ over suspend/resume.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Enable button press detection for headsets by using the ovcd IRQ to get
notified of button presses.
This is modelled after (almost exactly copied from) the button press code
for the rt5640 which has identical ovcd hardware.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow the machine driver to disable jack-detect over a suspend/resume by
calling snd_soc_component_set_jack(NULL).
Note this renames rt5651_set_jack, where all the jack-enable work was done
to rt5651_enable_jack_detect. This function can now no longer fail as it
does not request the IRQ anymore. It can still be passed an invalid jack
source, but that should never happen, so this is now logged and treated as
no jack source.
Cc: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On removal we must free the IRQ *before* cancelling the jack-detect work,
so that the jack-detect work cannot be rescheduled by the IRQ.
Before this commit we were cancelling the jack-detect work from the
driver remove callback, while relying on devm to free the IRQ, which
happens after the remove callback.
This is the wrong order. This commit uses a devm-action to register
a devm callback which cancels the work, before requesting the IRQ
(devm tears things down in reverse order). This also allows us to
remove the now empty remove driver callback.
Cc: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The rt5651 does not have a built-in speaker amplifier, so it is often
used together with an external amplifier. On Cherry Trail boards this
external amplifier's enable pin is driven through a GPIO, which is
given as the first GPIO in the ACPI resources of the codec fwnode.
This commit adds support to the bytcr_rt5651 for this GPIO, fixing
the speaker not working on CHT devices with a rt5651 codec.
Cc: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the getting of the codec_dev, to add device-props to it, out of
byt_rt5651_add_codec_device_props() and into its caller,
snd_byt_rt5651_mc_probe().
This is a preparation patch for adding support for an external amplifier
enable GPIO, which requires further accesses to the codec_dev.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove is_valleyview helper, this is not necessary, we can simply call
x86_match_cpu() directly instead.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a quirk for the Lenovo Miix2 8 tablet, this tablet uses a digital
mic on DMIC1 and has a mono-speaker. The jack-detect uses the default
settings..
Reported-and-tested-by: russianneuromancer@ya.ru
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The playback DAI is connected to the DSP and the DSP might be sourcing
signals from the playback stream. Add a DAPM route between the two to make
sure that the playback DAI is powered up, when the DSP is active.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Alexandru Ardelean <alexandru.ardelean@analog.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For the moment, we can't enable CONFIG_SND_PXA_SOC_SSP unless we are
building for ARM PXA or MMP:
WARNING: unmet direct dependencies detected for PXA_SSP
Depends on [n]: PLAT_PXA [=n]
Selected by [y]:
- SND_PXA_SOC_SSP [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y]
This adds an explicit dependency for it.
Fixes: 0a94cf3457 ("ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The format specifier "%p" can leak kernel addresses.
Use "%pK" instead.
Signed-off-by: Benjamin Gaignard <benjamin.gaignard@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.
Notice that such constant is used in a context that expects an
expression of type u64 (64 bits, unsigned) and the following
expression is currently being evaluated using 32-bit arithmetic:
256 * fs * 2 * mclk_src_scaling[i].param
Addresses-Coverity-ID: 1432039 ("Unintentional integer overflow")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow setting a clock called 'extclk' in the device of the ssp-dai
device. If specified, this clock will be set to the mclk rate from the
DAI's .set_sysclk() callback. The DAI will also configure itself to
use that external clock.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
To comply with the style of all kernel messages, add newline
to the end of every message.
Fixes: 70fb10529f ("ASoC: rsnd: add MIX (Mixer) support")
Signed-off-by: Andrew Gabbasov <andrew_gabbasov@mentor.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suffix ULL to constant 64 in order to give the compiler complete
information about the proper arithmetic to use.
Notice that such constant is used in a context that expects an
expression of type u64 (64 bits, unsigned) and the following
expression is currently being evaluated using 32-bit arithmetic:
rate[index] * txclk_df * 64
Addresses-Coverity-ID: 1222129 ("Unintentional integer overflow")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
num_channels for slim dais are aready set int set_channel_map,
do not overwrite them in hw_params.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch add routings mixer controls for slim rx ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to SLIMbus TX dais in AFE module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Existing code already has support for SLIMbus TX and RX, only thing
that was missing from TX side was mapping between virtual to actual
DSP port ids.
This patch adds those mappings.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The XRUN trigger from the driver should be done via
snd_pcm_stop_xrun(). It simplifies the locking as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The XRUN trigger from the driver should be done via
snd_pcm_stop_xrun(). It fixes the missing stream locking as a gratis,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The XRUN trigger from the driver should be done via
snd_pcm_stop_xrun(). It fixes the missing stream locking as a gratis,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace open-codes with the standard snd_pcm_stop_xrun() helper.
It simplifies codes a lot.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.
Notice that such constant is used in a context that expects an
expression of type u64 (64 bits, unsigned) and the following
expression is currently being evaluated using 32-bit arithmetic:
256 * fs * 2 * mclk_src_scaling[i].param
Addresses-Coverity-ID: 1339616 ("Unintentional integer overflow")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Machine drivers statically define a number of DAI links that currently
cannot be changed or removed by topology. This means PCMs and platform
components cannot be changed by topology at runtime AND machine drivers
are tightly coupled to topology.
This patch allows topology to override the machine driver DAI link config
in order to reuse machine drivers with different topologies and platform
components. The patch supports :-
1) create new FE PCMs with a topology defined PCM ID.
2) destroy existing static FE PCMs
3) change the platform component driver.
4) assign any new HW params fixups.
5) assign a new card name prefix to differentiate this topology to userspace.
The patch requires no changes to the machine drivers, but does add some
platform component flags that the platform component driver can assign
before loading topologies.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The generated clock (gclk) driver is able to set aclk as its parent and
change its rate alone, if needed. This means that our driver no longer
needs to configure aclk and we can let gclk select and configure its
clock source.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the es7154 which is basically an es7134 with an
embedded power amplifier and lower maximum sample rate
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop AVDD in favor of PVDD to match the names used in the datasheet
and only claim PVDD on the es7154. The es7134 and es7144 don't have
a separate supply for the digital I/O.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the I2S channel names are fixed, and DMA data flow order is
consistent (ch1 then ch2), we can simplify channel start order:
start the upstream channel and then the downstream channel for both
playback and capture cases.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The pointer() callback gets its value by reading the I2S BYTE_COUNT
register. This is a 64-bit runnning transaction counter. If a
transaction was aborted in the middle of a sample buffer, the counter will
stop counting on a number divisible by the buffer size. Since we actually
use it as a pointer into an aligned buffer, however, we do want to ensure
that it always starts at a number divisible by the buffer size when
starting a transaction, hence we reset it whenever starting a transaction.
To accomplish this, it wasn't necessary to zero bytescount at the
termination of each transaction, so remove this unnecessary code.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On capture, audio data is first copied from I2S to ACP memory, and then
from ACP to SYSRAM. The I2S_TO_ACP_DMA interrupt fires on every sample
transferred from I2S to ACP memory. That is it fires ~48000 times per
second when capturing @ 48 kHz. Since we don't do anything on this
interrupt anyway, disable it to save quite a few unnecessary interrupts.
The real "work" (calling snd_pcm_period_elapsed()) is done when transfer
from ACP to SYSRAM is complete.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
On capture, audio data is first copied from I2S to ACP memory, and then
to SYSRAM. For each step the channel number increases, so the names in
the driver were wrong.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is always correct to subtract out the starting bytescount value. Even
in the case of 2^64 byte rollover (292 Million Years in the future
@ 48000 Hz) the math still works out.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 6b116dfb46 ("ASoC: AMD: make channel 1 dma as circular") made
both channels circular, so this comment and logic no longer applies. Always
stop ch2 (the channel closest to the output) before ch1. This ensures
that the downstream circular DMA channel does not continue to play/capture
repeated samples after the upstream circular DMA channel has already
stopped.
Signed-off-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation to enabling -Wimplicit-fallthrough, mark switch cases
where we are expecting to fall through.
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the everest es7241 which is a simple 2 channels
analog to digital converter.
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We are currently using power saving mode for button detection.
However, it will impact the headset recording performance.
This patch will switch button detection to normal mode in capture
and switch to power saving mode in the end of capture.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
