To support DP MST audio, new pin verbs/params are added. This patch
adds helper functions to do following:
o To set a specific port
o To get the currently selected port
o To get the length of port.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The FSL SAI can support up to 32 channels using TDM. Report that value so
they can actually be used.
Tested using 8 channels.
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The pclk is necessary for zx2967 I2S controller. the driver
currently doesn't handle it. This is something we need to fix.
In turn, the driver supports zx296718's I2S controller.
By the way, this patch also change the clock name from tx to wclk
to make it clear.
Signed-off-by: Baoyou Xie <baoyou.xie@linaro.org>
Reviewed-by: Shawn Guo <shawnguo@kernel.org>
Reviewed-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
While unregistering the hdmi-codec, the hdmi device list must be
cleaned up. It avoid kernel page fault when registering again the
hdmi-codec.
Signed-off-by: Vincent Abriou <vincent.abriou@st.com>
Reviewed-by: Philipp Zabel <p.zabel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
As the 64-bit Allwinner H5 SoC has the same analog codec part (also the
same digital part) as H3, enable the driver to be built on ARM64
Allwinner platform, so that it can be used on H5.
Signed-off-by: Icenowy Zheng <icenowy@aosc.xyz>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
To handle jack event and configuration of the pin widget for MST
capable pin, this patch adds:
o Flag to identify the pin is MST capable.
o In notify callback(), based on the pipe and port information marks if
the port is mst_capable. In case of non MST, port is defaulted to zero.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To support MST moved pin to port, this changes the routes based on
port. So change the route in nau88l25_ssm4567 machine.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To support MST moved pin to port, this changes the routes based on
port. So change the route in nau88l25_max98357a machine.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To support MST moved pin to port, this changes the routes based on
port. So change the route in bxt_rt298 machine.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With MST each pin contains several ports to which device can be
connected.
As a preparatory work to support DP MST this patch adds below changes:
1. Defines the port structure and moves all stream related information
like ELD, converter list, chmap to port.
2. Creates ports for each pin based on the max_ports support.
3. Based on Pin-Port combination creates DAPM Mux widget instead of Pin
to allow user to select a converter.
4. Port zero is the default port when pin does not support MST.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The two front-ends to the codec can now be selected individually, but fail to
build when the bus support is missing:
sound/built-in.o: In function `es8328_spi_probe':
es8328-spi.c:(.text+0x125854): undefined reference to `__devm_regmap_init_spi'
sound/built-in.o: In function `es8328_spi_driver_init':
es8328-spi.c:(.init.text+0x3589): undefined reference to `__spi_register_driver'
Related to this, the added dependency on SND_SOC_ES8328 breaks:
warning: (SND_SOC_ALL_CODECS) selects SND_SOC_ES8328_I2C which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && SND_SOC_ES8328 && I2C)
This adds the respective Kconfig dependencies and changes SND_SOC_ES8328 to a hidden
symbol that is selected implicitly by the two more specific options, as we do for
some other codecs. We have to remove the 'depends on' for SND_SOC_IMX_ES8328 in the
same step to avoid a recursive dependency.
Fixes: aa00f2c8af ("ASoC: Allow to select ES8328_I2C and ES8328_SPI directly")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch create entry in sysfs file system to report the
platform_id = "pci-id-oem_id-oem_table_id-oem_revision"
for board identification.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Sodhi, VunnyX <vunnyx.sodhi@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver is used for Rockchip rk3288-based boards using a configurable
analog output (can be an headphone) and the built-in HDMI audio output
that is part of the RK3288 SoCs and use the Alsa HDMI codec driver. For
some rk3288-based boards the analog output and the hdmi audio are plugged
on the same i2s line, so we have to do the same in the driver by using a
DAI link CPU to multicodecs. This configuration can be found for example
on the Radxa Rock2 or the Firefly-RK3288.
This commit is based on the initial work that was done by Sjoerd Simons
<sjoerd.simons@collabora.com> with some improvements.
Signed-off-by: Romain Perier <romain.perier@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, we have to select these symbols explictly via Kconfig, from
another entry. If we plan to use generic audio drivers like
simple-audio-card, the user need to be able to enable these symbols
directly via the menuconfig.
This commit also fixes unmet dependencies to SND_SOC_IMX_ES8328 caused
by these changes.
Signed-off-by: Romain Perier <romain.perier@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, the function that changes the DAI format only supports master
mode. Trying to use a slave mode exits the function with -EINVAL and
leave the codec misconfigured. This commits adds support for enabling
the slave mode.
Signed-off-by: Romain Perier <romain.perier@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HDMI registers channel map controls per PCM. As PCMs are not
registered during dai_link init callback, store the pcm ids and
codec DAIs during this init callback.
Register for late probe and call the jack_init API which also
registers channel map in the late probe callback handler.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To support MST moved pin to port, this changes the routes based on
port. So change the route in bxt_da7219_max98357a machine.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To support MST moved pin to port, this changes the routes based on
port. So change the route in skl_rt286 machine.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HDMI registers channel map controls per pcm. As PCMs are not
registered during dai_link init callback, store the pcm ids and
codec DAIs during this init callback.
Register for late probe and call the jack_init API which registers
channel map in the late probe callback handler.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Channel info is part of the pcm parameter and channel map control is
created for each pcm. So move channel info to pcm instead of pin
structure and the mutex lock to pcm.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is a relict of 6553bf06a3 ("ASoC: Don't try to register debugfs
entries if the parent does not exist").
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some SoCs have a reset line that must be asserted/deasserted.
This patch adds a quirk to handle the new compatible
"allwinner,sun6i-a31-i2s" which will deassert the reset
line on probe function and assert it on remove's one.
This new compatible is useful in case of A33 codec driver, for example.
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>
Add the sun8i audio codec which handles the digital register of
A33 codec.
The driver handles only the basic playback from the DAC to headphones.
All other features (microphone, capture, etc) will be added later.
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>
When playing a sound for the first time, a short delay, where the audio
file is not played, can be noticed.
On a second play (right after), the sound is played correctly.
If we wait a short time (~5 sec which corresponds to the aplay
timeout), the delay is back.
This patch fixes it by using an event on headphone amplifier.
It allows to keep the amplifier enable while playing a sound.
A delay of 700ms allows to wait that the amplifier is powered-up
before playing the sound.
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>
Sync convert rate settings should be availabled *after* Playing.
Thus, src->sync should be reset first of init function.
Otherwise, it will set remaining settings when it start playing.
This patch fixes it. Thanks to Yokoyama-san
Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Missed unused variable in previous changes, oops.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Optional gpio handling should not cause an error status and prevent
probing if it's missing. Remove error return for -ENOENT case and
move error message to dev_info
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can get audio errors if hitting deeper idle states on omaps:
[alsa.c:230] error: Fatal problem with alsa output, error -5.
[audio.c:614] error: Error in writing audio (Input/output error?)!
This seems to happen with off mode idle enabled as power for the
whole SoC may get cut off between filling the McBSP fifo using DMA.
While active DMA blocks deeper idle states in hardware, McBSP
activity does not seem to do so.
Basing the QoS latency calculation on the FIFO size, threshold,
sample rate, and channels.
Based on the original patch by Tony Lindgren
Link: https://patchwork.kernel.org/patch/9305867/
Signed-off-by: Matt Ranostay <matt@ranostay.consulting>
Signed-off-by: Liam Breck <kernel@networkimprov.net>
Tested-by: Tony Lindgren <tony@atomide.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The combination of dev_err() and __func__ make most of these log prints
over 100 chars long. Remove the usage of __func__ to clean the kernel
log and as the usage is not necessary to identify the individual log
prints.
Signed-off-by: Bjorn Andersson <bjorn.andersson@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Clean up the clock calling code by removing numerous IS_ERR() checks by
just assigning the clock NULL; as this turn all used functions in the
clk API to nops.
Also include the word "optional" in the error message when failing to acquire
the optional osr clocks.
Signed-off-by: Bjorn Andersson <bjorn.andersson@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This macro is unused since commit e369bd006f ("ASoC: wm8741: Allow master
clock switching").
Signed-off-by: Sergej Sawazki <ce3a@gmx.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_card_jack_new function can call snd_soc_jack_add_pins for
you, so pass directly the pins struct when you create the new jack.
Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove default and set I2S mode correctly both on codec and
cpu sides
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use ACPI ID 10EC3270 to load machine driver for cht-bsw-rt5645
and add reference to 3270 to use the rt5645 mode
Tested on Asus T100HA
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some platforms use AIF2, use routing information to set ASRC as needed
Suggested-by: Bard Liao <bardliao@realtek.com>
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This driver may be used on Baytrail CR platforms where SSP2 is
not available.
Add quirks and routing detection based on work done for RT5640.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALC3270 is a low-cost version of RT5645, add ACPI ID
to enable probe and use rt5645 codec driver
Tested on Asus T100HA
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the BIOS reports this codec as RT5640 but it's a rt5670. Use the
quirk mechanism to use the cht_bsw_rt5672 machine driver
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix classic issue of having multiple codecs listed in DSDT
but a single one actually enabled. The previous code did
not handle such errors and could also lead to uninitalized
configurations
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use machine driver initially defined for CherryTrail
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
add ACPI ID 10EC5648 found e.g on Asus X205TA and use
rt5645 driver
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the BIOS incorrectly reports this codec as 5640 but it is
really a rt5670
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
RT5651 is used on some Cherrytrail platforms, add the ACPI
ID in machine table.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=156191
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add machine entry for HP X2 Pavilion 10-p100.
This notebook contains rt5640 codec, but with ACPI ID "10EC3276".
Signed-off-by: Alexandrov Stansilav <neko@nya.ai>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add ACPI ID "10EC3276" for sound card found on notebook HP Pavilion X2 10-p000.
ACPI DSDT Table on this device describes this card as ALC3276, but it is in fact rt5640.
Signed-off-by: Alexandrov Stansilav <neko@nya.ai>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing code assumes a 19.2 MHz MCLK as the default
hardware configuration. This is valid for CherryTrail but
not for Baytrail.
Add explicit MCLK configuration to set the 19.2 clock on/off
depending on DAPM events.
This is a prerequisite step to enable devices with Baytrail
and RT5645 such as Asus X205TA
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current frame sync polarity definitions are inconsistent in the
Atom/DPCM driver, fix to align with regular ASoC definitions and
update code in platform and machine drivers for RT5640 and RT5651.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch corrects an omission in bytcr_rt5640 and bytcr_rt5651.
All existing machine drivers shall not use .pm_ops to avoid a double
suspend, as initially implemented by 3f2dcbeaeb
("ASoC: Intel: Remove soc pm handling to allow platform driver handle it").
Reported-by: Shrirang Bagul <shrirang.bagul@canonical.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Most of the devices are using stereo speakers so media_loop1 and
sprot_loop default mode should be stereo.
As per default all the routing UCM configuration doesn't enable Post
processing loops it is not impacting curent configurations.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit 8f5ebb1bee
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_card_name")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit 8f5ebb1bee.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit b6defcca0a
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_prefix")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit b6defcca0a.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit 1ef5bcd57b
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_simple_widgets")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit 1ef5bcd57b.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit 7364c8dc25
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_routing")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit 7364c8dc25.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
of_match_device could return NULL, and so can cause a NULL
pointer dereference later.
Signed-off-by: Shailendra Verma <shailendra.v@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
CQ0093VC is no longer dependent on MFD_DAVINCI_VOICECODEC,
let's remove it. Otherwise, we can't compile it by COMPILE_TEST
on non-DAVINCE platform
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DMA channel(stream tag) used by the HDA link need to programmed in
codec so that codec receives packet from the link associated with the
same channel.
DMA channel is allocated in link BE dai hw_params, the same needs to be
set for the BE codec dai. Instead of using get/set dma_data(), use
dai_ops snd_soc_dai_set_tdm_slot() to set the stream tag.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of hdmi connect/disconnect or when stream need to be route to
multiple monitors, corresponding port and audio infoframe needs to be
reconfigured. Currently all the configuration are done in DAI ops which
results in silence playback.
So use dapm widget event handlers to program audio infoframe and enable
/disable port configuration when widget is power on/off.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently we are not disabling MEM_ENA on the error path, we should
really do this to unwind the state back to how it was. This patch adds a
clear of MEM_ENA on the error path, again there is no major issues
caused by this minor fix.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The booted and running flags should really only be set once all the
steps at that power level have been complete. Currently operations can
fail after the flags have been set, which would leave us in an
inconsistent state where the flags are set but the things expected to
reach that level have not happened. Whilst there isn't really any major
impact from this it is best to clean it up.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent refactoring overlooked some places which should be covered by
the pwr_lock, all code that affects or depends on the power status of
the DSP should be covered, this patch adds the missing coverage.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit c6644119a3 and
restores the ability to specify DMA channel names per DAI dma_data.
Unfortunately the functionality removed in the patch being reverted
cannot be entirely replaced by specifying DMA channel names in struct
snd_dmaengine_pcm_config as that does not cover devices with more than
2 DMA channels.
Together with patch "ASoC: Revert "samsung: Remove unneeded
initialization of chan_name"" this fixes broken sound on the s3c24xx
SoC platforms.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit cdaf9af1ea
which breaks I2S support on the non-DT Samsung SoC platforms,
since the default "tx", "rx" DMA channel names for playback
and capture streams or custom channel names in struct
snd_dmaengine_pcm_config are supported in the ASoC dmaengine
module only for devicetree booting case.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit adds a compatible string for everest,es8388. This is
an audio codec that is compatible with es8328.
Signed-off-by: Romain Perier <romain.perier@collabora.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils is getting clk by of_clk_get(), but didn't call
clk_free(). Now we can use devm_get_clk_from_child() for this purpose.
Let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 5f222a292 ("ASoC: rsnd: use for_each_rsnd_mod_xxx() ...")
modifies rsnd_dai_call() to use for_each_rsnd_mod_arrays().
Current rsnd is incrementing iterator in rsnd_mod_next(),
but the iterator will indicate +1 position in for_each loop in
this case. Incremental position should be inside for()
Reported-by: Hoan Nguyen An <na-hoan@jinso.co.jp>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add user interface to provide channel mapping.
In a first step this control is read only.
As TLV type, the control provides all configuration available for
HDMI sink(ELD), and provides current channel mapping selected by codec
based on ELD and number of channels specified by user on open.
When control is called before the number of the channel is specified
(i.e. hw_params is set), it returns all channels set to UNKNOWN.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During probe, DAIs can need to perform some actions that requests
the knowledge of the pcm runtime handle.
The callback is called during DAIs linking, after PCM device creation.
For instance this can be used to add relationship between a DAI pcm
control and the pcm device.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With codec read sometimes the pin_sense shows invalid monitor present
and eld_valid. Currently driver polls for few times to get the valid
eld data.
To avoid the latency, Instead of reading ELD from codec, read it
directly from the display driver using audio component framework.
and removed the unused direct codec helper functions.
Signed-off-by: Sandeep Tayal <sandeepx.tayal@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 66feeec9322132689d42723df2537d60f96f8e44
"RFC: ASoC: dapm: handle probe deferrals"
forgot a to update some two sites where the call
was used. The static codechecks quickly found them.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Fixes: 66feeec93221 ("RFC: ASoC: dapm: handle probe deferrals")
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The H3 SoC uses the same SPDIF block as found in earlier SoCs, but its
TXFIFO is mapped to another address.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 5f222a292 ("ASoC: rsnd: use for_each_rsnd_mod_xxx() ...")
modifies rsnd_dai_call() to use for_each_rsnd_mod_arrays().
Current rsnd is incrementing iterator in rsnd_mod_next(),
but the iterator will indicate +1 position in for_each loop in
this case. Incremental position should be inside for()
Reported-by: Hoan Nguyen An <na-hoan@jinso.co.jp>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When Ref capture is used during S0IX, only the DSP pipelines
are needed, thus remove the ignore_suspend for WoV streams so
that DMA can be suspended, but keep them for WoV endpoints.
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of enabling pin and cvt in pcm_open(), need to restore pin and
cvt state after system resume to restart the playback which is
paused/stopped before system suspend.
So enable pin and cvt in playback_prepare and call prepare when trigger
cmd is paused/started and resume to reconfigure pin and cvt.
Signed-off-by: Sachin Mokashi <sachinx.mokashi@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In system suspend, the firmware pipelines will be deleted and there
is no need to save the pipeline context. Driver will save the DPIB and
LPIB pointers in suspend.
In system resume, the firmware pipelines will be created again and the
RD/RW pointers in the Firmware buffer points to the base address. So
need to fetch the non-played data again to firmware buffer. LPIB
indicates the HW rendered position.
Instead of setting DPIB as resume point, set it to LPIB to restore from
the HW render position so that DMA would fetch the non-played data one
more time.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When pipe is pass-through, BE and FE modules are defined inside
a pipe, reset of pipe will be done in FE DAI prepare. So don't
reset in the BE prepare.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
arizona_spk_init uses snd_soc_dapm_new_control which since
commit 37e1df8c95 ("ASoC: dapm: handle probe deferrals") will
occasionally request a probe deferral. Which means we should propagate the
error out of our driver from it.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As done previously for sun4i-codec, the DMA maxburst of 4
is not supported by every SoCs so the DMA controller engine
returns "unsupported value".
As a maxburst of 8 is supported by all variants, this patch
increases it to 8.
For more details, see commit from Chen-Yu Tsai:
commit 730e2dd0cb ("ASoC: sun4i-codec: Increase DMA max burst to 8")
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SAIF base oversample rates are either 512*fs or 384*fs. An additional
divider exists within the SAIF to generate sub-multiples of these two base
rates if MCLK is required by the codec.
* The sub-rates for the 512x base rate are: 256x, 128x, 64x, and 32x.
* The sub-rates for the 384x base rate are: 192x, 96x, and 48x.
Setting the base rate depending on the modulo operation with 32 and 48
give wrong results for some mclk.
If mclk=18.432MHz both modulo operations results in 0. As testing the
result with 32 is done first, a wrong base rate of 512*fs is set instead
of the correct 384*fs.
Fix this by setting the base rate depending on the calculated sub-rate.
Signed-off-by: Jörg Krause <joerg.krause@embedded.rocks>
Signed-off-by: Mark Brown <broonie@kernel.org>
If SAIF0 is used in master and SAIF1 in slave mode setting the SAIF1
register in mxs_saif_set_dai_fmt() does not have any effect on the
interface as the clk gate needs to be cleared before the register can be
written.
Signed-off-by: Jörg Krause <joerg.krause@embedded.rocks>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/rt5659.c:4236:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This starts to handle probe deferrals on regulators and clocks
on the ASoC DAPM.
I came to this patch after audio stopped working on Ux500 ages
ago and I finally looked into it to see what is wrong. I had
messages like this in the console since a while back:
ab8500-codec.0: ASoC: Failed to request audioclk: -517
ab8500-codec.0: ASoC: Failed to create DAPM control audioclk
ab8500-codec.0: Failed to create new controls -12
snd-soc-mop500.0: ASoC: failed to instantiate card -12
snd-soc-mop500.0: Error: snd_soc_register_card failed (-12)!
snd-soc-mop500: probe of snd-soc-mop500.0 failed with error -12
Apparently because the widget table for the codec looks like
this (sound/soc/codecs/ab8500-codec.c):
static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = {
/* Clocks */
SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"),
/* Regulators */
SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0, 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0, 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0, 0),
SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0, 0),
So when we call snd_soc_register_codec() and any of these widgets
get a deferred probe we do not get an -EPROBE_DEFER (-517) back as
we should and instead we just fail. Apparently the code assumes
that clocks and regulators must be available at this point and
not defer.
After this patch it rather looks like this:
ab8500-codec.0: Failed to create new controls -517
snd-soc-mop500.0: ASoC: failed to instantiate card -517
snd-soc-mop500.0: Error: snd_soc_register_card failed (-517)!
(...)
abx500-clk.0: registered clocks for ab850x
snd-soc-mop500.0: ab8500-codec-dai.0 <-> ux500-msp-i2s.1 mapping ok
snd-soc-mop500.0: ab8500-codec-dai.1 <-> ux500-msp-i2s.3 mapping ok
I'm pretty happy about the patch as it it, but I'm a bit
uncertain on how to proceed: there are a lot of users of the
external functions snd_soc_dapm_new_control() (111 sites)
and that will now return an occassional error pointer, which
is not handled in the calling sites.
I want an indication from the maintainers whether I should just
go in and augment all these call sites, or if deferred probe
is frowned upon when it leads to this much overhead.
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel Broadwell machine driver will call API snd_soc_set_dmi_name() to
use DMI info to make the sound card long name.
For example, here are the changed long name for two Broadwell-based
machines:
Dell XPS-13(2015): DellInc.-XPS139343-01-0310JH
Intel WilsonBeach: Intel Corp.-BroadwellClientplatform-0.1-WilsonBeachSDS
They still share the same card name "broadwell-rt286".
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel DSP platform drivers are used by many different devices but are
difficult for userspace to differentiate. This patch adds an API to allow
the DMI name to be used in the sound card long name, thereby helping
userspace load the correct UCM configuration. Usually machine drivers
uses their own name as the sound card name (short name), and leave the
long name and driver name blank. This API will use the DMI info like
vendor, product and board to make up the card long name. If the machine
driver has already explicitly set the long name, this API will do nothing.
This patch also allows for further differentiation as many devices that
share the same DMI name i.e. Minnowboards, UP boards may be configured
with different codecs or firmwares. The API supports flavoring the DMI
name into the card longname to provide the extra differentiation required
for these devices.
For Use Case Manager (UCM) in the user space, changing card long name by
this API is backward compatible, since the card name does not change. For
a given sound card, even if there is no device-specific UCM configuration
file that uses the card long name, UCM will fall back to load the default
configuration file that uses the card name.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For selected only options the explicit dependencies do not make much sense
becase Kbuild ignores them anyway. Remove them explicitly.
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Select DW_DMAC_CORE like the rest of glue drivers do, e.g.
drivers/dma/dw/Kconfig.
While here group selectors under SND_SOC_INTEL_HASWELL and
SND_SOC_INTEL_BAYTRAIL.
Make platforms, which are using a common SST firmware driver, to be
dependent on DMADEVICES.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Rename SND_SST_MFLD_PLATFORM to SND_SST_ATOM_HIFI2_PLATFORM to make it clear
that is not only about Medfield platform.
The new name is derived from Intel Atom and HiFi2. HiFi2 is the DSP version,
it's public information for Intel *Field/*Trail parts, see
https://www.alsa-project.org/main/index.php/Firmware. By combining HiFi2 with
Atom we get a unique non-ambiguous description of the core+DSP hardware for
Intel Medfield through Intel Cherrytrail.
Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_INTEL_SKYLAKE selects SND_SOC_INTEL_SST already. Thus no need to
duplicate. Remove duplications.
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ulseep_range() uses hrtimers and provides no advantage over msleep()
for larger delays. For this large delay msleep() is preferable.
Link: http://lkml.org/lkml/2017/1/11/377
Fixes: commit 2b26dd4c1f ("ASoC: rt5660: add rt5660 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop the const qualifier as it is being added by SOC_ENUM_DOUBLE_DECL()
already which is called by SOC_ENUM_SINGLE_DECL() here.
Fixes: commit 2b26dd4c1f ("ASoC: rt5660: add rt5660 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Drop the const qualifier as it is being added by SOC_ENUM_DOUBLE_DECL()
already which is called by SOC_ENUM_SINGLE_DECL() as well as the
double const by calls to SOC_VALUE_ENUM_SINGLE_DECL() via
SOC_VALUE_ENUM_DOUBLE_DECL).
Fixes: commit d3cb2de247 ("ASoC: rt5659: add rt5659 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Declar rt5659_i2c_driver, which is only being passed to
module_i2c_driver(rt5659_i2c_driver), static.
Fixes: commit d3cb2de247 ("ASoC: rt5659: add rt5659 codec driver")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
ulseep_range() uses hrtimers and provides no advantage over msleep()
for larger delays. For this large delay msleep() is preferable.
Fixes: commit d3cb2de247 ("ASoC: rt5659: add rt5659 codec driver")
Link: http://lkml.org/lkml/2017/1/11/377
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
ulseep_range() uses hrtimers and provides no advantage over msleep()
for larger delays. Fix up the 70/80ms delays here passing the "min"
value to msleep(). This reduces the load on the hrtimer subsystem.
Link: http://lkml.org/lkml/2017/1/11/377
Fixes: commit 246693ba7b ("ASoC: rt5640: change widget sequence for depop")
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace mdelay to msleep to avoid busy loop on ak4642_lout_event().
Otherwise, sometimes playback doesn't work correctly when pulseaudio
was used.
Signed-off-by: Harunobu Kurokawa <harunobu.kurokawa.dn@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAI ID defines are back from the time when DAIs were referenced by a
numerical ID. These days a string is used instead and the defines are
unused. The last user of these defines was removed in commit f0fba2ad1b
("ASoC: multi-component - ASoC Multi-Component Support"). So remove the
defines as well.
This also means the mpc5200_psc_ac97.h file no longer has any content and
can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
When trying to use simple card with wm8962 the following probe
error happens:
wm8731 0-001a: simple-card: set_sysclk error
In simple-card.c the snd_soc_dai_set_sysclk() function is called with
clk_id as 0, which is an invalid clock for wm8731.
Adjust the clocks source definitions in wm8731.h so that the simple
card driver can work successfully
Signed-off-by: Jörg Krause <joerg.krause@embedded.rocks>
Signed-off-by: Mark Brown <broonie@kernel.org>
No one is using snd_soc_platform_trigger().
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The mono differential output for "Line Out" downmixes the stereo audio
from the mixer, instead of just taking the left channel.
Add a route from the "Right Mixer" to "Line Out Source Playback Route"
through the "Mono Differential" path, so DAPM doesn't shut down
everything if the left channel is muted.
Fixes: 0f909f98d7 ("ASoC: sun4i-codec: Add support for A31 Line Out
playback")
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No existing platform is using .bespoke_trigger.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No existing platform is using .delay.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Sample size of 24 bits use in reality 32 bits for storage. We
can safelly enable this sample size and treat the data as
32 bits.
Tested in a x86_64 platform and in ARC AXS101 SDP platform.
Signed-off-by: Jose Abreu <joabreu@synopsys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Up until now PIO mode offered only playback support. With
this patch we add support for record mode. The PCM was
refactored so that we could reuse the existing infrastructure
without many changes.
We have support for 16 and 32 bits of sample size using
only 2 channels.
Tested in a x86_64 platform and in ARC AXS101 SDP platform.
Signed-off-by: Jose Abreu <joabreu@synopsys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The default capless power mode is low voltage mode. We should set
it to high voltage mode to get fair headphone performance.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The i2s clock pre-divider 1 is used for both i2s1 and sysclk.
The i2s1 is usually used for the main i2s and the pre-divider
will be set in hw_params function.
However, if i2s2 is used, the pre-divider is not set in the hw_params
function and the default value of i2s clock pre-divider 1 is too high
for sysclk and DMIC usage. Fix by overriding default divider value to 2.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=95681
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When multiple front-ends are using the same back-end, putting state of a
front-end to STOP state upon receiving pause command will result in backend
stream getting released by DPCM framework unintentionally. In order to
avoid backend to be released when another active front-end stream is
present, put the stream state to PAUSED state instead of STOP state.
Signed-off-by: Patrick Lai <plai@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Saved firmware ctx was not never released, so release Firmware
ctx in cleanup routine.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the optimized dsp_register_poll API to poll the DSP firmware
status register rather than open coding it.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Poll implementation is not quite accurate, especially for smaller
values of timeout or timeout values close to the actual timeout needed
Use jiffies to set the timeout value and time_before() to get the
accurate time. So update the dsp register poll implementation to
provide accurate timeout using jiffies.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of passing the topology manifest info directly to IPC library,
define the manifest info in topology and use this in IPC Library.
This will remove the dependency on topology interface definition with
IPC library.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As requirements to bring up audio paths are continuous getting tighter
and the DSP download to most ADSP devices happens over an external bus
it can become an important factor in the path bring up time. As such
sometimes it is a reasonable trade off to download the firmware ahead of
when it will be required and take a small hit on power consumption for
keeping the core powered up.
This "preloading" adds an additional control for each DSP core "DSPx
Preload Switch" that when set to true will power up the DSP core and
download the firmware currently selected in the "DSPx Firmware" control.
Whilst the core is preloaded the current firmware can not be changed and
the CODEC will be kept powered up and SYSCLK held on. Although future
improvements may allow the SYSCLK to be powered down as well because
the hardware only requires SYSCLK whilst the download is actually taking
place, but this is not covered in this series.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are literally dozens of Insyde devices with a different
name but with the same audio routing. Use a generic quirk to
match on vendor name only to avoid recurring edits of the
same thing.
Signed-off-by: youling257 <youling257@gmail.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Frequency value of zero did not make sense, use same 24.576MHz
setting and only change the clock source in idle mode
Suggested-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAI ID defines are back from the time when DAIs were referenced by a
numerical ID. These days a string is used instead and the defines are
unused. The last user of these defines was removed in commit f0fba2ad1b
("ASoC: multi-component - ASoC Multi-Component Support"). So remove the
defines as well.
This also means the pxa2xx-ac97.h file no longer has any content and can be
removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Tested-by: Robert Jarzmik <robert.jarzmik@free.fr> (for mioa701_wm9713)
Signed-off-by: Mark Brown <broonie@kernel.org>
The fsl_ssi fifo watermark is by default set to 2 free spaces (i.e.
activate DMA on FIFO when only 2 spaces are left.) This means the
DMA must service the fifo within 2 audio samples, which is just not
enough time for many use cases with high data rate. In many
configurations the audio channel slips (causing l/r swap in stereo
configurations, or channel slipping in multi-channel configurations).
This patch gives more breathing room and allows the SSI to operate
reliably by changing the fifio refill watermark to 8.
There is no change in behavior for older chips (with an 8-deep fifo).
Only the newer chips with a 15-deep fifo get the new behavior. I
suspect a new fifo depth setting could be optimized on the older
chips too, but I have not tested.
Signed-off-by: Caleb Crome <caleb@crome.org>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
"DAC L2 Power" and "DAC R2 Power" are used by both rt5639 and rt5640.
But it was defined in rt5640_specific_dapm_widgets[]. Move them to
rt5640_dapm_widgets will let both rt5639 and rt5640 can use it.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The clk_ref_div is not configured in the correct position of the
register. The patch fixes that clk_ref_div, Pre-Scalar, is assigned
the wrong value.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If a module is not available in a pipeline, fail safely rather than
causing oops.
Signed-off-by: G Kranthi <gudishax.kranthikumar@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It has been seen that some newer SoCs have a different TX FIFO
address and we already have the difference with the A31 requiring
a reset. Add a quirks structure so that these can be managed
easily.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm components are now handled by the ALSA SoC SPDIF DIT driver
so can be removed.
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When the codec is powered on, it's registers are in reset state as the
power off will do a soft reset of the codec.
After the register sync we need to add delay to remove the pop-noise on
stream start.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The RESET register only have one self clearing bit and it should not be
cached. If it is cached, when we sync the registers back to the chip we
will initiate a software reset as well, which is not desirable.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds pointer to I2S device to clk_register_* functions.
This in the future allow clock framework to ensure proper runtime state
of the I2S device during all operations on the clocks provided by I2S
module.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds handling of parent operational clock to runtime PM
callbacks. This way it is ensured that when I2S module is in runtime
suspended state, all its parent clocks are disabled and unprepared.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch moves saving and restoring I2S registers to runtime PM
operations, what prepares the driver to operate with audio power domain.
When support for audio power domain is enabled and the domain is being
turned off, the I2S module will loose its context (registers), so runtime
callbacks have to handle it. System sleep suspend/resume operation are
implemented on top of runtime PM operations with generic
pm_runtime_force_suspend/resume helpers.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds calls to pm_runtime_get/put to ensure that any access to
I2S registers is done with proper (active) runtime PM state of I2S device.
Till now the driver enabled runtime PM, but didn't manage the state during
driver operation. The driver worked fine only because the runtime PM
callbacks managed device clock, which was enabled all the time because of
the additional enable call in the driver's probe function.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some unknown (maybe historical?) reasons support for secondary I2S DAI
was implemented by adding additional virtual platform device, which was
then probed again with the main I2S driver. This pattern is really hard
to follow and provides no benefits, so lets remove this hack and register
both DAIs during linear probe of Exynos I2S controller driver.
Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently binding of auxiliary devices doesn't work as in
soc_bind_aux_dev() function a bound component is not being added
to any list and in soc_probe_aux_devices() we are trying to walk
the component_dev_list list to probe auxiliary components but
at that time this list doesn't contain any auxiliary components
since they are being added to the card only in soc_probe_component().
This patch adds a list to the card where are stored bound but not
probed auxiliary devices, so that all aux devices can be probed.
Fixes: 1a653aa447 "ASoC: core: replace aux_comp_list to component_dev_list"
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Protect against corrupt firmware files by ensuring that the length we
get for the data in a region actually lies within the available firmware
file data buffer.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit df1a2776a7 ("ASoC: Intel: bytcr_rt5640: add MCLK support")
was merged but the corresponding clock framework patches have not,
after being bumped from audio to clock to x86 domains. The missing
clock-related patches result in a regression starting with 4.9 with
the audio card not being created.
Rather than reverting this commit and all following updates already
queued up for 4.10, handle run-time dependency on MCLK and fall back
to the previous bit-clock mode. This provides the same functionality
as in 4.8 for Baytrail devices. On Baytrail-CR most devices remain
silent with this fallback but additional patches are needed anyway.
As suggested by Mark Brown, the fallback is only allowed with -ENOENT,
all other run-time errors, including -EPROBE_DEFER, will stop the probe
with no sound card registered.
This patch should be applied to -stable as well as ASoC 4.10 fixes
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For TDM mode, the I2S data out line can be shared between mutliple
codecs. In this scenario, only the active codec should be using
the line, and all others should be high impedance. However,
currently in the driver this configuration isn't set when capture
is inactive, and the line remains driven.
This patch updates the AIF_OUT widget to set the DAI output pin of
the device as high impedance when not in use.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Platform only has pcm_new/pcm_free feature,
but it should be supported on Component level. This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/wm0010.c does not use any miscdevice so this patch
remove this unnecessary inclusion.
Signed-off-by: Corentin Labbe <clabbe.montjoie@gmail.com>
Acked-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/rt5677-spi.c does not use any miscdevice so this patch
remove this unnecessary inclusion.
Signed-off-by: Corentin Labbe <clabbe.montjoie@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/rt5514-spi.c does not use any miscdevice so this patch
remove this unnecessary inclusion.
Signed-off-by: Corentin Labbe <clabbe.montjoie@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Removed the unused function skl_get_format as the format is calculated
directly using the HDA core API.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds helper function to configure the host/link DMA when
the DMA is in decoupled mode.
Next patch adds the usage of this helper routines for configuring
DMA in Mixer event handler.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If system is suspended when PCM was paused/stopped, restart doesn't
configure DMA as it is we are in Pause state and results in IO error
eventually.
Configure host/link DMA before initializing DSP Gateway copier module
instead of DAI prepare(). So moved DMA configuration to mixer PRE_PMD
widget handler instead of DAI prepare.
This uses previously added new API to do the configuration and removes
old DAI prepare code.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To configure Host/Link DMA, additionally link index and format
are required based on the hw params. So added these parameters in
the pipe params and in hw_params the pipe params are updated.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This breaks devicetree compatibility, but in this case that is ok. All
affected units are either on my desk, or running an even older version
of the driver that is not compatible with the upstreamed version anyway
(and when these other units are eventually updated, they will get a
fresh dtb as well, so that is not a significant problem either).
All of that is of course assuming that noone else has managed to build
something that can use this driver, but that seems extremely improbable.
Signed-off-by: Peter Rosin <peda@axentia.se>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds a control to allow swapping HiFi DAC Left/Right channels.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver was checking for non-NULL address of struct's members:
- s3c_audio_pdata->type (union),
- s3c_audio_pdata->type.i2s (embedded struct).
This is pointless as these will be always non-NULL. The 's3c_audio_pdata'
is always initialized in static memory so it will be zeroed.
Additionally the 'type' member was an union with only one member.
It is safe to reorganize the structures to get rid of useless union and
checks for addresses to fix the coccinelle warning:
>> sound/soc/samsung/i2s.c:1270:2-4: ERROR: test of a variable/field address
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Reviewed-by: Bartlomiej Zolnierkiewicz <b.zolnierkie@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Both SND_SOC_SMARTQ and SND_SOC_SAMSUNG_TM2_WM5110
use gpio/consumer.h
This patch adds GPIOLIB || COMPILE_TEST to Kconfig entries
to fix runtime dependency.
See commit 638f958bae
("extcon: Allow compile test of GPIO consumers if !GPIOLIB")
for similar problem and explanations.
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Reported-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the following build errors on X86_32 !GPIOLIB
sound/soc/samsung/tm2_wm5110.c:220:3: error: implicit declaration
of function 'gpiod_set_value_cansleep' [-Werror=implicit-function-declaration]
sound/soc/samsung/tm2_wm5110.c:438:24: error: implicit declaration
of function 'devm_gpiod_get' [-Werror=implicit-function-declaration]
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
kcontrol->private_value is being kfree'd after kcontrol has been freed
(in previous call to snd_ctl_remove). Instead, fix this by kfreeing
the private_value before kcontrol.
CoverityScan CID#1388311 "Read from pointer after free"
Fixes: eea3dd4f12 ("ASoC: topology: Only free TLV for volume mixers of a widget")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In zx_i2s_hw_params(), 'format' is initialized and assigned bits based on
params_format, but never used. So remove it.
sound/soc/zte/zx296702-i2s.c: In function ‘zx_i2s_hw_params’:
sound/soc/zte/zx296702-i2s.c:228:21: warning: variable ‘format’ set but not used [-Wunused-but-set-variable]
unsigned long val, format;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In smdk_hw_params(), 'bfs' is initialized and assigned bits based on
params_width, but never used.
We could have removed the whole switch case but then driver might be
relying on checking bits, so I have kept the case for now.
sound/soc/samsung/smdk_wm8580.c: In function ‘smdk_hw_params’:
sound/soc/samsung/smdk_wm8580.c:35:6: warning: variable ‘bfs’ set but not used [-Wunused-but-set-variable]
int bfs, rfs, ret;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In sst_media_close(), 'ret_val' is initialized and assigned as return value
of stream ops close but never used. So remove it.
ound/soc/intel/atom/sst-mfld-platform-pcm.c: In function ‘sst_media_close’:
sound/soc/intel/atom/sst-mfld-platform-pcm.c:360:6: warning: variable ‘ret_val’ set but not used [-Wunused-but-set-variable]
int ret_val = 0, str_id;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In img_prl_out_hw_params(), 'format' is initialized but never used.
So remove it.
sound/soc/img/img-parallel-out.c: In function ‘img_prl_out_hw_params’:
sound/soc/img/img-parallel-out.c:126:19: warning: variable ‘format’ set but not used [-Wunused-but-set-variable]
snd_pcm_format_t format;
Cc: Damien.Horsley <Damien.Horsley@imgtec.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In pcm3168a_hw_params(), 'format' is initialized but never used.
sound/soc/codecs/pcm3168a.c: In function ‘pcm3168a_hw_params’:
sound/soc/codecs/pcm3168a.c:405:19: warning: variable ‘format’ set but not
used [-Wunused-but-set-variable]
snd_pcm_format_t format;
Cc: Damien.Horsley <Damien.Horsley@imgtec.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In max9867_dai_set_fmt(), 'ret' is initialized as return value of
regmap_raw_write() but never checked, so remove this and assignement.
sound/soc/codecs/max9867.c: In function ‘max9867_dai_set_fmt’:
sound/soc/codecs/max9867.c:312:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable]
int ret;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In adau17x1_pll_event(), 'ret' is initialized as return value of
regmap_raw_write() but never checked, so remove this and assignement.
sound/soc/codecs/adau17x1.c: In function ‘adau17x1_pll_event’:
sound/soc/codecs/adau17x1.c:68:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable]
int ret;
Cc: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In acp_dma_hw_params(), 'dma_buffer' is initialized, but not used. So
remove it.
sound/soc/amd/acp-pcm-dma.c: In function ‘acp_dma_hw_params’:
sound/soc/amd/acp-pcm-dma.c:673:25: warning: variable ‘dma_buffer’ set but not used [-Wunused-but-set-variable]
struct snd_dma_buffer *dma_buffer;
Cc: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In max98090_probe(), code checks for micbias being out of range. The
'micbias' variable in unsigned and checked against M98090_MBVSEL_2V2 which
is zero, so remove this check.
sound/soc/codecs/max98090.c: In function ‘max98090_probe’:
sound/soc/codecs/max98090.c:2459:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
} else if (micbias < M98090_MBVSEL_2V2 || micbias > M98090_MBVSEL_2V8) {
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In soc_tplg_pcm_elems_load, a variable 'err' is initialized but not
used.
It is assigned return values for pcm_new_ver() but never checked, so
remove it.
sound/soc/soc-topology.c: In function ‘soc_tplg_pcm_elems_load’:
sound/soc/soc-topology.c:1865:9: warning: variable ‘err’ set but not used [-Wunused-but-set-variable]
int i, err;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In process_fw_async_msg(), a variable 'msg_high' is initialized but
not used. So remove it.
sound/soc/intel/atom/sst/sst_ipc.c: In function ‘process_fw_async_msg’:
sound/soc/intel/atom/sst/sst_ipc.c:263:24: warning: variable ‘msg_high’ set but not used [-Wunused-but-set-variable]
union ipc_header_high msg_high;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In sst_free_stream(), a variable 'ops' is initialized but
not used. So remove it.
sound/soc/intel/atom/sst/sst_stream.c: In function ‘sst_free_stream’:
sound/soc/intel/atom/sst/sst_stream.c:397:24: warning: variable ‘ops’ set but not used [-Wunused-but-set-variable]
struct intel_sst_ops *ops;
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In skl_tplg_mixer_dapm_post_pmd_event(), a variable 'ret' is initialized but
not used.
We don't check return of skl_delete_pipe, so remove the assignment as
well, so remove this variable.
sound/soc/intel/skylake/skl-topology.c: In function ‘skl_tplg_mixer_dapm_post_pmd_event’:
sound/soc/intel/skylake/skl-topology.c:976:6: warning: variable ‘ret’ set but not used [-Wunused-but-set-variable]
int ret = 0;
^
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On an error, snd_ctl_add already free's kctrl, so calling snd_ctl_free_one
to free it again leads to a double free error. Fix this by removing
the extraneous snd_ctl_free_one call.
Issue found using static analysis with CoverityScan, CID 1372908
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No dramatic changes are found in this development cycle, but as usual,
many commits are applied in a wide range of drivers.
Most of big changes are in ASoC, where a few bits of framework work
and quite a lot of cleanups and improvements to existing code have
been done. The rest are usual stuff, a few HD-audio and USB-audio
quirks and fixes, as well as the drop of kthread usages in the whole
subsystem.
Below are some highlights:
ASoC:
- Support for stereo DAPM controls
- Some initial work on the of-graph sound card
- regmap conversions of the remaining AC'97 drivers
- A new version of the topology ABI; this should be backward compatible
- Updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona,
Intel skylake, atom-sst
- New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and
Realtek RT5665
USB-audio:
- Yet another race fix at disconnection
- Tolerated packet size calculation for some Android devices
- Quirks for Axe-Fx II, QuickCam, TEAC 501/503
HD-audio:
- Improvement of Dell pin fixup mapping
- Quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic
Misc:
- Replace all kthread usages with simple works
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Merge tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"No dramatic changes are found in this development cycle, but as usual,
many commits are applied in a wide range of drivers.
Most of big changes are in ASoC, where a few bits of framework work
and quite a lot of cleanups and improvements to existing code have
been done. The rest are usual stuff, a few HD-audio and USB-audio
quirks and fixes, as well as the drop of kthread usages in the whole
subsystem.
Below are some highlights:
ASoC:
- support for stereo DAPM controls
- some initial work on the of-graph sound card
- regmap conversions of the remaining AC'97 drivers
- a new version of the topology ABI; this should be backward
compatible
- updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona,
Intel skylake, atom-sst
- new drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and
Realtek RT5665
USB-audio:
- yet another race fix at disconnection
- tolerated packet size calculation for some Android devices
- quirks for Axe-Fx II, QuickCam, TEAC 501/503
HD-audio:
- improvement of Dell pin fixup mapping
- quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic
Misc:
- replace all kthread usages with simple works"
* tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (296 commits)
ALSA: hiface: Fix M2Tech hiFace driver sampling rate change
ALSA: usb-audio: Eliminate noise at the start of DSD playback.
ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC
ASoC: wm_adsp: wm_adsp_buf_alloc should use kfree in error path
ASoC: topology: avoid uninitialized kcontrol_type
ALSA: usb-audio: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks
ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
ASoC: zte: spdif: correct ZX_SPDIF_CLK_RAT define
ASoC: zte: spdif and i2s drivers are not zx296702 specific
ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting
ASoC: rsnd: enable/disable ADG when suspend/resume timing
ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params
ALSA: cs46xx: add a new line
ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture
ASoC: nau8825: disable sinc filter for high THD of ADC
ALSA: usb-audio: more tolerant packetsize
ALSA: usb-audio: avoid setting of sample rate multiple times on bus
ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting
ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO
ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq
...
We can no longer rely on the return value of
devm_snd_dmaengine_pcm_register(...) to check if the DMA
handle is declared in the DT.
Previously this check activated PIO mode but currently
dma_request_chan returns either a valid channel or -EPROBE_DEFER.
In order to activate PIO mode check instead if the interrupt
line is declared. This reflects better what is documented in
the DT bindings (see Documentation/devicetree/bindings/sound/
designware-i2s.txt).
Also, initialize use_pio variable which was never being set
causing PIO mode to never work.
Signed-off-by: Jose Abreu <joabreu@synopsys.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
buf was allocated by kzalloc() so it should be passed to kfree()
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When num_kcontrols is zero, widget->dobj.widget.kcontrol_type
gets set to an uninitialized local variable:
sound/soc/soc-topology.c: In function 'soc_tplg_dapm_widget_create':
sound/soc/soc-topology.c:1566:36: error: 'kcontrol_type' may be used uninitialized in this function [-Werror=maybe-uninitialized]
I could not figure out which of the valid types would be appropriate
here, so this sets it to '0', which is invalid but at least well-defined
here. There is probably a better way to address the issue.
Fixes: eea3dd4f12 ("ASoC: topology: Only free TLV for volume mixers of a widget")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Mark Brown <broonie@kernel.org>
The macro ZX_SPDIF_CLK_RAT should be 2 instead of 4. With this
fix, we can get correct audio output on HDMI through SPDIF interface.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
ZTE ZX SPDIF and I2S drivers can work on not only ZX296702 but also
other ZTE ZX family SoCs like ZX296718, which is an arm64 platform.
Let's make a few renaming and tweak the Kconfig a bit to get the drivers
available for other ZTE ZX platforms.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Reviewed-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver setups BRGCKR/BRRA/BRRB when .probe timing.
But it breaks sound after Suspend/Resume. These should be setups
every start timing.
This patch is tested on R-Car Gen3 Salvator-X board
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Gaku Inami <gaku.inami.xw@bp.renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsnd driver enables ADG clock when .probe timing,
but it breaks sound after Suspend/Resume. These should be setups
every suspend/resume timing too.
This patch is tested on R-Car Gen3 Salvator-X board
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Gaku Inami <gaku.inami.xw@bp.renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ssi->usrcnt will be updated on snd_soc_dai_ops::trigger,
but snd_pcm_ops::hw_params will be called *before* it.
Thus, ssi->usrcnt is still 0 when 1st call.
rsnd_ssi_hw_params() needs to check its called count, this means
trigger should be if (ssi->usrcnt) instead of if (ssi->usrcnt > 1).
Reported-by: Nguyen Viet Dung <nv-dung@jinso.co.jp>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch updates FE channel constraints & BE fixup to support
quad channel DMIC capture.
DMIC pin's BE fixup is configured based on channel input, i.e.
either stereo or quad.
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This bit will enable 4th order SINC filter.
=1, filter will enable; but it consumes higher power.
=0, the sinc filter is disable, and it should always keep 0 value to
get high THD.
Therefor, disable the filter when codec initiation for better
performance when recording.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The logic of "value = ~CS35L34_MCLK_DIV & CS35L34_MCLK_RATE_XXXXXX;" is
unnecessary complex. By setting CS35L34_MCLK_DIV | CS35L34_MCLK_RATE_MASK
as the mask for regmap_update_bits() call, what the code does is exactly
the same as setting value = CS35L34_MCLK_RATE_XXXXXX.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After the boot of the SST FW the firmware version is send back
to the driver. This patch is saving the FW version inside the
driver.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch is adding a sysfs entry in order to be able to get
access to SST FW version.
Signed-off-by: Sebastien Guiriec <sebastien.guiriec@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Using regmap_update_bits(..., mask, 1) with 'mask' following (1 << k)
and k greater than 0 is wrong. Indeed, _regmap_update_bits will perform
(mask & 1), which results in 0 if LSB of mask is 0. Thus the call
regmap_update_bits(..., mask, 1) is in reality equivalent to
regmap_update_bits(..., mask, 0).
In such a case, the correct use is regmap_update_bits(..., mask, mask).
This driver is performing such a mistake with the CS42L56_AIN*_REF_MASK
masks, which equal 0x10, 0x20, 0x40 and 0x80. Fix the driver to make it
consistent with the API. Please note that this change is untested,
as I do not have this piece of hardware. Testers are welcome!
Signed-off-by: Florian Vaussard <florian.vaussard@heig-vd.ch>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch will check the type of embedded controls for a widget, and
only free the TLV of volume mixers. Bytes controls don't have TLV.
Just free the private value which is used as struct soc_mixer_control
for volume mixers or soc_bytes_ext for bytes controls. No need to cast
to these types before freeing it.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch can create multiple enumerated mixer controls for a widget.
Previously topology kernel driver assumes a widget can have only one
emumerated mixer control. We need to remove this restriction for Broxton.
Its firmware modules (widgets) may need multiple enum controls based on
the channel and MIC combination.
No ABI change is needed. The ABI allows a widget to embed multiple
controls.
Reported-by: G Kranthi <gudishax.kranthikumar@intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple_card_utils was created as simple_card_core in 1st prototype,
and current code still have it. Let's tidyup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SSIU was controlled by SSI before, but
commit c7f69ab53("ASoC: rsnd: use mod base common method on SSIU")
separated it into ssiu.c
But, it didn't care about rsnd_get_dalign() for judging SSI_BUSIF_DALIGN
register value which changes the stream data order.
This function will be called from cmd/src/ssiu now, but current code
still cares ssi, not ssiu.
This patch fix it up
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Backmerge tag 'v4.9-rc8' into drm-next
Linux 4.9-rc8
Daniel requested this so we could apply some follow on fixes cleanly to -next.
This patch adds the sound machine driver for the TM2 and TM2E boards.
Speaker and headphone playback, Main Mic capture, Bluetooth, Voice
call and external accessory are supported.
Signed-off-by: Inha Song <ideal.song@samsung.com>
[k.kozlowski: rebased on 4.1]
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
[s.nawrocki: rebased to 4.7, adjustment to the ASoC core changes,
removed unused ops and direct calls to the max98504 function,
added parsing of "audio-amplifier" and "audio-codec"
properties, added TDM API calls, switched to gpiod API]
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Codec only has suspend/resume feature,
but it should be supported on Component level. This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now, Card has component_dev_list, we can replace aux_comp_list
to component_dev_list with new auxiliary flags
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current Card has Codec list (= codec_dev_list), but Codec will be
removed in the future. Because of this reason, this patch adds
new Component list in Card, and replace Codec list.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The codec on the H3 is similar to the one found on the A31. One key
difference is the analog path controls are routed through the PRCM
block. This is supported by the sun8i-codec-analog driver, and tied
into this codec driver with the audio card's aux_dev.
In addition, the H3 has no HP (headphone) and HBIAS support, and no
MIC3 input. The FIFO related registers are slightly rearranged.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Rob Herring <robh@kernel.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The codec in the A23 is similar to the one found on the A31. One key
difference is the analog path controls are routed through the PRCM
block. This is supported by the sun8i-codec-analog driver, and tied
into this codec driver with the audio card's aux_dev.
In addition, the A23 does not have LINEOUT, and it does not support
headset jack detection or buttons.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Rob Herring <robh@kernel.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of hard-coded "i2c-10EC5670:00", use the translation helper to
avoid the mismatch between i2c-codec and ACPI strings just like what
we've done for bytcr_rt5640. This gives more robust binding on funky
devices like Dell Wyse 3040.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is an issue about pop noise in NAU88L25 as follows.
Issue 54078: Chell_headphone pop back from S3
(1)Play directly to hw, bypassing CRAS:
sox -b 16 -n -t alsa hw:0,0 synth sine 200 sine 200
(2)Close lid or powerd_dbus_suspend, then press a key to resume.
(3)no audio after resume
(4)Audio will be back after close then reopen the pcm device.
After verification, we find one defect is that semaphone lock is not
long enough and expired. In this situation, the playback comes back
early but pauses a while to wait for the crosstalk detection done.
But the detection spends too much time and lock time is up. Therefore,
the playback and jack detection sequence interfere to each other.
That breaks sequence and makes noise. The driver extends the lock
time for the issue.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The usage pattern of kthread worker in Intel SST drivers can be
replaced gracefully with the normal workqueue, which is more light-
weight and easier to manage in general. Let's do it.
While in the replacement, move the schedule_work() call inside the
spinlock for excluding the race, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to modify the state of some of our own pins and are currently
not taking account that the pin name may have a name_prefix applied
to it.
Replace the snd_soc_dapm_x_pin functions with the equivalent
snd_soc_component_x_pin functions so that any name_prefix will be
handled automatically.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It's often the case that a codec driver will need to control its
own pins. However, if a name_prefix has been applied to this codec it
must be included in the name passed to any of the snd_soc_dapm_x_pin()
functions.
The behaviour of the existing pin control functions is reasonable, since
you may want to search for a fully-specified name within the scope of an
entire card. This means that we can't apply the prefix in these functions
because it will break card-scope searches.
Constructing a prefixed string "manually" in codec drivers leads to a lot
of repetition of the same code.
To make this tidier in codec drivers this patch adds a new set of
equivalent functions that take a struct snd_soc_component instead of a
dapm context and automatically add the component's name_prefix to the
given name. This makes it a simple change in codec drivers to be
prefix-safe.
The new functions are not quite trivial enough to be inlines and the
compiler won't be able to compile-away any part of them.
Although it looks somewhat inefficient to have to allocate a temporary
buffer and combine strings, the current design of the widget list
doesn't lend itself to a more optimized implementation - it's a single
list of all widgets on a card and is searched linearly for a matching
string. As pin state changes are generally low-frequency events it's
unlikely to be a significant issue - at least not enough to rewrite the
widget list handling just for this.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5670 driver supports also RT5672 codec, but its ACPI ID is missing.
This was found on Dell Wyse 3040 box.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Dell Wyse 3040 requires the platform workaround for enabling dmic and
jack detection as defined for Braswell CRB.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the missing ACPI ID 10EC5672 for Cherry Trail, which bounds with
cht-bsw-rt5672 driver. This combination was found on Dell Wyse 3040.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
drm/qxl: various bugfixes and cleanups,
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Merge tag 'drm-qemu-20161121' of git://git.kraxel.org/linux into drm-next
drm/virtio: fix busid in a different way, allocate more vbufs.
drm/qxl: various bugfixes and cleanups,
* tag 'drm-qemu-20161121' of git://git.kraxel.org/linux: (224 commits)
drm/virtio: allocate some extra bufs
qxl: Allow resolution which are not multiple of 8
qxl: Don't notify userspace when monitors config is unchanged
qxl: Remove qxl_bo_init() return value
qxl: Call qxl_gem_{init, fini}
qxl: Add missing '\n' to qxl_io_log() call
qxl: Remove unused prototype
qxl: Mark some internal functions as static
Revert "drm: virtio: reinstate drm_virtio_set_busid()"
drm/virtio: fix busid regression
drm: re-export drm_dev_set_unique
Linux 4.9-rc5
gp8psk: Fix DVB frontend attach
gp8psk: fix gp8psk_usb_in_op() logic
dvb-usb: move data_mutex to struct dvb_usb_device
iio: maxim_thermocouple: detect invalid storage size in read()
aoe: fix crash in page count manipulation
lightnvm: invalid offset calculation for lba_shift
Kbuild: enable -Wmaybe-uninitialized warnings by default
pcmcia: fix return value of soc_pcmcia_regulator_set
...
Disable IRQ when jack is NULL in rt298_mic_detect. So machine driver
can use it in suspend/resume.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel sst driver spews lots of kernel messages in INFO level;
typically, it gives a kernel message at each time it allocates or
frees a stream, or at each time when a stream is started or stopped.
This will piles up quickly, and it's almost useless for normal users,
so let's hide them to KERN_DEBUG level.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Functions are given external linkage by default making the extern's
unnecessary, as such remove them.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Functions are given external linkage by default making the extern's
unnecessary, as such remove them.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that I2S conflig blob structure is no longer in use, remove it.
Signed-off-by: GuruprasadX Pawse <guruprasadx.pawse@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
I2S blob config structure can change between DSP copier module versions.
We are not modifying these structures anyway and NHLT blobs specify them
properly, so use the blob directly to send the dma_control ipc instead
of using the predefined I2S blob config structure.
Signed-off-by: GuruprasadX Pawse <guruprasadx.pawse@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Oddly enough, my version of GCC misses this uninitialized variable.
Fixes: ba2ff3027b ("ASoC: sunxi: Add support for A23/A33/H3 codec's analog path controls")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Acked-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fields of structs with global storage are implicitly initialized to 0/NULL,
there is usually no need to do this explicitly.
Removing the initialization of the legacy suspend/resume callback fields
also gets the driver ready for the day when they are eventually removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Resources should be requested when the device is probed on the control bus
rather then when the CODEC is bound to the sound card. This allows things
like probe deferring and device managed allocations to work.
So move the GPIO request calls from the CODEC probe to the bus probe and
also make them managed along the way.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAI ID defines are back from the time when DAIs were referenced by a
numerical ID. These days a string is used for matching instead and the
defines are unused. The last user of these defines was removed in commit
f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component Support"). So
remove the defines as well.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
We are not checking the return value from a call to wm_adsp_buffer_init
it looks like this used to be returned at the bottom of the function but
probably got missed when more error paths were added. This patch adds
the appropriate error check.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The internal codec on A23/A33/H3 is split into 2 parts. The
analog path controls are routed through an embedded custom register
bus accessed through the PRCM block.
The SoCs share a common set of inputs, outputs, and audio paths.
The following table lists the differences.
----------------------------------------
| Feature \ SoC | A23 | A33 | H3 |
----------------------------------------
| Headphone | v | v | |
----------------------------------------
| Line Out | | | v |
----------------------------------------
| Phone In/Out | v | v | |
----------------------------------------
Add an ASoC component driver for it. This should be tied to the codec
audio card as an auxiliary device. This patch adds the commont paths
and controls, and variant specific headphone out and line out.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
IOMMU needs DMA mapping function to use it. One solution is that
we can use DMA mapped dev on snd_pcm_lib_preallocate_pages_for_all()
for SNDRV_DMA_TYPE_DEV. But pcm_new and dma map timing are mismatched.
Thus, this patch uses SNDRV_DMA_TYPE_CONTINUOUS for pcm_new,
and use dma_sync_single_for_xxx() for each transfer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current Renesas Sound driver requests DMA channel when .probe timing,
and release it when .remove timing. And use DMA on .start/.stop
But, Audio DMAC power ON was handled when request timing (= .probe),
and power OFF was when release timing (= .remove).
This means Audio DMAC power is always ON during driver was enabled.
The best choice to solve this issue is that DMAEngine side handle
this. But current DMAEngine API design can't solve atmic/non-atmic
context issue for power ON/OFF. So next better choice is sound
driver request/release DMA channel each time. This patch do it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The kcontrol pointer in wm_coeff_ctl is not used now.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We were adding firmware controls as card controls (using
snd_soc_add_codec_controls). The DSP is part of a specific codec so
we should be adding them as codec controls. Adding as codec controls
also means that if the codec has a name_prefix it will be added to
the control name, which won't happen when adding as a card control.
As that was the only use of the card pointer in struct wm_adsp it can
be removed.
For ADSP2 codecs a wm_adsp2_codec_probe() was added since the original
control handling was written, and that's the logical place to store a
pointer to the codec rather than delaying it until the codec is
powered-up.
For ADSP1 we don't use a codec_probe() stage so the codec pointer
initialization replaces the original card pointer initialization in
wm_adsp1_event().
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
cht_bsw_rt5645 driver allocates the own codec_id string but doesn't
release it. For simplicity, put the string in cht_mc_private; then
the string is allocated in a shot and released altogether.
Fixes: c8560b7c91 ("ASoC: cht_bsw_rt5645: Fix writing to string literal")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The I2S sound drivers for SmartQ board and WM8580 codec can be compile
tested to increase build coverage.
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of build time, Samsung ASoC drivers have rather runtime
dependency on Exynos or other Samsung platforms. For building they
require Common Clock Framework. If it is provided they could be compile
tested to increase build coverage.
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
MACH_SMDKC100, MACH_SMDKV210 and MACH_SMDKC110 are no longer supported
so we can drop the dead code. After this the driver no longer
differentiates between machines (S3C24xx machines are not supported by
it) so there is no need to override I2S device id in cpu_dai_name and
SEC_PLAYBACK dai_link can be removed as well.
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
MACH_SMDKC100 was removed in commit b8529ec1c1 ("ARM: S5PC100: no more
support S5PC100 SoC"). MACH_SMDKV210 and MACH_SMDKC110 in commit
28c8331d38 ("ARM: S5PV210: Remove support for board files").
Signed-off-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
dpcm_state_string() returns a pointer to a string literal. Modifying a
string literal causes undefined behaviour. So make the return type of the
function const char * to make it explicit that the returned value should
not be modified.
This patch is purely cosmetic, none of the users of dpcm_state_string()
attempt to modify the returned content.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm8580 driver doesn't work without CONFIG_I2C anyway.
Let's remove #if IS_ENABLED(CONFIG_I2C), and use module_i2c_driver().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm8523 driver doesn't work without CONFIG_I2C anyway.
Let's remove #if IS_ENABLED(CONFIG_I2C), and use module_i2c_driver().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The control list may contain shutdown controls for firmwares that are
not currently active, attempting to write this will at best fail. To
avoid this issue we skip any control that is not active.
Fixes: commit f4f0c4c60c ("ASoC: wm_adsp: Signal firmware shutdown
through event control")
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm9081 driver doesn't work without CONFIG_I2C anyway.
Let's remove #if IS_ENABLED(CONFIG_I2C)
And, this patch adds "depends on I2C" to Kconfig
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
uda1380 driver doesn't work without CONFIG_I2C anyway.
Let's remove #if IS_ENABLED(CONFIG_I2C), and use module_i2c_driver().
And, this patch adds "depends on I2C" to Kconfig
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The TSE-850 is an FM Transmitter Station Equipment, designed to generate
baseband signals for FM, mainly the DARC subcarrier, but other signals
are also possible.
Signed-off-by: Peter Rosin <peda@axentia.se>
Signed-off-by: Mark Brown <broonie@kernel.org>
When this driver masters BCLK and/or LRCLK, and noone has stated
differently, assume that all the bits of a frame are used.
This relieves the cpu dai users from the duty to fill in the dividers for
the common case.
Signed-off-by: Peter Rosin <peda@axentia.se>
Signed-off-by: Mark Brown <broonie@kernel.org>
const char * const [] is the preferred type for static string arrays since
this states explicitly that the individual entries are not going to be
changed. Due to limitations in the ASoC API it was not possible to use it
for enum text arrays. Commit 87023ff74b ("ASoC: Declare const properly
for enum texts") changed this, but most drivers still use 'const char
* []' as the type for their enum text arrays.
Change these occurrences of 'static * const char * []' to 'static const
char * const []'.
The conversion was done automatically using the following coccinelle
semantic patch:
// <smpl>
@disable optional_qualifier@
identifier s;
@@
static
-const char *
+const char * const
s[] = ...;
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In commit 022d00ee0b ("ASoC: lpass-platform: Fix broken pcm data
usage") the stream specific information initialization was broken, with
the dma channel information not being initialized if there was no
alloc_dma_channel() helper function.
Before that, the DMA channel number was implicitly initialized to zero
because the backing store was allocated with devm_kzalloc(). When the
init code was rewritten, that implicit initialization was lost, and gcc
rightfully complains about an uninitialized variable being used.
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The lookup of the base register corresponding to a control is
duplicated in read and write so factor it out into a separate
function.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
const char * const [] is the preferred type for static string arrays since
this states explicitly that the individual entries are not going to be
changed. Due to limitations in the ASoC API it was not possible to use it
for enum text arrays. Commit 87023ff74 ('ASoC: Declare const properly for
enum texts') changed this, but most drivers still use 'const char
* []' as the type for their enum text arrays.
Change these occurrences of 'static * const char * []' to 'static const
char * const []'.
The conversion was done automatically using the following coccinelle semantic
patch:
// <smpl>
@disable optional_qualifier@
identifier s;
@@
static
-const char *
+const char * const
s[] = ...;
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a missing whitespace in the dev_err message between
"will" and "lead". Add the whitespace.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC Compress ops have only platform ops and no DAI ops unlike PCM device
where we have both platform ops as well as DAI ops.
So add compress dai ops and add this new structure to the ASoC core to make
compressed devices a first class ASoC citizen
Again like PCM ops, drivers are free to implement either or both of
these ops based on device needs.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Most Baytrail-CR devices use analog differential microphones,
modify capture default to avoid DMI quirks. Keep digital mics
for all other configurations.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Baytrail-CR devices usually expose information in the DSDT
which can be used to auto-detect AIF1/AIF2 connections.
The CHAN package contains two integers, the first one describes
the AIF number (1: AIF1, 2: AIF2) and the second the MCLK
value (ignored in this patch)
For example the following information is found in Lenovo 100s:
Device (RTEK) {
[...]
Name (CHAN, Package (0x02)
{
One,
0x017D7840
})
While on Asus T100TAF the package values are:
Name (CHAN, Package (0x02)
{
0x02,
0x017D7840
})
This patch relies on the new common routine to extract
a package exposed by a device indexed with the HID value.
The CHAN package contents are queried from the machine driver
and stored in a structure.
If this auto-detection fails (missing or bad package in the
BIOS), the routing falls back to SSP0-AIF2.
Note that quirks may still be needed to support mono speakers
or microphone, but this should reduce the number of issues with
Baytrail significantly.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a new common routine to extract a package exposed by a
device indexed with the HID value. The functionality is
implemented without assumptions on the package type or
structure to allow for reuse. The caller is responsible for
defining the name and allocating structures to store the
results, ACPICA will complain in case of type mismatches
or buffer size issues.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAC3101 is mostly identical to DAC3100 with the exception that it has
stereo speaker AMP instead of mono used in DAC3100.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch handles publishing acknowledged controls through ALSA.
These controls allow user-side to send events to the firmware and
wait for the firmware to acknowledge it.
Note that although acked controls only operate in the direction
host->firmware, and therefore they are write-only as seen from user-
side code, we have to make them readable to account for all the code
out there that assumes that ALSA controls are always readable (amixer
for example.)
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for firmware controls marked SYSTEM. These are
internal to the driver-firmware interface and do not have
a user-accessible ALSA control.
Signed-off-by: Stuart Henderson <stuarth@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch factors out converting a memory region type into
a name string, mainly so that it can be used in log commands.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the firmware has any system event signalling controls, signal
them during DSP PRE_PMD to tell the firmware it is about to be stopped.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SND_SOC_SMDK_WM8580_PCM Kconfig entry depends on either MACH_SMDKV210
or MACH_SMDKC110. Both of which were removed in commit 28c8331d38 ("ARM:
S5PV210: Remove support for board files") over two years ago. The driver
has been unselectable ever since.
Considering the lack of complaints about this it can be concluded that the
driver is unused and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Krzysztof Kozlowski <krzk@kernel.org>
Reviewed-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit a076d41823 ("ASoC: samsung: Drop AC97 drivers") removed some
unused code and the associated Kconfig options, but left those options
referenced in the Makefile. Remove the leftover references in the
Makefile.
Signed-off-by: Valentin Rothberg <valentinrothberg@gmail.com>
Reviewed-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver fine-tune some parameters to improve FLL performance.
Those items have description as follow.
(1)ICTRL_LATCH: FLL DSP speed capability control
When FLL running at high frequency with long decimal number, DSP needs
to operate at high speed. FLL DSP can optimize between performance and
power consumption by ICTRL_LATCH.(111 has highest power consumption.)
The default setting can be used to reduce power.
(2)CUTOFF500: loop filter cutoff frequency at 500Khz
It will give the best FLL performance but highest power consumption
to enable the cutoff frequency. FLL Loop Filter enable to reduce FLL
output noise, especially,(DCO frequency)/(FLL input reference frequency)
is not a integer.
(3)GAIN_ERR: FLL gain error correction threshold setting
The threshold is comparison between DCO and target frequency.
The value 1111 has the most sensitive threshold, that is, 1111 can have
the most accurate DCO to target frequency. However, the gain error setting
conditionally and inversely depends on FLL input reference clock rate.
Higher FLL reference input frequency can only set lower gain error, such
as 0000 for input reference from MCLK=12.288Mhz. On the other side, if FLL
reference input is from Frame Sync, 48KHz, higher error gain can apply
such as 1111.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Over Sampling Rate (OSR) is defined as CLK_ADC frequency divided by the
audio sample rate (Fs).
OSR = CLK_ADC / FS
The available OSRs are 32, 64, 128 or 256. Note that the OSR and Fs
values must be selected such that the maximum frequency of CLK_ADC
is less than 6.144 MHz. It is recommended to match the relationship
between OSR and clock SRC according to following Table.
ADC_RATE: 00(OSR=32) | CLK_ADC_SRC: 11(CODEC 1/8)
ADC_RATE: 01(OSR=64) | CLK_ADC_SRC: 10(CODEC1/4)
ADC_RATE: 10(OSR=128) | CLK_ADC_SRC: 01(CODEC 1/2)
ADC_RATE: 11(OSR=256) | CLK_ADC_SRC: 00(CODEC CLK)
The over sampling rate about DAC follows the same rule with ADCs.
The driver changes the OSR to 64 value when initiation for better FLL
performance and applies the dynamic SRC change by different OSR.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the short Frame Sync detection logic enabled, the logic will check the
short frame sync threshold. If frame sync is less than the setting;
for example, frame sync less than 252 MCLK, the short frame sync signal is
flagged, digital filter temporary mute and skip that data.
If the system was intended for sampling rate change which could create
temporary short frame sync and not enough MIPS to run the digital filter.
But the situation doesn't happen in ALSA architecure. Thus the Frame Sync
is always stable, then no require to do the detection. Therefore,
the dirver disables the function for better performance.
Signed-off-by: John Hsu <KCHSU0@nuvoton.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is assuming that the card related information is located on
"card" node, but graph case doesn't have it.
This patch adds node parameter to adjust for graph support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is assuming that the card related information is located on
"card" node, but graph case doesn't have it.
This patch adds node parameter to adjust for graph support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is assuming that the card related information is located on
"card" node, but graph case doesn't have it.
This patch adds node parameter to adjust for graph support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_get_dai_name() is used from snd_soc_of_get_dai_name(),
and it is assuming that DT is using "sound-dai" / "#sound-dai-cells".
But graph base DT is using "remote-endpoint". This patch makes
snd_soc_get_dai_name() non static for graph support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is assuming that the card related information is located on
"card" node, but graph case doesn't have it.
This patch adds node parameter to adjust for graph support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The compatible table was not updated when the support for DAC3100 was added.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The call to arizona_init_notifiers was only added for CODECs that are
generating voice trigger events, however, this is somewhat annoying
for machine drivers that might be used with multiple CODECs as they
need to conditionally register for the notifier, depending on the
CODEC being attached.
As the cost of initialising the notifier is so minimal, and we may
well add other events in the future that apply to more CODECs, simply
do this for all Arizona CODECs.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
These functions are very thin wrappers around core functions, so they
make sense as inline functions. Also making them inline avoids build
issues in the case where the machine driver is built in but the CODEC
is built as a module.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The core expects "const char * const" and "unsigned int" for enum
controls, the mixer control definitions in wm2200 use "const char *"
and "int". This patch corrects the type of these arrays.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/cs42l42.c:1972:3-8: 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
CC: James Schulman <james.schulman@cirrus.com>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
While going to suspend, if we have any pending D0i3 work scheduled,
flush that and force the DSP to goto D0i3 mode before going to suspend.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We were invoking pci_disable_device() while going to suspend-to-idle and
pci_enable_device() while coming back to active state.
Turns out that we do not need these calls as we only need system to be
wake capable when in suspend-to-idle state. The wake capability is
already done by enable_irq_wake() calls, so remove these unwanted calls
in driver.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Check if the name strings are properly terminated, and only use valid
name strings to find existing physical DAI links to configure.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
pm8916_wcd_analog_enable_micbias_int1() should set micbias1_cap_mode
rather than micbias2_cap_mode.
Also change the order of pm8916_wcd_analog_enable_micbias_int1/init2
functions for better readability.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix build errors in sound/soc/fsl/fsl_spdif.c by selecting BITREVERSE.
Fixes these build errors:
sound/built-in.o: In function `spdif_write_channel_status':
fsl_spdif.c:(.text+0xbe39d): undefined reference to `byte_rev_table'
fsl_spdif.c:(.text+0xbe3a8): undefined reference to `byte_rev_table'
fsl_spdif.c:(.text+0xbe3be): undefined reference to `byte_rev_table'
fsl_spdif.c:(.text+0xbe3d8): undefined reference to `byte_rev_table'
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The later Allwinner SoCs have a dedicated reset controller, and
peripherals have dedicated reset controls which need to be deasserted
before the associated peripheral can be used.
Add support for this to the quirks structure and probe/remove functions.
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>
The A31's internal codec capture path has a mixer in front of the ADC
for each channel, capable of selecting various inputs, including
microphones, line in, phone in, and the main output mixer.
This patch adds the various controls, widgets and routes needed for
audio capture from the already supported inputs on the A31.
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>
Support was added to allow location of both CPU and CODEC components
of a DAI link from their parent's of_node if they did not have an
of_node themselves in this commit:
commit 3e0aa8d83b ("ASoC: core: If component doesn't have of_node
use parent's node instead")
However this leaves platforms as something of a special case as the
major DAI component that doesn't do this. Since this is useful for MFD
devices which often utilise a single device tree entry for the whole
device, add support for looking up platforms from the parent's of_node
as well.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Using simple-card with the wm8978 doesn't work because simple card calls
set_sysclk on the clock index 0, which is not the MCLK in the WM8978.
Adjust the clock definition so that the clock 0 is the MCLK.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In our i2s driver, we were previously trying to guess which oversample the
user wanted to use by looking at the rate and trying to max it.
However, the cards, and especially simple-card with its mclk-fs property
will already provide the expected oversample ratio by using the set_sysclk
callback.
We can thus implement it and remove the logic to deal with the runtime
guess.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for Cirrus Logic CS42L42 codec. SoundWire support
is not enabled. Features support for I2C control and I2S audio.
Signed-off-by: James Schulman <james.schulman@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A bugfix accidentally removed the implicit initialization of the
dma channel number, causing undefined behavior when
v->alloc_dma_channel is NULL:
sound/soc/qcom/lpass-platform.c: In function ‘lpass_platform_pcmops_open’:
sound/soc/qcom/lpass-platform.c:83:29: error: ‘dma_ch’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
This adds back an explicit initialization to zero, restoring the
previous behavior for that case.
Fixes: 022d00ee0b ("ASoC: lpass-platform: Fix broken pcm data usage")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Kenneth Westfield <kwestfie@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31 SoC's codec has various inputs, outputs and microphone bias
supplies. These can be routed on the board in different ways, such as:
- HPCOM may be connected to have the headphone DC coupled.
- Microphones all use the MBIAS main microphone supply or one mic may
use the HBIAS supply, which supports headset detection and buttons.
- Line Out may be routed to an audio jack, or an onboard speaker amp
with power controls.
Add support for specifying the audio routes in the device tree.
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>
The A31 internal codec has 3 microphone outputs, of which MIC2 and MIC3
are muxed internally. The resulting two microphone inputs have separate
gain controls and mixer inputs.
The codec also has 2 microphone bias pins. HBIAS is specifically for the
headphone jack, which also supports headphone detection and control
buttons. These extra functions are not supported yet. The other, MBIAS,
is for all other analog microphones.
There is also mention of digital microphone support, but documentation
is scarce, and no hardware with it is available.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31 integrated codec has a second "Line Out" output which does not
include an integrated amplifier in its path. This path does have a
separate volume control.
This patch adds support for the playback path from the DAC to the Line
Out pins.
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>
Since the regmap conversion ac97_read/ac97_write are just simple wrappers
around snd_soc_read/snd_soc_write. Use those instead directly and remove
the wrappers.
Also use snd_soc_update_bits() where appropriate.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the stac9766 driver still uses custom snd_soc_codec_driver IO
callbacks. This has been deprecated for a while, so convert the stac9766
driver to use regmap for its IO.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The stac9766 driver has a header file that defines 3 register locations.
Move these to the main source file since it is not really worth it having a
separate file for them. The header file is now empty and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The AC'97 standard defines paging support for the register range 0x60-0x6f.
Meaning registers in this window are mapped to different physical registers
depending on the setting of the page select register (0x24).
The stac9766 implements support for switching between page 0 and page 1
depending on the addressed register. But the driver never accesses any
registers from page 1, in addition page 0 is the page selected by default.
Considering the development history it is unlikely that the driver will see
any new features that require paging support. Removing the paging support
makes transitioning the driver to regmap a bit more straight forward.
The default register value table is update to contain the values from page
0, rather than page 1.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAI ID defines are back from the time when DAIs were referenced by a
numerical ID. These days a string is used instead and the defines are
unused.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a new flag bit SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP to link flags.
If a link is used for voice wake up, users can set this flag bit and
topology will set the link's 'ignore_suspend' to true.
This ABI update is backward compatible.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Code refactoring. These functions and variables are for configuring
physical DAIs, not only backend DAIs since users may not need DPCM.
So remove 'be' from the function names, and rename variables 'be'
to 'd' or 'dai'.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Rename the ABI struct and type because they are for configuring physical
DAIs, not only backend DAIs since users may not need DPCM:
- Rename struct snd_soc_tplg_be_dai to snd_soc_tplg_dai.
- Rename type SND_SOC_TPLG_TYPE_BE_DAI to SND_SOC_TPLG_TYPE_DAI.
This code refactoring is backward compatible because:
- Both layout of the struct and type value has no change. Kernel can
find the same type value and map to same data layout.
- This struct is not in ABI v4 at all. Now the user space uses ABI v4.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Just code refactoring. The function soc_tplg_link_create() will
create a front end link, not a physical link. So rename it to
soc_tplg_fe_link_create().
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology will find an existing physical link (including BE link for
DPCM) by checking its ID, name and stream name, and configure its physical
audio format and flags.
This support is backward compatible for old ABI v4.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Define the API to find an existing DAI link of the soc card by matching
the ID, name and stream name.
Some cards may use unique ID for each DAI link, so matching ID is enough,
and name or stream name are not necessary. But user need to specify name
or stream name as well if not sure whether link ID is unique since most
cards use 0 as the default link ID.
Topology can use this API to find an existing BE link and configure it.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch renames msm8916_wcd_digital_parse_dt() to
msm8916_wcd_digital_get_clks() as the function is not directly dealing
with dt parsing.
No functional changes done.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the code which is not parsing dt from pm8916_wcd_analog_parse_dt()
to make it clear to reader.
No functional changes done.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The A31 integrated codec has a stereo "Line In" input. Add support for
it to the playback paths.
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>
The A31 has a similar codec to the A10/A20. The PCM parts are very
similar, with different register offsets. The analog paths are very
different. There are more inputs and outputs. The ADC mux has been
replaced with a proper mixer.
This patch adds support for the basic playback path of the A31 codec,
from the DAC to the headphones. Headphone detection, microphone,
signaling, other inputs/outputs and capture will be added later.
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>
According to the DMA engine API documentation, maxburst denotes the
largest possible size of a single transfer, so as not to overflow
destination FIFOs as explained in this excerpt from dmaengine.h
* @src_maxburst: the maximum number of words (note: words, as in
* units of the src_addr_width member, not bytes) that can be sent
* in one burst to the device. Typically something like half the
* FIFO depth on I/O peripherals so you don't overflow it. This
* may or may not be applicable on memory sources.
* @dst_maxburst: same as src_maxburst but for destination target
* mutatis mutandis.
The TX FIFO is 64 samples deep for stereo, and the RX FIFO is 16
samples deep. So maxburst could be 32 and 8 for TX and RX respectively.
Unfortunately the sunxi DMA controller driver takes maxburst as
the requested burst size, rather than a limit, and returns an error
for unsupported values. The original value was 4, but some later
SoCs do not officially support this burst size.
This patch increases maxburst on the TX side to 8, which is supported
by all variants of the sunxi DMA controller.
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>
The A31 has a similar codec to the A10/A20. The PCM parts are very
similar, with just different register offsets. The analog paths are
very different. There are more inputs and outputs.
The A31s, A23, and H3 have a similar PCM interface, again with register
offsets slightly rearranged. The analog path controls, while very
similar between them and the A31, have been moved a separate bus which
is accessed through a message box like interface in the PRCM address
range. This would be handled by a separate auxiliary device tied in
through the device tree in its supporting create_card function.
The quirks structure is expanded to include different register offsets
and separate callbacks for creating the ASoC card. The regmap_config,
quirks, and of_device_match tables have been moved to facilitate this.
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>
This revises existing comments in the register definition macros
section, and adds a few more, so that readers can clearly identify
the types of control registers.
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>
The audio codec on later Allwinner SoCs have a different layout and
audio path compared to the A10/A20. However the PCM parts are still
the same.
The different layout and audio paths mean we need a different
create_card function for different families, so they can create
DAPM endpoint widgets and routes.
This patch moves the regmap configs, quirks and of_device_id
structures to just before the probe function, so we can, among other
things, include a pointer for the create_card function. None of the
lines of code were changed.
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>
For device opened/closed, we check the D0i3 capability for the device
and invoke skl_tplg_d0i3_get/put, which counts the use case based on the
mode supported.
These counters are then used to decide if the device can enter D0i3 mode
of streaming or non-streaming or no D0i3.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Not all use cases can support Doi3. Only certain use cases like hot word
detection, deep buffering can support D0i3 based on resource requirement.
So, pass the D0i3 capability for the FE/BE copier using topology. This will
be used to take a decision for D0i3 mode entry/exit.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For D0i3, we need to tell DSP to run the pipelines in LP mode. This
information is kept in topology and passed to driver as an attribute
for pipe.
So add a new tuple for lpmode and program the pipe based on value set.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver needs two DSP callback, one to set D0i0 (active) and D0i3
(low-power) states.
Add these callbacks in dsp ops and implement them for broxton platforms.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To set the controller in D0i3 mode, the driver needs to set D0i3C
register after DSP is quiesced. Since the D0iX entry/exit is done by IPC,
add this as callback so that it can be invoked from IPC module.
Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada@intel.com>
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio DSP supports intermediate power states between D0 and D3
states. These states are D0i0 and D0i3 states.
Collectively we refer these two states as D0iX states.
To set or wake up from these states, driver also needs to send an IPC "Set D0iX
IPC" before doing anything else.
Add support for this new IPC messages.
Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada@intel.com>
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If the DSP is in low power mode, it needs to be woken up by a "wake" IPC
to set it into the D0 state before we can send any other IPC command.
The call flow is that the driver calls sst_ipc_tx_message_wait() to send any
IPC and this call checks if the device is in low power mode and in that
case we need to send the wake IPC.
So add a new IPC nopm variant which can be called from driver and
doesn't check for power state (as we already know that) and avoids
circular dependency of again checking power state.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The i2s driver was only implementing playback for now. Implement capture to
make sure that's not a limitation anymore.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the remaining update to PCM ABI object of version 5.
The flags will be applied to FE (Front End) links and can also be used
by physical links. The private data is reserved for future extension, so
offset update will add the private data size.
Now user space is using ABI v4, and the previous patch "ASoC: topology:
make PCM backward compatible from ABI v4" can assure the backward
compatibility.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Users start to use topology ABI from v4. ABI v5 updated existing manifest
and PCM elements. Two previous patches can support these ABI updates in a
backward compatible way. So if the topology file from user space is
generated by ABI v4, kernel will no longer quit but continue parsing.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
User space may not always set a valid FE DAI driver's name, FE DAI link's
name, stream name or cpu DAI name. In such cases, there are all ZERO in
these name string buffers of a topology PCM object. This patch will only
duplicate valid name strings for kernel FE DAI driver and DAI link when
creating them from topology, and free the name strings when destroying
them.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for old version 4 of PCMs (FE DAI & DAI links).
Topology ABI v5 added new fields to stream caps and thus changed PCMs.
Since user space may still uses v4, kernel will check the ABI version by
comparing the object size. If user space uses v4 of PCMs, kernel will
create the latest version of PCMs from the old version, and use the new
version internally to create FE DAI & DAI links. Because these new created
PCM elements will be freed later, kernel need duplicate the name strings
of DAI driver and DAI links when creating them.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
add depends on HAS_DMA to Kconfig. This fixes error reported
by kbuild test robot when building for ARCH=m32r:
ERROR: "bad_dma_ops" [sound/soc/bcm/snd-soc-cygnus.ko] undefined!
Signed-off-by: Scott Branden <scott.branden@broadcom.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
While DAPM is mono or single channel, its controls can be shared between
widgets, such as sharing one stereo mixer control between the left and
right channel widgets. An example such as the following routes
[Line In Left]----------<Line In Playback Switch>-------[Left Mixer]
^
^ ^ | ^
(inputs) (paths) <shared stereo mixer control> (outputs)
v v | v
v
[Line In Right]---------<Line In Playback Switch>-------[Right Mixer]
where we have separate widgets and paths for the left and right channels
from "Line In" to "Mixer", but a shared stereo mixer control for the
2 paths.
This patch introduces support for such shared mixer controls, allowing
more than 1 path to be attached to a single stereo control, and being
able to control left/right channels independently.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
