Some of the default value on rt298_index_def are incorrect. Change
them to the correct value.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Gen1 SRU support was created for preparation of Gen2 SRC support,
but no-one is using this feature (sampling rate convert) on Gen1.
BockW had used SRU before, but it was pass through mode.
This means it is same as SSI. And BockW "platform base" code was
removed from upstream code. It is now supported via DT, but it doesn't
use SRU. More detail, r8a7778.dtsi has "rcar_sound,src" entry, but
no-one is using this feature today. SRU probing has no relation to this
removing. This means there is no effect for DT compatibility, no issues
on upstream kernel.
Gen2 SRC was created from Gen1 SRU, these are similar but not same IP.
Keeping Gen1 SRU in current driver is a little bit difficult,
and no-one is using it today. Gen1 sound is still supported via SSI.
Gen1 SRU support will be removed in the next kernel version.
This patch announces it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is not necessary to set registers volatile. So, return false
for default case of rt298_volatile_register.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This ensures that the link is not requesting any clock and the
PLL can turn off. The link is powered when controller is brought
out of reset.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On runtime pm resume, we need to download the firmware, also on
suspend we need to ensure all the interrupts from controller and
DSP are disabled.
Also since we download the firmware on resume, we don't need to do
so on init, so remove that bit
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Like we have in legacy mode HDA driver, we need to check the
status bit and handle interrupt only when it is not zero or all
bits set. We typically see the status as all 1's when controller
resumes from suspend, So add the check here as well and don't
handle for these cases.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skylake driver will set the SPA bit to 0 to turn off the DSP core.
Driver will poll the Current Power Active (CPA) bit to match the
Set Power Active (SPA) bit value. When CPA bit matches the value
of SPA bit, the achieved power state has reached.
In case of DSP power down, register that was polled is SPA
instead of CPA. This patch corrects the register to be polled
in case of DSP power down.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We were getting build warning about "Section mismatch".
dmi_platform_intel_broadwell is being referenced from the probe function
rt5645_i2c_probe(), but dmi_platform_intel_broadwell was marked with
__initdata.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Reviewed-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a driver for the SPDIF transceiver available on RK3066, RK3188 and
RK3288. Heavily based on the rockchip i2s driver.
Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
dev_info is too noisy for tplg wiget loading, so move it to
debug level
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Load and Initialize Non HDA Link Table in Skylake driver
to get platform configuration.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Initialize and creates DSP controls if processing pipe capability
is supported by HW. Updates the dma_id, hw_params to module param
to be used when DSP module has to be configured.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SKL driver does not code DSP topology in driver. It uses the
newly added ASoC topology core to parse the topology information
(controls, widgets and map) from topology binary.
Each topology element passed private data which contains
information that driver used to identify the module instance
within firmware and send IPCs for that module to DSP firmware
along with parameters.
This patch adds init routine to invoke topology load and callback
for topology creation.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For FE and BE, the PCM parameters come from FE and BE hw_params
values passed. For a FE we convert the FE params to DSP expected
module format and pass to DSP. For a BE we need to find the
gateway settings (i2s/PDM) to be applied. These are queried from
NHLT table and applied.
Further for BE based on direction the settings are applied as
either source or destination parameters.
These helpers here allow the format to be calculated and queried
as per firmware format.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Skylake driver topology model tries to model the firmware
rule for pipeline and module creation.
The creation rule is:
- Create Pipe
- Add modules to Pipe
- Connect the modules (bind)
- Start the pipes
Similarly destroy rule is:
- Stop the pipe
- Disconnect it (unbind)
- Delete the pipe
In driver we use Mixer, as there will always be ONE mixer in a
pipeline to model a pipe. The modules in pipe are modelled as PGA
widgets. The DAPM sequencing rules (mixer and then PGA) are used
to create the sequence DSP expects as depicted above, and then
widget handlers for PMU and PMD events help in that.
This patch adds widget event handlers for PRE/POST PMU and
PRE/POST PMD event for mixer and pga modules. These event
handlers invoke pipeline creation, destroy, module creation,
module bind, unbind and pipeline bind unbind
Event handler sequencing is implement to target the DSP FW
sequence expectations to enable path from source to sink pipe for
Playback/Capture.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To configure a module, driver needs to send input and output PCM
params for a module in DSP. The FE PCM params come from hw_params
ie from user, for a BE they also come from hw_params but from
BE-link fixups.
So based on PCM params required driver has to find a converter
module (src/updown/format) and then do the conversion and
calculate PCM params in these pipelines In this patch we add the
helper modules which allow driver to do these calculations.
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SKL driver needs to instantiate pipelines and modules in the DSP.
The topology in the DSP is modelled as DAPM graph with a PGA
representing a module instance and mixer representing a pipeline
for a group of modules along with the mixer itself.
Here we start adding building block for handling these. We add
resource checks (memory/compute) for pipelines, find the modules
in a pipeline, init modules in a pipe and lastly bind/unbind
modules in a pipe These will be used by pipe event handlers in
subsequent patches
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver now can make use of mclk data, if provided, to set, enable
and disable the clock source. As part of this, the choice to
enable clock squaring is dealt with as part of dai_sysclk() call
rather than as platform data.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for DT bindings in the codec driver.
As part of this support, the mclk data can now be provided and
used to control the mclk during codec operation.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 3d7608e4c1 ("ARM: shmobile: bockw: remove legacy
board file and config"), Renesas R-Car SoCs are only supported in
generic DT-only ARM multi-platform builds. The driver doesn't need to
use platform data anymore, hence remove platform data configuration.
Move <sound/rcar_snd.h> to sound/soc/sh/rcar/, as it's no longer needed
by platform code.
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
Should only try to enable/disable the provided mclk, during bias
level changes, if it's not NULL. Also return value of
clk_prepare_enable() should be checked and dealt with accordingly.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use of_match_ptr() to handle non-DT kernel scenario where match
table should be NULL.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Most of the boards have their headphone jack directly connected to the
matching pins of the SoCs. Since most of the time we will have the same
routing path, it makes no sense to put that in the DTS, since it will only
be some useless duplication there.
It also fixes the following warning messages that were seen so far, on
boards where we were using the bindings in the documentation example.
sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Left -> direct -> Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Right -> direct -> Headphone Jack
Reported-by: Priit Laes <plaes@plaes.org>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/nau8825.c:1096:3-8: 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: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of hardconding a platform data for dw_dmac let's use it's own
autoconfiguration feature. Thus, remove hardcoded values.
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make a copy of memory for index_cache rather than directly use the
rt286_index_def to avoid run time error.
Fixes: c418a84a8c ("ASoC: Constify reg_default tables")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AD1939 is missed from the table, so add it.
AD1936 and AD1937 are controlled by I2C interface, so remove them.
Fixes: e5224f58e3 ("ASoC: ad193x: add support to ad1934")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The minimum volume level for the TAS2552 (control register value 0x00)
is -7dB however the driver declares it as -0.07dB.
Running amixer before the patch reports:
dBscale-min=-0.07dB,step=1.00dB,mute=0
Running amixer with the patch applied reports:
dBscale-min=-7.00dB,step=1.00dB,mute=0
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Change the rockchip i2s object name (and thus module name) from the
rather generic snd-soc-i2s to the more specific snd-soc-rockchip-i2s
Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
The AD1934 codec has no ADC feature. Hence it register mapping is slightly
different from the register mapping of other members of the AD193x family.
Some ASoC controls and widgets are related to the DAC feature so are not
relevant in the case of an AD1934 codec.
Signed-off-by: Cyrille Pitchen <cyrille.pitchen@atmel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, ESAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power
source of modules during system suspend and resume procedure. Thus,
SSI needs to save all the values of registers before the system
suspend and restore them after the system resume.
The register SFCSR is volatile, but some bits in it need to be
recovered after suspend/resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SPDIF needs to save
all the values of registers before the system suspend and restore them after
the system resume.
The SRPC register should be volatile, LOCK bit is set by the hardware.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add snd_soc_pm_ops in machine driver to make the trigger suspend/resume
be called in suspend/resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
dw i2s controller can work in slave mode, codec being master.
dw i2s is made to support master/slave operation, by reading dwc
register.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently if bulk_enable() of supplies fails, the code still goes on
to try and put the device into active state, and set expected IO
voltage of the device. This doesn't really make sense so code now
returns on bulk_enable() failure, with an error message. Also,
to help with debug, failure to get supplies also provides an error
message.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch adds the codec reset setting in the shutdown function to prevent
the weird sound of the headphone happened by rebooting.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Its a bit odd that debugfs_create_bool() takes 'u32 *' as an argument,
when all it needs is a boolean pointer.
It would be better to update this API to make it accept 'bool *'
instead, as that will make it more consistent and often more convenient.
Over that bool takes just a byte.
That required updates to all user sites as well, in the same commit
updating the API. regmap core was also using
debugfs_{read|write}_file_bool(), directly and variable types were
updated for that to be bool as well.
Signed-off-by: Viresh Kumar <viresh.kumar@linaro.org>
Acked-by: Mark Brown <broonie@kernel.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The codec supports 4 channel recording with TDM on AIF1.
This patch modifies the DAI capability to allow it.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds support for the DA7219 audio codec with built-in advanced
accessory detect features.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Designware I2S uses tx empty and rx available signals as the DMA
handshaking signals. during music playing, if XRUN occurs,
i2s_stop() function will be executed and both tx and rx irq are
masked, when music continues to be played, i2s_start() is executed
but both tx and rx irq are not unmasked which cause I2S stop
sending DMA handshaking signal to DMA controller, and it finally
causes music playing will be stopped once XRUN occurs for the first
time.
[On list discussion suggests this may be partly a race condition on slow
systems -- broonie]
Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The struct snd_soc_tplg_pcm_dai is renamed to snd_soc_tplg_pcm.
This struct will now be used to handle data related to PCMs
(FE DAI & DAI links). It's not for BE, because BE DAI mappings will be
provided by ACPI/FDT data.
Remove the unused struct snd_soc_tplg_pcm_cfg_caps. We are using
snd_soc_tplg_stream and snd_soc_stream_caps instead.
Bump ABI version to 4.
Signed-off-by: Vedang Patel <vedang.patel@intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds convenience defines for declaring a gain control that
has an input mux. These blocks are functionally equivalent to
the existing mixer blocks but can only have a single input
active at once.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
from Designware I2S datasheet, tx/rx XRUN irq is cleared by
reading register TOR/ROR, rather than by writing into them.
Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The current code writes a set of registers that are reserved on the
tlc320aic3104. The change skips those registers for that IC.
Signed-off-by: Rick Mann <rmann@latencyzero.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes the old PXA DMA API usage and switches over to
generic functions provided by snd-soc-dmaengine-pcm.
More cleanups may be done on top of this, and some function stubs can
now be removed completetly. However, the intention here was to keep
the transition as small as possible.
This was tested on the mioa701 pxa27x board.
Signed-off-by: Daniel Mack <zonque@gmail.com>
[trivial change from mmp-dma to pxa-dma]
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
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Merge tag 'asoc-fix-v4.3-rc2' into asoc-pxa
ASoC: Fixes for v4.3
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
The SND_PCM_RATE_KNOT covers all the rate settings, even though some that
we don't support, while we also list all the rate we support. Simply remove
it.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current code, disregarding the clk_set_rate error code, was always
returning -EINVAL. Fix that and return the code in order to have more clue
about what's going on.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
pm_runtime_enable is called in probe to enable runtime PM
for wm8962 codec, but pm_runtime_disable isn't called in remove
callback, nor is called in error path if probe fails after runtime
PM is enabled, this causes unbalanced pm_runtime_enable.
This patch Adds pm_runtime_disable in remove callback and error path,
to balance pm_runtime_enable.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_DAIFMT_{IB|NB}_{IF|NF} are defined as inverting or not BCLK or
FRM relatively to what is standard for the specified DAI hardware audio
format. Consequently, the absolute polarities of these signals cannot be
derived only from these settings as this driver did. The format has to
be taken into account too.
This fixes inverted left/right channels in I²S mode.
Signed-off-by: Benoît Thébaudeau <benoit@wsystem.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit b4508d0f95 ("ASoC: db1200: Use static DAI format setup") switched
the db1200 driver over to using static DAI format setup instead of a
callback function. But the commit only added the dai_fmt field to one of
the three DAI links in the driver. This breaks audio on db1300 and db1550.
Add the two missing dai_fmt settings to fix the issue.
Fixes: b4508d0f95 ("ASoC: db1200: Use static DAI format setup")
Reported-by: Manuel Lauss <manuel.lauss@gmail.com>
Tested-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Also move the include of sound/hda_verbs.h to rl6347a.h because it is used
in rl6347a.h.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM receive and transmit DMA requestor lines were reverted, breaking the
PCM playback interface for PXA platforms using the sound/soc/ variant
instead of the sound/arm variant.
The commit below shows the inversion in the requestor lines.
Fixes: d65a14587a ("ASoC: pxa: use snd_dmaengine_dai_dma_data")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Broadwell can not triger the IRQ falling and rising simultaneously,
so it can not detect jack-in and jack-out simultaneously.
We add a flag "jd_invert" to platform data. If this flag is set,
codec IRQ will be set to invert that forces IRQ as pulse when jack-in
and jack-out.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add DMI data for Buddy project.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current code incorrectly treats dai format for AC97 as bit mask
whereas it's actually an integer value. This causes DAI formats
other than AC97 (e.g. DSP_B) to trigger AC97 related code,
which is incorrect and breaks functionality. This patch fixes
the code to correctly compare values to determine AC97 or not.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the capability to use multiple codecs on the same DAI linke where
one codec is used for playback and another one is used for capture.
Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS
microphone for capture.
No verification is made that the playback and capture codec setups are
compatible in terms of number of TDM slots (where applicable).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The hdmi stub codec has not been used since refactoring of OMAP HDMI
audio support.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to create CPU DAI for each endpoint instance. For this we
should have one DMIC DAI, one HDA DAI and SSP DAI. Thus, DMIC23,
HDA-SPK/AMIC was not required so this patch removes them
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the error path so that we can free the allocated memory on the error
path instead of releasing them individually on each error.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have requested for the firmware but we have missed releasing it both
on success and on error path.
While checking the code it turned out that the requested firmware is not
even used. More over the same firmware is being loaded by
wm0010_stage2_load().
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the max register value of mic boost pga should be 3.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm8962 can't support 64k sample rate. When playing a 64KHz wave file,
'Unsupported rate 64000Hz' will be prompted.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make SND_SOC_ROCKCHIP_MAX98090 and SND_SOC_ROCKCHIP_RT5645 depend on
CLKDEV_LOOKUP to fix below build warning:
warning: (SND_SOC_ROCKCHIP_MAX98090 && SND_SOC_ROCKCHIP_RT5645) selects
SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && !M68K &&
!UML && SND && SND_SOC && CLKDEV_LOOKUP && SND_SOC_ROCKCHIP)
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The '\n' at the end of the format string is not needed. It adds an extra
line break when doing
cat /proc/interrupts
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/sunxi/sun4i-codec.c:708:3-8: 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: Emilio López <emilio@elopez.com.ar>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The slot_width is for essentially same thing. Instead of storing
bclk_lrclk_ratio, just store the slot_width. Comments has been updated
accordingly and some variable names changed to more descriptive.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Apply PTR_ERR to the value that was recently assigned.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x,y;
@@
if (IS_ERR(x) || ...) {
... when any
when != IS_ERR(...)
(
PTR_ERR(x)
|
* PTR_ERR(y)
)
... when any
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Store return value of of_get_property() to a pointer of __be32 type as
device tree has big endian type. This fixes a sparse warning couple of
lines later when be32_to_cpup() is used to convert from big endian to
cpu endian.
The whole conversion is not really necessary, as we are only checking
if the value is zero or not, but I wanted to add it to remind in the
future that the data has to be converted before use. Compiler should
optimize the unnecessary operations away.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Earlier revisions of the wm5110/8280 silicon require a slightly more
complex procedure to enable analogue inputs. This patch adds this into
the driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We will occasionally require to take different action based on if an
input is analog or digital so add a helper function to return if an
input is analog.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the addition of the WILL_PMU / WILL_PMD several of the switches in
arizona.c no longer cover all cases or have a default case. Whilst this
isn't causing any problems in the interests of robustness add default
cases.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous fix of pxa library support, which was introduced to fix the
library dependency, broke the previous SoC behavior, where a machine
code binding pxa2xx-ac97 with a coded relied on :
- sound/soc/pxa/pxa2xx-ac97.c
- sound/soc/codecs/XXX.c
For example, the mioa701_wm9713.c machine code is currently broken. The
"select ARM" statement wrongly selects the soc/arm/pxa2xx-ac97 for
compilation, as per an unfortunate fate SND_PXA2XX_AC97 is both declared
in sound/arm/Kconfig and sound/soc/pxa/Kconfig.
Fix this by ensuring that SND_PXA2XX_SOC correctly triggers the correct
pxa2xx-ac97 compilation.
Fixes: 846172dfe3 ("ASoC: fix SND_PXA2XX_LIB Kconfig warning")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") added AUDIO_CLKOUT
support for ADG. But single AUDIO_CLKOUT needs clkout_name[CLKOUT],
not clkout_name[i]. Kernel will have NULL pointer access without this
patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
71a0138ab("ASoC: ak4642: enable to use MCKO as fixed rate output
pin on DT") added new FS() macro, but x86 already has it in
arch/x86/include/uapi/asm/ptrace-abi.h
This patch exchange FS() to FSs() to avoid redefinition warning
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") uses
of_clk_add_provider() which is requesting struct clk_onecell_data.
But it is COMMON_CLK feature. SND_SOC_RCAR depends on COMMON_CLK
This patch also solved compile error of 7486d80f7("ASoC: rsnd: remove
unneeded sh_clk header")
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The sun4i, sun5i and sun7i SoC families have a built-in codec, capable
of both audio capture and playback.
While this is called a codec by Allwinner, it really is an in-SoC
combination of a codec and a DAI, with its own DAC/ADC and amplifiers
in a single memory-mapped controller.
The capture part has been left out for now, and will be added eventually.
Signed-off-by: Emilio López <emilio@elopez.com.ar>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The function get_current_pipe_id() was not being used.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds DT binding for explicitly choosing a tdm mask for DAI and uses it
in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been
changed.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Implements set_tdm_slot() callback for mcasp. Channel constraints are
updated according to the configured tdm mask and slots each time
set_tdm_slot() is called. The special case when slot width is set to
zero is allowed and it means that slot width is the same as the sample
width.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Before this patch the set_tdm_slots() callback did not store the value
of slot width anywhere. The tdm support only worked if selected slot
width was equal to the sample width. With this patch all sample widths
that fit into the slot width are supported. There unused bits are
filled unnecessarily in the capture direction, but the other end of
the i2s bus should be able to ignore them.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_snd_dmaengine_pcm_register() is guarded by
CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound has AUDIO_CLKOUT (in Gen1/Gen2) AUDIO_CLKOUT1/2/3 (in Gen3)
This patch support these patches as clock provider.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
undefined clock is not error. Accept such case. And this is prepare
for clock out support in the same time.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It didn't have "\n", and indicated different data
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ADG clock calculation needs ADG and SSI settings.
Thus, SSI side clock request function depends on ADG settings.
After reconsideration, we can close this method inside ADG.
This function uses new method. And it becomes preparation for
AUDIO_CLKOUT support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ADG is special IP since it doesn't have MSTP. And now, ADG has common
mod base register access. We can now setup ADG initial setting when
probe timing. It is needed if sound card is based on AUIDO_CLK which
is used as Master clock.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound has SSI/SRC/DVC/MIX/ADG modules, and these have original
register mapping. Thus this driver is using regmap field, and each module
is using it based on each module ID.
Sometimes, each module needs other module to controlling. but current each
function is using just "mod" as parameter name. This is confusable.
For example, if SSI0 and SRC2 are connected, and if SRC module function
has bug of module access, and if it needs to control connected SSI,
SRC function will access to SSI2 (It should access to SSI0, but it uses
SRC's ID 2). This is easy to happen in current driver style.
To avoid this kind of confusable trouble, this patch adds module confirm
macro for debug purpose.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound has ADG IP, but it is special device.
(It is clock generater, and it doesn't need MSTP)
Thus, ADG didn't use mod base common method on rsnd driver.
But it can be confusable point. Let's use common method
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound driver has SSI/SRC/DVC/CTU/MIX, and these are controlled
as modules. And these module are member of each modules's private data.
It used own method to get module pointer, but Let's use common method
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound Gen3 is updated version of Gen2. We need to update
driver for it, but basically it should works as Gen2 compatible.
This is initial support for Gen3. Gen3 specific feature will be
incrementally added in the future
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sh_clk header is not needed, and it will create confusion.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_src_pcm_new() is used only from Gen2. make it clear in function name,
and remove unneeded Gen1 check.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_dma_to_xxx() macro should exist in same place
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ak4642 chip can output clock via MCKO pin as audio reference clock.
This patch supports it on DT
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The TX and RX direction share the same bit clock and frame sync, so
the samplerate must be the same to both directions.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
These platform drivers have a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix below build warning:
sound/soc/au1x/psc-i2s.c: In function 'au1xpsc_i2s_drvprobe':
sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret' [-Wunused-variable]
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_snd_dmaengine_pcm_register() is guarded by
CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In capture, there is chance that hw_ptr reported at IRQ is
a little smaller than period_size due to internal AFE buffer.
In the case of ping-pong buffer:
|xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------|
hw_ptr < period_size
This available buffer will not be read since its size is smaller than
avail_min (which is period_size by default), and read thread continues
to sleep. If the next hw_ptr is just a little larger than buffer_size,
overrun occurs. One more period can hold the possible unread buffer.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The FIFO threshold for McASP should be <=[tx/rx]numevt so the initial value
for the refining should meet this requirement as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The input PGAs have a gain range from -17.25dB to +30dB in 0.75dB steps.
The boost stage can provide additional gain. For line inputs, -12dB to
+6dB gain is available on the boost mixer. For micphone inputs, it can
provide up to +29dB additional gain from the microphone PGA.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Bard Liao