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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 3e0aa8d83bf8 ("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: 022d00ee0b55 ("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>
Currently the ALSA proc handler allows read or write even if the proc
file were write-only or read-only. It's mostly harmless, does thing
but allocating memory and ignores the input/output. But it doesn't
tell user about the invalid use, and it's confusing and inconsistent
in comparison with other proc files.
This patch adds some sanity checks and let the proc handler returning
an -EIO error when the invalid read/write is performed.
Cc: <stable@vger.kernel.org> # v4.2+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ALSA proc handler allows currently the write in the unlimited size
until kmalloc() fails. But basically the write is supposed to be only
for small inputs, mostly for one line inputs, and we don't have to
handle too large sizes at all. Since the kmalloc error results in the
kernel warning, it's better to limit the size beforehand.
This patch adds the limit of 16kB, which must be large enough for the
currently existing code.
Cc: stable@vger.kernel.org # v4.2+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Backmerge tag 'v4.9-rc4' into drm-next
Linux 4.9-rc4
This is needed for nouveau development.
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>
