This patch adds the OMAP4 ABE platform driver. This driver defines and
exports control for the DSP Frontend and Backend routing.
TODO: cleanup
Signed-off-by: Liam Girdwood <lrg@ti.com>
This allows the DSP core to call a beskope trigger() call
on DAIs and platforms that require it.
TODO: move into DSP patch series.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Allow DAI's to be hostless so that no PCM data is sent between DAI
and CPU. This allows for power savings as there is no DMA or CPU
interaction required.
TODO: we shouldn't need to allocate a PAGE for a dummy DMA buffer.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Allow some PCM devices to be hostless, i.e. there is no PCM data transferred
to or from the host CPU. This can be used to minimise power on systems since
the CPU can idle/sleep during the PCM device operation (e.g. a phone call
where the DAI is between a MODEM and DSP)
TODO: cleanup, look at adding a read/write blocker.
Singed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
OMAP4 ABE requires dynamic minors in order to support all its
PCM devices.
TODO: Move this into DSP core patch series.
Signed-off-by: Liam Girdwood <lrg@ti.com>
DSP components can have DAIs that can exist in either the CODEC driver or
platform DAI driver. Simplify DAI refcounting logic by ref counting all
(CODEC and platform) DAIs in each sound card.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Add DSP support to ASoC core.
This adds a NULL CODEC driver allow binding and configuration of
DSP DAIs, configures the PCM for DSP, initilaises the DSP state
variables and managed the DSP DAI link registration.
TODO: come cleanup and refactor some variable names.
TODO: split out with previous patch into smaller patches.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Export inline DAI PCM operations to allow DSP core to individually call BE PCM
operations and ....
Allow machine drivers to specifify DSP FE and BE DAI links.
Add FE and BE state information to PCM rtd.
TODO: split out this into smaller patches.
Signed-off-by: Liam Girdwood <lrg@ti.com>
This adds ASoC core support for internal and external DSPs and represents them
to the audio user as a CODEC like device with mixers, muxes and runtime audio
route changing.
The DSP core allows DSP DAIs to be dynamically re-routed at runtime between the
PCM device end (or Frontend - FE) and the physical DAI (Backend - BE) using
regular kcontrols (just like a hardware CODEC).
The DSP core also deal with DSP FE+BE suspend and resume PM ops.
Signed-off-by: Liam Girdwood <lrg@ti.com>
In preparation for ASoC DSP support.
Allow for the operation of custom mixer and mux DAPM widgets that can call
snd_soc_dapm_mixer_update_power() and snd_soc_dapm_mux_update_power() directly
after updating their status.
This adds two previously static calls to the public DAPM API.
Signed-off-by: Liam Girdwood <lrg@ti.com>
In preparation for ASoC DSP support.
Add a DAPM API call to determine whether a DAPM audio path is valid between
source and sink widgets. This also takes into account all kcontrol mux and mixer
settings in between the source and sink widgets to validate the audio path.
This will be used by the DSP core to determine the runtime DAI mappings
between FE and BE DAIs in order to run PCM operations.
Signed-off-by: Liam Girdwood <lrg@ti.com>
In preparation for ASoC DSP support.
Allow platform drivers to export widgets and kcontrols.
This is required to support DSP dynamic DAI routing and power management where
platform DSP drivers have kcontrols, DAPM widgets etc.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Add widget IO in preparation to support platform driver DAPM widgets with IO.
This provides a generic widget IO operation wrapper in preparation for
decouplement of register IO from CODEC drivers.
Signed-off-by: Liam Girdwood <lrg@ti.com>
In preparation for ASoC DSP support.
The new DSP core allows DSP DAIs to be dynamically re-routed at runtime
between the PCM device end (or Frontend - FE) and the physical DAI
(Backend - BE) using regular kcontrols (just like a hardware CODEC).
The DSP core therefore must be able to call PCM operations for both the
Frontend and Backend(s) DAIs at the same time.
Currently we have a global pcm_mutex that is used to serialise
the ASoC PCM operations. This patch removes the global mutex
and adds a mutex per RTD allowing the PCM operations to be reentrant and
allow control of more than one DAI at at time. e.g. a frontend DSP hw_params()
could configure multiple backend hw_params() with similar or different
hw parameters at the same time.
Also fix the naming of soc_pcm_close.
Signed-off-by: Liam Girdwood <lrg@ti.com>
In preparation for ASoC DSP support.
This adds a callback function to be called at the completion of a DAPM stream
event.
This can be used by DSP components to perform calculations based on DAPM graphs.
Signed-off-by: Liam Girdwood <lrg@ti.com>
In preparation for ASoC DSP support.
Currently pcm_new() passes in card, pcm and DAI. Refactor this to only pass
in rtd since rtd contains card, pcm and DAI. Struct rtd also contain other
members too that are useful to DSPs.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Some ASoC components depend on other ASoC components to provide clocks and
power resources in order to probe().
Provide a method to allow some components to be probed() later and be also
removed() earlier.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Currently CODEC and platform drivers have their module reference count
incremented soc_probe_dai_link() whilst CPU DAI drivers have their reference
count incremented in soc_bind_dai_link().
CPU DAIs should have their reference count incremented in soc_probe_dai_link()
just like the CODEC and platform drivers.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Remove dependency between pll (hppll, lppll) and headset power
mode (low-power, high-performance), as headset power mode can
be used with any pll.
A new control is created to allow headset power mode configuration
from userspace. Changing headset power mode during earpiece related
usecases is not propagated down to the codec as earpiece requires
HS DAC in HP mode.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Convert TWL6040 CODEC driver into a TWL6040 MFD child, it implies
that MFD-level operations like register accesses, clock setting
and power management are done through MFD APIs, not directly by
CODEC driver anymore. To avoid conflicts with the other MFD child,
vibrator registers are skipped in CODEC driver.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allign the platform data names for twl4030 audio submodule:
twl4030_audio_data: for the core MFD driver
twl4030_codec_data: for ASoC codec driver
twl4030_vibra_data: for the input/ForceFeedback driver
To avoid breakage, change all depending drivers, files
to use the new types.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Rename the driver, and header file from twl4030-codec to
twl4030-audio.
To avoid breakage change depending drivers at the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
CC: Misael Lopez Cruz <misael.lopez@ti.com>
The WM8994 and WM8958 series of devices have two MICBIAS supplies rather
than one, the current widget actually manages the microphone detection
control register bit (which is managed separately by the relevant API).
Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2
widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's harmless but annyoing.
sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
