In some cases it is necessary to before additional operations after the
device has been initialized and before the device is registered. This can
for example be resetting the device.
This patch introduces a new function snd_soc_alloc_ac97_codec() which is
similar to snd_soc_new_ac97_codec() except that it does not register the
device. Any users of snd_soc_alloc_ac97_codec() are responsible for calling
device_add() manually.
Fixes: 6794f709b7 ("ASoC: ac97: Drop delayed device registration")
Reported-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Queues are used both for scheduling playback events and for assigning
timestamps to recorded events, so it is easy to need quite a lot of
them, especially on a multi-user system. Additionally, the actual
queue objects are allocated dynamically, so it does not really make
sense to have a low limit. Increase it to something still sane.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are no more users of this field left so it can finally be removed.
New users should use snd_soc_dapm_to_codec(w->dapm);
The reason why it is removed is because it doesn't fit to well anymore in
the componentized ASoC hierarchy, where DAPM works on the snd_soc_component
level. And the alternative of snd_soc_dapm_to_codec(w->dapm) typically
generates the same amount of code, so there is really no reason to keep it.
For automatic conversion the following coccinelle semantic patch can be used:
// <smpl>
@@
struct snd_soc_dapm_widget *w;
@@
-w->codec
+snd_soc_dapm_to_codec(w->dapm)
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Call clk_prepare_enable() and clk_disable_unprepare() for cpu dai
clock and codec dai clock in dai statup and shutdown callbacks. This
to make sure the related clock are enabled when the audio device is
used.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixes the following compilation error:
include/sound/compress_driver.h: In function ‘snd_compr_drain_notify’:
include/sound/compress_driver.h:177:2: error: implicit declaration of function ‘snd_BUG_ON’ [-Werror=implicit-function-declaration]
if (snd_BUG_ON(!stream))
snd_BUG_ON() is defined in sound/core.h but the file is not included explicitly,
so include it.
Signed-off-by: Qais Yousef <qais.yousef@imgtec.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
L/R channel will be switched if under/over flow error happen on
Renesas R-Car sound device by the HW bugs. Then, HW restart is required
for salvage. This patch add salvage support for SRC.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch adds the MICBIAS VDD setting in the platform data. It can be set to
1V8 or 3V3 in the MICBIAS VDD.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For some setups it is necessary to change the DAI link format at runtime.
This patch factors out the code that does the initial static DAI link format
configuration into a separate helper function which can be used board
drivers as well.
This allows board drivers that have to change the DAI link format at runtime
to reuse it instead of having to manually change the format on all DAIs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove function declarations for functions that don't have a matching
implementation.
For snd_pcm_build_linear_format the implementation was removed in
64d27f96cb ("[ALSA] Support 3-bytes 24bit format in PCM OSS
emulation"). All the others never had one (as far as git history goes).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The various PCM and hwdep allocation functions in this driver take a pointer
to a pointer of a PCM/hwdep where if this parameter is provided the newly
allocated object is stored. All callers pass NULL though, so remove the
parameter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most callers of snd_wss_pcm(), snd_wss_timer() and snd_cs4236_pcm() pass
NULL as the last parameter, some callers pass a pointer but never use it
after the function has been called and only a few callers pass a pointer and
actually use it. The later is only the case for snd_wss_pcm() for
snd_cs4236_pcm() and it is possible to get the same PCM object by accessing
the pcm field of the snd_wss struct that was passed as the first parameter.
This function removes the last parameters from the functions mentioned above
and updates the callers which used it to use chip->pcm instead. This allows
us to slightly simplify the functions since they don't have to check and set
the last parameter anymore which makes the code slightly shorter and
cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All callers of snd_sb16dsp_pcm() always pass the pcm field of the first
parameter as the last parameter. Simplify the function by moving this inside
the function itself. This makes the code a bit shorter and cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_sb8dsp_pcm() and snd_sb8dsp_midi() take a pointer to a pointer of a
PCM/MIDI where if this parameter is provided the newly allocated object is
stored. All callers pass NULL though, so remove the parameter. This makes
the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_gf1_pcm_new() and snd_gf1_rawmidi_new() take a pointer to a pointer of a
PCM/MIDI where if this parameter is provided the newly allocated object is
stored. All callers pass NULL though, so remove the parameter. This makes
the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_es1688_pcm() takes a pointer to a pointer of a PCM where if this
parameter is provided the newly allocated PCM is stored. This PCM is also
available from the pcm field of the snd_es1688 struct that got passed to the
same function. This patch updates all callers which passed a pointer to use
that field instead and then removes the parameter from the function. This
makes the code a bit shorter and cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ad1816a_pcm() and snd_ad1816a_timer() take a pointer to a pointer of a
PCM/timer where if this parameter is provided the newly allocated object is
stored. All callers pass NULL though, so remove the parameter. This makes
the code a bit cleaner and shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain.
Some audio devices require notification of drain events
in order to properly drain and shutdown an audio stream.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
__fls has the same semantics as ld2, so there is no need to re-implement it.
Furthermore a lot of architectures have custom implementations of __fls that
are able to use special hardware instructions to compute the result. This
makes the code slightly shorter and faster.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The difference between __ffs and ffs is that ffs will return a one based
index whereas __ffs will return a zero based index. Furthermore ffs will
check if the passed value is zero and return zero in that case, whereas
__ffs behavior is undefined if the passed parameter is 0.
Since we already check if the mask is 0 before calling ffs and also subtract
1 from the result __ffs is the better choice.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hw_params struct has a parameter that contains the period size in bytes.
This can be used instead of deriving the value from other parameters. This
is similar to e.g. params_buffer_bytes()
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use static inline functions instead of macros for the remaining params_*()
helpers that have not been converted yet. This is slightly cleaner and
offers better type safety.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both SNDRV_PCM_IOCTL1_FALSE and SNDRV_PCM_IOCTL1_TRUE are unused and have in
fact never been used (at least as far as the git history goes).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Those two functions are not used anywhere and also their name is a bit to
generic to be in a global header, so remove them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a copy and paste error in the kernel doc description for the params_*()
functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For legacy reasons the ASoC framework assumes that a CODEC INPUT or OUTPUT
widget that is not explicitly connected to a external source or sink is
potentially connected to a source or a sink and hence the framework treats
the widget itself as source (for INPUT) or sink (for OUTPUT). For this
reason a INPUT or OUTPUT widget that is really not connected needs to be
explicitly marked as so.
Setting the card's fully_routed flag will cause the ASoC core, once that all
widgets and routes have been registered, to go through the list of all
widgets and mark all INPUT and OUTPUT that are not externally connected as
non-connected. This essentially negates the default behaviour of treating
INPUT or OUTPUT widgets without external routes as sources or sinks.
This patch takes a different approach while getting the same result. Instead
of first marking INPUT and OUTPUT widgets as sinks/sources and then later
marking them as non-connected, just never mark them as a sink or a source if
the fully_routed flag is set on a card.
This requires a lot less code and also results in a slightly faster card
initialization since there is no need to iterate over all widgets and check
whether the INPUT and OUTPUT widgets are connected or not.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This became a fairly large pull request. In addition to the usual
driver updates / fixes, there have been a high amount of cleanups in
ASoC area, as well as control API helpers and kernel documentations
fixes touching through the whole tree.
In the driver side, the biggest changes are the support for new Intel
SoC found on new x86 machines, and the updates of FireWire dice and
oxfw drivers.
Some remarkable items are below:
* ALSA core
- PCM mmap code cleanup, removal of arch-dependent codes
- PCM xrun injection support
- PCM hwptr tracepoint support
- Refactoring of snd_pcm_action(), simplification of PCM locking
- Robustified sequecner auto-load functionality
- New control API helpers and lots of cleanups along with them
- Lots of kerneldoc fixes and cleanups
* USB-audio
- The mixer resume code was largely rewritten, and the devices with
quirks are resumed properly.
- New hardware support: Focusrite Scarlett, Digidesign Mbox1,
Denon/Marantz DACs, Zoom R16/24
* FireWire
- DICE driver updates with better duplex and sync support, including
MIDI support
- New OXFW driver for Oxford Semiconductor FW970/971 chipset,
including the previous LaCie Speakers device. Fullduplex and MIDI
support included as well as DICE driver.
* HD-audio
- Refactoring the driver-caps quirk handling in snd-hda-intel
- More consistent control names representing the topology better
- Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
fix, ASUS Z99He laptop EAPD
* ASoC
- Conversion of AC'97 drivers to use regmap, bringing us closer to
the removal of the ASoC level I/O code
- Clean up a lot of old drivers that were open coding things that
have subsequently been implemented in the core
- Some DAPM performance improvements
- Removal of the now seldom used CODEC mutex
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers
- Support for Samsung boards using rt5631 as the CODEC
- Removal of the obsolete AFEB9260 machine driver
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks
* Others
- ASIHPI driver update and cleanups
- Lots of dev_*() printk conversions
- Lots of trivial cleanups for the codes spotted by Coccinelle
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Merge tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became a fairly large pull request. In addition to the usual
driver updates / fixes, there have been a high amount of cleanups in
ASoC area, as well as control API helpers and kernel documentations
fixes touching through the whole tree.
In the driver side, the biggest changes are the support for new Intel
SoC found on new x86 machines, and the updates of FireWire dice and
oxfw drivers.
Some remarkable items are below:
ALSA core:
- PCM mmap code cleanup, removal of arch-dependent codes
- PCM xrun injection support
- PCM hwptr tracepoint support
- Refactoring of snd_pcm_action(), simplification of PCM locking
- Robustified sequecner auto-load functionality
- New control API helpers and lots of cleanups along with them
- Lots of kerneldoc fixes and cleanups
USB-audio:
- The mixer resume code was largely rewritten, and the devices with
quirks are resumed properly.
- New hardware support: Focusrite Scarlett, Digidesign Mbox1,
Denon/Marantz DACs, Zoom R16/24
FireWire:
- DICE driver updates with better duplex and sync support, including
MIDI support
- New OXFW driver for Oxford Semiconductor FW970/971 chipset,
including the previous LaCie Speakers device. Fullduplex and MIDI
support included as well as DICE driver.
HD-audio:
- Refactoring the driver-caps quirk handling in snd-hda-intel
- More consistent control names representing the topology better
- Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
fix, ASUS Z99He laptop EAPD
ASoC:
- Conversion of AC'97 drivers to use regmap, bringing us closer to
the removal of the ASoC level I/O code
- Clean up a lot of old drivers that were open coding things that
have subsequently been implemented in the core
- Some DAPM performance improvements
- Removal of the now seldom used CODEC mutex
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers
- Support for Samsung boards using rt5631 as the CODEC
- Removal of the obsolete AFEB9260 machine driver
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks
Others:
- ASIHPI driver update and cleanups
- Lots of dev_*() printk conversions
- Lots of trivial cleanups for the codes spotted by Coccinelle"
* tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (594 commits)
ALSA: pcxhr: NULL dereference on probe failure
ALSA: lola: NULL dereference on probe failure
ALSA: hda - Add "eapd" model string for AD1986A codec
ALSA: hda - Add EAPD fixup for ASUS Z99He laptop
ALSA: oxfw: Add hwdep interface
ALSA: oxfw: Add support for capture/playback MIDI messages
ALSA: oxfw: add support for capturing PCM samples
ALSA: oxfw: Add support AMDTP in-stream
ALSA: oxfw: Add support for Behringer/Mackie devices
ALSA: oxfw: Change the way to start stream
ALSA: oxfw: Add proc interface for debugging purpose
ALSA: oxfw: Change the way to make PCM rules/constraints
ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation
ALSA: oxfw: Change the way to name card
ALSA: dice: Add support for MIDI capture/playback
ALSA: dice: Add support for capturing PCM samples
ALSA: dice: Support for non SYT-Match sampling clock source mode
ALSA: dice: Add support for duplex streams with synchronization
ALSA: dice: Change the way to start stream
ALSA: jack: Add dummy snd_jack_set_key() definition
...
* support for mx6sl and mx6sx
* OMAP HDMI audio rewrite to make it finally work
* OMAP video PLL work to prepare for new DRA7xx SoCs
* simplefb DT related improvements
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Merge tag 'fbdev-3.19' of git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux
Pull fbdev updates from Tomi Valkeinen:
- support for mx6sl and mx6sx
- OMAP HDMI audio rewrite to make it finally work
- OMAP video PLL work to prepare for new DRA7xx SoCs
- simplefb DT related improvements
* tag 'fbdev-3.19' of git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux: (81 commits)
video: uvesafb: Deletion of an unnecessary check before the function call "platform_device_put"
video: fbdev-VIA: Deletion of an unnecessary check before the function call "framebuffer_release"
video: fbdev-MMP: Deletion of an unnecessary check before the function call "mmp_unregister_path"
video: mx3fb: Deletion of an unnecessary check before the function call "backlight_device_unregister"
video: fbdev-OMAP2: Deletion of unnecessary checks before the function call "i2c_put_adapter"
video: fbdev-SIS: Deletion of unnecessary checks before the function call "pci_dev_put"
video: smscufx: Deletion of unnecessary checks before the function call "vfree"
video: udlfb: Deletion of unnecessary checks before the function call "vfree"
video: uvesafb: Deletion of an unnecessary check before the function call "uvesafb_free"
video: fbdev-LCDC: Deletion of an unnecessary check before the function call "vfree"
video: fbdev: arkfb: suppress build warning
video: fbdev: s3fb: suppress build warning
video: fbdev: vt8623fb: suppress build warning
OMAPDSS: hdmi5: Fix bit field for IEC958_AES2_CON_SOURCE
OMAPDSS: hdmi: Remove __exit qualifier from hdmi_uninit_output()
OMAPDSS: hdmi5: Change hdmi_wp idlemode to to no_idle for audio playback
OMAPDSS: Remove all references to obsolete HDMI audio callbacks
ASoC: omap: Remove obsolete HDMI audio code and Kconfig options
OMAPDSS: hdmi5: Register ASoC platform device for omap hdmi audio
OMAPDSS: hdmi5: Remove callbacks for the old ASoC DAI driver
...
For fixing a build error with CONFIG_SND_JACK=n
sound/soc/codecs/ts3a227e.c:223:2: error: implicit declaration of function ‘snd_jack_set_key’ [-Werror=implicit-function-declaration]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots and lots of changes this time around, the usual set of driver
updates and a huge bulk of cleanups from Lars-Peter. Probably the most
interesting thing for most users is the Intel driver updates which will
(with some more machine integration work) enable support for newer x86
laptops.
- Conversion of AC'97 drivers to use regmap, bringing us closer to the
removal of the ASoC level I/O code.
- Clean up a lot of old drivers that were open coding things that have
subsequently been implemented in the core.
- Some DAPM performance improvements.
- Removal of the now seldom used CODEC mutex.
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers.
- Support for Samsung boards using rt5631 as the CODEC.
- Removal of the obsolete AFEB9260 machine driver.
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks.
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Merge tag 'asoc-v3.19' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.19
Lots and lots of changes this time around, the usual set of driver
updates and a huge bulk of cleanups from Lars-Peter. Probably the most
interesting thing for most users is the Intel driver updates which will
(with some more machine integration work) enable support for newer x86
laptops.
- Conversion of AC'97 drivers to use regmap, bringing us closer to the
removal of the ASoC level I/O code.
- Clean up a lot of old drivers that were open coding things that have
subsequently been implemented in the core.
- Some DAPM performance improvements.
- Removal of the now seldom used CODEC mutex.
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers.
- Support for Samsung boards using rt5631 as the CODEC.
- Removal of the obsolete AFEB9260 machine driver.
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks.
Current R-Car sound SSI PIO/DMA mode are using interrupt.
it is no longer "xxx_pio_xxx", rename it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The platform device should only be registered from OMAPDSS HDMI
driver. The platform driver registers and unregisters all ASoC
components needed for OMAP HDMI audio.
The hdmi audio driver implements cpu-dai component using the callbacks
provided by OMAPDSS and registers the component under DSS HDMI
device. Omap-pcm is registered for platform component also under DSS
HDMI device. Dummy codec is used as as codec component. The hdmi audio
driver implements also the card and registers it under its own
platform device.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ti.com>
The commit [7a2e9ddc: ALSA: usb-audio: Add native DSD support for
Denon/Marantz DACs] requires the new format definition that has
landed only in for-next branch.
As Russell King's explained it, there should not be pointers to
struct device_node:
"struct device_node is a ref-counted structure. That means if you
store a reference to it, you should "get" it, and you should "put"
it once you've done. The act of "put"ing the pointed-to structure
involves writing to that structure, so it is totally unappropriate
to store a device_node structure as a const pointer. It forces you
to have to cast it back to a non-const pointer at various points
in time to use various OF function calls."
[This isn't quite the application here, we're not geting or putting the
pointer though we did add some other users who call non-const OF
functions -- broonie]
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
If regmap is disabled there will be no users of the ASoC regmap helpers.
Furthermore regmap_exit() will no be defined causing the following compile
error:
sound/soc/soc-core.c: In function 'snd_soc_component_exit_regmap':
sound/soc/soc-core.c:2645:2: error: implicit declaration of function
'regmap_exit' [-Werror=implicit-function-declaration]
So disable the helpers if regmap is disabled.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 20feb88198 ASoC: Add helper functions for deferred regmap setup")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch exports a core function which handles the DT description
of multi-codec links (as: "sound-dai = <&hdmi 0>, <&spdif_codec>;")
and creates a CODEC component array in the DAI link.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are 3 JD modes in RT5645. This patch configure register
values according to platform data.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to its documentation the is_powered_on_standby field of the
uda134x platform data is supposed to prevent the the driver from shutting
down the ADC and DAC in standby mode. This behavior was broken in commit
commit f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component Support")
almost 5 years ago and all the flag does now is cause the driver to go to
SND_SOC_BIAS_ON in probe, just for the ASoC core to put it back into
SND_SOC_BIAS_STANDBY right after probe.
Apparently the intended behavior has not been missed, so just remove
is_powered_on_standby from the platform data struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds
DSD_U16_BE and DSD_U32_BE sample formats.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Acked-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some drivers (most notably the AC'97 drivers) do not have access to their
regmap struct when the component/codec is registered. For those drivers the
automatic regmap setup will not work and needs to be done manually,
typically from the component/CODEC drivers probe callback.
This patch adds a set of helper function to handle deferred regmap
initialization as well as early regmap tear-down.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that the ASoC core no longer needs a handle to the AC'97 device that is
associated with a CODEC we can remove it from the snd_soc_codec struct and
push it into the individual driver state structs like we do for other
communication buses. Doing so creates a clean separation between the AC'97
bus support and the ASoC core.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Setting the ac97_control field on a CPU DAI tells the ASoC core that this
DAI in addition to audio data also transports control data to the CODEC.
This causes the core to suspend the DAI after the CODEC and resume it before
the CODEC so communication to the CODEC is still possible. This is not
necessarily something that is specific to AC'97 and can be used by other
buses with the same requirement. This patch renames the flag from
ac97_control to bus_control to make this explicit.
While we are at it also change the type from int to bool.
The following semantich patch was used for automatic conversion of the
drivers:
// <smpl>
@@
identifier drv;
@@
struct snd_soc_dai_driver drv = {
- .ac97_control
+ .bus_control
=
- 1
+ true
};
// </smpl>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have all the information and dependencies we need to initialize and
register the device available in snd_soc_new_ac97_codec(). So there is no
need to delay the device registration until after the card itself as been
registered.
This makes the code significantly simpler and also makes it possible to use
the AC'97 device in the CODECs probe function. The later will be required to
be able to convert the AC'97 CODEC drivers to regmap.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This has no users since commit f0fba2ad1b ("ASoC: multi-component - ASoC
Multi-Component Support") which was almost 5 years ago. Given that this runs
after CODEC probe functions have been run it also doesn't seem to be that
useful.
So drop it altogether to make the code simpler.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
We always pass soc_ac97_ops to snd_soc_new_ac97_codec(). So instead of
allocating a snd_ac97_bus in snd_soc_new_ac97_codec() just use a static one
that gets initialized when snd_soc_set_ac97_ops() is called.
Also drop the device number parameter from snd_soc_new_ac97_codec(). We
currently only support one device per bus and all drivers pass 0 for the
device number. And if we should ever support multiple devices per bus it
wouldn't be up to individual AC'97 device drivers to pick their number, but
rather either the AC'97 adapter driver or the core code will assign them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the AC'97 support is splattered all throughout soc-core.c. Some
parts are #ifdef'd some parts are not. This patch moves the AC'97 support to
its own file, this should make the code a bit more clearer and also makes it
possible to easily not compile it into the kernel when not needed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a new helper function snd_pcm_stop_xrun() to the standard sequnce
lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the
existing open codes with this helper.
The function checks the PCM running state to prevent setting the wrong
state, too, for more safety.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This function has no more non regmap user, which means we can remove the
implementation of the function and associated functions and structure
fields.
For convenience we keep a static inline version of the function that
forwards calls to regcache_sync() unconditionally.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5645 codec support jack detection function. The patch will set
related registers if JD function is used.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DPCM can update the FE/BE connection states totally asynchronously
from the FE's PCM state. Most of FE/BE state changes are protected by
mutex, so that they won't race, but there are still some actions that
are uncovered. For example, suppose to switch a BE while a FE's
stream is running. This would call soc_dpcm_runtime_update(), which
sets FE's runtime_update flag, then sets up and starts BEs, and clears
FE's runtime_update flag again.
When a device emits XRUN during this operation, the PCM core triggers
snd_pcm_stop(XRUN). Since the trigger action is an atomic ops, this
isn't blocked by the mutex, thus it kicks off DPCM's trigger action.
It eventually updates and clears FE's runtime_update flag while
soc_dpcm_runtime_update() is running concurrently, and it results in
confusion.
Usually, for avoiding such a race, we take a lock. There is a PCM
stream lock for that purpose. However, as already mentioned, the
trigger action is atomic, and we can't take the lock for the whole
soc_dpcm_runtime_update() or other operations that include the lengthy
jobs like hw_params or prepare.
This patch provides an alternative solution. This adds a way to defer
the conflicting trigger callback to be executed at the end of FE/BE
state changes. For doing it, two things are introduced:
- Each runtime_update state change of FEs is protected via PCM stream
lock.
- The FE's trigger callback checks the runtime_update flag. If it's
not set, the trigger action is executed there. If set, mark the
pending trigger action and returns immediately.
- At the exit of runtime_update state change, it checks whether the
pending trigger is present. If yes, it executes the trigger action
at this point.
Reported-and-tested-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
This patch adds a new proc entry for PCM substreams to inject an
XRUN. When a PCM substream is running and any value is written to its
xrun_injection proc file, the driver triggers XRUN. This is a useful
feature for debugging XRUN and error handling code paths.
Note that this entry is enabled only when CONFIG_SND_PCM_XRUN_DEBUG is
set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The card field of the snd_soc_dai field is very rarely used. We can use
dai->component->card instead and remove the card field from the snd_soc_dai
struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Typically a DAI does not need direct access to the platform. Currently the
only user of this field is in a platform driver where we have a more direct
way of getting a pointer to the platform. This patch updates the driver to
use the more direct way and then removes the platform field from the
snd_soc_dai struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Those are unused and new drivers should use device driver suspend/resume.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
... and add proper kerneldoc comments.
There is no big reason to keep them as macros. Static inline
functions are safer in general, and suitable for kerneldoc, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some structure documentation was not right so fix it now
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit f227b88f0f ("ASoC: core: Add signed register volume control logic")
added support for signed control to the generic volsw control handler.
This makes it possible to use them for the S8 control as well, rather than
having to use a custom control handler implementation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently we cache the number of input and output paths going to/from a
widget only within a power update sequence. But not in between power update
sequences.
But we know how changes to the DAPM graph affect the number of input (form a
source) and output (to a sink) paths of a widget and only need to
recalculate them if a operation has been performed that might have changed
them.
* Adding/removing or connecting/disconnecting a path means that the for
the source of the path the number of output paths can change and for
the sink the number of input paths can change.
* Connecting/disconnecting a widget has the same effect has connecting/
disconnecting all paths of the widget. So for the widget itself the
number of inputs and outputs can change, for all sinks of the widget
the number of inputs can change and for all sources of the widget the
number of outputs can change.
* Activating/Deactivating a stream can either change the number of
outputs on the sources of the widget associated with the stream or the
number of inputs on the sinks.
Instead of always invalidating all cached numbers of input and output paths
for each power up or down sequence this patch restructures the code to only
invalidate the cached numbers when a operation that might change them has
been performed. This can greatly reduce the number of DAPM power checks for
some very common operations.
Since per DAPM operation typically only either change the number of inputs
or outputs the number of path checks is reduced by at least 50%. The number
of neighbor checks is also reduced about the same percentage, but since the
number of neighbors encountered when walking from sink to source is not the
same as when walking from source to sink the actual numbers will slightly
vary from card to card (e.g. for a mixer we see 1 neighbor when walking from
source to sink, but the number of inputs neighbors when walking from source
to sink).
Bigger improvements can be observed for widgets with multiple connected
inputs and output (e.g. mixers probably being the most widespread form of
this). Previously we had to re-calculate the number of inputs and outputs
on all input and output paths. With this change we only have to re-calculate
the number of outputs on the input path that got changed and the number of
inputs on the output paths.
E.g. imagine the following example:
A --> B ----.
v
M --> N --> Z <-- S <-- R
|
v
X
Widget Z has multiple input paths, if any change was made that cause Z to be
marked as dirty the power state of Z has to be re-computed. This requires to
know the number of inputs and outputs of Z, which requires to know the
number of inputs and outputs of all widgets on all paths from or to Z.
Previously this meant re-computing all inputs and outputs of all the path
going into or out of Z. With this patch in place only paths that actually
have changed need to be re-computed.
If the system is idle (or the part of the system affected by the changed
path) the number of path checks drops to either 0 or 1, regardless of how
large or complex the DAPM context is. 0 if there is no connected sink and no
connected source. 1 if there is either a connected source or sink, but not
both. The number of neighbor checks again will scale accordingly and will be
a constant number that is the number of inputs or outputs of the widget for
which we did the path check.
When loading a state file or switching between different profiles typically
multiple mixer and mux settings are changed, so we see the benefit of this
patch multiplied for these kinds of operations.
Testing with the ADAU1761 shows the following changes in DAPM stats for
changing a single Mixer switch for a Mixer with 5 inputs while the DAPM
context is idle.
Power Path Neighbour
Before: 2 12 30
After: 2 1 2
For the same switch, but with a active playback stream the stat changed are
as follows.
Power Path Neighbour
Before: 10 20 54
After: 10 7 21
Cumulative numbers for switching the audio profile which changes 7 controls
while the system is idle:
Power Path Neighbour
Before: 16 80 170
After: 16 7 23
Cumulative numbers for switching the audio profile which changes 7 controls
while playback is active:
Power Path Neighbour
Before: 51 123 273
After: 51 29 109
Starting (or stopping) the playback stream:
Power Path Neighbour
Before: 34 34 117
After: 34 17 69
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The state of endpoint widgets is affected by that card's power state.
Endpoint widgets that do no have the ignore_suspend flag set will be
considered inactive during suspend. So they have to be re-checked and marked
dirty after the card's power state changes. Currently the input and output
widgets are marked dirty instead, this works most of the time since
typically a path from one endpoint to another will go via a input or output
widget. But marking the endpoints dirty is technically more correct and will
also work for odd corner cases.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Supply widgets do not count towards the input and output widgets of their
neighbors and for supply widgets themselves we do not care for the number
of input or output paths. This means that a path that connects to a supply
widget effectively behaves the same as a path that as the weak property set.
This patch adds a new path flag that gets set to true when the path is
connected to at least one supply widget. If a path with the flag set is
encountered in is_connected_{input,output}_ep() is is skipped in the same
way that weak paths are skipped. This slightly brings down the number of
path checks.
Since both the weak and the supply flag are implemented as bitfields which
are stored in the same word there is no runtime overhead due to checking
both rather than just one and also the size of the path struct is not
increased by this patch. Another advantage is that we do not have to handle
supply widgets in is_connected_{input,output}_ep() anymore since it will
never be called for supply widgets. The only exception is from
dapm_widget_power_read_file() where a check is added to special case supply
widgets.
Testing with the ADAU1761, which has a handful of supply widgets, shows the
following changes in the DAPM stats for a playback stream start.
Power Path Neighbour
Before: 34 78 117
After: 34 48 117
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAPM widgets can be classified into four categories:
* supply: Supply widgets do not affect the power state of their
non-supply widget neighbors and unlike other widgets a
supply widget is not powered up when it is on an active
path, but when at least on of its neighbors is powered up.
* source: A source is a widget that receives data from outside the
DAPM graph or generates data. This can for example be a
microphone, the playback DMA or a signal generator. A source
widget will be considered powered up if there is an active
path to a sink widget.
* sink: A sink is a widget that transmits data to somewhere outside
of the DAPM graph. This can e.g. be a speaker or the capture
DMA. A sink widget will be considered powered up if there is
an active path from a source widget.
* normal: Normal widgets are widgets not covered by the categories
above. A normal widget will be considered powered up if it
is on an active path between a source widget and a sink
widget.
The way the number of input and output paths for a widget is calculated
depends on its category. There are a bunch of factors which decide which
category a widget is. Currently there is no formal classification of these
categories and we calculate the category of the widget based on these
factors whenever we want to know it. This is at least once for every widget
during each power update sequence. The factors which determine the category
of the widgets are mostly static though and if at all change rather seldom.
This patch introduces three new per widget flags, one for each of non-normal
widgets categories. Instead of re-computing the category each time we want
to know them the flags will be checked. For the majority of widgets the
category is solely determined by the widget id, which means it never changes
and only has to be set once when the widget is created. The only widgets
with dynamic categories are:
snd_soc_dapm_dai_out: Is considered a sink iff the capture stream is
active, otherwise normal.
snd_soc_dapm_dai_in: Is considered a source iff the playback stream
is active, otherwise normal.
snd_soc_dapm_input: Is considered a sink iff it has no outgoing
paths, otherwise normal.
snd_soc_dapm_output: Is considered a source iff it has no incoming
paths, otherwise normal.
snd_soc_dapm_line: Is considered a sink iff it has no outgoing paths
and is considered a source iff it has no incoming paths,
otherwise normal.
For snd_soc_dapm_dai_out/snd_soc_dapm_dai_in widgets the category will be
updated when a stream is started or stopped. For the other dynamic widgets
the category will be updated when a path connecting to it is added or
removed.
Introducing those new widget categories allows to make
is_connected_{output,input}_ep, which are among the hottest paths of the
DAPM algorithm, more generic and significantly shorter.
The before and after sizes for is_connected_{output,input}_ep are:
On ARM (defconfig + CONFIG_SND_SOC):
function old new delta
is_connected_output_ep 480 340 -140
is_connected_input_ep 456 352 -104
On amd64 (defconfig + CONFIG_SND_SOC):
function old new delta
is_connected_output_ep 579 427 -152
is_connected_input_ep 563 427 -136
Which is about a 25%-30% decrease, other architectures are expected to have
similar numbers. At the same time the size of the snd_soc_dapm_widget struct
does not change since the new flags are stored in the same word as the
existing flags.
Note: that since the per widget 'ext' flag was only used to decide whether a
snd_soc_dapm_input or snd_soc_dapm_output widget was a source or a sink it
is now unused and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The 'walked' flag was used to avoid walking paths that have already been
walked. But since we started caching the number of inputs and outputs of a
path we never actually get into a situation where we try to walk a path that
has the 'walked' flag set.
There are two cases in which we can end up walking a path multiple times
within a single run of is_connected_output_ep() or is_connected_input_ep().
1) If a path splits up and rejoins later:
.--> C ---v
A -> B E --> F
'--> D ---^
When walking from A to F we'll end up at E twice, once via C and once via D.
But since we do a depth first search we'll fully discover the path and
initialize the number of outputs/inputs of the widget the first time we get
there. The second time we get there we'll use the cached value and not
bother to check any of the paths again. So we'll never see a path where
'walked' is set in this case.
2) If there is a circle:
A --> B <-- C <-.--> F
'--> D ---'
When walking from A to F we'll end up twice at B. But since there is a
circle the 'walking' flag will still be set on B once we get there the
second time. This means we won't look at any of it's outgoing paths. So in
this case we won't ever see a path where 'walked' is set either.
So it is safe to remove the flag. This on one hand means we remove some
always true checks from one of the hottest paths of the DAPM algorithm and
on the other hand means we do not have to do the tedious clearing of the
flag after checking the number inputs or outputs of a widget.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This allows to enable Mic Jack detection feature
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Modified-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
this is a series of patches to just convert the plain info callback
for enum ctl elements to snd_ctl_elem_info(). Also, it includes the
extension of snd_ctl_elem_info(), for catching the unexpected string
cut-off and handling the zero items.
gpio_config is array of 6 elements that allows to set GPIO as
floating, pullup, pulldown.
Sponsored: Google ChromeOS
Signed-off-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull slave-dmaengine updates from Vinod Koul:
"For dmaengine contributions we have:
- designware cleanup by Andy
- my series moving device_control users to dmanegine_xxx APIs for
later removal of device_control API
- minor fixes spread over drivers mainly mv_xor, pl330, mmp, imx-sdma
etc"
* 'for-linus' of git://git.infradead.org/users/vkoul/slave-dma: (60 commits)
serial: atmel: add missing dmaengine header
dmaengine: remove FSLDMA_EXTERNAL_START
dmaengine: freescale: remove FSLDMA_EXTERNAL_START control method
carma-fpga: move to fsl_dma_external_start()
carma-fpga: use dmaengine_xxx() API
dmaengine: freescale: add and export fsl_dma_external_start()
dmaengine: add dmaengine_prep_dma_sg() helper
video: mx3fb: use dmaengine_terminate_all() API
serial: sh-sci: use dmaengine_terminate_all() API
net: ks8842: use dmaengine_terminate_all() API
mtd: sh_flctl: use dmaengine_terminate_all() API
mtd: fsmc_nand: use dmaengine_terminate_all() API
V4L2: mx3_camer: use dmaengine_pause() API
dmaengine: coh901318: use dmaengine_terminate_all() API
pata_arasan_cf: use dmaengine_terminate_all() API
dmaengine: edma: check for echan->edesc => NULL in edma_dma_pause()
dmaengine: dw: export probe()/remove() and Co to users
dmaengine: dw: enable and disable controller when needed
dmaengine: dw: always export dw_dma_{en,dis}able
dmaengine: dw: introduce dw_dma_on() helper
...
Currently the sequencer module binding is performed independently from
the card module itself. The reason behind it is to keep the sequencer
stuff optional and allow the system running without it (e.g. for using
PCM or rawmidi only). This works in most cases, but a remaining
problem is that the binding isn't done automatically when a new driver
module is probed. Typically this becomes visible when a hotplug
driver like usb audio is used.
This patch tries to address this and other potential issues. First,
the seq-binder (seq_device.c) tries to load a missing driver module at
creating a new device object. This is done asynchronously in a workq
for avoiding the deadlock (modprobe call in module init path).
This action, however, should be enabled only when the sequencer stuff
was already initialized, i.e. snd-seq module was already loaded. For
that, a new function, snd_seq_autoload_init() is introduced here; this
clears the blocking of autoloading, and also tries to load all pending
driver modules.
Reported-by: Adam Goode <agoode@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- More componentisation work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes and
enhancements for the associated CODEC drivers, this is going to need
a lot quirks over time due to the lack of any firmware description of
the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest
Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX
processors.
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Merge tag 'asoc-v3.18' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.18
- More componentisation work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes and
enhancements for the associated CODEC drivers, this is going to need
a lot quirks over time due to the lack of any firmware description of
the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest
Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX
processors.
The DSP driver needs to know widget control value in its event handler for
widgets like mixers. This is required in the subsequent patches
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 only remaining user of the CODEC pointer in the DAPM struct is to
initialize the CODEC pointer in the widget struct. The later is scheduled
for removal, but has still a few users left. For now use
dapm->component->codec to initialize it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Rewrite VXpocket and VX222 drivers to use the new PCM nonatomic ops.
The former irq tasklet is replaced with a threaded irq handler, and
the tasklet for the PCM delayed start is simply merged into the normal
PCM trigger, as well as the replacement of spinlock with mutex.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a common storage for platform data related structures and definitions
inside kernel source tree. The patch moves file from include/linux to
include/linux/platform_data and renames it acoordingly. The users are also
updated.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-by: Viresh Kumar <viresh.kumar@linaro.org>
[For the arch/avr32/.* and .*sound/atmel.*]
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
XMOS based USB DACs with native DSD support expose this feature via a USB
alternate setting. The audio format is either 32-bit raw or a 32-bit PCM format.
To utilize this feature on linux this patch introduces a new 32-bit DSD
sampleformat DSD_U32_LE.
A follow up patch will add a quirk for XMOS based devices to utilize the new format.
Further patches will add support to alsa-lib.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are no users left and new users should rather use the component_driver
struct embedded in the snd_soc_platform_driver struct to do this. E.g.:
static const struct snd_soc_platform_driver foobar_driver = {
.component_driver = {
.dapm_widgets = ...,
.num_dapm_widgets = ...,
...,
},
...
};
instead of
static const struct snd_soc_platform_driver foobar_driver = {
.dapm_widgets = ...,
.num_dapm_widgets = ...,
...
};
This also allows us to remove the steal_sibling_dai_widgets hack.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add Add snd_soc_component_{get,set}_drvdata() similar to
snd_soc_codec_{get,set}_drvdata() and snd_soc_platform_{get,set}_drvdata().
Also update them to use the new functions internally.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is a substantial amount of drivers that in go to SND_SOC_BIAS_OFF on
suspend and go back to SND_SOC_BIAS_SUSPEND on resume (Often this is even
the only thing done in the suspend and resume handlers). This patch
introduces a new suspend_bias_off flag, which when set by a driver will let
the ASoC core automatically put the device's DAPM context at the
SND_SOC_BIAS_OFF level during suspend. Once the device is resumed the DAPM
context will go back to SND_SOC_BIAS_STANDBY (if the context is idle,
otherwise to SND_SOC_BIAS_ON).
This will allow us to remove a fair bit of duplicated code from the drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous commit for the non-atomic PCM ops added more codes to
snd_pcm_stream_lock() and its variants. Since they are inlined
functions, it resulted in a significant code size bloat. For reducing
the size bloat, this patch changes the inline functions to the normal
function calls. The export of rwlock and rwsem are removed as well,
since they are referred only in pcm_native.c now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, many PCM operations are performed in a critical section
protected by spinlock, typically the trigger and pointer callbacks are
assumed to be atomic. This is basically because some trigger action
(e.g. PCM stop after drain or xrun) is done in the interrupt handler.
If a driver runs in a threaded irq, however, this doesn't have to be
atomic. And many devices want to handle trigger in a non-atomic
context due to lengthy communications.
This patch tries all PCM calls operational in non-atomic context.
What it does is very simple: replaces the substream spinlock with the
corresponding substream mutex when pcm->nonatomic flag is set. The
driver that wants to use the non-atomic PCM ops just needs to set the
flag and keep the rest as is. (Of course, it must not handle any PCM
ops in irq context.)
Note that the code doesn't check whether it's atomic-safe or not, but
trust in 100% that the driver sets pcm->nonatomic correctly.
One possible problem is the case where linked PCM substreams have
inconsistent nonatomic states. For avoiding this, snd_pcm_link()
returns an error if one tries to link an inconsistent PCM substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are no real users for cache_only in "struct snd_soc_codec" so remove
it and needless debugfs node.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are two pins can be used for rt5677's DMIC2 clock. This patch
add the select options for it.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There were never any actual users of this in upstream and by we have with
regmap a replacement in place, which should be used by new drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
So far regmap is only automatically initialized for CODECs. Now that we have the
infrastructure in place to let components have DAPM widgets and controls that
want to use the generic regmap based IO also make sure to automatically
initialize regmap for all components.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Given that the component is the containing structure it makes more sense to pass
the component rather than the DAPM context to the AUX dev init callback.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch makes it possible to register arbitrary components as a AUX dev
for a card. This was previously only possible for CODEC components. With
componentization having made it possible for components to have DAPM contexts
and controls there is no reason why AUX devs should be artificially limited to
snd_soc_codec devices.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that we have a unified probe and remove path make sure to call them for all
components. soc_{probe,remove}_component are responsible for setting up the DAPM
context for the component, initialize the component prefix, manage the debugfs
entries as well as do the registration of table based controls and DAPM
elements. They also call the component drivers probe and remove callbacks. This
patch makes these things available for generic snd_soc_component drivers rather
than only having them for snd_soc_codec and snd_soc_platform drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform and CODEC probe and remove code is now largely identical. This
patch consolidates it at the component level.
The resulting code is slightly larger due to all the boiler plate code setting
up the indirection for the table based control and DAPM registration. Once all
drivers have been update to no longer use the snd_soc_codec_driver and
snd_soc_platform_driver specific fields for this the indirection can be removed
again.
This patch contains two noteworthy hacks that are only meant to be temporary to
be able to update drivers and the core in separate incremental patches.
The first hack is related to that some DPCM platforms expect that the DAPM
widgets for the DAIs of a snd_soc_component are created in the DAPM context of
the snd_soc_platform that has the same parent device. For handling this the
steal_sibling_dai_widgets attribute is introduced. It gets set for
snd_soc_platforms that register DAPM elements. When creating the DAI widgets for
a component this flag is checked and if it is found on one of the siblings the
component will not create any DAI widgets in its own DAPM context. If the
attribute is set on a platform it will look for siblings components and create
DAI widgets for them in its own context. The fix for this will be to update
the offending drivers to only register a single component rather than two.
The second hack deals with the fact that the ASoC card suspend and resume code
still needs a list of CODECs that have been registered for the card. To handle
this the generic probe and remove path have a check to see if the component is
CODEC and if yes add/remove it to the card's CODEC list. While it is possible to
clean up the suspend/resume code to not need the CODEC list anymore this is a
bit of a chicken and egg problem since it will become easier to clean up the
suspend/resume code once there is a unified component layer.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The debugfs registration is mostly identical between platforms and CODECs. This
patches consolidates the two implementations at the component level.
Unfortunately there are still a couple of CODEC specific debugfs files that are
related to legacy ASoC IO that need to be registered. For this a new callback is
added to the component struct that will be initialized when a CODEC is
registered and will be used to register the CODEC specific files. Once there are
no drivers left using legacy IO this can be removed again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit 7523a271 - "ASoC: core: add a helper for extended byte controls using
TLV" introduced support for TLV byte controls but had a typo for the info
function, so fix the same
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@linaro.org>
Dropping the const qualifiers prevents "passing argument 1 of ‘of_node_put’
discards ‘const’ qualifier from pointer target type" type warnings when
compiling the code dropping reference to cpu_of_node, codec_of_node or
platform_of_node with with an of_node_put() function call.
This lets us to avoid casting to struct device_node * or caching variables
internally in drivers just to be able to properly drop a reference to the
OF node on clean up paths.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This has been a pretty exciting release in terms of the framework, we've
finally got support for multiple CODECs attached to a single DAI link
which has been something there's been interest in as long as I've been
working on ASoC. A big thanks to Benoit and Misael for their work on
this.
Otherwise it's been a fairly standard release for development, including
more componentisation work from Lars-Peter and a good selection of both
CODEC and CPU drivers.
- Support for multiple CODECs attached to a single DAI, enabling
systems with for example multiple DAC/speaker drivers on a single
link, contributed by Benoit Cousson based on work from Misael Lopez
Cruz.
- Support for byte controls larger than 256 bytes based on the use of
TLVs contributed by Omair Mohammed Abdullah.
- More componentisation work from Lars-Peter Clausen.
- The remainder of the conversions of CODEC drivers to params_width()
- Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
TAS2552.
- Lots of updates and fixes, especially to the DaVinci, Intel,
Freescale, Realtek, and rcar drivers.
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Merge tag 'asoc-v3.17' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.17
This has been a pretty exciting release in terms of the framework, we've
finally got support for multiple CODECs attached to a single DAI link
which has been something there's been interest in as long as I've been
working on ASoC. A big thanks to Benoit and Misael for their work on
this.
Otherwise it's been a fairly standard release for development, including
more componentisation work from Lars-Peter and a good selection of both
CODEC and CPU drivers.
- Support for multiple CODECs attached to a single DAI, enabling
systems with for example multiple DAC/speaker drivers on a single
link, contributed by Benoit Cousson based on work from Misael Lopez
Cruz.
- Support for byte controls larger than 256 bytes based on the use of
TLVs contributed by Omair Mohammed Abdullah.
- More componentisation work from Lars-Peter Clausen.
- The remainder of the conversions of CODEC drivers to params_width()
- Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
TAS2552.
- Lots of updates and fixes, especially to the DaVinci, Intel,
Freescale, Realtek, and rcar drivers.
snd_soc_open() will trigger pm_runtime resume() which will then enable
the regulator and initialization. So we should make sure the MCLK is
enabled before this resume().
Previously we let the machine driver get the clock and enable it in
its probe(). However, considering about power saving, it'll be better
to enable it when it's going to be used and disable it after using.
So this patch just simply adds clk_get() and clk_enable() to WM8962
driver. Meanwhile, it marks clock pointer to NULL if no clock assigned
to it so it will not break any current function.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have now everything in place to actual let a component register controls. Add
a function which allows to do so.
Also update snd_soc_add_codec_controls() and snd_soc_platform_controls() to use
this new function internally. And while we are at it also change the
num_controls parameter of those two functions from int to unsigned int.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Both the snd_soc_codec and snd_soc_platform struct do have a pointer to the
parent card and both handle this pointer in mostly the same way. This patch
moves the card field to the component level which will allow further code
consolidation between platforms and CODECS.
Since there are only a handful of users of the snd_soc_codec struct's card field
(and none of the snd_soc_platform's) these are update in this patch as well,
which allows it to be removed from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform_dev_list was added in commit f0fba2ad1b ("ASoC: multi-component -
ASoC Multi-Component Support") and while platforms are added and remove from
that list it is otherwise unused. This patch removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Support the TI TAS2552 Class D amplifier.
The TAS2552 is a high efficiency Class-D audio
power amplifier with advanced battery current
management and an integrated Class-G boost
The device constantly measures the
current and voltage across the load and provides a
digital stream of this information.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a function helper to factorize the hw_params code.
Suggested by Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
DAI link assumes a one to one mapping between CPU DAI and CODEC. In
some cases, the same CPU DAI can be connected to several codecs.
This is the case for example, if you connect two mono codecs to the
same I2S link in order to have a stereo card.
The current ASoC implementation does not allow such setup.
Add support for DAI links composed of a single CPU DAI and multiple
CODECs. Sound cards have to pass the CODECs array in the corresponding
DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in
this array is described in the same manner single CODEC DAIs are
(either DT/OF node or codec_name).
Multi-codec links are not supported in the case of CODEC to CODEC links.
Just print a warning if it happens.
Based on an original code done by Misael.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Fabien Parent <fparent@baylibre.com>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALSA supports arbitrary length TLVs for each kcontrol that can be used
to pass metadata about the control (e.g. volumes, enum information). The
same transport mechanism is now used for arbitrary length data by
defining a new helper.
Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For applications which need to synchronise with external timebases such
as broadcast TV applications the kernel monotonic time is not optimal as
it includes adjustments from NTP and so may still include discontinuities
due to that. A raw monotonic time which does not include any adjustments
is available in the kernel from getrawmonotonic() so provide userspace with
a new timestamp type SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW which provides
timestamps based on this as an option.
[dropped tstamp_type assignment code, as it's no longer needed -- tiwai]
Reported-by: Daniel Thompson <daniel.thompson@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds an equivalent of SOC_ENUM_EXT for value enums
Strictly speaking SOC_ENUM_EXT can also be used to define
a value enum since the only difference is the get and set
functions. But this doesn't look good in code because it is
inconsistent with the normal control definitions. Adding a
specific SOC_VALUE_ENUM_EXT is better for code clarity.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds Realtek ALC286 codec driver.
ALC286 is a dual mode codec, which can run as HD-A or I2S mode.
It is controlled by HD-A verb commands via I2C protocol.
The following is the I/O difference between ALC286 and general I2S codecs.
1. A HD-A verb command contains three parts, NID, VID, and PID.
And an I2S command contains only two parts: address and data.
2. Not only the register address is written, but the read command also
includes the entire write command.
3. rt286 uses different registers for read and write the same bits.
We map verb command to regmap structure. However, we read most registers from
cache to prevent the asymmetry read/write issue in rt286.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a minimum support of Realtek ALC5670 codec.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Sound data needs to be sent to R-Car sound SSI when playback.
But, there are 2 interfaces for it.
1st is SSITDR/SSIRDR which are mapped on SSI.
2nd is SSIn_BUSIF which are mapped on SSIU.
2nd SSIn_BUSIF is used when DMA transfer,
and it is always used if sound data came from via SRC.
But, we can use it when SSI+DMA case too.
(Current driver is assuming 1st SSITDR/SSIRDR for it)
2nd SSIn_BUSIF can be used as FIFO.
This is very helpful/useful for SSI+DMA.
But DMA address / DMA ID are not same between 1st/2nd cases.
This patch care about these settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform field in the snd_soc_dapm_widget and snd_soc_dapm_context structs
is now unused can be removed. New code that wants to get the platform for a
widget or dapm context should use snd_soc_dapm_to_platform(w->dapm) or
snd_soc_dapm_to_platform(dapm).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DAI DAPM context was added in commit be09ad90 ("ASoC: core: Add platform DAI
widget mapping") and the only user was removed again in commit ae10e7e8f ("ASoC:
core: Only add platform DAI widgets once."). Now that we have a per component
DAPM context it is unlikely that we'll need the DAI DAPM context again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds stream_event() and seq_notifier() callbacks similar to those
found in the snd_soc_codec_driver and snd_soc_platform driver struct to the
snd_soc_component_driver struct. This is meant to unify the handling of these
callbacks across different types of components and will eventually allow their
removal from the CODEC and platfrom driver structs.
The new callbacks are slightly different from the old ones in that they take a
snd_soc_component as a parameter rather than a snd_soc_dapm_context. This was
done since otherwise casting from the DAPM context to the component would
typically be the first thing to do in the callback. And the interface becomes
slightly cleaner by passing a snd_soc_component to all callbacks in the
snd_soc_component_driver struct.
The patch also already removes the stream_event() callback from the
snd_soc_codec_driver and snd_soc_platform_driver structs as it is currently
unused.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_soc_platform dapm field is not accessed outside of the ASoC core. Switch
it over to using the snd_soc_component DAPM context.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds full DAPM support at the component level. Previously there was
only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor
MUXs) for platforms. Having DAPM support at the component level will allow all
types of components to use DAPM and also help in consolidating the DAPM support
between CODECs and platforms.
Since the DAPM context is directly embedded into the snd_soc_codec and
snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of
drivers moving the field just right now is not possible without causing code
churn. The approach this patch takes is to add two new fields to the component
struct. One field which is the pointer to the actual DAPM context used by the
component and one DAPM context that will be used as the default if no other
context was specified. For CODECs and platforms the pointer is initialized to
point to the CODEC or platform DAPM context. All generic code when referencing
a component's DAPM struct will go via the pointer. This will make it possible to
eventually seamlessly move the DAPM context from snd_soc_codec and
snd_soc_platform struct over once all direct references have been eliminated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently the DAPM code directly looks at the CODEC driver struct to get a
handle to the set_bias_level() callback. This patch adds a new set_bias_level()
callback to the DAPM context struct. The DAPM code will use this new callback
instead of the CODEC callback. For CODECs the new callback is set up to call the
CODEC specific set_bias_level callback(). Not looking directly at the CODEC
driver struct will allow non CODEC DAPM contexts to implement a set_bias_level()
callback.
This is also similar to how the seq_notifier() and stream_event() callbacks are
currently handled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently only pins in CODEC DAPM contexts are automatically marked as
non-connected if the card has the fully_routed flag set. This makes sense since
widgets which qualify for auto-disconnection are only found in CODEC DAPM
contexts. But with componentisation this is going to change, so consider all
widgets for auto-disconnection.
Also it is probably faster to walk the widgets list only once rather than once
for each CODEC.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The component struct already has a name and id field which are initialized to
the same values as the same fields in the CODEC and platform structs. So remove
them from the CODEC and platform structs and used the ones from the component
struct instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move the name_prefix from the CODEC struct to the component struct. This will
eventually allow to specify prefixes for all types of components. It is also
necessary to make the DAPM code component type independent (i.e. a DAPM context
does not need to know whether it belongs to a CODEC or a platform or something
else).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Lars-Peter Clausen