Add a callback fro clients for notification about DAPM route loading and
unloading.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Give topology clients more access to the topology data by passing index,
pcm, link_config and dai_driver to clients. This allows clients to fully
instantiate and track topology objects.
The SOF driver is the first user of these new APIs and needs them to build
component topology driver and FW objects.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Machine drivers statically define a number of DAI links that currently
cannot be changed or removed by topology. This means PCMs and platform
components cannot be changed by topology at runtime AND machine drivers
are tightly coupled to topology.
This patch allows topology to override the machine driver DAI link config
in order to reuse machine drivers with different topologies and platform
components. The patch supports :-
1) create new FE PCMs with a topology defined PCM ID.
2) destroy existing static FE PCMs
3) change the platform component driver.
4) assign any new HW params fixups.
The patch requires no changes to the machine drivers, but does add some
platform component flags that the platform component driver can assign
before loading topologies.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The emu10k1 fx8010 code allocates each irq resource dynamically and
links to the list at PCM trigger callback. Due to the nature of
trigger callback, the allocation is done with GFP_ATOMIC, hence it
may fail more often. Moreover, the irq resource isn't big at all, and
using the kmalloc for this won't save many bytes, either.
This patch removes the dynamic allocation and embeds the irq resource
into struct snd_emu10k1_fx8010_pcm.irq field instead of keeping a
pointer. As a result, it simplifies the code and removes the
unnecessary GFP_ATOMIC usage.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This became a large update. The changes are scattered widely,
and majority of them are attributed to ASoC componentization.
The gitk output made me dizzy, but it's slightly better than
London tube.
OK, below are some highlights:
- Continued hardening works in ALSA PCM core; most of the
existing syzkaller reports should have been covered.
- USB-audio got the initial USB Audio Class 3 support, as well
as UAC2 jack detection support and more DSD-device support.
- ASoC componentization: finally each individual driver was
converted to components framework, which is more future-proof
for further works. Most of conversations were systematic.
- Lots of fixes for Intel Baytrail / Cherrytrail devices with
Realtek codecs, typically tablets and small PCs.
- Fixes / cleanups for Samsung Odroid systems
- Cleanups in Freescale SSI driver
- New ASoC drivers:
* AKM AK4458 and AK5558 codecs
* A few AMD based machine drivers
* Intel Kabylake machine drivers
* Maxim MAX9759 codec
* Motorola CPCAP codec
* Socionext Uniphier SoCs
* TI PCM1789 and TDA7419 codecs
- Retirement of Blackfin drivers along with architecture removal.
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Merge tag 'sound-4.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became a large update. The changes are scattered widely, and the
majority of them are attributed to ASoC componentization. The gitk
output made me dizzy, but it's slightly better than London tube.
OK, below are some highlights:
- Continued hardening works in ALSA PCM core; most of the existing
syzkaller reports should have been covered.
- USB-audio got the initial USB Audio Class 3 support, as well as
UAC2 jack detection support and more DSD-device support.
- ASoC componentization: finally each individual driver was converted
to components framework, which is more future-proof for further
works. Most of conversations were systematic.
- Lots of fixes for Intel Baytrail / Cherrytrail devices with Realtek
codecs, typically tablets and small PCs.
- Fixes / cleanups for Samsung Odroid systems
- Cleanups in Freescale SSI driver
- New ASoC drivers:
* AKM AK4458 and AK5558 codecs
* A few AMD based machine drivers
* Intel Kabylake machine drivers
* Maxim MAX9759 codec
* Motorola CPCAP codec
* Socionext Uniphier SoCs
* TI PCM1789 and TDA7419 codecs
- Retirement of Blackfin drivers along with architecture removal"
* tag 'sound-4.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (497 commits)
ALSA: pcm: Fix UAF at PCM release via PCM timer access
ALSA: usb-audio: silence a static checker warning
ASoC: tscs42xx: Remove owner assignment from i2c_driver
ASoC: mediatek: remove "simple-mfd" in the example
ASoC: cpcap: replace codec to component
ASoC: Intel: bytcr_rt5651: don't use codec anymore
ASoC: amd: don't use codec anymore
ALSA: usb-audio: fix memory leak on cval
ALSA: pcm: Fix mutex unbalance in OSS emulation ioctls
ASoC: topology: Fix kcontrol name string handling
ALSA: aloop: Mark paused device as inactive
ALSA: usb-audio: update clock valid control
ALSA: usb-audio: UAC2 jack detection
ALSA: pcm: Return -EBUSY for OSS ioctls changing busy streams
ALSA: pcm: Avoid potential races between OSS ioctls and read/write
ALSA: usb-audio: Integrate native DSD support for ITF-USB based DACs.
ALSA: usb-audio: FIX native DSD support for TEAC UD-501 DAC
ALSA: usb-audio: Add native DSD support for Luxman DA-06
ALSA: usb-audio: fix uac control query argument
ASoC: nau8824: recover system clock when device changes
...
This is a *very* big release for ASoC. Not much change in the core but
there s the transition of all the individual drivers over to components
which is intended to support further core work. The goal is to make it
easier to do further core work by removing the need to special case all
the different driver classes in the core, many of the devices end up
being used in multiple roles in modern systems.
We also have quite a lot of new drivers added this month of all kinds,
quite a few for simple devices but also some more advanced ones with
more substantial code.
- The biggest thing is the huge series from Morimoto-san which
converted everything over to components. This is a huge change by
code volume but was fairly mechanical
- Many fixes for some of the Realtek based Baytrail systems covering
both the CODECs and the CPUs, contributed by Hans de Goode.
- Lots of cleanups for Samsung based Odroid systems from Sylwester
Nawrocki.
- The Freescale SSI driver also got a lot of cleanups from Nicolin
Chen.
- The Blackfin drivers have been removed as part of the removal of the
architecture.
- New drivers for AKM AK4458 and AK5558, several AMD based machines,
several Intel based machines, Maxim MAX9759, Motorola CPCAP,
Socionext Uniphier SoCs, and TI PCM1789 and TDA7419
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Merge tag 'asoc-v4.17' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.17
This is a *very* big release for ASoC. Not much change in the core but
there s the transition of all the individual drivers over to components
which is intended to support further core work. The goal is to make it
easier to do further core work by removing the need to special case all
the different driver classes in the core, many of the devices end up
being used in multiple roles in modern systems.
We also have quite a lot of new drivers added this month of all kinds,
quite a few for simple devices but also some more advanced ones with
more substantial code.
- The biggest thing is the huge series from Morimoto-san which
converted everything over to components. This is a huge change by
code volume but was fairly mechanical
- Many fixes for some of the Realtek based Baytrail systems covering
both the CODECs and the CPUs, contributed by Hans de Goode.
- Lots of cleanups for Samsung based Odroid systems from Sylwester
Nawrocki.
- The Freescale SSI driver also got a lot of cleanups from Nicolin
Chen.
- The Blackfin drivers have been removed as part of the removal of the
architecture.
- New drivers for AKM AK4458 and AK5558, several AMD based machines,
several Intel based machines, Maxim MAX9759, Motorola CPCAP,
Socionext Uniphier SoCs, and TI PCM1789 and TDA7419
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Backmerge tag 'v4.16-rc7' into drm-next
Linux 4.16-rc7
This was requested by Daniel, and things were getting
a bit hard to reconcile, most of the conflicts were
trivial though.
OSS PCM stream management isn't modal but it allows ioctls issued at
any time for changing the parameters. In the previous hardening
patch ("ALSA: pcm: Avoid potential races between OSS ioctls and
read/write"), we covered these races and prevent the corruption by
protecting the concurrent accesses via params_lock mutex. However,
this means that some ioctls that try to change the stream parameter
(e.g. channels or format) would be blocked until the read/write
finishes, and it may take really long.
Basically changing the parameter while reading/writing is an invalid
operation, hence it's even more user-friendly from the API POV if it
returns -EBUSY in such a situation.
This patch adds such checks in the relevant ioctls with the addition
of read/write access refcount.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Back in 2013, runtime PM for GPUs with integrated HDA controller was
introduced with commits 0d69704ae3 ("gpu/vga_switcheroo: add driver
control power feature. (v3)") and 246efa4a07 ("snd/hda: add runtime
suspend/resume on optimus support (v4)").
Briefly, the idea was that the HDA controller is forced on and off in
unison with the GPU.
The original code is mostly still in place even though it was never a
100% perfect solution: E.g. on access to the HDA controller, the GPU
is powered up via vga_switcheroo_runtime_resume_hdmi_audio() but there
are no provisions to keep it resumed until access to the HDA controller
has ceased: The GPU autosuspends after 5 seconds, rendering the HDA
controller inaccessible.
Additionally, a kludge is required when hda_intel.c probes: It has to
check whether the GPU is powered down (check_hdmi_disabled()) and defer
probing if so.
However in the meantime (in v4.10) the driver core has gained a feature
called device links which promises to solve such issues in a clean way:
It allows us to declare a dependency from the HDA controller (consumer)
to the GPU (supplier). The PM core then automagically ensures that the
GPU is runtime resumed as long as the HDA controller's ->probe hook is
executed and whenever the HDA controller is accessed.
By default, the HDA controller has a dependency on its parent, a PCIe
Root Port. Adding a device link creates another dependency on its
sibling:
PCIe Root Port
^ ^
| |
| |
HDA ===> GPU
The device link is not only used for runtime PM, it also guarantees that
on system sleep, the HDA controller suspends before the GPU and resumes
after the GPU, and on system shutdown the HDA controller's ->shutdown
hook is executed before the one of the GPU. It is a complete solution.
Using this functionality is as simple as calling device_link_add(),
which results in a dmesg entry like this:
pci 0000:01:00.1: Linked as a consumer to 0000:01:00.0
The code for the GPU-governed audio power management can thus be removed
(except where it's still needed for legacy manual power control).
The device link is added in a PCI quirk rather than in hda_intel.c.
It is therefore legal for the GPU to runtime suspend to D3cold even if
the HDA controller is not bound to a driver or if CONFIG_SND_HDA_INTEL
is not enabled, for accesses to the HDA controller will cause the GPU to
wake up regardless if they're occurring outside of hda_intel.c (think
config space readout via sysfs).
Contrary to the previous implementation, the HDA controller's power
state is now self-governed, rather than GPU-governed, whereas the GPU's
power state is no longer fully self-governed. (The HDA controller needs
to runtime suspend before the GPU can.)
It is thus crucial that runtime PM is always activated on the HDA
controller even if CONFIG_SND_HDA_POWER_SAVE_DEFAULT is set to 0 (which
is the default), lest the GPU stays awake. This is achieved by setting
the auto_runtime_pm flag on every codec and the AZX_DCAPS_PM_RUNTIME
flag on the HDA controller.
A side effect is that power consumption might be reduced if the GPU is
in use but the HDA controller is not, because the HDA controller is now
allowed to go to D3hot. Before, it was forced to stay in D0 as long as
the GPU was in use. (There is no reduction in power consumption on my
Nvidia GK107, but there might be on other chips.)
The code paths for legacy manual power control are adjusted such that
runtime PM is disabled during power off, thereby preventing the PM core
from resuming the HDA controller.
Note that the device link is not only added on vga_switcheroo capable
systems, but for *any* GPU with integrated HDA controller. The idea is
that the HDA controller streams audio via connectors located on the GPU,
so the GPU needs to be on for the HDA controller to do anything useful.
This commit implicitly fixes an unbalanced runtime PM ref upon unbind of
hda_intel.c: On ->probe, a runtime PM ref was previously released under
the condition "azx_has_pm_runtime(chip) || hda->use_vga_switcheroo", but
on ->remove a runtime PM ref was only acquired under the first of those
conditions. Thus, binding and unbinding the driver twice on a
vga_switcheroo capable system caused the runtime PM refcount to drop
below zero. The issue is resolved because the AZX_DCAPS_PM_RUNTIME flag
is now always set if use_vga_switcheroo is true.
For more information on device links please refer to:
https://www.kernel.org/doc/html/latest/driver-api/device_link.html
Documentation/driver-api/device_link.rst
Cc: Dave Airlie <airlied@redhat.com>
Cc: Ben Skeggs <bskeggs@redhat.com>
Cc: Alex Deucher <alexander.deucher@amd.com>
Cc: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
Acked-by: Bjorn Helgaas <bhelgaas@google.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Peter Wu <peter@lekensteyn.nl>
Tested-by: Kai Heng Feng <kai.heng.feng@canonical.com> # AMD PowerXpress
Tested-by: Mike Lothian <mike@fireburn.co.uk> # AMD PowerXpress
Tested-by: Denis Lisov <dennis.lissov@gmail.com> # Nvidia Optimus
Tested-by: Peter Wu <peter@lekensteyn.nl> # Nvidia Optimus
Tested-by: Lukas Wunner <lukas@wunner.de> # MacBook Pro
Signed-off-by: Lukas Wunner <lukas@wunner.de>
Link: https://patchwork.freedesktop.org/patch/msgid/51bd38360ff502a8c42b1ebf4405ee1d3f27118d.1520068884.git.lukas@wunner.de
Purpose of having pinctrl dapm is to dynamically put the pins in
low power state when they are not actively used by the audio and
saving power.
Without this each driver has to set the pinctrl states, either
during probe or dynamically depending on the callbacks received
from ASoC core.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move the definitions of constants used in the dt-bindings from
include/sound/rt5651.h to include/dt-bindings/sound/rt5651.h.
As dt-bindings headers may also be parsed by the dt-compiler, they cannot
use enums, only defines, so this commit also changes the code declaring
the constants to use defines.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The code for dereferencing device nodes in the 'codecs' array is moved
to a separate function so we can avoid open coding that in drivers.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a need to use DA7219 as DAI clock master for other codecs
within a system, which means that the DAI clocks are required to
remain, regardless of whether the codec is actually running
playback/capture. To be able to expose control of the DAI clocking
the common clock framework has been employed.
The current implementation adds a simple clock gate for enabling
and disabling the DAI clocks, with no rate control supported
(this is still handled through standard hw_params() functions as
before). If DT is enabled then the clock is added to the OF
providers list, otherwise a clkdev lookup is used.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
OVer-Current-Detection (OVCD) for the micbias current is used to detect if
an inserted jack is a headset or headphones (mic shorted to ground).
The threshold for at which current the OVCD triggers on the rt5651 is not
only controlled by setting the absolute current limit, but also by setting
a scale factor which applies to the limit. Testing has shown that we need
to set both (depending on the board).
This commit adds support for the sofar unused OVCD scale-factor register
and adds support for specifying non-default values for it through the
"realtek,over-current-scale-factor" device-property.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Configure the jack-detect source through a device-property which can be
set by code outside of the codec driver. Rather then putting platform
specific DMI quirks inside the generic codec driver.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are no in tree users of platform-data for the rt5651 codec driver,
so lets remove support for it.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow dmaengine client to optionally register a processing callback.
This callback is intended to apply processing
on samples in buffer copied from/to user space, before/after DMA transfer.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Audigy 2 CA0102 chip (but most likely others from the emu10k1 family,
too) has a problem that from time to time it likes to do few DMA reads a
bit beyond its normal allocation and gets very confused if these reads get
blocked by a IOMMU.
For the first (reserved) page this happens multiple times at every
playback, for various synth pages it happens randomly, rarely for PCM
playback buffers and the page table memory itself.
All these reads seem to follow a similar pattern, observed read offsets
beyond the allocation end were 0x00, 0x40, 0x80 and 0xc0 (PCI cache line
multiples), so it looks like the device tries to accesses up to 256 extra
bytes.
As a workaround let's widen these DMA allocations by an extra page if we
detect that the device is behind a non-passthrough IOMMU (the DMA memory
should be relatively plenty on IOMMU systems).
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The emu10k1-family chips need the first page (index 0) reserved in their
page tables for some reason (every emu10k1 driver I've checked does this
without much of an explanation).
Using the first page for normal samples results in a broken playback.
However, we already have a dummy page allocated - so called "silent page"
and, in fact, had always been setting it as the first page in the chip page
table because an initialization of every entry of the page table to point
to a silent page happens after and overwrites the reserved_page allocation.
So the only thing remaining to remove the reserved_page allocation is a
trivial change to the page allocation logic to ignore the first page entry
and start its allocations from the second entry (index 1).
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current sync_power_state is local to hda code, moving it
core so that other users apart from hda legacy can use it.
The helper function ensures the actual state reaches the target state.
Signed-off-by: Abhijeet Kumar <abhijeet.kumar@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now platform can be replaced to component, let's do it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now we can replace Codec to Component. Let's do it.
Note:
xxx_codec_xxx() -> xxx_component_xxx()
.idle_bias_off = 0 -> .idle_bias_on = 1
.ignore_pmdown_time = 0 -> .use_pmdown_time = 1
- -> .endianness = 1
- -> .non_legacy_dai_naming = 1
To keep compatibilty, this patch adds snd_soc_xxx_ac97_codec()
macro. These will be removed when all codec code was removed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
With the merge window having been delayed for another week here's
another batch of updates that came in during that week. There's a few
important fixes in here, mainly a fix for I/O on a number of devices
caused by some of the component rework and a fix for a potential issue
if more than one component in a link provides compressed operations.
The I/O fixes are particularly important as the problem causes a power
regression on a number of OMAP platforms.
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Merge tag 'asoc-v4.16-5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound
Pull more ASoC updates from Mark Brown:
"With the merge window having been delayed for another week here's
another batch of updates that came in during that week.
There's a few important fixes in here, mainly a fix for I/O on a
number of devices caused by some of the component rework and a fix for
a potential issue if more than one component in a link provides
compressed operations. The I/O fixes are particularly important as the
problem causes a power regression on a number of OMAP platforms"
* tag 'asoc-v4.16-5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound: (22 commits)
ASoC: stm32: add of dependency for stm32 drivers
ASoC: mt8173-rt5650: fix child-node lookup
ASoC: dapm: fix debugfs read using path->connected
ASoC: compress: Fixup error messages
ASoC: compress: Remove some extraneous blank lines
ASoC: compress: Correct handling of copy callback
ASoC: Intel: kbl: Enable mclk and ssp sclk early
ASoC: Intel: Skylake: Add extended I2S config blob support in Clock driver
ASoC: Intel: Skylake: Add ssp clock driver
ASoC: Fix twl4030 and 6040 regression by adding back read and write
ASoC: sun8i-codec: Add ADC support for a33
ASoC: rockchip: Use dummy_dai for rt5514 dsp dailink
ASoC: soc-pcm: rename .pmdown_time to .use_pmdown_time for Component
ASoC: ak4613: call dummy write for PW_MGMT1/3 when Playback
ASoC: soc-pcm: don't call flush_delayed_work() many times in soc_pcm_private_free()
ASoC: soc-core: snd_soc_rtdcom_lookup() cares component driver name
ASoC: sam9x5_wm8731: Drop 'ASoC' prefix from error messages
ASoC: sam9g20_wm8731: use dev_*() logging functions
ASoC: max98373 Changed SPDX header in C++ comments style
ASoC: dmic: Fix check of return value from read of 'num-channels'
...
Pull poll annotations from Al Viro:
"This introduces a __bitwise type for POLL### bitmap, and propagates
the annotations through the tree. Most of that stuff is as simple as
'make ->poll() instances return __poll_t and do the same to local
variables used to hold the future return value'.
Some of the obvious brainos found in process are fixed (e.g. POLLIN
misspelled as POLL_IN). At that point the amount of sparse warnings is
low and most of them are for genuine bugs - e.g. ->poll() instance
deciding to return -EINVAL instead of a bitmap. I hadn't touched those
in this series - it's large enough as it is.
Another problem it has caught was eventpoll() ABI mess; select.c and
eventpoll.c assumed that corresponding POLL### and EPOLL### were
equal. That's true for some, but not all of them - EPOLL### are
arch-independent, but POLL### are not.
The last commit in this series separates userland POLL### values from
the (now arch-independent) kernel-side ones, converting between them
in the few places where they are copied to/from userland. AFAICS, this
is the least disruptive fix preserving poll(2) ABI and making epoll()
work on all architectures.
As it is, it's simply broken on sparc - try to give it EPOLLWRNORM and
it will trigger only on what would've triggered EPOLLWRBAND on other
architectures. EPOLLWRBAND and EPOLLRDHUP, OTOH, are never triggered
at all on sparc. With this patch they should work consistently on all
architectures"
* 'misc.poll' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs: (37 commits)
make kernel-side POLL... arch-independent
eventpoll: no need to mask the result of epi_item_poll() again
eventpoll: constify struct epoll_event pointers
debugging printk in sg_poll() uses %x to print POLL... bitmap
annotate poll(2) guts
9p: untangle ->poll() mess
->si_band gets POLL... bitmap stored into a user-visible long field
ring_buffer_poll_wait() return value used as return value of ->poll()
the rest of drivers/*: annotate ->poll() instances
media: annotate ->poll() instances
fs: annotate ->poll() instances
ipc, kernel, mm: annotate ->poll() instances
net: annotate ->poll() instances
apparmor: annotate ->poll() instances
tomoyo: annotate ->poll() instances
sound: annotate ->poll() instances
acpi: annotate ->poll() instances
crypto: annotate ->poll() instances
block: annotate ->poll() instances
x86: annotate ->poll() instances
...
commit fbb16563c6 ("ASoC: snd_soc_component_driver has pmdown_time")
added new .pmdown_time which is for inverted version of current
.ignore_pmdown_time
But it is confusable name. Let's rename it to .use_pmdown_time
Reported-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Codec only has .read/.write callback.
Codec will be merged into Component in next generation ALSA SoC,
thus current Codec specific feature need to be merged into it.
This is glue patch for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of home grown snd_soc_acpi_find_name_from_hid() use
acpi_dev_get_first_match_name().
Tested-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove hard-codec [16] array size, replace with clearer description and
dependency on ACPI_ID_LEN
No functionality change
Suggested-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
0-day reports compilation issues with non-ACPI platforms.
In file included from sound/soc/soc-acpi.c:17:0:
>> include/sound/soc-acpi.h:36:46: error: 'ACPI_ID_LEN' undeclared here
(not in a function); did you mean 'ACPI_FILE'?
snd_soc_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN])
sound/soc/soc-acpi.c: At top level:
>> sound/soc/soc-acpi.c:174:16: error: expected declaration specifiers or
'...' before string constant
MODULE_LICENSE("GPL v2");
Add missing include files.
Fixes: 7feb2f786a ("ASoC: move ACPI common code out of Intel/sst tree")
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Replace snd_soc_acpi_check_hid() with the generic acpi_dev_present()
and remove the now unused snd_soc_acpi_check_hid function. This should
have no functional change.
Signed-off-by: Jeremy Cline <jeremy@jcline.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC has some duplicate parameter.
snd_soc_dai::symmetric_xxx are one of them.
commit f0fba2ad1b ("ASoC: multi-component - ASoC Multi-Component
Support") moved "symmetric_xxx" flags from snd_soc_dai to
snd_soc_dai_driver. No one is using snd_soc_dai::symmetric_xxx now.
Let's remove it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
"dai_list" and "num_dai" on snd_soc_component are related
parameter. Let's arrange these.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC has some duplicate parameter.
snd_soc_component::dai_drv is one of them.
Each DAI is keeping its driver as snd_soc_dai::driver,
and component has dai_list.
This means, we can reach to each DAI and its driver by using dai_link.
Thus, there is no need to keep DAI driver pointer on component.
Let's remove it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The GPD win and pocket devices both use the same codec setup and both
have too generic dmi strings making snd_soc_set_dmi_name() not work.
As these devices have only a single speaker we want a separate ucm
file for them, which requires a unique long_name, use the existing
GPD quirk handling to also provide a unique long_name.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch renames all the variable instances of hdac_device with hdev
to prepare the code base to remove the usage of hdac_ext_device
data structures done in the following patches. Existing code uses hdev
and hdac as variable names for hdac_device as well as hdac_ext_device,
which creates confusion.
Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This format is similar to existing SNDRV_PCM_FORMAT_{S,U}20_3 that keep
20-bit PCM samples in 3 bytes, however i.MX6 platform SSI FIFO does not
allow 3-byte accesses (including DMA) so a 4-byte (more conventional)
format is needed for it.
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now, we have snd_card_disconnect_sync() on ALSA framework.
snd_soc_disconnect_sync() is ASoC version of it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The most common place to find POLL... bitmaps: return values
of ->poll() and its subsystem counterparts.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
The previous fix for addressing the breakage in vmaster slave
initialization, commit a91d66129f ("ALSA: hda - Fix incorrect TLV
callback check introduced during set_fs() removal"), introduced a new
helper to process over each slave kctl. However, this helper passes
only the original kctl, not the virtual slave kctl. As a result,
HD-audio driver (which is the only user so far) couldn't initialize
the slave correctly because it's trying to update the value directly
with the original kctl, not with the mapped kctl.
This patch fixes the situation again by passing both the mapped slaved
and original slave kctls to the function. Luckily there is a single
caller as of now, so changing the call signature is no big matter.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=197959
Fixes: a91d66129f ("ALSA: hda - Fix incorrect TLV callback check introduced during set_fs() removal")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The biggest thing this release has been the conversion of the AC98 bus
to the driver model, that's been a long time coming so thanks to Robert
Jarzmik for his dedication there. Due to there being some AC97 MFD
there's a few fairly large changes in input and the MFD layer, mainly to
the wm97xx driver.
There's also some drivers/drm changes to support the new AMD Stoney
platform, these are shared with the DRM subsystem and should be being
merged via both.
Within the subsystem the overwhelming bulk of the changes is in the
Intel drivers which continue to need lots of cleanups and fixes, this
release they've also gained support for their open source firmware.
There's also some large changs in the core as Morimoto-san continues to
mirror operations into the component level in preparation for conversion
of drivers to that.
- The AC97 bus has finally caught up with the driver model thanks to
some dedicated and persistent work from Robert Jarzmik.
- Continued work from Morimoto-san on moving us towards being able to
use components for everything.
- Lots of cleanups for the Intel platform code, including support for
their open source audio firmware.
- Support for scaling MCLK with sample rate in simple-card.
- Support for AMD Stoney platform.
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Merge tag 'asoc-v4.15' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.15
The biggest thing this release has been the conversion of the AC98 bus
to the driver model, that's been a long time coming so thanks to Robert
Jarzmik for his dedication there. Due to there being some AC97 MFD
there's a few fairly large changes in input and the MFD layer, mainly to
the wm97xx driver.
There's also some drivers/drm changes to support the new AMD Stoney
platform, these are shared with the DRM subsystem and should be being
merged via both.
Within the subsystem the overwhelming bulk of the changes is in the
Intel drivers which continue to need lots of cleanups and fixes, this
release they've also gained support for their open source firmware.
There's also some large changs in the core as Morimoto-san continues to
mirror operations into the component level in preparation for conversion
of drivers to that.
- The AC97 bus has finally caught up with the driver model thanks to
some dedicated and persistent work from Robert Jarzmik.
- Continued work from Morimoto-san on moving us towards being able to
use components for everything.
- Lots of cleanups for the Intel platform code, including support for
their open source audio firmware.
- Support for scaling MCLK with sample rate in simple-card.
- Support for AMD Stoney platform.
snd_soc_dapm_kcontrol_codec() (= for Codec) will be removed soon.
This patch Component version of it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_cache_sync() (= for Codec) will be removed soon.
This patch Component version of it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_codec_xxx_bias_level() (= for Codec) will be removed soon.
This patch Component version of it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current codec drivers are using snd_soc_read(). It will be replaced
into snd_soc_component_read(), but these 2 are using different style.
For example, it will be
- val = snd_soc_read(xxx, reg);
+ ret = snd_soc_component_read(xxx, reg, &val);
+ if (ret < 0) {
+ ...
+ }
To more smooth replace, let's add snd_soc_component_read32
which is copied from snd_soc_read()
- val = snd_soc_read(xxx, reg);
+ val = snd_soc_component_read32(xxx, reg);
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the option for the DSP clock that can be calibrated by the other clock
source.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent fix for adding rwsem nesting annotation was using the given
"hop" argument as the lock subclass key. Although the idea itself
works, it may trigger a kernel warning like:
BUG: looking up invalid subclass: 8
....
since the lockdep has a smaller number of subclasses (8) than we
currently allow for the hops there (10).
The current definition is merely a sanity check for avoiding the too
deep delivery paths, and the 8 hops are already enough. So, as a
quick fix, just follow the max hops as same as the max lockdep
subclasses.
Fixes: 1f20f9ff57 ("ALSA: seq: Fix nested rwsem annotation for lockdep splat")
Reported-by: syzbot <syzkaller@googlegroups.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we allow unlimited number of timer instances, and it may
bring the system hogging way too much CPU when too many timer
instances are opened and processed concurrently. This may end up with
a soft-lockup report as triggered by syzkaller, especially when
hrtimer backend is deployed.
Since such insane number of instances aren't demanded by the normal
use case of ALSA sequencer and it merely opens a risk only for abuse,
this patch introduces the upper limit for the number of instances per
timer backend. As default, it's set to 1000, but for the fine-grained
timer like hrtimer, it's set to 100.
Reported-by: syzbot
Tested-by: Jérôme Glisse <jglisse@redhat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many source files in the tree are missing licensing information, which
makes it harder for compliance tools to determine the correct license.
By default all files without license information are under the default
license of the kernel, which is GPL version 2.
Update the files which contain no license information with the 'GPL-2.0'
SPDX license identifier. The SPDX identifier is a legally binding
shorthand, which can be used instead of the full boiler plate text.
This patch is based on work done by Thomas Gleixner and Kate Stewart and
Philippe Ombredanne.
How this work was done:
Patches were generated and checked against linux-4.14-rc6 for a subset of
the use cases:
- file had no licensing information it it.
- file was a */uapi/* one with no licensing information in it,
- file was a */uapi/* one with existing licensing information,
Further patches will be generated in subsequent months to fix up cases
where non-standard license headers were used, and references to license
had to be inferred by heuristics based on keywords.
The analysis to determine which SPDX License Identifier to be applied to
a file was done in a spreadsheet of side by side results from of the
output of two independent scanners (ScanCode & Windriver) producing SPDX
tag:value files created by Philippe Ombredanne. Philippe prepared the
base worksheet, and did an initial spot review of a few 1000 files.
The 4.13 kernel was the starting point of the analysis with 60,537 files
assessed. Kate Stewart did a file by file comparison of the scanner
results in the spreadsheet to determine which SPDX license identifier(s)
to be applied to the file. She confirmed any determination that was not
immediately clear with lawyers working with the Linux Foundation.
Criteria used to select files for SPDX license identifier tagging was:
- Files considered eligible had to be source code files.
- Make and config files were included as candidates if they contained >5
lines of source
- File already had some variant of a license header in it (even if <5
lines).
All documentation files were explicitly excluded.
The following heuristics were used to determine which SPDX license
identifiers to apply.
- when both scanners couldn't find any license traces, file was
considered to have no license information in it, and the top level
COPYING file license applied.
For non */uapi/* files that summary was:
SPDX license identifier # files
---------------------------------------------------|-------
GPL-2.0 11139
and resulted in the first patch in this series.
If that file was a */uapi/* path one, it was "GPL-2.0 WITH
Linux-syscall-note" otherwise it was "GPL-2.0". Results of that was:
SPDX license identifier # files
---------------------------------------------------|-------
GPL-2.0 WITH Linux-syscall-note 930
and resulted in the second patch in this series.
- if a file had some form of licensing information in it, and was one
of the */uapi/* ones, it was denoted with the Linux-syscall-note if
any GPL family license was found in the file or had no licensing in
it (per prior point). Results summary:
SPDX license identifier # files
---------------------------------------------------|------
GPL-2.0 WITH Linux-syscall-note 270
GPL-2.0+ WITH Linux-syscall-note 169
((GPL-2.0 WITH Linux-syscall-note) OR BSD-2-Clause) 21
((GPL-2.0 WITH Linux-syscall-note) OR BSD-3-Clause) 17
LGPL-2.1+ WITH Linux-syscall-note 15
GPL-1.0+ WITH Linux-syscall-note 14
((GPL-2.0+ WITH Linux-syscall-note) OR BSD-3-Clause) 5
LGPL-2.0+ WITH Linux-syscall-note 4
LGPL-2.1 WITH Linux-syscall-note 3
((GPL-2.0 WITH Linux-syscall-note) OR MIT) 3
((GPL-2.0 WITH Linux-syscall-note) AND MIT) 1
and that resulted in the third patch in this series.
- when the two scanners agreed on the detected license(s), that became
the concluded license(s).
- when there was disagreement between the two scanners (one detected a
license but the other didn't, or they both detected different
licenses) a manual inspection of the file occurred.
- In most cases a manual inspection of the information in the file
resulted in a clear resolution of the license that should apply (and
which scanner probably needed to revisit its heuristics).
- When it was not immediately clear, the license identifier was
confirmed with lawyers working with the Linux Foundation.
- If there was any question as to the appropriate license identifier,
the file was flagged for further research and to be revisited later
in time.
In total, over 70 hours of logged manual review was done on the
spreadsheet to determine the SPDX license identifiers to apply to the
source files by Kate, Philippe, Thomas and, in some cases, confirmation
by lawyers working with the Linux Foundation.
Kate also obtained a third independent scan of the 4.13 code base from
FOSSology, and compared selected files where the other two scanners
disagreed against that SPDX file, to see if there was new insights. The
Windriver scanner is based on an older version of FOSSology in part, so
they are related.
Thomas did random spot checks in about 500 files from the spreadsheets
for the uapi headers and agreed with SPDX license identifier in the
files he inspected. For the non-uapi files Thomas did random spot checks
in about 15000 files.
In initial set of patches against 4.14-rc6, 3 files were found to have
copy/paste license identifier errors, and have been fixed to reflect the
correct identifier.
Additionally Philippe spent 10 hours this week doing a detailed manual
inspection and review of the 12,461 patched files from the initial patch
version early this week with:
- a full scancode scan run, collecting the matched texts, detected
license ids and scores
- reviewing anything where there was a license detected (about 500+
files) to ensure that the applied SPDX license was correct
- reviewing anything where there was no detection but the patch license
was not GPL-2.0 WITH Linux-syscall-note to ensure that the applied
SPDX license was correct
This produced a worksheet with 20 files needing minor correction. This
worksheet was then exported into 3 different .csv files for the
different types of files to be modified.
These .csv files were then reviewed by Greg. Thomas wrote a script to
parse the csv files and add the proper SPDX tag to the file, in the
format that the file expected. This script was further refined by Greg
based on the output to detect more types of files automatically and to
distinguish between header and source .c files (which need different
comment types.) Finally Greg ran the script using the .csv files to
generate the patches.
Reviewed-by: Kate Stewart <kstewart@linuxfoundation.org>
Reviewed-by: Philippe Ombredanne <pombredanne@nexb.com>
Reviewed-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Enable jack detection for the RT5651 codec on the JD* pins.
Signed-off-by: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In preparation for unconditionally passing the struct timer_list pointer to
all timer callbacks, switch to using the new timer_setup() and from_timer()
to pass the timer pointer explicitly.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Codec will be replaced into Component, then Codec side
doesn't use legacy_dai_naming on snd_soc_register_dais().
This patch adds new non_legacy_dai_naming flag on Component driver
and use converted its value for snd_soc_register_dais().
When Codec is replaced into Component, Codec driver needs
to have non_legacy_dai_naming = 1 flags.
Existing CPU side of course doesn't have this flag, thus CPU calls
it as true.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Codec will be replaced into Component, then Codec side only
needs to call fixup_codec_formats() at this point.
This patch adds new endianness flag on Component driver
and call convert_endianness_formats() (= was fixup_codec_format())
if endianness was true.
When Codec is replaced into Component, Codec driver needs
to have endianness = 1 flags.
Existing CPU side of course doesn't have this flag, thus CPU doesn't
call it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_runtime_ignore_pmdown_time() tallys all Codec and
CPU's "ignore_pmdown_time". Now, CPU (= via compoent)
ignore_pmdown_time is fixed as "true". Codec's one is copied from Codec
driver. This means Codec side default is "false".
Current all Codec driver will be replaced into Component, thus, we can
use for_each_rtdcom() for this totalization. This patch adds new
"pmdown_time" on Component driver. Its inverted value will be used
for this "ignore" totalizaton.
Of course all existing Component driver doesn't have its settings now,
thus, all existing "pmdown_time" is "false". This means all
Components will ignore pmdown time. This is current CPU behavior.
To keep compatibility, snd_soc_runtime_ignore_pmdown_time() totalize
Component's inverted "pmdown_time" (= total will be true) and
Codec's "ignore_pmdown_time" (= depends on Codec driver settings).
Because It is using AND operation, its result is based on Codec driver
settings only.
This means this operation can keep compatibility and doesn't have
nonconformity.
When we replace Codec to Component, the driver which has
".ignore_pmdown_time = true" will be just removed,
and the driver which doesn't have it will have new
".pmdown_time = true".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Platform will be replaced into Component in the future.
snd_soc_platform_driver has snd_compr_ops, but snd_soc_component_driver
doesn't have. To prepare for replacing, this patch adds snd_compr_ops on
component driver.
platform will be replaced into component, and its code will be removed.
But during replacing, both platform and component process code exists.
To keep compatibility, to avoid platform NULL access and to avoid
platform/component duplicate operation during replacing process, this
patch has such code. Some of this code will be removed when platform was
removed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Platform will be replaced into Component in the future.
snd_soc_platform_driver has snd_pcm_ops, but snd_soc_component_driver
doesn't have it. To prepare for replacing, this patch adds snd_pcm_ops
on component driver.
platform will be replaced into component, and its code will be removed.
But during replacing, both platform and component process code exists.
To keep compatibility, to avoid platform NULL access and to avoid
platform/component duplicate operation during replacing process, this
patch has such code. Some of this code will be removed when platform was
removed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
First step of cleaning, move all tables to soc-acpi-intel-match module.
The tables remain in separate files per platform to keep them
manageable. Skylake+ platforms are still handled elsewhere since
there is no conflict with SOF for now, but this will have to be
handled at a later point.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To prepare for SOF integration, we need new fields in the machine table.
It is intended that the same table is used for both closed-source and
open-source firmware to avoid repeating ACPI-related information
multiple times
No functional change
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ACPI support is not specific to the Intel/SST driver. Move the enumeration
and matching code which is not hardware-dependent to sound/soc and rename
relevant sst_acpi_ structures and functions with snd_soc_acpi_ prefix
soc-acpi.h is protected by a #ifndef __LINUX_SND_SOC_ACPI_H for
consistency with all other SoC .h files:
grep -L __LINUX include/sound/soc* | wc -l
0
grep __LINUX include/sound/soc* | wc -l
14
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The refresh of HD-audio widget sysfs kobjects via
snd_hdac_refresh_widget_sysfs() is slightly racy.
The driver recreates the whole tree from scratch after deleting the
whole. When CONFIG_DEBUG_KOBJECT_RELEASE option is used, kobject
release doesn't happen immediately but delayed, while the re-creation
of the same named kobject happens soon after invoking kobject_put().
This may end up with the conflicts of duplicated kobjects, as found in
the bug report below.
In this patch, we take another approach to refresh the tree: instead
of recreating the whole tree, just add the new nodes and delete the
non-existing nodes. Since the refresh happens only once at
initialization, no longer race would happen.
Along with the code change, merge snd_hdac_refresh_widget_sysfs() with
the existing snd_hdac_refresh_widgets() with an additional bool flag
for simplifying the code.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=197307
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a helper function to remove duplicate code used in machine drivers
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit 99b5c5bb9a ("ALSA: hda - Remove the use of set_fs()")
converted the get_kctl_0dB_offset() call for killing set_fs() usage in
HD-audio codec code. The conversion assumed that the TLV callback
used in HD-audio code is only snd_hda_mixer_amp() and applies the TLV
calculation locally.
Although this assumption is correct, and all slave kctls are actually
with that callback, the current code is still utterly buggy; it
doesn't hit this condition and falls back to the next check. It's
because the function gets called after adding slave kctls to vmaster.
By assigning a slave kctl, the slave kctl object is faked inside
vmaster code, and the whole kctl ops are overridden. Thus the
callback op points to a different value from what we've assumed.
More badly, as reported by the KERNEXEC and UDEREF features of PaX,
the code flow turns into the unexpected pitfall. The next fallback
check is SNDRV_CTL_ELEM_ACCESS_TLV_READ access bit, and this always
hits for each kctl with TLV. Then it evaluates the callback function
pointer wrongly as if it were a TLV array. Although currently its
side-effect is fairly limited, this incorrect reference may lead to an
unpleasant result.
For addressing the regression, this patch introduces a new helper to
vmaster code, snd_ctl_apply_vmaster_slaves(). This works similarly
like the existing map_slaves() in hda_codec.c: it loops over the slave
list of the given master, and applies the given function to each
slave. Then the initializer function receives the right kctl object
and we can compare the correct pointer instead of the faked one.
Also, for catching the similar breakage in future, give an error
message when the unexpected TLV callback is found and bail out
immediately.
Fixes: 99b5c5bb9a ("ALSA: hda - Remove the use of set_fs()")
Reported-by: PaX Team <pageexec@freemail.hu>
Cc: <stable@vger.kernel.org> # v4.13
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case of user unbind ALSA driver during playing back / capturing,
each driver needs to stop and remove it correctly. One note here is
that we can't cancel from remove function in such case, because
unbind operation doesn't check return value from remove function.
So, we *must* stop and remove in this case.
For this purpose, we need to sync (= wait) until the all top-level
operations are canceled at remove function.
For example, snd_card_free() processes the disconnection procedure at
first, then waits for the completion. That's how the hot-unplug works
safely. It's implemented, at least, in the top-level driver removal.
Now for the lower level driver, we need a similar strategy. Notify to
the toplevel for hot-unplug (disconnect in ALSA), and sync with the
stop operation, then continue the rest of its own remove procedure.
This patch adds snd_card_disconnect_sync(), and driver can use it from
remove function.
Note: the "lower level" driver here refers to a middle layer driver
(e.g. ASoC components) that can be unbound freely during operation.
Most of legacy ALSA helper drivers don't have such a problem because
they can't be unbound.
Note#2: snd_card_disconnect_sync() merely calls snd_card_disconnect()
and syncs with closing all pending files. It takes only the files
opened by user-space into account, and doesn't care about object
refcounts. (The latter is handled by snd_card_free() completion call,
BTW.) Also, the function doesn't free resources by itself.
Tested-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA SoC platform/codec will be replaced to component soon.
But, some function exist in "platform" doesn't exist in "component".
Current soc-core has snd_soc_register_component(), but
doesn't have snd_soc_add_component() like snd_soc_add_platform().
This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The event handler in the virmidi sequencer code takes a read-lock for
the linked list traverse, while it's calling snd_seq_dump_var_event()
in the loop. The latter function may expand the user-space data
depending on the event type. It eventually invokes copy_from_user(),
which might be a potential dead-lock.
The sequencer core guarantees that the user-space data is passed only
with atomic=0 argument, but snd_virmidi_dev_receive_event() ignores it
and always takes read-lock(). For avoiding the problem above, this
patch introduces rwsem for non-atomic case, while keeping rwlock for
atomic case.
Also while we're at it: the superfluous irq flags is dropped in
snd_virmidi_input_open().
Reported-by: Jia-Ju Bai <baijiaju1990@163.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes, mostly with stable ones:
- X32 ABI fix for PCM;
likely not so many people suffer from it, but still better to fix
- Two minor kernel warning fixes on USB audio devices spotted by
syzkaller
- Regression fix of echoaudio due to its inconsistent dimension
- Fix for HBR support on Intel DP audio, on some recent chips
- USB-audio quirk for yet another Plantronics devices
- Fix for potential double-fetch in ASIHPI FIFO queue
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Merge tag 'sound-4.14-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes, mostly with stable ones:
- X32 ABI fix for PCM; likely not so many people suffer from it, but
still better to fix
- Two minor kernel warning fixes on USB audio devices spotted by
syzkaller
- Regression fix of echoaudio due to its inconsistent dimension
- Fix for HBR support on Intel DP audio, on some recent chips
- USB-audio quirk for yet another Plantronics devices
- Fix for potential double-fetch in ASIHPI FIFO queue"
* tag 'sound-4.14-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usx2y: Suppress kernel warning at page allocation failures
Revert "ALSA: echoaudio: purge contradictions between dimension matrix members and total number of members"
ALSA: usb-audio: Check out-of-bounds access by corrupted buffer descriptor
ALSA: pcm: Fix structure definition for X32 ABI
ALSA: usb-audio: Add sample rate quirk for Plantronics C310/C520-M
ALSA: hda - program ICT bits to support HBR audio
ALSA: asihpi: fix a potential double-fetch bug when copying puhm
ALSA: compress: Remove unused variable
ALSA SoC platform/codec will be replaced to component soon.
This means 1 device might have multiple components. But current
unregister component function only checks "dev" to find it.
This means, unexpected component might be unregistered by current
function.
But, it is no problem if driver registered only 1 component.
To prepare avoid this issue, this patch adds new component
lookup function. it finds component by "dev" and "driver name".
Here, the reason why it uses "driver name" is that "component name"
was created by fmt_single_name() and difficult to use it from driver.
Driver of course knows its "driver name", thus, using it is more easy.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Codec only has set_bias_level feature.
Codec will be merged into Component in next generation ALSA SoC,
thus current Codec specific feature need to be merged into it.
This is glue patch for it.
Codec driver has .idle_bias_off for dapm bias. But Component
driver doesn't have it, and dapm->idle_bias_off is set as "true".
To keep compatibility, this patch adds "idle_bias_on" instead of
".idle_bias_off" on Component driver.
dapm->idle_bias_off will be set by inverted idle_bias_on.
When we replace Codec to Component, the driver which has
".idle_bias_off = true" is just remove it,
and the driver which doesn't have it will have new
".idle_bias_on = true".
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Platform only has pcm_new/pcm_free feature,
but it should be supported on Component level. This patch adds it.
The v1 was added commit 99b04f4c40 ("ASoC: add Component level
pcm_new/pcm_free") but it called all "card" connected component's
pcm_new/free, it was wrong.
This patch calls "rtd" connected component.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 44c07365e9 ("ASoC: add Component level set_jack") added new
snd_soc_component_set_jack(), but it didn't add definition on soc.h.
This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On recent Intel platforms (Haswell, Broadwell, Skylake, ApolloLake,
KabyLake, ...), the IEC Coding Type (ICT) bitfield in the Digital
Converter Control #3 needs to be set explicitly for HDMI/DisplayPort
High Bit Rate (HBR) audio playback to work. This was not required in
earlier platforms when HBR was first introduced. The ICT bits are
defined in Section 7.3.3.9 of the HDaudio 1.0a specification.
Since the ICT bitfield was not specified for HDAudio 1.0 devices
(before 2009), we only program it on machines more recent than
Haswell.
We tested that this fix is not needed on Baytrail-I (MinnowBoard
Turbot) and believe by extension it also does not apply to Braswell.
[ Moved AC_VERB_SET_DIGI_CONVERT_3 definition to the right place
by tiwai ]
Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=98797
Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support the function of impedance sensing. It could be set the matrix row
number of the impedance sensing table and the related parameters in the
DTS.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have touched quite a lot of files but with fewer changes at this
cycle; as you can see, most of changes are trivial fixes, especially
constification patches. Among the massive attacks by constification
gangs, we had a few core changes (mostly for ASoC core), as well the
fixes and the updates by major vendors. Some highlights are below:
ALSA core:
- Fix possible races in control API user-TLV codes
- Small cleanup of PCM core
ASoC:
- Continued work for componentization; still half-baked, but we're
certainly progressing
- Use of devres for jack detection GPIOs, rather as a cleanup
- Jack detection support for Qualcomm MSM8916
- Support for Allwinner H3, Cirrus Logic CS43130, Intel Kabylake
systems with RT5663, Realtek RT274, TI TLV320AIC32x6 and Wolfson
WM8523
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Merge tag 'sound-4.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"We have touched quite a lot of files but with fewer changes at this
cycle; as you can see, most of changes are trivial fixes, especially
constification patches.
Among the massive attacks by constification gangs, we had a few core
changes (mostly for ASoC core), as well the fixes and the updates by
major vendors.
Some highlights:
ALSA core:
- Fix possible races in control API user-TLV codes
- Small cleanup of PCM core
ASoC:
- Continued work for componentization; still half-baked, but we're
certainly progressing
- Use of devres for jack detection GPIOs, rather as a cleanup
- Jack detection support for Qualcomm MSM8916
- Support for Allwinner H3, Cirrus Logic CS43130, Intel Kabylake
systems with RT5663, Realtek RT274, TI TLV320AIC32x6 and Wolfson
WM8523"
* tag 'sound-4.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (512 commits)
ALSA: hda/ca0132 - Fix memory leak at error path
ALSA: hda: Fix forget to free resource in error handling code path in hda_codec_driver_probe
ASoC: cs43130: Fix unused compiler warnings for PM runtime
ASoC: cs43130: Fix possible Oops with invalid dev_id
ASoC: cs43130: fix spelling mistake: "irq_occurrance" -> "irq_occurrence"
ALSA: atmel: Remove leftovers of AVR32 removal
ALSA: atmel: convert AC97c driver to GPIO descriptor API
ALSA: hda/realtek - Enable jack detection function for Intel ALC700
ALSA: hda: Fix regression of hdmi eld control created based on invalid pcm
ASoC: Intel: Skylake: Add IPC to configure the copier secondary pins
ASoC: add missing compile rule for max98371
ASoC: add missing compile rule for sirf-audio-codec
ASoC: add missing compile rule for max98371
ASoC: cs43130: Add devicetree bindings for CS43130
ASoC: cs43130: Add support for CS43130 codec
ASoC: make clock direction configurable in asoc-simple
ALSA: ctxfi: Remove null check before kfree
ASoC: max98927: Changed device property read function
ASoC: max98927: Modified DAPM widget and map to enable/disable VI sense path
ASoC: max98927: Added PM suspend and resume function
...
All pxa library functions don't use the input parameters for nothing but
slot number. This simplifies their prototypes, and makes them usable by
both the legacy ac97 bus and the new ac97 bus.
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
AC97 is a bus for sound usage. It enables for a AC97 AC-Link to link one
controller to 0 to 4 AC97 codecs.
The goal of this new implementation is to implement a device/driver
model for AC97, with an automatic scan of the bus and automatic
discovery of AC97 codec devices.
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Split out from the ac97_codec.h the ac97 generic registers, which can be
used by a codec, typically a generic ac97 codec, and by the ac97 bus, to
scan an ac97 AC-Link.
This split encompasses all the AC97 standard registers, but not the
codec specific ones.
In order to have a clean split between former ac97 bus implementation
and the new coming one in sound/ac97, it is safer to not include any
former ac97 includes, excepting in sound/ac97/compat.c.
Amongst the thing to isolate :
- don't have the struct snd_ac97 in sound/ac97/* (except compat.c) to not
be "fooled" by a definition which would come with ac97_codec.h by
"chance".
- don't have to have snd_a97_*() functions, as they rely on struct
snd_ac97.
- don't want the struct snd_ac97_bus_ops, there is a new one
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is quite a large release by volume of patches and diff, a lot of
that is mechanical cleanup patches but it's great to also see a range of
vendors actively working on adding new features and fixing issues in
their drivers. Intel and Realtek have been especially active here.
- Continued work towards moving everything to the component model from
Morimoto-san.
- Use of devres for jack detection GPIOs, eliminating some potential
resource leaks.
- Jack detection support for Qualcomm MSM8916.
- Support for Allwinner H3, Cirrus Logic CS43130, Intel Kabylake
systems with RT5663, Realtek RT274, TI TLV320AIC32x6 and Wolfson
WM8523.
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Merge tag 'asoc-v4.14' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.14
This is quite a large release by volume of patches and diff, a lot of
that is mechanical cleanup patches but it's great to also see a range of
vendors actively working on adding new features and fixing issues in
their drivers. Intel and Realtek have been especially active here.
- Continued work towards moving everything to the component model from
Morimoto-san.
- Use of devres for jack detection GPIOs, eliminating some potential
resource leaks.
- Jack detection support for Qualcomm MSM8916.
- Support for Allwinner H3, Cirrus Logic CS43130, Intel Kabylake
systems with RT5663, Realtek RT274, TI TLV320AIC32x6 and Wolfson
WM8523.
The ALSA related include header files are left overs after the commit
020c5260c2 ("ALSA: atmel: Remove AVR32 bits from the driver")
Fixes: 020c5260c2 ("ALSA: atmel: Remove AVR32 bits from the driver")
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some CPU drivers (e. g. davinci-mcasp) may require the system clock to
be configured as OUT, while there's no good way currently to set
SND_SOC_CLK_OUT in simple-soc driver if the clock is fixed-rate.
This patch makes asoc_simple_card_init_dai() initialize clock to
SND_SOCK_CLK_OUT if explicitly stated in the relevant dts file. This
change is transparent and doesn't change the default behavior.
Signed-off-by: Vitaly Wool <vitaly.wool@konsulko.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Codec only has set_jack feature.
Codec will be merged into Component in next generation ALSA SoC,
thus current Codec specific feature need to be merged into it.
This is glue patch for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Codec only has set_pll feature.
Codec will be merged into Component in next generation ALSA SoC,
thus current Codec specific feature need to be merged into it.
This is glue patch for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Codec only has set_sysclk feature.
Codec will be merged into Component in next generation ALSA SoC,
thus current Codec specific feature need to be merged into it.
This is glue patch for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently we're taking power_lock at each card component for assuring
the power-up sequence, but it doesn't help anything in the
implementation at the moment: it just serializes unnecessarily the
callers, but it doesn't protect about the power state change itself.
It used to have some usefulness in the early days where we managed the
PM manually. But now the suspend/resume core procedure is beyond our
hands, and power_lock lost its meaning.
This patch drops the power_lock from allover the places.
There shouldn't be any issues by this change, as it's no helper
regarding the power state change. Rather we'll get better performance
by removing the serialization; which is the only slight concern of any
behavior change, but it can't be a showstopper, after all.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_component and snd_soc_component_driver both have
dapm_routes/num_dapm_routes, but these are duplicated.
Let's remove duplicated definition.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_component and snd_soc_component_driver both have
dapm_widgets/num_dapm_widgets, but these are duplicated.
Let's remove duplicated definition.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_component and snd_soc_component_driver both have
controls/num_controls, but these are duplicated.
Let's remove duplicated definition.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We're keeping in a good shape, this batch contains just a few small
fixes (a regression fix for ASoC rt5677 codec, NULL dereference and
error-path fixes in firewire, and a corner-case ioctl error fix for
user TLV), as well as usual quirks for USB-audio and HD-audio.
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Merge tag 'sound-4.13-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into asoc-rt5677
sound fixes for 4.13-rc7
We're keeping in a good shape, this batch contains just a few small
fixes (a regression fix for ASoC rt5677 codec, NULL dereference and
error-path fixes in firewire, and a corner-case ioctl error fix for
user TLV), as well as usual quirks for USB-audio and HD-audio.
The HDMI audio driver only needs to know which generation of HDMI
transmitter it deals with, not the detailed SoC model. Pass the version
number as an integer to prepare for removal of the OMAP SoC version from
the omapdrm driver.
Signed-off-by: Laurent Pinchart <laurent.pinchart@ideasonboard.com>
Acked-by: Mark Brown <broonie@kernel.org>
Acked-by: Bartlomiej Zolnierkiewicz <b.zolnierkie@samsung.com>
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ti.com>
Basically, current ALSA SoC framework is based on CPU/Codec/Platform,
but its operation doesn't have consistent.
Thus, source code was unreadable, and difficult to understand.
This patch connects each component (= CPU/Codec/Platform) to rtd by
using snd_soc_rtdcom_add(), and convert uneven operations to consistent
operation.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_pcm_runtime has platform / codec pointers, and we could
use these specific pointer. But these will be replaced to more generic
"component" soon, and will need more generic method to get each
connected component pointer from rtd.
This patch adds new snd_soc_rtdcom_xxx() to connect/disconnect
component to rtd. It means same as previous "platform" / "codec"
pointer style, but more generic.
We can find necessary component pointer from rtd by using component
driver name on snd_soc_rtdcom_lookup().
Here, the reason why it uses "driver name" is that "component name"
was created by fmt_single_name() and difficult to use it from driver.
Driver of course knows its "driver name", thus, using it is more easy.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need to have the *debugfs_reg dentry member as part of
the snd_soc_codec structure as its only usage is inside
soc_init_codec_debugfs().
Use a local dentry variable instead.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
At a previous commit, concurrent requests for TLV data are maintained
exclusively between read requests and write/command requests. TLV
callback handlers in each driver has no risk from concurrent access for
reference/change.
In current implementation, 'struct snd_card' has a mutex to control
concurrent accesses to user-defined element sets. This commit obsoletes it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch seprates the DC offset between headphone and headset.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
'debugfs_dpcm_state' member from structure snd_soc_pcm_runtime
is never used at all, so it is safe to remove it.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_codec .cache_bypass related operation code has been removed.
Let's remove remaining code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no user of legacy platform data.
Remove separate header and hide its content inside module sources.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the ability to configure the MFP1->MFP5 registers
as GPIOs. In addition adding ALSA controls to get and set
the GPIO state.
Per the data sheet each MFP can be configured as a GPIO
input only, output only or either an input or output.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 99b04f4c40 ("ASoC: add Component level
pcm_new/pcm_free"), which started calling the pcm_new callback for every
component in a *card* when creating a new pcm, something which does not
seem to make any sense.
This specifically led to memory leaks in systems with more than one
platform component and where DMA memory is allocated in the
platform-driver callback. For example, when both mcasp devices are being
used on an am335x board, DMA memory would be allocated twice for every
DAI link during probe.
When CONFIG_SND_VERBOSE_PROCFS was set this fortunately also led to
warnings such as:
WARNING: CPU: 0 PID: 565 at ../fs/proc/generic.c:346 proc_register+0x110/0x154
proc_dir_entry 'sub0/prealloc' already registered
Since there seems to be no users of the new component callbacks, and the
current implementation introduced a regression, let's revert the
offending commit for now.
Fixes: 99b04f4c40 ("ASoC: add Component level pcm_new/pcm_free")
Signed-off-by: Johan Hovold <johan@kernel.org>
Reviewed-by: Linus Walleij <linus.walleij@linaro.org>
Tested-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable <stable@vger.kernel.org> # 4.10
The patch adds the manual offset field in the devicetree to compensate the
DC offset that will be different between the PCB layout. It only can be
measured by the real production.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The big news with this release is the of-graph card, this provides a
replacement for simple-card that is much more flexibile and scalable,
allowing many more systems to use a generic sound card than was possible
before:
- The of-graph card, finally merged after a long and dedicated effort
by Morimoto-san.
- New widget types intended mainly for use with DSPs.
- New drivers for Allwinner V3s SoCs, Ensonic ES8316, several classes
of x86 machine, Rockchip PDM controllers, STM32 I2S and S/PDIF
controllers and ZTE AUD96P22 CODECs.
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Merge tag 'asoc-v4.13' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.13
The big news with this release is the of-graph card, this provides a
replacement for simple-card that is much more flexibile and scalable,
allowing many more systems to use a generic sound card than was possible
before:
- The of-graph card, finally merged after a long and dedicated effort
by Morimoto-san.
- New widget types intended mainly for use with DSPs.
- New drivers for Allwinner V3s SoCs, Ensonic ES8316, several classes
of x86 machine, Rockchip PDM controllers, STM32 I2S and S/PDIF
controllers and ZTE AUD96P22 CODECs.
Add some DAPM widget types to better support the construction of DAPM
graphs within DSPs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Added quirk DW_I2S_QUIRK_16BIT_IDX_OVERRIDE to Designware
driver. This quirk will set idx value to 1.
By setting this quirk, it will override supported format
as 16 bit resolution and bus width as 2 Bytes.
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The flag will invert jd1_1 status. Which will be used if the jack
connector is normal closed.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The jd_invert flag is actually used for level triggered IRQ. Rename
it to let code more readable.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Tested-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have an ioctl to inform the PCM protocol version the running kernel
supports, but there is no way to know which protocol version the
user-space can understand. This lack of information caused headaches
in the past when we tried to extend the ABI. For example, because we
couldn't guarantee the validity of the reserved bytes, we had to
introduce a new ioctl SNDRV_PCM_IOCTL_STATUS_EXT for assigning a few
new fields in the formerly reserved bits. If we could know that it's
a new alsa-lib, we could assume the availability of the new fields,
thus we could have reused the existing SNDRV_PCM_IOCTL_STATUS.
In order to improve the ABI extensibility, this patch adds a new ioctl
for user-space to inform its supporting protocol version to the
kernel. By reporting the supported protocol from user-space, the
kernel can judge which feature should be provided and which not.
With the addition of the new ioctl, the PCM protocol version is bumped
to 2.0.14, too. User-space checks the kernel protocol version via
SNDRV_PCM_INFO_PVERSION, then it sets the supported version back via
SNDRV_PCM_INFO_USER_PVERSION.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current simple card drivers are parsing widgets on each own driver
(only simple-card at this point, but will be supported on all drivers)
Encapsulation is one of simple card util's purpose.
Let's add asoc_simple_card_of_parse_widgets for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple card drivers are parsing routing on each own driver.
Encapsulation is one of simple card util's purpose.
Let's add asoc_simple_card_of_parse_routing for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple/audio scu card drivers are supporting same
convert-rate/convert-channels on DT, but doesn't use same function
for it.
Encapsulation is one of simple card util's purpose.
Let's add asoc_simple_card_parse_convert/asoc_simple_card_convert_fixup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SNDRV_PCM_IOCTL1_GSTATE was firstly introduced in v0.9.0, however never
be used and the purpose is missing.
This commit removes the long-abandoned command, bye.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drivers can implement 'struct snd_pcm_ops.ioctl' to handle some requests
from ALSA PCM core. These requests are internal purpose in kernel land.
Usually common set of operations are used for it.
SNDRV_PCM_IOCTL1_INFO is one of the requests. According to code comment,
it has been obsoleted in the old days.
We can see old releases in ftp.alsa-project.org. The command was firstly
introduced in v0.5.0 release as SND_PCM_IOCTL1_INFO, to allow drivers to
fill data of 'struct snd_pcm_channel_info' type. In v0.9.0 release,
this was obsoleted by the other commands for ioctl(2) such as
SNDRV_PCM_IOCTL_CHANNEL_INFO.
This commit removes the long-abandoned command, bye.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current simple card drivers are using asoc_simple_dai's tx_slot_mask,
rx_slot_mask, slots, slot_width directly to parse TDM.
Encapsulation is one of simple card util's purpose.
Let's add asoc_simple_card_of_parse_tdm for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils sets asoc_simple_dai::clk via
asoc_simple_card_parse_clk().
Current simple card drivers are using it directly for
clk_enable/disable.
Encapsulation is one of simple card util's purpose.
Let's encapsulate it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently OSS sequencer emulation is tied with ALSA sequencer core,
both are built in the same level; i.e. when CONFIG_SND_SEQUENCER=y,
the OSS sequencer emulation is also always built-in, even though the
functionality can be built as an individual module.
This patch changes the rule and allows users to build snd-seq-oss
module while others are built-in. Essentially, it's just a few simple
changes in Kconfig and Makefile. Some driver codes like opl3 need to
convert from the simple ifdef to IS_ENABLED(). But that's all.
You might wonder how about the dependency: right, it can be messy, but
it still works. Since we rewrote the sequencer binding with the
standard bus, the driver can be bound at any time on demand. So, the
synthesizer driver module can be loaded individually from the OSS
emulation core before/after it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current topology only allows for widget configuration before the widget
is registered. This patch also allows further configuration and usage
after registration is complete.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should be declaring snd_kcontrol_new and soc_dai_link as both are
used within this header so need to be declared.
[Reworded commit message to indicate this wasn't an immediate build
failure -- broonie]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Let's share same debug message for sysclk
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now all materials are ready, let's allow the direct in-kernel
read/write, i.e. a kernel-space buffer is passed for read or write,
instead of the normal user-space buffer. This feature is used by OSS
layer and UAC1 driver, for example.
The __snd_pcm_lib_xfer() takes in_kernel argument that indicates the
in-kernel buffer copy. When this flag is set, another transfer code
is used. It's either via copy_kernel PCM ops or the normal memcpy(),
depending on the driver setup.
As external API, snd_pcm_kernel_read(), *_write() and other variants
are provided.
That's all. This support is really simple because of the code
refactoring until now.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both __snd_pcm_lib_read() and __snd_pcm_write() functions have almost
the same code to loop over samples. For simplification, this patch
unifies both as the single helper, __snd_pcm_lib_xfer().
Other than that, there should be no functional change by this patch.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make snd_pcm_lib_read() and *_write() static inline functions that
call the common helper functions directly. This reduces a slight
amount of codes, and at the same time, it's a preparation for the
further cleanups / fixes.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that all users of old copy and silence ops have been converted to
the new PCM ops, the old stuff can be retired and go away.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For supporting the explicit in-kernel copy of PCM buffer data, and
also for further code refactoring, three new PCM ops, copy_user,
copy_kernel and fill_silence, are introduced. The old copy and
silence ops will be deprecated and removed later once when all callers
are converted.
The copy_kernel ops is the new one, and it's supposed to transfer the
PCM data from the given kernel buffer to the hardware ring-buffer (or
vice-versa depending on the stream direction), while the copy_user ops
is equivalent with the former copy ops, to transfer the data from the
user-space buffer.
The major difference of the new copy_* and fill_silence ops from the
previous ops is that the new ops take bytes instead of frames for size
and position arguments. It has two merits: first, it allows the
callback implementation often simpler (just call directly memcpy() &
co), and second, it may unify the implementations of both interleaved
and non-interleaved cases, as we'll see in the later patch.
As of this stage, copy_kernel ops isn't referred yet, but only
copy_user is used.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Several files are used to construct PCM core module, a.k.a snd-pcm.
Although available APIs are described in 'include/sound/pcm.h', some of
them are not exported as symbols in kernel space. Such APIs are just for
module local usage.
This commit adds module local header file and move some function prototypes
into it so that scopes of them are controlled properly and developers
get no confusion from unavailable symbols.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The indirect-PCM helper codes have an implicit assumption that the
appl_ptr always increases. But the PCM core may deal with the
decrement of appl_ptr via rewind ioctls, and it may screw up the
buffer pointer management.
This patch adds the negative appl_ptr diff in transfer functions and
let returning an error instead of always accepting the appl_ptr
updates. The callers are usually PCM ack callbacks, and they pass the
error to the upper layer accordingly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA SoC needs to know connected DAI ID for probing.
It is not a big problem if device/driver was only for sound,
but getting DAI ID will be difficult if device includes both
Video/Sound, like HDMI.
To solve this issue, this patch adds new .get_dai_id callback
on hdmi_codec_ops
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA SoC needs to know connected DAI ID for detecting.
It is not a big problem if device/driver was only for sound,
but getting DAI ID will be difficult if device includes both
Video/Sound, like HDMI.
To solve this issue, this patch adds new snd_soc_get_dai_id() and
its related .of_xlate_dai_id callback on component driver.
In below case, we can handle Sound port (= port@2) as ID = 0
if .of_xlate_dai_id has its support.
hdmi {
port@0 { /* VIDEO */ };
port@1 { /* VIDEO */ };
port@2 { /* SOUND */ };
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the Boost Inductor parameters based off the size of the inductor
on the HW setup
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card already has asoc_simple_card_parse_dai(),
but graph base parsing needs graph specific version of it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The markup inside the #if 0 comment actually refers to a
kernel-doc markup. As we're getting rid of DocBook update it.
Signed-off-by: Mauro Carvalho Chehab <mchehab@s-opensource.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM notifier code is used only by OSS emulation layer, so we can
build it conditionally for reducing the size.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ak4113, ak4114 and ak4117 i2c drivers have some tricky codes to
access the struct fields in the callback. This can be simplified by
replacing the struct fields with the array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A quiet release for the core, but lots of new drivers this time around:
- A new, generalized, API for hooking up jacks which makes it easier to
write generic machine drivers for simple cases.
- Continuing fixes for issues with the x86 CPU drivers.
- New drivers for Cirrus CS35L35, DIO DIO2125, Everest ES7132,
HiSilicon hi6210, Maxim MAX98927, MT2701 systems with WM8960, Nuvoton
NAU8824, Odroid systems, ST STM32 SAI controllers and x86 systems with
DA7213
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Merge tag 'asoc-v4.12' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.12
A quiet release for the core, but lots of new drivers this time around:
- A new, generalized, API for hooking up jacks which makes it easier to
write generic machine drivers for simple cases.
- Continuing fixes for issues with the x86 CPU drivers.
- New drivers for Cirrus CS35L35, DIO DIO2125, Everest ES7132,
HiSilicon hi6210, Maxim MAX98927, MT2701 systems with WM8960, Nuvoton
NAU8824, Odroid systems, ST STM32 SAI controllers and x86 systems with
DA7213
For systems without DMI, it makes no sense to have the code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On the chip the IMON signal is a full 24-bits however normally only
some of the bits will be sent over the bus. The chip provides a field
to select which bits of the IMON will be sent back, this is the only
feedback signal that has this feature.
Add an additional entry to the cirrus,imon device tree property to
allow the IMON scale parameter to be passed.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are many codecs with the capability of jack detection. Usually,
we create a jack on machine driver but there is no common function for
machine driver to deliver the jack pointer to codec driver.
snd_soc_codec_set_jack can be used for delivering the jack pointer to
codec driver and enable the jack detection function. To make it work,
codec driver need to define a callback function to receive the jack
pointer and do all necessary procedure for enabling jack detection.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch refines the definition of AZX_MLCTL_SPA and AZX_MLCTL_CPA
and add more definitions of ML registers
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When using an external boost supply the PDN_DONE bit is not set, update
the handling in this case to use to use an appropriate fixed delay.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The macros _snd_hdac_chip_read() and *_write() expand to different
types (b,w,l) per their argument. They were thought to be used only
internally for other snd_hdac_chip_*() macros, but in some situations
we need to call these directly, and they are way too ugly.
Instead of saving a few lines, we just write these macros explicitly
with the types, so that they can be used in a saner way.
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For GPIO-backed pins that are not configured as wakeup sources, we may
miss change in their state that happens while system is suspended. Let's
use PM notifier to refresh their state upon resume.
Signed-off-by: Dmitry Torokhov <dtor@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 1a653aa447 ("ASoC: core: replace aux_comp_list to ...")
tried to replace aux_comp_list to component_dev_list,
but it failed because of binding timing. Thus, Sylwester fixuped it by
commit d2e3a1358c ("ASoC: Fix binding and probing of auxiliary...").
One of main purpose of commit 1a653aa447 ("ASoC: core: replace...")
was remove replaceable list (= list_aux) from snd_soc_component by using
new "auxiliary" flags (but it failed).
Because of this background, current code has reborned card_aux_list
(= same as original list_aux), and almost pointless "auxiliary" flags.
Let's remove pointless "auxiliary" flags by this patch
This means, it is same as revert both
commit 1a653aa447 ("ASoC: core: replace aux_comp_list to ...") and
commit d2e3a1358c ("ASoC: Fix binding and probing of auxiliary...").
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for setting how the I2S pins are driven in unused slots,
currently the chip will just use the default of drive 0, however this
causes issues when multiple devices are attached to the same bus.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for the Cirrus Logic
CS35L35 9V Boosted Amplifier
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In the following patches we are going to remove various headers
from sched.h and other headers that sched.h includes.
To make those patches build cleanly prepare the scene by adding
dependencies to various files that learned to rely on those
to-be-removed dependencies.
These changes all make sense standalone: they add a header for
a data type that a particular .c or .h file is using.
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Cc: Mike Galbraith <efault@gmx.de>
Cc: Peter Zijlstra <peterz@infradead.org>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Ingo Molnar <mingo@kernel.org>
These are the current source files that should not have
executable attributes set.
[ Normally this would be sent through Andrew Morton's tree
but his quilt tools don't like permission only patches. ]
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Another release that's mainly focused on drivers rather than core
changes, highlights include:
- A huge batch of updates to the Intel drivers, mainly around
DisplayPort and HDMI with some additional board support too.
- Channel mapping support for HDMI.
- Support for AllWinner A31 and A33, Everest Semiconductor ES8328,
Nuvoton NAU8540.
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Merge tag 'asoc-v4.11' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.11
Another release that's mainly focused on drivers rather than core
changes, highlights include:
- A huge batch of updates to the Intel drivers, mainly around
DisplayPort and HDMI with some additional board support too.
- Channel mapping support for HDMI.
- Support for AllWinner A31 and A33, Everest Semiconductor ES8328,
Nuvoton NAU8540.
... so that the driver can avoid ifdef's for the dead PM callbacks.
The compiler should optimize them out in anyway.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit 8f5ebb1bee
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_card_name")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit 8f5ebb1bee.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit b6defcca0a
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_prefix")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit b6defcca0a.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit 1ef5bcd57b
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_simple_widgets")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit 1ef5bcd57b.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Because prototype of OF-graph sound card support didn't have Sound Card
node, commit 7364c8dc25
("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_routing")
adjusted to it on each functions.
But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound
card has node. Thus, this commit became no longer needed.
This reverts commit 7364c8dc25.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit c6644119a3 and
restores the ability to specify DMA channel names per DAI dma_data.
Unfortunately the functionality removed in the patch being reverted
cannot be entirely replaced by specifying DMA channel names in struct
snd_dmaengine_pcm_config as that does not cover devices with more than
2 DMA channels.
Together with patch "ASoC: Revert "samsung: Remove unneeded
initialization of chan_name"" this fixes broken sound on the s3c24xx
SoC platforms.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card-utils is getting clk by of_clk_get(), but didn't call
clk_free(). Now we can use devm_get_clk_from_child() for this purpose.
Let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During probe, DAIs can need to perform some actions that requests
the knowledge of the pcm runtime handle.
The callback is called during DAIs linking, after PCM device creation.
For instance this can be used to add relationship between a DAI pcm
control and the pcm device.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel DSP platform drivers are used by many different devices but are
difficult for userspace to differentiate. This patch adds an API to allow
the DMI name to be used in the sound card long name, thereby helping
userspace load the correct UCM configuration. Usually machine drivers
uses their own name as the sound card name (short name), and leave the
long name and driver name blank. This API will use the DMI info like
vendor, product and board to make up the card long name. If the machine
driver has already explicitly set the long name, this API will do nothing.
This patch also allows for further differentiation as many devices that
share the same DMI name i.e. Minnowboards, UP boards may be configured
with different codecs or firmwares. The API supports flavoring the DMI
name into the card longname to provide the extra differentiation required
for these devices.
For Use Case Manager (UCM) in the user space, changing card long name by
this API is backward compatible, since the card name does not change. For
a given sound card, even if there is no device-specific UCM configuration
file that uses the card long name, UCM will fall back to load the default
configuration file that uses the card name.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make snd_rawmidi_substream.ops to be a const pointer to be safer and
allow more optimization. The patches to constify each rawmidi ops
will follow.
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No one is using snd_soc_platform_trigger().
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No existing platform is using .bespoke_trigger.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No existing platform is using .delay.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently binding of auxiliary devices doesn't work as in
soc_bind_aux_dev() function a bound component is not being added
to any list and in soc_probe_aux_devices() we are trying to walk
the component_dev_list list to probe auxiliary components but
at that time this list doesn't contain any auxiliary components
since they are being added to the card only in soc_probe_component().
This patch adds a list to the card where are stored bound but not
probed auxiliary devices, so that all aux devices can be probed.
Fixes: 1a653aa447 "ASoC: core: replace aux_comp_list to component_dev_list"
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is a fix for Linux 4.10-rc1.
In C language specification, a bit-field is interpreted as a signed or
unsigned integer type consisting of the specified number of bits.
In GCC manual, the range of a signed bit field of N bits is from
-(2^N) / 2 to ((2^N) / 2) - 1
https://www.gnu.org/software/gnu-c-manual/gnu-c-manual.html#Bit-Fields
Therefore, when defined as 1 bit-field with signed type, variables can
represents -1 and 0.
The snd-soc-hdmi-codec module includes a structure which has signed type
members with bit-fields. Codes of this module assign 0 and 1 to the
members. This seems to result in implementation-dependent behaviours.
As of v4.10-rc1 merge window, outside of sound subsystem, this structure
is referred by below GPU modules.
- tda998x
- sti-drm
- mediatek-drm-hdmi
- msm
As long as I review their codes relevant to the structure, the structure
members are used just for condition statements and printk formats.
My proposal of change is a bit intrusive to the printk formats but this
may be acceptable.
Totally, it's reasonable to use unsigned type for the structure members.
This bug is detected by Sparse, static code analyzer with below warnings.
./include/sound/hdmi-codec.h:39:26: error: dubious one-bit signed bitfield
./include/sound/hdmi-codec.h:40:28: error: dubious one-bit signed bitfield
./include/sound/hdmi-codec.h:41:29: error: dubious one-bit signed bitfield
./include/sound/hdmi-codec.h:42:31: error: dubious one-bit signed bitfield
Fixes: 09184118a8 ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
CC: stable@vger.kernel.org
In current ALSA SoC, Platform only has pcm_new/pcm_free feature,
but it should be supported on Component level. This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
No dramatic changes are found in this development cycle, but as usual,
many commits are applied in a wide range of drivers.
Most of big changes are in ASoC, where a few bits of framework work
and quite a lot of cleanups and improvements to existing code have
been done. The rest are usual stuff, a few HD-audio and USB-audio
quirks and fixes, as well as the drop of kthread usages in the whole
subsystem.
Below are some highlights:
ASoC:
- Support for stereo DAPM controls
- Some initial work on the of-graph sound card
- regmap conversions of the remaining AC'97 drivers
- A new version of the topology ABI; this should be backward compatible
- Updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona,
Intel skylake, atom-sst
- New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and
Realtek RT5665
USB-audio:
- Yet another race fix at disconnection
- Tolerated packet size calculation for some Android devices
- Quirks for Axe-Fx II, QuickCam, TEAC 501/503
HD-audio:
- Improvement of Dell pin fixup mapping
- Quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic
Misc:
- Replace all kthread usages with simple works
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Merge tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"No dramatic changes are found in this development cycle, but as usual,
many commits are applied in a wide range of drivers.
Most of big changes are in ASoC, where a few bits of framework work
and quite a lot of cleanups and improvements to existing code have
been done. The rest are usual stuff, a few HD-audio and USB-audio
quirks and fixes, as well as the drop of kthread usages in the whole
subsystem.
Below are some highlights:
ASoC:
- support for stereo DAPM controls
- some initial work on the of-graph sound card
- regmap conversions of the remaining AC'97 drivers
- a new version of the topology ABI; this should be backward
compatible
- updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona,
Intel skylake, atom-sst
- new drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and
Realtek RT5665
USB-audio:
- yet another race fix at disconnection
- tolerated packet size calculation for some Android devices
- quirks for Axe-Fx II, QuickCam, TEAC 501/503
HD-audio:
- improvement of Dell pin fixup mapping
- quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic
Misc:
- replace all kthread usages with simple works"
* tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (296 commits)
ALSA: hiface: Fix M2Tech hiFace driver sampling rate change
ALSA: usb-audio: Eliminate noise at the start of DSD playback.
ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC
ASoC: wm_adsp: wm_adsp_buf_alloc should use kfree in error path
ASoC: topology: avoid uninitialized kcontrol_type
ALSA: usb-audio: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks
ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
ASoC: zte: spdif: correct ZX_SPDIF_CLK_RAT define
ASoC: zte: spdif and i2s drivers are not zx296702 specific
ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting
ASoC: rsnd: enable/disable ADG when suspend/resume timing
ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params
ALSA: cs46xx: add a new line
ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture
ASoC: nau8825: disable sinc filter for high THD of ADC
ALSA: usb-audio: more tolerant packetsize
ALSA: usb-audio: avoid setting of sample rate multiple times on bus
ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting
ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO
ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq
...
There's been a few bits of framework work this time around and quite a
lot of cleanups and improvements to existing code:
- Support for stereo DAPM controls from Chen-yu Tsai.
- Some initial work on the of-graph sound card from Morimoto-san, the
main bulk of this is currently in binding review.
- Lots of Renesas cleanups from Morimoto-san and sunxi work from
Chen-yu Tsai.
- regmap conversions of the remaining AC'97 drivers from Lars-Peter
Clausen.
- A new version of the topology ABI from Mengdong Lin.
- New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and Realtek
RT5665.
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Merge tag 'asoc-v4.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.10
There's been a few bits of framework work this time around and quite a
lot of cleanups and improvements to existing code:
- Support for stereo DAPM controls from Chen-yu Tsai.
- Some initial work on the of-graph sound card from Morimoto-san, the
main bulk of this is currently in binding review.
- Lots of Renesas cleanups from Morimoto-san and sunxi work from
Chen-yu Tsai.
- regmap conversions of the remaining AC'97 drivers from Lars-Peter
Clausen.
- A new version of the topology ABI from Mengdong Lin.
- New drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and Realtek
RT5665.
This patch will check the type of embedded controls for a widget, and
only free the TLV of volume mixers. Bytes controls don't have TLV.
Just free the private value which is used as struct soc_mixer_control
for volume mixers or soc_bytes_ext for bytes controls. No need to cast
to these types before freeing it.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple_card_utils was created as simple_card_core in 1st prototype,
and current code still have it. Let's tidyup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In current ALSA SoC, Codec only has suspend/resume feature,
but it should be supported on Component level. This patch adds it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now, Card has component_dev_list, we can replace aux_comp_list
to component_dev_list with new auxiliary flags
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current Card has Codec list (= codec_dev_list), but Codec will be
removed in the future. Because of this reason, this patch adds
new Component list in Card, and replace Codec list.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It's often the case that a codec driver will need to control its
own pins. However, if a name_prefix has been applied to this codec it
must be included in the name passed to any of the snd_soc_dapm_x_pin()
functions.
The behaviour of the existing pin control functions is reasonable, since
you may want to search for a fully-specified name within the scope of an
entire card. This means that we can't apply the prefix in these functions
because it will break card-scope searches.
Constructing a prefixed string "manually" in codec drivers leads to a lot
of repetition of the same code.
To make this tidier in codec drivers this patch adds a new set of
equivalent functions that take a struct snd_soc_component instead of a
dapm context and automatically add the component's name_prefix to the
given name. This makes it a simple change in codec drivers to be
prefix-safe.
The new functions are not quite trivial enough to be inlines and the
compiler won't be able to compile-away any part of them.
Although it looks somewhat inefficient to have to allocate a temporary
buffer and combine strings, the current design of the widget list
doesn't lend itself to a more optimized implementation - it's a single
list of all widgets on a card and is searched linearly for a matching
string. As pin state changes are generally low-frequency events it's
unlikely to be a significant issue - at least not enough to rewrite the
widget list handling just for this.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch is a cleanup of EMU1010 dock probing code in emu10k1 driver
to use work instead of kthread in a loop. The work is lighter and
easier to control than kthread, in general.
Instead of a loop with the explicit sleep, we do simply
delayed-schedule the work. At suspend/resume callbacks, the work is
canceled and restarted, respectively.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Several lines in sound/core.h get the kernel-doc warnings like
./include/sound/core.h:323: warning: No description found for parameter '...'
where we use define like foo(x, args...) and "args" isn't mentioned in
the comments. As an easy workaround, use simple __VA_ARGS__ for VLA
in macros.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some fields in struct snd_compr have no corresponding comments, and
the kernel-doc complains like:
./include/sound/compress_driver.h:162: warning: No description found for parameter 'id[64]'
./include/sound/compress_driver.h:162: warning: No description found for parameter 'proc_root'
./include/sound/compress_driver.h:162: warning: No description found for parameter 'proc_info_entry'
Actually all these are internal elements, just put "private:" comment
so that they will be ignored.
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC Compress ops have only platform ops and no DAI ops unlike PCM device
where we have both platform ops as well as DAI ops.
So add compress dai ops and add this new structure to the ASoC core to make
compressed devices a first class ASoC citizen
Again like PCM ops, drivers are free to implement either or both of
these ops based on device needs.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is assuming that the card related information is located on
"card" node, but graph case doesn't have it.
This patch adds node parameter to adjust for graph support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is assuming that the card related information is located on
"card" node, but graph case doesn't have it.
This patch adds node parameter to adjust for graph support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is assuming that the card related information is located on
"card" node, but graph case doesn't have it.
This patch adds node parameter to adjust for graph support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_get_dai_name() is used from snd_soc_of_get_dai_name(),
and it is assuming that DT is using "sound-dai" / "#sound-dai-cells".
But graph base DT is using "remote-endpoint". This patch makes
snd_soc_get_dai_name() non static for graph support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is assuming that the card related information is located on
"card" node, but graph case doesn't have it.
This patch adds node parameter to adjust for graph support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
A DAPM_DOUBLE_R control type can be used for dual channel mixer input
selectors / mute controls across 2 registers, possibly toggling both
channels together.
The control is meant to be shared by 2 widgets, 1 for each channel,
such that the mixer control exposed to userspace remains a combined
stereo control.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
A DAPM_DOUBLE control type can be used for dual channel mixer input
selectors / mute controls in one register, possibly toggling both
channels together.
The control is meant to be shared by 2 widgets, 1 for each channel,
such that the mixer control exposed to userspace remains a combined
stereo control.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
To support double channel shared controls split across 2 registers, one
for each channel, we must be able to update both registers together.
Add a second set of register fields to struct snd_soc_dapm_update, and
update the DAPM control writeback (put) callbacks to support this.
For codecs that use custom events which call into DAPM to do updates,
also clear struct snd_soc_dapm_update before using it, so the second
set of fields remains clean.
Signed-off-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 194c7dea00
"ASoC: dmaengine: add custom DMA config to snd_dmaengine_pcm_config"
custom DMA channels can be also specified in chan_names[] field of
struct snd_dmaengine_pcm_config. This patch removes chan_name field
of struct snd_dmaengine_dai_dma_data as it is now unused.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Due to the DMIC that needs time to initial after the MCLK is provided, the
field of delay time is implemented by the platform data.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
- first slice of the gvt device model (Zhenyu et al)
- compression support for gpu error states (Chris)
- sunset clause on gpu errors resulting in dmesg noise telling users
how to report them
- .rodata diet from Tvrtko
- switch over lots of macros to only take dev_priv (Tvrtko)
- underrun suppression for dp link training (Ville)
- lspcon (hmdi 2.0 on skl/bxt) support from Shashank Sharma, polish
from Jani
- gen9 wm fixes from Paulo&Lyude
- updated ddi programming for kbl (Rodrigo)
- respect alternate aux/ddc pins (from vbt) for all ddi ports (Ville)
* tag 'drm-intel-next-2016-10-24' of git://anongit.freedesktop.org/drm-intel: (227 commits)
drm/i915: Update DRIVER_DATE to 20161024
drm/i915: Stop setting SNB min-freq-table 0 on powersave setup
drm/i915/dp: add lane_count check in intel_dp_check_link_status
drm/i915: Fix whitespace issues
drm/i915: Clean up DDI DDC/AUX CH sanitation
drm/i915: Respect alternate_ddc_pin for all DDI ports
drm/i915: Respect alternate_aux_channel for all DDI ports
drm/i915/gen9: Remove WaEnableYV12BugFixInHalfSliceChicken7
drm/i915: KBL - Recommended buffer translation programming for DisplayPort
drm/i915: Move down skl/kbl ddi iboost and n_edp_entires fixup
drm/i915: Add a sunset clause to GPU hang logging
drm/i915: Stop reporting error details in dmesg as well as the error-state
drm/i915/gvt: do not ignore return value of create_scratch_page
drm/i915/gvt: fix spare warnings on odd constant _Bool cast
drm/i915/gvt: mark symbols static where possible
drm/i915/gvt: fix sparse warnings on different address spaces
drm/i915/gvt: properly access enabled intel_engine_cs
drm/i915/gvt: Remove defunct vmap_batch()
drm/i915/gvt: Use common mapping routines for shadow_bb object
drm/i915/gvt: Use common mapping routines for indirect_ctx object
...
commit f2ed6b0764 ("ASoC: Make aux_dev more like a generic component")
removed its usecase. No one is using it now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can reduce struct size in certain environment.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can reduce struct size in certain environment.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current snd_soc_dai_link is already using many bit fields.
Let's use it for playback_only/capture_only too.
We can reduce struct size in certain environment.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Define DAI physical PCM data formats for user space, so users can
configure the formats of backends by topology (e.g. the DAI format
to set on backend link init).
The kernel will also refer to these formats.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Initial commit of the Cirrus Logic cs35l34 8V boosted class D
amplifier.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Apart from the cleanups done by Morimoto-san this has very much been a
driver focused release with very little generic change:
- A big factoring out of the simple-card code to allow it to be shared
more with the rcar generic card from Kuninori Morimoto.
- Removal of some operations duplicated on the CODEC level, again by
Kuninori Morimoto.
- Lots more machine support for x86 systems.
- New drivers for Nuvoton NAU88C10, Realtek RT5660 and RT5663.
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Merge tag 'asoc-v4.9' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.9
Apart from the cleanups done by Morimoto-san this has very much been a
driver focused release with very little generic change:
- A big factoring out of the simple-card code to allow it to be shared
more with the rcar generic card from Kuninori Morimoto.
- Removal of some operations duplicated on the CODEC level, again by
Kuninori Morimoto.
- Lots more machine support for x86 systems.
- New drivers for Nuvoton NAU88C10, Realtek RT5660 and RT5663.
Currently if AAD is enabled in the device, during system suspend
the feature remains, regardless of whether the codec is a wake-up
source or not. This means some additional power is being used
which is unnecessary, and can causes issues with some platforms'
IRQ handlers where state changes during system suspend aren't
captured.
This patch updates the driver to disable AAD during suspend, if
we're not a wake-up source, and then re-enables this on resume.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In commit bf1d1c9b61 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()"), the new
macro was added so that "dB range information can be specified without
having to count the items manually for TLV_DB_RANGE_HEAD()". In short,
TLV_DB_RANGE_HEAD macro was obsoleted.
In commit 46e860f768 ("ALSA: rename TLV-related macros so that they're
friendly to user applications"), TLV-related macros are exposed for
applications in user land to get content of data structured by
Type/Length/Value shape. The commit managed to expose TLV-related macros
as many as possible, while obsoleted TLV_DB_RANGE_HEAD() was included to
the list of exposed macros.
This situation brings some confusions to application developers because
they might think all exposed macros have their own purpose and useful for
applications.
For the reason, this commit moves TLV_DB_RANGE_HEAD macro from UAPI header
to a header for kernel land, again. The above commit is done within the
same development period for kernel 4.9, thus not published yet. This
commit might certainly brings no confusions to user land.
Reference: commit bf1d1c9b61 ("ALSA: tlv: add DECLARE_TLV_DB_RANGE()")
Reference: commit 46e860f768 ("ALSA: rename TLV-related macros so that they're friendly to user applications")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DP MST provides the capability to send multiple video and audio streams
through a single port. This requires the API's between i915 and audio
drivers to distinguish between multiple audio capable displays that can be
connected to a port. Currently only the port identity is shared in the
APIs. This patch adds support for MST with an additional parameter
'int pipe'. The existing parameter 'port' does not change it's meaning.
pipe =
MST : display pipe that the stream originates from
Non-MST : -1
Affected APIs:
struct i915_audio_component_ops
- int (*sync_audio_rate)(struct device *, int port, int rate);
+ int (*sync_audio_rate)(struct device *, int port, int pipe,
+ int rate);
- int (*get_eld)(struct device *, int port, bool *enabled,
- unsigned char *buf, int max_bytes);
+ int (*get_eld)(struct device *, int port, int pipe,
+ bool *enabled, unsigned char *buf, int max_bytes);
struct i915_audio_component_audio_ops
- void (*pin_eld_notify)(void *audio_ptr, int port);
+ void (*pin_eld_notify)(void *audio_ptr, int port, int pipe);
This patch makes dummy changes in the audio drivers (thanks Libin) for
build to succeed. The audio side drivers will send the right 'pipe' values
for MST in patches that will follow.
v2:
Renamed the new API parameter from 'dev_id' to 'pipe'. (Jim, Ville)
Included Asoc driver API compatibility changes from Jeeja.
Added WARN_ON() for invalid pipe in get_saved_encoder(). (Takashi)
Added comment for av_enc_map[] definition. (Takashi)
v3:
Fixed logic error introduced while renaming 'dev_id' as 'pipe' (Ville)
Renamed get_saved_encoder() to get_saved_enc() to reduce line length
v4:
Rebased.
Parameter check for pipe < -1 values in get_saved_enc() (Ville)
Switched to for_each_pipe() in get_saved_enc() (Ville)
Renamed 'pipe' to 'dev_id' in audio side code (Takashi)
v5:
Included a comment for the dev_id arg. (Libin)
Signed-off-by: Dhinakaran Pandiyan <dhinakaran.pandiyan@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Ville Syrjälä <ville.syrjala@linux.intel.com>
Signed-off-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Link: http://patchwork.freedesktop.org/patch/msgid/1474488168-2343-1-git-send-email-dhinakaran.pandiyan@intel.com
In a previous commit, some macros newly appeared to UAPI header for TLV
packet. These macros have short names and they easily bring name conflist
to applications. The conflict can be avoided to rename them with a proper
prefix.
For this purpose, this commit renames these macros with prefix
'SNDRV_CTL_TLVD_'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ALSA control interface, each element set can have threshold level
information. This information is transferred between drivers/applications,
in a shape of tlv packet. The layout of this packet is defined in
'uapi/sound/asound.h' (struct snd_ctl_tlv):
struct snd_ctl_tlv {
unsigned int numid;
unsigned int length;
unsigned int tlv[0];
};
Data in the payload (struct snd_ctl_tlv.tlv) is expected to be filled
according to our own protocol. This protocol is described in
'include/sound/tlv.h'. A layout of the payload is expected as:
struct snd_ctl_tlv.tlv[0]: one of SNDRV_CTL_TLVT_XXX
struct snd_ctl_tlv.tlv[1]: Length of data
struct snd_ctl_tlv.tlv[2...]: data
Unfortunately, the macro is not exported to user land yet, thus
applications cannot get to know the protocol.
Additionally, ALSA control core has a feature called as 'user-defined'
element set. This allows applications to add/remove arbitrary element sets
with elements to control devices. Elements in the element set can be
operated by the same way as the ones added by in-kernel implementation.
For threshold level information of 'user-defined' element set, applications
need to register the information to an element set. However, as described
above, layout of the payload is closed in kernel land. This is quite
inconvenient, too.
This commit moves the protocol to UAPI header for TLV.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
asoc_simple_card_set_dailink_name() uses devm_kvasprintf() to format
some of its arguments. Adding a __printf attribute to this function
makes it possible to detect at compile-time errors related to format
strings.
Signed-off-by: Nicolas Iooss <nicolas.iooss_linux@m4x.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card needs to decrease the reference count of the device nodes.
This patch makes this method simple style standard.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card needs remove dai_link->cpu_dai_name if it CPU was
single DAI.
This patch makes this method simple style standard.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card is assuming that sometimes platform and cpu are same.
This patch makes this method simple style standard.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card is supporting clock/tdm slot initialization.
This patch makes this method simple style standard.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_pcm_set_drvdata() will set driver data to rtd->dev,
but driver data of rtd->dev is already used as "rtd" on
soc_post_component_init().
static int soc_post_component_init(xxx)
{
...
dev_set_drvdata(rtd->dev, rtd);
...
}
To remove confusion, this patch removes snd_soc_pcm_set/get_drvdata().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skylake onwards HDA controller supports new capabilities like
Global Time Stamping (GTS) capability. So add support to parse
these new capabilities.
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the unused one as we have moved it up to hdac core.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDA capability introduced recently are move to hdac core so that it can
be used by legacy driver as well. Also move the capability pointers up
to hdac_bus object.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
codec driver and component driver has duplicated callback functions,
and codec side functions are just copied to component side when
register timing. This was quick-hack, but no longer needed.
This patch removes codec side duplicated callback function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The power() callback has always been empty so just remove it.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds aptional GPIO bit-bang based callback implementations
for setting CLK, DATA and MODE L3 bus lines. It is added here
to avoid possible duplicate implementations across users of
the bus.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
simple-card needs to get its dai name and endpoint node.
This patch makes it simple style standard
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card can get clock via DT clocks or
"system-clock-frequency" property.
This patch makes it simple style standard
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge drm updates from Dave Airlie:
"This is the main drm pull request for 4.8.
I'm down with a cold at the moment so hopefully this isn't in too bad
a state, I finished pulling stuff last week mostly (nouveau fixes just
went in today), so only this message should be influenced by illness.
Apologies to anyone who's major feature I missed :-)
Core:
Lockless GEM BO freeing
Non-blocking atomic work
Documentation changes (rst/sphinx)
Prep for new fencing changes
Simple display helpers
Master/auth changes
Register/unregister rework
Loads of trivial patches/fixes.
New stuff:
ARM Mali display driver (not the 3D chip)
sii902x RGB->HDMI bridge
Panel:
Support for new panels
Improved backlight support
Bridge:
Convert ADV7511 to bridge driver
ADV7533 support
TC358767 (DSI/DPI to eDP) encoder chip support
i915:
BXT support enabled by default
GVT-g infrastructure
GuC command submission and fixes
BXT workarounds
SKL/BKL workarounds
Demidlayering device registration
Thundering herd fixes
Missing pci ids
Atomic updates
amdgpu/radeon:
ATPX improvements for better dGPU power control on PX systems
New power features for CZ/BR/ST
Pipelined BO moves and evictions in TTM
GPU scheduler improvements
GPU reset improvements
Overclocking on dGPUs with amdgpu
Polaris powermanagement enabled
nouveau:
GK20A/GM20B volt and clock improvements.
Initial support for GP100/GP104 GPUs, GP104 will not yet support
acceleration due to NVIDIA having not released firmware for them as of yet.
exynos:
Exynos5433 SoC with IOMMU support.
vc4:
Shader validation for branching
imx-drm:
Atomic mode setting conversion
Reworked DMFC FIFO allocation
External bridge support
analogix-dp:
RK3399 eDP support
Lots of fixes.
rockchip:
Lots of small fixes.
msm:
DT bindings cleanups
Shrinker and madvise support
ASoC HDMI codec support
tegra:
Host1x driver cleanups
SOR reworking for DP support
Runtime PM support
omapdrm:
PLL enhancements
Header refactoring
Gamma table support
arcgpu:
Simulator support
virtio-gpu:
Atomic modesetting fixes.
rcar-du:
Misc fixes.
mediatek:
MT8173 HDMI support
sti:
ASOC HDMI codec support
Minor fixes
fsl-dcu:
Suspend/resume support
Bridge support
amdkfd:
Minor fixes.
etnaviv:
Enable GPU clock gating
hisilicon:
Vblank and other fixes"
* tag 'drm-for-v4.8' of git://people.freedesktop.org/~airlied/linux: (1575 commits)
drm/nouveau/gr/nv3x: fix instobj write offsets in gr setup
drm/nouveau/acpi: fix lockup with PCIe runtime PM
drm/nouveau/acpi: check for function 0x1B before using it
drm/nouveau/acpi: return supported DSM functions
drm/nouveau/acpi: ensure matching ACPI handle and supported functions
drm/nouveau/fbcon: fix font width not divisible by 8
drm/amd/powerplay: remove enable_clock_power_gatings_tasks from initialize and resume events
drm/amd/powerplay: move clockgating to after ungating power in pp for uvd/vce
drm/amdgpu: add query device id and revision id into system info entry at CGS
drm/amdgpu: add new definition in bif header
drm/amd/powerplay: rename smum header guards
drm/amdgpu: enable UVD context buffer for older HW
drm/amdgpu: fix default UVD context size
drm/amdgpu: fix incorrect type of info_id
drm/amdgpu: make amdgpu_cgs_call_acpi_method as static
drm/amdgpu: comment out unused defaults_staturn_pro static const structure to fix the build
drm/amdgpu: enable UVD VM only on polaris
drm/amdgpu: increase timeout of IB test
drm/amdgpu: add destroy session when generate VCE destroy msg.
drm/amd: fix deadlock of job_list_lock V2
...
simple-card needs to get its card name.
This patch makes this method simple style standard.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current simple-card is creating dai_link->name / dai_link->stream_name.
These are based on CPU + Codec name, or "fe.CPU" or "be.Codec" if it
was DPCM.
This patch adds asoc_simple_card_set_dailink_name() and set dailink
name as common method.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is useful outside the core, when one dapm element is added
at a time.
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>
Allow the HDMI CODECs to get private data passed in in callbacks.
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Merge tag 'asoc-hdmi-codec-pdata' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into drm-next
ASoC: Add private data for HDMI CODEC callbacks
Allow the HDMI CODECs to get private data passed in in callbacks.
[airlied:
Add STI/mediatek patches from Arnd for drivers merged later in drm tree.]
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Dave Airlie <airlied@redhat.com>
* tag 'asoc-hdmi-codec-pdata' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound:
ASoC: hdmi-codec: callback function will be called with private data
We can use simpel utils asoc_simple_card_parse_daifmt().
Let's use it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ALSA SoC has simple-card driver which is supporting both
platform and DT probe.
Now, some sound cards driver are created based on simple-card.
They have similar feature or function, but implemented separately
on each drivers. This is a waste of code.
OTOH, merging these driver into same driver is highly risk,
because it will be very difficult to keep compatibility.
More over, ALSA SoC want to have graph base of DT feature in the
future. Maybe it want to use simple-card like feature / function.
Because of these background, this patch creates simple-card
helper utils, and provides common function to each drivers.
1st is asoc_simple_card_parse_daifmt()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current hdmi-codec driver is assuming that it will be registered
from HDMI driver. Because of this assumption, each callback function
has struct device pointer which is parent device (= HDMI).
Then, it can use dev_get_drvdata() to get private data.
OTOH, on some SoC/HDMI case, SoC has VIDEO/SOUND and HDMI IPs.
This case, it needs SoC VIDEO, SoC SOUND and HDMI video, HDMI codec
driver. In DesignWare HDMI IP case, SoC VIDEO (= DRM/KMS) driver tries
to bind DesignWare HDMI video driver, and HDMI codec driver
(= hdmi-codec). This case, above "parent device" of HDMI codec driver
is DRM/KMS driver and its "device" already has private data.
And, from DT and ASoC CPU/Codec/Card binding point of view, HDMI codec
(= hdmi-codec) needs to have "parent device" (= DRM/KMS), otherwise,
it never detect sound card.
Because of these reasons, some driver can't use dev_get_drvdata() to
get private data on hdmi-codec driver. This patch add new void pointer
on hdmi_codec_pdata for private data, and callback function will be
called with it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Initial commit of the Cirrus Logic cs35l33 8V boosted class D
amplifier.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, the avail IOCTL doesn't pass any error status, which
means typically on error it simply shows no data available. This
can lead to situations where user-space is waiting indefinitely
for data that will never come as the DSP has suffered an
unrecoverable error.
Add snd_compr_stop_error which end drivers can call to indicate
the stream has suffered an unrecoverable error and stop it. The
avail and poll IOCTLs are then updated to report if the stream is
in an error state to user-space. Allowing the error to propagate
out. Processing of the actual snd_compr_stop needs to be deferred
to a worker thread as the end driver may detect the errors during
an existing operation callback.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Clean up the header files regarding to hdmi audio so the omap-hdmi-audio.h
file will only need to include the platform_data/omapdss.h file.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
CC: Mark Brown <broonie@kernel.org>
CC: Jyri Sarha <jsarha@ti.com>
CC: Liam Girdwood <liam.r.girdwood@linux.intel.com>
This patch introduces SOC_SINGLE_S8_TLV() macro for volume control
on chips which supports both negative and positive gains with sign
bit on a 8 bit register, Gain ranges from -128 to +127 with a
predefined step size.
Currently we only have support to DOUBLE_S8_TLV() which does not fit
for cases where we just have separate gain control register for each
channel.
One of the Qualcomm SOC msm8916 has such gain control register whose gain
range is from -38.4dB to +38.4dB with step size of 0.3dB.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Certain situations may warrant examining DAPM paths only to a certain
arbitrary point, as opposed to always following them to the end. For
instance, when establishing a connection between a front-end DAI link
and a back-end DAI link in a DPCM path, it does not make sense to walk
the DAPM graph beyond the first widget associated with a back-end link.
This patch introduces a mechanism which lets a user of
dai_get_connected_widgets supply a function which will be called for
every node during the graph walk. When invoked, this function can
execute arbitrary logic to decide whether the walk, given a DAPM widget
and walk direction, should be terminated at that point or continued
as normal.
Signed-off-by: Piotr Stankiewicz <piotrs@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The updates this time around are almost all driver code:
- Further slow progress on the topology code.
- Substantial updates and improvements for the da7219, es8328, fsl-ssi
Intel and rcar drivers.
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Merge tag 'asoc-v4.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.7
The updates this time around are almost all driver code:
- Further slow progress on the topology code.
- Substantial updates and improvements for the da7219, es8328, fsl-ssi
Intel and rcar drivers.
The HDA links can be switched off when not is use, similarly
command DMA can be stopped as well. This calls for a reference
counting mechanism on the link by it's users to manage the link
power. The DMA can be turned off when all links are off
For this we add two APIs
snd_hdac_ext_bus_link_get
snd_hdac_ext_bus_link_put
They help users to turn up/down link and manage the DMA as well
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 086d7f804e ("ASoC: Convert WM8962 MICBIAS to a supply widget",
2011-09-23) says:
A supply widget is generally clearer than a MICBIAS widget and a
mic bias is just a type of supply so use a supply widget for the
MICBIAS. This also avoids confusion with the routing when
connected to multiple inputs.
but this has never been documented as a policy. Add some comments to
make it clear.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
dmaengine_pcm currently only supports setups where FIFO reads/writes
correspond to exactly one sample, eg 16-bit sample data is transferred
via 16-bit FIFO accesses, 32-bit data via 32-bit accesses.
This patch adds support for setups with fixed width FIFOs where
multiple samples are packed into a larger word.
For example setups with a 32-bit wide FIFO register that expect
16-bit sample transfers to be done with the left+right sample data
packed into a 32-bit word.
Support for packed transfers is controlled via the
SND_DMAENGINE_PCM_DAI_FLAG_PACK flag in snd_dmaengine_dai_dma_data.flags
If this flag is set dmaengine_pcm doesn't put any restriction on the
supported formats and sets the DMA transfer width to undefined.
This means control over the constraints is now transferred to the DAI
driver and it's responsible to provide proper configuration and
check for possible corner cases that aren't handled by the ALSA core.
Signed-off-by: Matthias Reichl <hias@horus.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Martin Sperl <kernel@martin.sperl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent bug report suggests that BCLK setup for i915 HSW/BDW needs
to be updated at each HDMI hotplug, not only at initialization and
resume. That is, we need to update HSW_EM4 and HSW_EM5 registers at
ELD notification, too. Otherwise the HDMI audio may be out of sync
and played in a wrong pitch.
However, the HDA codec driver has no access to the controller
registers, and currently the code managing these registers is in
hda_intel.c, i.e. local to the controller driver. For allowing the
explicit BCLK update from the codec driver, as in this patch, the
former haswell_set_bclk() in hda_intel.c is moved to hdac_i915.c and
exposed as snd_hdac_i915_set_bclk(). This is called from both the HDA
controller driver and intel_pin_eld_notify() in HDMI codec driver.
Along with this change, snd_hdac_get_display_clk() gets dropped as
it's no longer used.
Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=91410
Cc: <stable@vger.kernel.org> # v4.5+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio driver uses regmap cache bypass feature for reading a raw
value without the cache. But this is racy since both the cached and
the uncached reads may occur concurrently. The former is done via the
normal control API access while the latter comes from the proc file
read.
Even though the regmap itself has the protection against the
concurrent accesses, the flag set/reset is done without the
protection, so it may lead to inconsistent state of bypass flag that
doesn't match with the current read and occasionally result in a
kernel WARNING like:
WARNING: CPU: 3 PID: 2731 at drivers/base/regmap/regcache.c:499 regcache_cache_only+0x78/0x93
One way to work around such a problem is to wrap with a mutex. But in
this case, the solution is simpler: for the uncached read, we just
skip the regmap and directly calls its accessor. The verb execution
there is protected by itself, so basically it's safe to call
individually.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=116171
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This API can be used by topology to find an existing BE dai by name
and further configure it.
Topology will also check DAI ID to avoid wrong match.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The generic ID can be used by topology:
- Toplogy can create FE links and set their ID, machine drivers will
be notified and check this ID for machine-specific init.
- Toplogy can use the ID to find existing BE & CC links and further
configure them.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The hdmi-codec is a platform device driver to be registered from
drivers of external HDMI encoders with I2S and/or spdif interface. The
driver in turn registers an ASoC codec for the HDMI encoder's audio
functionality.
The structures and definitions in the API header are mostly redundant
copies of similar structures in ASoC headers. This is on purpose to
avoid direct dependencies to ASoC structures in video side driver.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Acked-by: PC Liao <pc.liao@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add IEC958 channel status helper that gets the audio properties from
snd_pcm_hw_params instead of snd_pcm_runtime. This is needed to
produce the channel status bits already in audio stream configuration
phase.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The existing TLV callback implementation copies all of the
cea_channel_speaker_allocation map table to the TLV container
irrespective of what is reported by sink. This is of little use
to the userspace application.
With this patch, it parses the spk_alloc block as queried from
the ELD, and copies only the corresponding mapping channel
allocation entries from the cea channel speaker allocation table.
Thus the user can parse the TLV container to identify sink's
capability and set the channel map accordingly.
It shouldn't impact the behavior in AMD chipset, as this makes
use of already parsed spk alloc block to calculate the channel
map.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Intel IronLake and ValleyView platforms have different HDMI widget pin
and digital port mapping from other newer ones. The recent ones
(HSW+) have NID 0x05 to 0x07 for port B to port D, while these chips
have NID 0x04 to 0x06.
For adapting this mapping, pass the codec object instead of the bus
object to snd_hdac_sync_audio_rate() and snd_hdac_acomp_get_eld() so
that they can check the codec ID and calculate the mapping properly.
The changes in the HDMI codec driver side will follow in the later
patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The main thing in terms of the core this time around has been some
additional framework work for dynamic topologies (though we *still*
don't appear to have a stable ABI for the topology code, it's probably
worth considering if this will ever happen...). Otherwise the work has
almost all been in the drivers:
- HDMI support for Sky Lake, along with other fixes and enhancements
for the Intel drivers.
- Lots of improvements to the Renesas drivers.
- Capture support for Qualcomm drivers.
- Support for TI DaVinci DRA7xxx devices.
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs.
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514.
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Merge tag 'asoc-v4.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.6
The main thing in terms of the core this time around has been some
additional framework work for dynamic topologies (though we *still*
don't appear to have a stable ABI for the topology code, it's probably
worth considering if this will ever happen...). Otherwise the work has
almost all been in the drivers:
- HDMI support for Sky Lake, along with other fixes and enhancements
for the Intel drivers.
- Lots of improvements to the Renesas drivers.
- Capture support for Qualcomm drivers.
- Support for TI DaVinci DRA7xxx devices.
- New machine drivers for Freescale systems with Cirrus CODECs,
Mediatek systems with RT5650 CODECs.
- New CPU drivers for Allwinner S/PDIF controllers
- New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514.
HD-audio driver has a mechanism to trigger the runtime resume
automatically at accessing the verbs. This auto-resume, however,
causes the mutex deadlock when invoked from the regmap handler since
the regmap keeps the mutex while auto-resuming. For avoiding that,
there is some tricky check in the HDA regmap handler to return -EAGAIN
error to back-off when the codec is powered down. Then the caller of
regmap r/w will retry after properly turning on the codec power.
This works in most cases, but there seems a slight race between the
codec power check and the actual on-demand auto-resume trigger. This
resulted in the lockdep splat, eventually leading to a real deadlock.
This patch tries to address the race window by getting the runtime PM
refcount at the check time using pm_runtime_get_if_in_use(). With
this call, we can keep the power on only when the codec has been
already turned on, and back off if not.
For keeping the code consistency, the code touching the runtime PM is
stored in hdac_device.c although it's used only locally in
hdac_regmap.c.
Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Chmap helper arguments are modified to use either hdac_device
object or hdac_chmap object instead of codec specific object.
With this moving these APIs to core will be easier.
Helper added to access a specific channel_allocation object
instead of directly accessing.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add slot and channel count programming to hdmi_chmap object and
move the chmap_ops to core. Use register_chmap_ops API to
register for default ops. Override specific chmap ops in the
driver.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With this chmap object is added as private data and new ops are
added to access driver specific chmap.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
chmap object represents multichannel capability and contains chmap
ops. Legacy driver is updated to use this.
With next set of patches chmap object is moved to common to be
reused by other drivers (ex: skylake ASoC hdmi driver).
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the interrupt handler of HD-audio driver assumes that no irq
update is needed while processing the irq. But in reality, it has
been confirmed that the HW irq is issued even during the irq
handling. Since we clear the irq status at the beginning, process the
interrupt, then exits from the handler, the lately issued interrupt is
left untouched without being properly processed.
This patch changes the interrupt handler code to loop over the
check-and-process. The handler tries repeatedly as long as the IRQ
status are turned on, and either stream or CORB/RIRB is handled.
For checking the stream handling, snd_hdac_bus_handle_stream_irq()
returns a value indicating the stream indices bits. Other than that,
the change is only in the irq handler itself.
Reported-by: Libin Yang <libin.yang@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the recent integration of kctl jack and input jack layers, we
can basically build the jack layer even without input devices. That
is, the jack layer itself can be built with conditional to enable the
input device support or not, while the users may enable always
CONFIG_SND_JACK unconditionally.
For achieving it, this patch changes the following:
- A new Kconfig, CONFIG_SND_JACK_INPUT_DEV, was introduced to indicate
whether the jack layer supports the input device,
- A few items in snd_jack struct and relevant codes are conditionally
built upon CONFIG_SND_JACK_INPUT_DEV,
- The users of CONFIG_SND_JACK drop the messy dependency on
CONFIG_INPUT.
This change also automagically fixes a potential bug in HD-audio
driver Arnd reported, where the NULL or uninitialized jack instance is
dereferenced.
Reported-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Topology will also create FE DAI links dynamically from the PCM
objects. These links will be removed when the component is removed
and its topology info is unloaded.
The component driver can implement link_load/unload ops for extra
intialization (e.g. error check) and destruction.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Topology will create FE DAIs dynamically from the PCM objects,
and register them to the component.
A PCM topoplogy object describes a FE DAI and DAI link. Later
patch will add FE DAI links as well.
Change tplg load ops for DAI:
- Only process a DAI.
- Pass the DAI driver pointer to the component driver for
extra initialization.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This helper function can convert a given sample rate range to
SNDRV_PCM_RATE_xxx bits.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A kernel WARNING in snd_rawmidi_transmit_ack() is triggered by
syzkaller fuzzer:
WARNING: CPU: 1 PID: 20739 at sound/core/rawmidi.c:1136
Call Trace:
[< inline >] __dump_stack lib/dump_stack.c:15
[<ffffffff82999e2d>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50
[<ffffffff81352089>] warn_slowpath_common+0xd9/0x140 kernel/panic.c:482
[<ffffffff813522b9>] warn_slowpath_null+0x29/0x30 kernel/panic.c:515
[<ffffffff84f80bd5>] snd_rawmidi_transmit_ack+0x275/0x400 sound/core/rawmidi.c:1136
[<ffffffff84fdb3c1>] snd_virmidi_output_trigger+0x4b1/0x5a0 sound/core/seq/seq_virmidi.c:163
[< inline >] snd_rawmidi_output_trigger sound/core/rawmidi.c:150
[<ffffffff84f87ed9>] snd_rawmidi_kernel_write1+0x549/0x780 sound/core/rawmidi.c:1223
[<ffffffff84f89fd3>] snd_rawmidi_write+0x543/0xb30 sound/core/rawmidi.c:1273
[<ffffffff817b0323>] __vfs_write+0x113/0x480 fs/read_write.c:528
[<ffffffff817b1db7>] vfs_write+0x167/0x4a0 fs/read_write.c:577
[< inline >] SYSC_write fs/read_write.c:624
[<ffffffff817b50a1>] SyS_write+0x111/0x220 fs/read_write.c:616
[<ffffffff86336c36>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185
Also a similar warning is found but in another path:
Call Trace:
[< inline >] __dump_stack lib/dump_stack.c:15
[<ffffffff82be2c0d>] dump_stack+0x6f/0xa2 lib/dump_stack.c:50
[<ffffffff81355139>] warn_slowpath_common+0xd9/0x140 kernel/panic.c:482
[<ffffffff81355369>] warn_slowpath_null+0x29/0x30 kernel/panic.c:515
[<ffffffff8527e69a>] rawmidi_transmit_ack+0x24a/0x3b0 sound/core/rawmidi.c:1133
[<ffffffff8527e851>] snd_rawmidi_transmit_ack+0x51/0x80 sound/core/rawmidi.c:1163
[<ffffffff852d9046>] snd_virmidi_output_trigger+0x2b6/0x570 sound/core/seq/seq_virmidi.c:185
[< inline >] snd_rawmidi_output_trigger sound/core/rawmidi.c:150
[<ffffffff85285a0b>] snd_rawmidi_kernel_write1+0x4bb/0x760 sound/core/rawmidi.c:1252
[<ffffffff85287b73>] snd_rawmidi_write+0x543/0xb30 sound/core/rawmidi.c:1302
[<ffffffff817ba5f3>] __vfs_write+0x113/0x480 fs/read_write.c:528
[<ffffffff817bc087>] vfs_write+0x167/0x4a0 fs/read_write.c:577
[< inline >] SYSC_write fs/read_write.c:624
[<ffffffff817bf371>] SyS_write+0x111/0x220 fs/read_write.c:616
[<ffffffff86660276>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185
In the former case, the reason is that virmidi has an open code
calling snd_rawmidi_transmit_ack() with the value calculated outside
the spinlock. We may use snd_rawmidi_transmit() in a loop just for
consuming the input data, but even there, there is a race between
snd_rawmidi_transmit_peek() and snd_rawmidi_tranmit_ack().
Similarly in the latter case, it calls snd_rawmidi_transmit_peek() and
snd_rawmidi_tranmit_ack() separately without protection, so they are
racy as well.
The patch tries to address these issues by the following ways:
- Introduce the unlocked versions of snd_rawmidi_transmit_peek() and
snd_rawmidi_transmit_ack() to be called inside the explicit lock.
- Rewrite snd_rawmidi_transmit() to be race-free (the former case).
- Make the split calls (the latter case) protected in the rawmidi spin
lock.
BugLink: http://lkml.kernel.org/r/CACT4Y+YPq1+cYLkadwjWa5XjzF1_Vki1eHnVn-Lm0hzhSpu5PA@mail.gmail.com
BugLink: http://lkml.kernel.org/r/CACT4Y+acG4iyphdOZx47Nyq_VHGbpJQK-6xNpiqUjaZYqsXOGw@mail.gmail.com
Reported-by: Dmitry Vyukov <dvyukov@google.com>
Tested-by: Dmitry Vyukov <dvyukov@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of the previous ugly hack, introduce a new op, disconnect, to
snd_timer_instance object for handling the wake up of pending tasks
more cleanly.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=109431
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the info in /proc/interrupts doesn't allow to figure out which
interrupt belongs to which card (HDMI, PCH, ..).
Therefore add card details to the interrupt description.
With the patch the info in /proc/interrupts looks like this:
PCI-MSI 442368-edge snd_hda_intel:card1
PCI-MSI 49152-edge snd_hda_intel:card0
NOTE: this patch adds the new irq_descr field snd_card struct that is
filled automatically at a card object creation. This can be used
generically for other drivers as well. The changes for others will
follow later -- tiwai
Signed-off-by: Heiner Kallweit <hkallweit1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch more updates for v4.5, mainly driver work:
- More topology API enhancements from Mengdong Lin working towards
making everything more component based and being able to specify PCM
links via topology.
- Large sets driver updates from Cirrus, Intel (mainly more Skylake
support) and Renesas.
- New drivers for AMD ACP, Atmel PDMIC, Dialog DA7218, Imagination
Technologies SoC IPs, Rockchip RK3036 Inno CODEC and Texas Instruments
PCM3168A.
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Merge tag 'asoc-v4.4-rc8' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Last updates for v4.5
A bunch more updates for v4.5, mainly driver work:
- More topology API enhancements from Mengdong Lin working towards
making everything more component based and being able to specify PCM
links via topology.
- Large sets driver updates from Cirrus, Intel (mainly more Skylake
support) and Renesas.
- New driver for AMD ACP
- Rename PCM1792a driver to be generic pcm179x
This is quite a busy release on the driver front with a lot of new
drivers being added but comparatively quiet on the core side with only
one big change going in and that a fairly straightforward refactoring.
- Conversion of the array of DAI links to a list by Mengdong Lin,
supporting dynamically adding and removing DAI links.
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production. We really need to get to the
point where that can be done.
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come.
- New drivers for a number of Imagination Technologies IPs.
- Lots of new features and cleanups for the Renesas drivers.
- ANC support for WM5110.
- New driver for Atmel class D speaker drivers.
- New drivers for Cirrus CS47L24 and WM1831.
- New driver for Dialog DA7128.
- New drivers for Realtek RT5659 and RT56156.
- New driver for Rockchip RK3036.
- New driver for TI PC3168A
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Merge tag 'asoc-v4.5' into asoc-next
ASoC: Updates for v4.5
This is quite a busy release on the driver front with a lot of new
drivers being added but comparatively quiet on the core side with only
one big change going in and that a fairly straightforward refactoring.
- Conversion of the array of DAI links to a list by Mengdong Lin,
supporting dynamically adding and removing DAI links.
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production. We really need to get to the
point where that can be done.
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come.
- New drivers for a number of Imagination Technologies IPs.
- Lots of new features and cleanups for the Renesas drivers.
- ANC support for WM5110.
- New driver for Atmel class D speaker drivers.
- New drivers for Cirrus CS47L24 and WM1831.
- New driver for Dialog DA7128.
- New drivers for Realtek RT5659 and RT56156.
- New driver for Rockchip RK3036.
- New driver for TI PC3168A
# gpg: Signature made Wed 23 Dec 2015 00:42:40 GMT using RSA key ID 5D5487D0
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
# gpg: WARNING: This key is not certified with a trusted signature!
# gpg: There is no indication that the signature belongs to the owner.
# Primary key fingerprint: 3F25 68AA C269 98F9 E813 A1C5 C3F4 36CA 30F5 D8EB
# Subkey fingerprint: ADE6 68AA 6757 18B5 9FE2 9FEA 24D6 8B72 5D54 87D0
aux_dev is mainly used by the machine driver to specify analog devices,
which are registered as codecs. Making it more like a generic component
can help the machine driver to use it to specify any component with
topology info by name.
Details:
- Remove the stub 'rtd_aux' array from the soc card.
- Add a list 'aux_comp_list' to store the components of aux_devs.
And add a list head 'list_aux' to struct snd_soc_component, for adding
such components to the above list.
- Add a 'init' ops to a component for machine specific init.
soc_bind_aux_dev() will set it to be aux_dev's init. And it will be
called when probing the component.
- soc_bind_aux_dev() will also search components by name of an aux_dev,
since it may not be a codec.
- Move probing of aux_devs before checking new DAI links brought by
topology.
- Move removal of aux_devs later than removal of links. Because topology
of aux components may register DAIs and the DAI drivers will go with
removal of the aux components, we want soc_remove_link_dais() to remove
the DAIs at first.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DWC for capture in ACP 2.x IP reports playback and capture capabilities
though it supports only capture. Added a quirk to override default value
to represent capture capability only.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Define API snd_soc_register_dai() to add a DAI dynamically and
create the DAI widgets. Topology can use this API to register DAIs
when probing a component with topology info. These DAIs's playback
& capture widgets will be freed when the sound card is unregistered
and the DAIs will be freed when cleaning up the component.
And a dobj is embedded into the struct snd_soc_dai_driver. Topology
can use the dobj to find the DAI drivers created by it and free them
when the topology component is removed.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have an API for powering down all links, we need a similar one
for powering up links, so add for power up as well
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skylake sports new capability of DMA resume, DRSM where we can
resume the DMA. This capability is defined by presence of
AZX_DRSM_CAP_ID.
If this capability is present, we use this capability.
So we add:
snd_hdac_ext_stream_drsm_enable() - DMA resume caps
snd_hdac_ext_stream_set_dpibr() - set the DMA position
snd_hdac_ext_stream_set_lpib() - set the lpib
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dapm_mutex_lock currently uses the un-nested call which can
cause lockdep warnings when called from control handlers (a relatively
common usage) and using modules. As creating the control causes a
potential mutex inversion with the handler, creating the control will
take the controls_rwsem under the dapm_mutex and accessing the control
will take the dapm_mutex under controls_rwsem.
All the users look like they want to be using the runtime class of the
lock anyway, so this patch just changes snd_soc_dapm_mutex_lock to use
the nested call, with the SND_SOC_DAPM_CLASS_RUNTIME class.
Fixes: f6d5e586b4 ("ASoC: dapm: Add helpers to lock/unlock DAPM mutex")
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is quite a busy release on the driver front with a lot of new
drivers being added but comparatively quiet on the core side with only
one big change going in and that a fairly straightforward refactoring.
- Conversion of the array of DAI links to a list by Mengdong Lin,
supporting dynamically adding and removing DAI links.
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production. We really need to get to the
point where that can be done.
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come.
- New drivers for a number of Imagination Technologies IPs.
- Lots of new features and cleanups for the Renesas drivers.
- ANC support for WM5110.
- New driver for Atmel class D speaker drivers.
- New drivers for Cirrus CS47L24 and WM1831.
- New driver for Dialog DA7128.
- New drivers for Realtek RT5659 and RT56156.
- New driver for Rockchip RK3036.
- New driver for TI PC3168A
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Merge tag 'asoc-v4.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.5
This is quite a busy release on the driver front with a lot of new
drivers being added but comparatively quiet on the core side with only
one big change going in and that a fairly straightforward refactoring.
- Conversion of the array of DAI links to a list by Mengdong Lin,
supporting dynamically adding and removing DAI links.
- Some more fixes for the topology code, though it is still not final
and ready for enabling in production. We really need to get to the
point where that can be done.
- A pile of changes for Intel SkyLake drivers which hopefully deliver
some useful initial functionality for systems with this chipset,
though there is more work still to come.
- New drivers for a number of Imagination Technologies IPs.
- Lots of new features and cleanups for the Renesas drivers.
- ANC support for WM5110.
- New driver for Atmel class D speaker drivers.
- New drivers for Cirrus CS47L24 and WM1831.
- New driver for Dialog DA7128.
- New drivers for Realtek RT5659 and RT56156.
- New driver for Rockchip RK3036.
- New driver for TI PC3168A
HW can provide 1.6V micbias level as well the existing levels
already provided in the driver. This patch adds support for 1.6V
to the DT binding.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In AB silicon, the internal LDO is not supported so remove
DT and driver references to this (digital voltage direct from
'VDD' supply)
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Rob Herring <robh@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add SOC_DOUBLE_STS macro for read-only volatile status controls
Signed-off-by: Damien.Horsley <Damien.Horsley@imgtec.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A couple of i915_audio_component ops have been added and accessed
directly from patch_hdmi.c. Ideally all these should be factored out
into hdac_i915.c.
This patch does it, adds two new helper functions for setting N/CTS
and fetching ELD bytes. One bonus is that the hackish widget vs port
mapping is also moved to hdac_i915.c, so that it can be fixed /
enhanced more cleanly.
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A machine driver can register the two ops.
When a DAI link is added or removed by a component's topology, the
ASoC core can call the ops to notify the machine driver for extra
intialization or destruction.
E.g. topology can create FE DAI links from a cpu DAI component, and
the machine driver may define an add_dai_link ops to set machine-specific
.init ops for the DAI link.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Implement a dai link list for the soc card.
Add APIs to add/remove a DAI links dynamically, e.g. by topology.
And a dobj is embedded into the struct snd_soc_dai_link. Topology can
use the dobj to find the links created by it and remove them when the
topology component is unloaded.
The predefined DAI links are reserved to keep backward compatibility.
And they will also be added to the list.
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DWC in ACP 2.x IP has different offsets for I2S_COMP_PARAM_* registers.
Added a quirk to support the same.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In BXT-P A0, HD-Audio DMA requests is later than expected,
and makes an audio stream sensitive to system latencies when
24/32 bits are playing.
Adjusting threshold of DMA fifo to force the DMA request
sooner to improve latency tolerance at the expense of power.
v2: move Intel specific code to hda_intel.c
Signed-off-by: Lu, Han <han.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds support for DA7217 and DA7218 audio codecs.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make snd_compress_new take an id string (like snd_pcm_new).
This string can be included in the procfs info.
This patch also updates soc_new_compress() to create an ID
based on the stream and dai name, as done for PCM streams.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements a procfs info file for compr nodes when
SND_VERBOSE_PROCFS is enabled. This is equivalent to what the PCM
core already does for pcm nodes.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_i2c_ops structures are never modified, so declare them as const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we have SND_SOC_BYTES_TLV control to lets devices have
larger size data sent, we do not need SND_SOC_BYTES_EXT with 512
byte limitation so mark it deprecated
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAPM models various widgets but lacks a sink widget.
DSPs can have modules which take audio data, process it and are
capable of generating events thus acting as a sink of data.
To make the dapm graph complete for such paths we need a dapm
sink widget for these modules, so add a SND_SOC_DAPM_SINK to
declare such a widget. This widget will be treated as
SND_SOC_DAPM_EP_SINK endpoint in the dapm graph
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_rawmidi_global_ops structures are never modified, so declare them
as const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This extends the structure definition of ext_device and adds
definition for dma_params which will be used when hdmi codec.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the number of DAI links is statically defined by the machine
driver at build time using an array. This makes it difficult to shrink/
grow the number of DAI links at runtime in order to reflect any changes
in topology.
We can change the DAI link array in the core to a list so that PCMs and
FE DAI links can be added and deleted at runtime to reflect changes in
use case and DSP topology. The machine driver can still register DAI links
as an array.
As the 1st step, this patch change the PCM runtime array to a list. A new
PCM runtime is added to the list when a DAI link is bound successfully.
Later patches will further implement the DAI link list.
More:
- define snd_soc_new/free_pcm_runtime() to create/free a runtime.
- define soc_add_pcm_runtime() to add a runtime to the rtd list.
- define soc_remove_pcm_runtimes() to clean up the runtime list.
- traverse the rtd list to probe the link components and dais.
- Add a field "num" to PCM runtime struct, used to specify the device
number when creating the pcm device, and for a soc card to access
its dai_props array.
- The following 3rd party machine/platform drivers iterate the rtd list
to check the runtimes:
sound/soc/intel/atom/sst-mfld-platform-pcm.c
sound/soc/intel/boards/cht_bsw_rt5645.c
sound/soc/intel/boards/cht_bsw_rt5672.c
sound/soc/intel/boards/cht_bsw_max98090_ti.c
Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The AC97 specification provides a guide for 16 GPIOs in the codecs. If
the gpiolib is compiled in the kernel, declare a gpio chip.
This was tested with a pxa27x board (mioa701) and a wm9713 codec.
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add kcontrol to the tlv callbacks in soc_bytes_ext, as it is
needed for referencing the corresponding control in the driver
code
Also fix the only upstream user in topology core
Signed-off-by: Mythri P K <mythri.p.k@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If there is anything in damp->path_source_cache or
damp->path_sink_cache, it can not be valid after the widgets have been
freed. Without this patch a repeated remove and load of a machine
driver may cause NULL pointer reference in dapm_wcache_lookup() when a
freed widget, not belonging to any list, is haunting in the wcache.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Reported-by: Felipe Balbi <balbi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ALSA core does not modify the constraints provided by a driver. Most
constraint helper functions already take a const pointer to the constraint
description, the exception at the moment being the ratden and ratnum
constraints. Make those const as well, this allows a driver to declare them
as const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not much core work here, a few small tweaks to interfaces but mainly the
changes here are driver ones. Highlights include:
- Updates to the topology userspace interface
- Big updates to the Renesas support from Morimoto-san
- Most of the support for Intel Sky Lake systems.
- New drivers for Asahi Kasei Microdevices AK4613, Allwinnner A10,
Cirrus Logic WM8998, Dialog DA7219, Nuvoton NAU8825 and Rockchip
S/PDIF.
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Merge tag 'asoc-v4.3-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.4
Not much core work here, a few small tweaks to interfaces but mainly the
changes here are driver ones. Highlights include:
- Updates to the topology userspace interface
- Big updates to the Renesas support from Morimoto-san
- Most of the support for Intel Sky Lake systems.
- New drivers for Asahi Kasei Microdevices AK4613, Allwinnner A10,
Cirrus Logic WM8998, Dialog DA7219, Nuvoton NAU8825 and Rockchip
S/PDIF.
- A new driver for the Atmel Class D speaker drivers
A first batch of updates targetted at v4.4. There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
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Merge tag 'asoc-v4.3-rc2' into asoc-next
ASoC: Updates for v4.4
A first batch of updates targetted at v4.4. There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
# gpg: Signature made Fri 25 Sep 2015 05:37:06 KST using RSA key ID 5D5487D0
# gpg: key CD7BEEBC: no public key for trusted key - skipped
# gpg: key CD7BEEBC marked as ultimately trusted
# gpg: key AF88CD16: no public key for trusted key - skipped
# gpg: key AF88CD16 marked as ultimately trusted
# gpg: key 16005C11: no public key for trusted key - skipped
# gpg: key 16005C11 marked as ultimately trusted
# gpg: key 5621E907: no public key for trusted key - skipped
# gpg: key 5621E907 marked as ultimately trusted
# gpg: key 5C6153AD: no public key for trusted key - skipped
# gpg: key 5C6153AD marked as ultimately trusted
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
Currently there is no clear definition of what FSYNC polarity is.
Different drivers use its own definition of what is "normal" and what is
"inverted" fsync. This leads to compatibility problems between drivers.
For example TegraX1 driver assumes that DSP-A format with frames
starting at rising FSYNC edge has "inverted" polarity,
while RT5677 assumes it is "normal" polarity.
Explicitly specify meaning of BCLK/FSYNC polarity to avoid future
compatibility problems.
Signed-off-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
While there is nothing wrong with the transfer_ack_begin and
transfer_ack_end callbacks per-se, the last documented user was part of the
alsa-driver 0.5.12a package, which was released 14 years ago and even
predates the upstream integration of the ALSA core and has subsequently
been superseded by newer alsa-driver releases.
This seems to indicate that there is no need for having these callbacks and
they are just cruft that can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Given a kcontrol, we may want to access the parent widget
and it's associated data. So export function to return it.
Signed-off-by: Mythri P K <mythri.p.k@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't always need soc-compress in soc, here add a config item
SND_SOC_COMPRESS, when nobody select it, the soc-compress will
not be compiled.
Here also change Kconfig to 'select SND_SOC_COMPRESS' for drivers
that needed soc-compress.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_limit_volume() operates on a card and the CODEC that is passed in
is only used to look up the card. Let it directly take the card instead.
This makes it possible to use it when no snd_soc_codec is available.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The recommended and most efficient way to constraint a configuration
parameter to a single value is to set the minimum and maximum allowed
values to the same value, i.e. calling snd_pcm_hw_constraint_minmax() with
the same value for min and max.
It is not necessarily obvious though that this is the approach that should
be taken and some drivers have come up with other ways of solving this
problem, e.g. installing a list constraint with a single item. List
constraints are dynamic constraints though and hence less efficient than
the static min-max constraint.
This patch introduces a new helper function called
snd_pcm_hw_constraint_single() which only takes a single value has the same
effect as calling snd_pcm_hw_constraint_minmax() with the same values for
min and max. But it is hopefully semantically more expressive, making it
clear that this is the preferred way of setting a single value constraint.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are 24 EQ registers not 25, I suspect this bug came about because
the registers start at EQ1 not zero. The bug is relatively harmless as
the extra register written is an unused one.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
With the new modalias infrastructure support added for hda, create a macro
for ext devices similar to legacy to add the device entry.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch provide a new common helper function,
snd_hdac_codec_modalias(), to give the codec modalias name string.
This function will be used by multiple places in the later patches.
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For generating modalias entries automatically, move the definition of
struct hda_device_id to linux/mod_devicetable.h and add the handling
of this record in file2alias helper. The new modalias is represented
with combination of vendor id, device id, and api version as
"hdaudio:vNrNaN".
This patch itself doesn't convert the existing modaliases. Since they
were added manually, this patch won't give any regression by itself at
this point.
[Modified the modalias format to adapt the api_version field, and drop
invalid ANY_ID definition by tiwai]
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For distinguishing the difference between HDA legacy and ext codec
driver entries, we need to expose the value corresponding to type
field. This patch adds a new field, api_version, to hda_device_id
struct, so that this information is embedded in modalias string.
Although the information is basically redundant (struct hdac_device
already has type field), the helper that extracts from
MODULE_DEVICE_TABLE() won't take it account except for the exported
table entries themselves. So we need to put the same information in
the table, too.
Reviewed-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Subhransu S Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SX_TLV controls are intended for situations where the register behind
the control has some non-zero value indicating the minimum gain
and then gains increasing from there and eventually overflowing through
zero.
Currently every CODEC implementing these controls specifies the minimum
as the non-zero value for the minimum and the maximum as the number of
gain settings available.
This means when the info callback subtracts the minimum value from the
maximum value to calculate the number of gain levels available it is
actually under reporting the available levels. This patch fixes this
issue by adding a new snd_soc_info_volsw_sx callback that does not
subtract the minimum value.
Fixes: 1d99f2436d ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV")
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Tested-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
PCM timer is not always used. For embedded device, we need an interface
to disable it when it is not needed, to shrink the kernel size and
memory footprint, here add CONFIG_SND_PCM_TIMER for it.
When both CONFIG_SND_PCM_TIMER and CONFIG_SND_TIMER is unselected,
about 25KB saving bonus we can get.
Please be noted that when disabled, those stubs who using pcm timer
(e.g. dmix, dsnoop & co) may work incorrectlly.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few multiple codec drivers do renaming the chip_name string but all
these are open-coded and some of them have even no error check. Let's
make common helpers to do it properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current codec helpers are local to hda code and needs to be moved to
core so that other users can use it.
The helpers to read/write the codec and to check the
power state of widgets is copied
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Driver now can make use of mclk data, if provided, to set, enable
and disable the clock source. As part of this, the choice to
enable clock squaring is dealt with as part of dai_sysclk() call
rather than as platform data.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 3d7608e4c1 ("ARM: shmobile: bockw: remove legacy
board file and config"), Renesas R-Car SoCs are only supported in
generic DT-only ARM multi-platform builds. The driver doesn't need to
use platform data anymore, hence remove platform data configuration.
Move <sound/rcar_snd.h> to sound/soc/sh/rcar/, as it's no longer needed
by platform code.
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
dw i2s controller can work in slave mode, codec being master.
dw i2s is made to support master/slave operation, by reading dwc
register.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds support for the DA7219 audio codec with built-in advanced
accessory detect features.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes the old PXA DMA API usage and switches over to
generic functions provided by snd-soc-dmaengine-pcm.
More cleanups may be done on top of this, and some function stubs can
now be removed completetly. However, the intention here was to keep
the transition as small as possible.
This was tested on the mioa701 pxa27x board.
Signed-off-by: Daniel Mack <zonque@gmail.com>
[trivial change from mmp-dma to pxa-dma]
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
_EXT version of SOC_DOUBLE_R required to allow for custom handlers.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A first batch of updates targetted at v4.4. There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
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Merge tag 'asoc-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.4
A first batch of updates targetted at v4.4. There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
Broadwell can not triger the IRQ falling and rising simultaneously,
so it can not detect jack-in and jack-out simultaneously.
We add a flag "jd_invert" to platform data. If this flag is set,
codec IRQ will be set to invert that forces IRQ as pulse when jack-in
and jack-out.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a version of the SOC_SINGLE_RANGE_TLV macro that allows a custom get
and put to be specified.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds DT binding for explicitly choosing a tdm mask for DAI and uses it
in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been
changed.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sh_clk header is not needed, and it will create confusion.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This structure was added by 4d96eb255c ('ALSA: pcm_lib - add possibility
to log last 10 DMA ring buffer positions') to store PCM pointers
information of latest 10 pointer movements (=XRUN_LOG_CNT). When
CONFIG_SND_PCM_XRUN_DEBUG is configured, 'struct snd_pcm_runtime' has
'hwptr_log' member with a pointer to the structure. When calling
xrun_log() in pcm_lib.c, the structure was allocated to the pointer.
When calling snd_pcm_detach_substream() in pcm.c, the allocated pointer
is released.
In f5914908a5 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints'),
the pointer logging is replaced with using Linux Kernel Tracepoints. The
structure was also removed, while it's just declared. The member and kfree
still remains.
This commit removes the member and related codes. I think this was
overlooked because it brings no errors/warnings to C compilers.
Fixes: f5914908a5 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This lets the interested codec be notified when an i915 pin/ELD
event happens.
[tiwai: Fixed a trivial build error for CONFIG_SND_HDA_I915=n]
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not many updates to the core here, but an awful lot of driver updates
this time round:
- Factoring out of AC'97 reset code into the core
- New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
RT298 and ST STI controllers.
- Machine drivers for Rockchip systems with MAX98090 and RT5645 and
RT5650.
- Initial driver support for Intel Skylake devices.
- A large number of cleanups for Lars-Peter Clausen and Axel Lin.
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Merge tag 'asoc-v4.2-rc8' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.3
Not many updates to the core here, but an awful lot of driver updates
this time round:
- Factoring out of AC'97 reset code into the core
- New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
RT298 and ST STI controllers.
- Machine drivers for Rockchip systems with MAX98090 and RT5645 and
RT5650.
- Initial driver support for Intel Skylake devices.
- A large number of cleanups for Lars-Peter Clausen and Axel Lin.
A number of functions and structures in the sound subsystem had incomplete
and/or obsolete DocBook comments, leading to warnings when the docs were
built. Correct those comments so that we can enjoy our audio in the
absence of warning noise.
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a helper to find the stream using stream tag and direction.
This is useful for drivers to query stream based on stream tag
and direction, fox example while downloading FW thru DSP loader
code
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
spbmaxfifo API is actually a query function not a set function so
name it snd_hdac_ext_stream_get_spbmaxfifo()
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow vendor drivers to define bespoke bytes ext handlers and IDs for
TLV bytes controls. And the topology core will bind these handlers by
matching IDs defined by the vendor driver and user space topology
data file.
And TLV callback binding is moved to soc_tplg_kcontrol_bind_io(). This
function process all handler binding now.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Vendor specific handlers should override standard handlers. So we can
handle things in the order from specific to generic.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The drivers need to set the spib and maxfifios values, so add
these new APIs snd_hdac_ext_stream_set_spib() and
snd_hdac_ext_stream_set_spbmaxfifo() APIs
For these APIs we also need to have spib and fifos pointer, so
add these to hdac_ext_stream and initialize them at stream init
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New HDA controllers like Skylake sport multiple HDA links, so we need a
helper to turn off all the links in one go while suspending the device so
add snd_hdac_ext_bus_link_power_down_all() API
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codecs like Intel HDMI by default do not show up all the pins, they
have to be manually enabled, so we need to refresh the codec widgets and
then recreate the sysfs tree. So add new API snd_hdac_refresh_widget_sysfs()
to do this. It should be be used by codec driver after sending magic verbs
to codec
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds new extended driver objects and API for registering the
extended devices.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds based hdac extended device object which will be used by
ASoC HDAC codecs
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDAC extended device objects are created by HDAC extended bus on probe.
When controller is removed they should be removed as well, so add API
snd_hdac_ext_bus_device_remove for this
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the topology code to be compiled out so that users who don't need
topology don't need to havve the code compiled in, saving them some
memory.
Some more configuration could be added to remove some of the hooks into
the core data structures but that is probably best done with some
refactoring to use functions to do the updates of the data structures
rather than ifdefing in the code as we'd need to do at the minute.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
After the recent cleanups and generalizations of the DAPM algorithm the
handling of input and output paths is now fully symmetric. This means by
making some slight changes to the data structure and using arrays with one
entry for each direction, rather than separate fields, it is possible to
create a generic implementation that is capable of handling both input and
output paths.
Unfortunately this generalization significantly increases the code size on
the hot path of is_connected_{input,output}_ep() and
dapm_widget_invalidate_{input,output}_paths(), which has a negative impact
on the overall performance. The inner loops of those functions are quite
small and the generic implementation adds extra pointer arithmetic in a few
places.
Testing on ARM shows that the combined code size of the specialized
functions is about 50% larger than the generalized function in relative
numbers. But in absolute numbers its less than 200 bytes, which is still
quite small. On the other hand the generalized function increases the
execution time of dapm_power_one_widget() by 30%. Given that this function
is one of the most often called functions of the DAPM framework the
trade-off of getting better performance at expense of generating slightly
larger code at seems to be worth it.
To avoid this still keep two versions of these functions around, one for
input and one for output. But have a generic implementation of the
algorithm which gets inlined by those two versions. And then let the
compiler take care of optimizing it and removing he extra instructions.
This still reduces the source code size as well as the makes making changes
to the implementation more straight forward since the same change does no
longer need to be done in two separate places. Also on the slow paths we
can use a generic implementations that handle both input and output paths.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add helper iterator macros for iterating over the source and sink paths of
widget. This will make it easier to change the implementation later on.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
AC97 devices need to be initially reset before they can be used. Currently
each driver does this on its own.
Add support for resetting the device to core in snd_soc_new_ac97_codec().
If the caller supplies a device ID and device ID mask the function will
reset the device and verify that it has the correct ID, if it does not a
error is returned.
This will allow to remove custom code with similar functionality from
individual drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is currently a lot of code duplication in ASoC drivers regarding the
reset handling of devices. This patch introduces a new generic reset
function in the generic AC'97 framework that can be used to replace most
the custom reset functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_tplg_widget_remove_all() has a verbatim copy of an older version of
the widget freeing code from dapm_free_widgets(). Add a new helper function
that takes care of freeing a widget and use it in both places.
This removes the duplicated code and also makes sure that future changes to
the widget freeing code only have to be made in one location.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds CTU (Channel Transfer Unit) support for rsnd driver.
But, it does nothing to data at this point, but is required for MIX
support.
CTU design is a little different from other IPs (CTU0 is including
CTU00 - CTU03, and CTU1 is including CTU10 - CTU13, these have different
register mapping) We need to care about it on this driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch changes the return type of snd_hdac_power_up/down() and
variants to pass the error code from the underlying
pm_runtime_get/put() calls. Currently they are ignored, but in most
places, these should be handled properly.
As an example, the regmap handler is updated to check the return value
and accesses the register only when the wakeup succeeds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current ASoC can add name_prefix for DAPM, and it is necessary for
route settings. This patch adds snd_soc_of_parse_audio_prefix() for
this purpose. It will be used with snd_soc_of_parse_audio_routing().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use the card field of a component to indicate whether it is bound or not.
This makes a certain sense given that the field contains the card the
component is bound to and a component can only be bound to one card at a
time. And it also requires to unset the card field when the component is
unbound from the card.
This makes the probded flag redundant and it can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are no more direct users of the snd_soc_codec DAPM field left. So we
can finally remove it and switch over to directly using the component DAPM
context and remove the dapm_ptr indirection.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
There's a bunch of additional updates and fixes that came in since my
orignal pull request here, including DT support for rt5645 and fairly
large serieses of cleanups and improvements to tas2552 and rcar.
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Merge tag 'asoc-v4.2-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Further updates for v4.2
There's a bunch of additional updates and fixes that came in since my
orignal pull request here, including DT support for rt5645 and fairly
large serieses of cleanups and improvements to tas2552 and rcar.
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
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Merge tag 'asoc-v4.2' into asoc-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
# gpg: Signature made Mon 08 Jun 2015 18:48:37 BST using RSA key ID 5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
In HDA extended bus the HDA link objects are created when multilink
capabilities are parsed. We need a routine which free up these link objects
for a bus. So add snd_hdac_link_free_all routine
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDAC extended core should create streams for an extended bus and also free
up those on cleanup. So introduce snd_hdac_ext_stream_init_all and
snd_hdac_stream_free_all routines
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we have the bus and controller code added to find and initialize
the extended capabilities. Now we need to use them in stream code to
decouple stream, manage links etc
So this patch adds the stream handling code for extended capabilities
introduced in preceding patches
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The controller needs to support the new capabilities and allow
reading, parsing and initializing of these capabilities, so this patch
does it
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new HDA controllers from Intel support new capabilities like
multilink, pipe processing, SPIB, GTS etc In order to use them we
create an extended HDA bus which embed the hdac bus and contains the
fields for extended configurations
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another regression by the transition to regmap cache; for better
usability, we had the fake mute control using the zero amp value for
Conexant codecs, and this was forgotten in the new hda core code.
Since the bits 4-7 are unused for the amp registers (as we follow the
syntax of AMP_GET verb), the bit 4 is now used to indicate the fake
mute. For setting this flag, snd_hda_codec_amp_update() becomes a
function from a simple macro. The bonus is that it gained a proper
function description.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move gpio to gpio_desc and use gpiod APIs in codec driver.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
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Merge tag 'asoc-v4.2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
Make sure userspace can define TLV controls for topology using the correct
type numbers and channel mappings.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The topology core parses the FW topology file for known block types and
instanciates any common ALSA/ASoC objects that it discovers. The core
also passes any block that is does not understand to client component
drivers for enumeration.
The core exports some APIs to client drivers in order to load and unload
firmware topology data as use case require.
Currently the core deals with the following object types :-
o kcontrols. This includes TLV, enumerated and bytes controls.
o DAPM widgets. All types with any associated kcontrol.
o DAPM graph.
o FE PCM. FE PCM capabilities and configuration can be defined.
o BE DAI Link. BE DAI link capabilities and configuration can be defined.
o Codec <-> codec style links capabilities and configuration.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ASoC topology UAPI header defines the structures
required to define any DSP firmware audio topology and control objects from
userspace.
The following objects are supported :-
o kcontrols including TLV controls.
o DAPM widgets and graph elements
o Vendor bespoke objects.
o Coefficient data
o FE PCM capabilities and config.
o BE link capabilities and config.
o Codec <-> codec link capabilities and config.
o Topology object manifest.
The file format is simple and divided into blocks for each object type and
each block has a header that defines it's size and type. Blocks can be in
any order of type and can either all be in a single file or spread across
more than one file. Blocks also have a group identifier ID so that they can
be loaded and unloaded by ID.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds new registers as per HD audio Spec like capability registers
for processing pipe, software position based FIFO, Multiple Links and Global
Time Synchronization.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDA codec drivers can be matched using vendor id and revision id typically.
So provide a match function which does this and is loaded when driver hasn't
provided one (default behaviour)
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Build emux_proc.o and drop the unneeded ifdefs.
Replace the left CONFIG_PROC with the new CONFIG_SND_PROC_FS.
Along with this, fix the build of emux_oss.o in Makefile, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We may disable proc fs only for sound part, to reduce ALSA
memory footprint. So add CONFIG_SND_PROC_FS and replace the
old CONFIG_PROC_FSs in alsa code.
With sound proc fs disabled, we can save about 9KB memory
size on X86_64 platform.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Reviewed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a helper to create the IEC958 channel status from an ALSA
snd_pcm_runtime structure, taking account of the sample rate and
sample size.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Reviwed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a helper for the EDID like data structure, which is typically passed
from a HDMI adapter to its associated audio driver. This informs the
audio driver of the capabilities of the attached HDMI sink.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Reviewed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current DPCM is caring only FE format. but it will be no sound
if FE/BE was below style, and user selects S24_LE format.
FE: S16_LE/S24_LE
BE: S16_LE
DPCM can rewrite the format, so basically we don't want to
constrain with the BE constraints. But sometimes it will be trouble.
This patch adds new .dpcm_merged_format on struct snd_soc_dai_link.
DPCM will use FE / BE merged format if .struct snd_soc_dai_link
has it. We can have other .dpcm_merged_xxx in the future
.dpcm_merged_foramt
.dpcm_merged_rate
.dpcm_merged_chan
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The file is moved to hda core and renamed to hdac_i915.c, so can be used
by both legacy HDA driver and new Skylake audio driver.
- Add snd_hdac_ prefix to the public APIs.
- The i915 audio component is moved to core bus and dynamically allocated.
- A static pointer hdac_acomp is used to help bind/unbind callbacks to get
this component, because the sound card's private_data is used by the azx
chip pointer, which is a legacy structure. It could be removed if private
_data changes to some core structure which can be extended to find the
bus.
- snd_hdac_get_display_clk() is added to get the display core clock for
HSW/BDW.
- haswell_set_bclk() is moved to hda_intel.c because it needs to write the
controller registers EM4/EM5, and only legacy HD-A needs it for HSW/BDW.
- Move definition of HSW/BDW-specific registers EM4/EM5 to hda_register.h
and rename them to HSW_EM4/HSW_EM5, because other HD-A controllers have
different layout for the extended mode registers.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some CODECs have a significant number of DAPM routes and for each route,
when it is added to the card, the entire card widget list must be
searched. When adding routes it is very likely, however, that adjacent
routes will require adjacent widgets. For example all the routes for a
mux are likely added in a block and the sink widget will be the same
each time and it is also quite likely that the source widgets are
sequential located in the widget list.
This patch adds a cache to the DAPM context, this cache will hold the
source and sink widgets from the last call to snd_soc_dapm_add_route for
that context. A small search of the widget list will be made from those
points for both the sink and source. Currently this search only checks
both the last widget and the one adjacent to it.
On wm8280 which has approximately 500 widgets and 30000 routes (one of
the largest CODECs in mainline), the number of paths that hit the cache
is 24000, which significantly improves probe time.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We can know if dmic is used by reading the value of dmic1_data_pin
and dmic2_data_pin. Also IRQ must be used if codec JD or button
detection function is used. So, dmic_en and en_jd_func can be remove
from platform data.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A demux is conceptually similar to a mux. Where a mux has multiple input
and one output and selects one of the inputs to be connected to the output,
the demux has one input and multiple outputs and selects one of the outputs
to which the input gets connected.
This similarity makes it straight forward to support them in DAPM using the
existing mux support, we only need to swap sinks and sources when initially
setting up the paths.
The only slightly tricky part is that there can only be one control per
path. Since mixers/muxes are at the sink of a path and a demux is at the
source and both types want a control it is not possible to directly connect
a demux output to a mixer/mux input. The patch adds some sanity checks to
make sure that this does not happen.
Drivers who want to model hardware which directly connects a demux output
to a mixer/mux input can do this by inserting a dummy widget between the
two. E.g.:
{ "Dummy", "Demux Control", "Demux" },
{ "Mixer", "Mixer Control", "Dummy" },
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Commit 57295073b6 ("ASoC: dapm: Implement mixer input auto-disable")
added support for autodisable controls, controls whose values are only
written to the hardware when their respective widgets are powered up.
But it only added support for controls based on the mixer abstraction.
This patch add support for mux controls (DAPM controls based on the
enum abstraction) to be auto-disabled as well. As each mux can only have
a single control, there is no need to tie the autodisable widget to the
inputs (as is done for the mixer controls) it can be tided directly to
the mux widget itself.
Note that it is assumed that the first entry in a autodisable mux
control will always represent the off state for the mux and is what the
mux will be set to whilst it is disabled.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
xnitmes is clearly intended to be xnitems, but all other macros just
refer to this as xitems, so change it to that.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
A few fixes for v4.1, none earth shattering and mostly driver related
except for one change to fix !PM builds for Intel platforms which is
done by adding stubs in the core so other platforms don't run into the
same issue.
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Merge tag 'asoc-v4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.1
A few fixes for v4.1, none earth shattering and mostly driver related
except for one change to fix !PM builds for Intel platforms which is
done by adding stubs in the core so other platforms don't run into the
same issue.
A flag "link_power_control" is added to indicate whether a codec needs to
control the link power. And a new bus ops link_power() is defined for the
codec to request to enable/disable the link power.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Looks like audigy emu10k2 (probably emu10k1 - sb live too) support two
modes for DMA. Second mode is useful for 64 bit os with more then 2 GB
of ram (fixes problems with big soundfont loading)
1) 32MB from 2 GB address space using 8192 pages (used now as default)
2) 16MB from 4 GB address space using 4096 pages
Mode is set using HCFG_EXPANDED_MEM flag in HCFG register.
Also format of emu10k2 page table is then different.
Signed-off-by: Peter Zubaj <pzubaj@marticonet.sk>
Tested-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
rajeev-dlh.kumar@st.com email-id doesn't exist anymore as I have left the
company. Replace ST's id with Rajeev Kumar <rajeevkumar.linux@gmail.com>
Signed-off-by: Rajeev Kumar <rajeevkumar.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently drivers are responsible for managing the bias_level field of
their DAPM context. The DAPM state itself is managed by the DAPM core
though and the core has certain expectations on how and when the bias_level
field should be updated. If drivers don't adhere to these undefined
behavior can occur.
This patch adds a few helper functions for manipulating the DAPM context
state, each function with a description on when it should be used and what
its effects are. This will also help us to move more of the bias_level
management from drivers to the DAPM core.
For convenience also add snd_soc_codec_* wrappers around these helpers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAPM context in the snd_soc_codec struct is redundant and scheduled to
be replaced by the DAPM context in the snd_soc_component struct. This patch
introduces a new helper function snd_soc_codec_get_dapm() which should be
used for getting the DAPM context for a CODEC rather then directly
accessing the dapm field. Once there are no more direct users of the dapm
field left it is possible to transparently switch all drivers to the
component DAPM context by updating snd_soc_codec_get_dapm() function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current struct snd_soc_dai_link has many members, but definition order
was random. Especially, bool / bit field are defined randomly.
This patch tidyups these definition order to calculate data alignment
easy.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Jack snd_kcontrols can now be created during snd_jack_new()
or by later calling snd_jack_add_new_kctls().
This patch creates the jacks during the initialisation stage
for both phantom and non phantom jacks.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dont create input devices for phantom jacks.
Here, we extend snd_jack_new() to support phantom jack creating:
pass in a bool param for [non-]phantom flag, and a bool param
initial_jack to indicate whether we need to create a kctl at
this stage.
We can also add a kctl to the jack after its created meaning we
can now integrate the HDA and ASoC jacks.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a static method get_available_index() to
allocate the index of new jack kcontrols and also adds
jack_kctl_name_gen() which is used to ensure compatibility
with jack naming by removing " Jack" from some incorrectly
passed names.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the ALSA jack core registers only input devices for each jack
registered. These jack input devices are not readable by userspace devices
that run as non root. This patch series will implement kctls inside the
core jack part, including kctls creating, status changing report, for both
HD-Audio and ASoC jack. This allows non root userspace to read jack status
and act on it.
This patch adds a new API called snd_jack_add_new_kctl(), which will create
a kcontrol, add it to the card, and also attach it to the jack kctl list.
This patch also initialises the jack kctl list after jack is newed, and
reports kctl status when jack insertion/removal events occur.
snd_jack_new() is updated in the following patches to also support creating
phantom jacks and jack kcontrols. We then remove these duplicated features
from HDA jack and have jack kctls handled by core throughout HDA and ASoC.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Modified-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Jie Yang <yang.jie@intel.com>
Reveiwed-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Whether residue can be reported or not is not a property of the audio
controller but of the DMA controller. The FLAG_NO_RESIDUE was initially
added when the DMAengine framework had no support for describing the residue
reporting capabilities of the controller. Support for this was added quite a
while ago and recently the DMAengine framework started to complain if a
driver does not describe its capabilities and a lot of patches have been
merged that add support for this where it was missing. So it should be safe
to assume that driver on actively used platforms properly implement the DMA
capabilities API.
This patch makes the FLAG_NO_RESIDUE internal and no longer allows audio
controller drivers to manually set the flag. If a DMA driver against
expectations does not support reporting its capabilities for now the generic
DMAengine PCM driver will now emit a warning and simply assume that residue
reporting is not supported. In the future this might be changed to aborting
with an error.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
On a 64-bit system there is a 32-bit hole in struct snd_pcm_constraint_list
and then 32-bit padding at the end. Reordering things slightly gets rid of
the hole and padding, reducing the size of the struct by 50% from its
original size.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On a 64-bit system there are two 32-bit holes due to the alignment of 64-bit
fields. Reordering things slightly gets rid of those holes, reducing the
size of the struct by 17% percent of its original size.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few minor cleanups:
- Move the call of snd_info_minor_register() into snd_info_init() so
that we can call all proc-related stuff in a shot
- Add missing __init prefix to snd_info_minor_register()
- Return an error properly from snd_oss_info_register()
- Drop snd_info_minor_unregister() that is superfluous now
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since each proc entry is freed automatically by the parent, we don't
have to take care of its life cycle any longer. This allows us to
reduce a few more lines of codes.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the way to manage the resource release of proc
files: namely, let snd_info_free_entry() freeing the whole children.
This makes it us possible to drop the snd_device_*() management. Then
snd_card_proc_new() becomes merely a wrapper to
snd_info_create_card_entry().
Together with this change, now you need to call snd_info_free_entry()
for a proc entry created via snd_card_proc_new(), while it was freed
via snd_device_free() beforehand.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
seq_file is _the_ standard interface for simple text proc files.
Though, we still need to support the binary proc files and the text
file write, and also we need to manage the device disconnection
gracefully. Thus this patch just replaces the text file read code
with seq_file while keeping the rest intact.
snd_iprintf() helper function is now a macro to expand itself to
seq_printf() to be compatible with the existing code. The seq_file
object is stored to the unused entry->rbuffer->buffer pointer.
When the output size is expected to be large (greater than PAGE_SIZE),
the driver should set entry->size field beforehand. Then the given
size will be preallocated and the multiple show calls can be avoided.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct small copy and paste error where autodisable was not being
enabled for the SOC_DAPM_SINGLE_TLV_AUTODISABLE control.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
When CONFIG_PM_SLEEP is not selected, calling funcs
snd_soc_suspend and _resume will generate a compiling
issue.
Here add static inline stub functions to fix it.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This will be used by hda controller driver to
setup stream params in prepare. This function will
setup the bdl and periods.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the controller helper codes to hda-core library.
The I/O access ops are added to the bus ops. The CORB/RIRB, the basic
attributes like irq# and iomap address, some locks and the list of
streams are added to the bus object, together with the stream object
and its helpers.
Currently the codes are just copied from the legacy driver, so you can
find duplicated codes in both directories. Only constants are removed
from the original hda_controller.h. More integration work will follow
in the later patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More updates for v4.1, pretty much all drivers:
- Lots of cleanups from Lars, mainly moving things from the CODEC level
to the card level.
- Continuing improvements to rcar from Morimoto-san, pcm512x from
Howard and Peter, the Intel platforms from Vinod, Jie, Jin and Han,
and to rt5670 from Bard.
- Support for some non-DSP Qualcomm platforms, Google's Storm
platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC.
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Merge tag 'asoc-v4.1-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.1
More updates for v4.1, pretty much all drivers:
- Lots of cleanups from Lars, mainly moving things from the CODEC level
to the card level.
- Continuing improvements to rcar from Morimoto-san, pcm512x from
Howard and Peter, the Intel platforms from Vinod, Jie, Jin and Han,
and to rt5670 from Bard.
- Support for some non-DSP Qualcomm platforms, Google's Storm
platform, Maxmim MAX98925 CODECs and the Ingenic JZ4780 SoC.
Although some races in runtime PM refcount was fixed by the commit
[664c715573c2: ALSA: hda - Work around races of power up/down with
runtime PM], there is still a race in the following case:
CPU0: CPU1 :
runtime suspend:
codec->in_pm = 1
snd_hdac_power_up_pm():
pm_runtime_get_sync() skipped
suspend finished:
codec->in_pm = 0
snd_hdac_power_down_pm():
pm_runtime_put_*() is called!
For avoiding this situation, increment in_pm flag atomically when it's
non-zero, and decrement accordingly, to ensure that in_pm is set
consistently for the whole concurrent operations.
Also, since atomic_inc_not_zero() and atomic_dec_if_positive() are
lengthy inline functions, move snd_hdac_power_up_pm() and _down_pm()
to sound/hda/hdac_device.c as no inline functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Failing to register the debugfs entries is not fatal and will not affect
normal operation of the sound card. Don't abort the card registration if
soc_dpcm_debugfs_add() fails.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, snd_hdac_power_up()/down() helpers checks whether the codec
is being in pm (suspend/resume), and skips the call of runtime get/put
during it. This is needed as there are lots of power up/down
sequences called in the paths that are also used in the PM itself. An
example is found in hda_codec.c::codec_exec_verb(), where this can
power up the codec while it may be called again in its power up
sequence, too.
The above works in most cases, but sometimes we really want to wait
for the real power up. For example, the control element get/put may
want explicit power up so that the value change is assured to reach to
the hardware. Using the current snd_hdac_power_up(), however,
results in a race, e.g. when it's called during the runtime suspend is
being performed. In the worst case, as found in patch_ca0132.c, it
can even lead to the deadlock because the code assumes the power up
while it was skipped due to the check above.
For dealing with such cases, this patch makes snd_hdac_power_up() and
_down() to two variants: with and without in_pm flag check. The
version with pm flag check is named as snd_hdac_power_up_pm() while
the version without pm flag check is still kept as
snd_hdac_power_up(). (Just because the usage of the former is fewer.)
Then finally, the patch replaces each call potentially done in PM with
the new _pm() variant.
In theory, we can implement a unified version -- if we can distinguish
the current context whether it's in the pm path. But such an
implementation is cumbersome, so leave the code like this a bit messy
way for now...
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96271
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we assumed that the node attributes like amp values remain
during the power state transition of the node itself. While this is
true for IDT/STAC codecs I've tested, but some other codecs don't seem
behaving in that way.
This patch implements a partial sync mechanism specific to the given
widget node. Now we've merged the regmap support, and it can be
easily written with regcache_sync_region().
Tested-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The delayed_work field in the snd_soc_dapm_context struct is now unused and
can be removed. Removing it reduces the size of the snd_soc_dapm_context
struct by ~50% from 100 bytes to 48 bytes.
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>
The only two users of the suspend_bias_level field were two rather old
drivers which weren't exactly doing things by the book. Those drivers have
been updated and field is now unused and can be removed.
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>
Having to set different formats on the CPU side and the CODEC side of a DAI
link is usually indication that something is terribly wrong and in most
cases is a result of a broken driver that implements a set_fmt() callback
which does not follow the specification. In the past this feature has been
used to work around broken drivers, rather than fixing them. We don't really
want to encourage this, so remove support for setting different formats on
both ends of the link.
Along the way switch to static DAI format setup by setting the the dai_fmt
field of the snd_soc_dai_link rather than calling snd_soc_dai_fmt().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The 16bit COEF read/write is pretty standard for many codecs, and they
can be cached in most cases -- more importantly, they need to be
restored at resume. For making this easier, add the cache support to
regmap. If the codec driver wants to cache the COEF access, set
codec->cache_coef flag and issue AC_VERB_GET_PROC_COEF with the coef
index in LSB 8 bits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio spec is inconvenient regarding the handling of stereo volume
controls. It can set and get only single channel at once (although
there is a special option to set the same value to both channels).
This patch provides a fake pseudo-register via the regmap access so
that the stereo channels can be read and written by a single call.
It'd be useful, for example, for implementing DAPM widgets.
A stereo amp pseudo register consists of the encoding like the normal
amp verbs but it has both SET_LEFT (bit 13) and SET_RIGHT (bit 12)
bits set. The regmap reads and writes a 16bit value for this pseudo
register where the upper 8bit is for the right chanel and the lower
8bit for the left channel.
Note that the driver doesn't recognize conflicts when both stereo and
mono channel registers are mixed. Mixing them would certainly confuse
the operation. So, use carefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Codecs may have own vendor-specific verbs, and we need to allow each
driver to give such verbs for cached accesses. Here a verb can be put
into a single array and looked through it at readable and writeable
callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp hash table was used for recording the cached reads of some
capability values like pin caps or amp caps. Now all these are moved
to regmap as well.
One addition to the regmap helper is codec->caps_overwriting flag.
This is set in snd_hdac_override_parm(), and the regmap helper accepts
any register while this flag is set, so that it can overwrite even the
read-only verb like AC_VERB_PARAMETERS. The flag is cleared
immediately in snd_hdac_override_parm(), as it's a once-off flag.
Along with these changes, the no longer needed amp hash and relevant
fields are removed from hda_codec struct now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes we need the uncached reads, e.g. for refreshing the tree.
This patch provides the helper function for that and uses it for
refreshing widgets, reading subtrees and the whole proc reads.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let's start converting the access functions to regmap.
The first one is the simplest, just converting the codec parameter
read helper function snd_hda_param_read().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds an infrastructure to support regmap-based verb
accesses. Because o the asymmetric nature of HD-audio verbs,
especially the amp verbs, we need to translate the verbs as a sort of
pseudo registers to be mapped uniquely in regmap.
In this patch, a pseudo register is built from the NID, the
AC_VERB_GET_* and 8bit parameters, i.e. almost in the form to be sent
to HD-audio bus but without codec address field. OTOH, for writing,
the same pseudo register is translated to AC_VERB_SET_* automatically.
The AC_VERB_SET_AMP_* verb is re-encoded from the corresponding
AC_VERB_GET_AMP_* verb and parameter at writing.
Some verbs has a single command for read but multiple for writes. A
write for such a verb is split automatically to multiple verbs.
The patch provides also a few handy helper functions. They are
designed to be accessible even without regmap. When no regmap is set
up (e.g. before the codec device instantiation), the direct hardware
access is used. Also, it tries to avoid the unnecessary power-up.
The power up/down sequence is performed only on demand.
The codec driver needs to call snd_hdac_regmap_exit() and
snd_hdac_regmap_exit() at probe and remove if it wants the regmap
access.
There is one flag added to hdac_device. When the flag lazy_cache is
set, regmap helper ignores a write for a suspended device and returns
as if it was actually written. It reduces the hardware access pretty
much, e.g. when adjusting the mixer volume while in idle. This
assumes that the driver will sync the cache later at resume properly,
so use it carefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add an overriding exec_verb op to struct hdac_device so that the call
via snd_hdac_exec_verb() can switch to a different route depending on
the setup. The codec driver sets this field so that it can handle the
errors or applying quirks appropriately. Furthermore, this mechanism
will be used for smooth transition for the regmap support in later
patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the sysfs files assigned to the codec device on the
bus which were formerly identical with hwdep sysfs files. Now it
shows only a few core parameter, vendor_id, subsystem_id, revision_id,
afg, mfg, vendor_name and chip_name.
In addition, now a widget tree is added to the bus device sysfs
directory for showing the widget topology and attributes. It's just a
flat tree consisting of subdirectories named as the widget NID
including various attributes like widget capability bits. The AFG
(usually NID 0x01) is always found there, and it contains always
amp_in_caps, amp_out_caps and power_caps files. Each of these
attributes show a single value. The rest are the widget nodes
belonging to that AFG. Note that the child node might not start from
0x02 but from another value like 0x0a.
Each child node may contain caps, pin_caps, amp_in_caps, amp_out_caps,
power_caps and connections files. The caps (representing the widget
capability bits) always contain a value. The rest may contain
value(s) if the attribute exists on the node. Only connections file
show multiple values while other attributes have zero or one single
value.
An example of ls -R output is like below:
% ls -R /sys/bus/hdaudio/devices/hdaudioC0D0/
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/:
01/ 04/ 07/ 0a/ 0d/ 10/ 13/ 16/ 19/ 1c/ 1f/ 22/
02/ 05/ 08/ 0b/ 0e/ 11/ 14/ 17/ 1a/ 1d/ 20/ 23/
03/ 06/ 09/ 0c/ 0f/ 12/ 15/ 18/ 1b/ 1e/ 21/
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/01:
amp_in_caps amp_out_caps power_caps
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/02:
amp_in_caps amp_out_caps caps connections pin_caps pin_cfg
power_caps
/sys/bus/hdaudio/devices/hdaudioC0D0/widgets/03:
.....
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now some codes and functionalities of hda_codec struct are moved to
hdac_device struct. A few basic attributes like the codec address,
vendor ID number, FG numbers, etc are moved to hdac_device, and they
are accessed like codec->core.addr. The basic verb exec functions are
moved, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few basic codes for communicating over HD-audio bus are moved to
struct hdac_bus now. It has only command and get_response ops in
addition to the unsolicited event handling.
Note that the codec-side tracing support is disabled temporarily
during this transition due to the code shuffling. It will be
re-enabled later once when all pieces are settled down.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define the common hd-audio driver and device types to bind over
snd_hda_bus_type publicly. This allows to implement other type of
device and driver code over hd-audio bus.
Now both struct hda_codec and struct hda_codec_driver inherit these
new struct hdac_device and struct hdac_driver, respectively.
The bus registration is done in subsys_initcall() to assure it
before any other driver registrations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dai-link params for codec-codec links were fixed. The fixed
link between codec and another chip which may be another codec,
baseband, bluetooth codec etc may require run time configuaration
changes. This change provides an optional alsa control to select
one of the params from a list of params.
Signed-off-by: Nikesh Oswal <nikesh@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dapm_kcontrol_codec() is a extremely simple function and inlining it
typically results in less code than necessary for calling the non-inlined
version of the function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds the IRQ function support of rt5670. We use a flag
named dev_gpio in platform data to inform codec driver if the IRQ
function is used or not. Also, we export rt5670_set_jack_detect
for machine driver to pass the jack point.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A selection of changes for v4.1 so far. The main things are:
- Move of jack registration to the card where it belongs.
- Support for DAPM routes specified by both the machine driver and DT.
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Merge tag 'asoc-v4.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Changes for v4.1
A selection of changes for v4.1 so far. The main things are:
- Move of jack registration to the card where it belongs.
- Support for DAPM routes specified by both the machine driver and DT.
There are no users of snd_soc_jack_new() left and new users should use
snd_soc_card_jack_new() instead. So remove the function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Jacks are typically card level elements, but are currently registered with a
CODEC. When it was originally introduced snd_soc_jack_new() took a
snd_soc_card as its parameter, but at that time DAPM was only implemented at
the CODEC level and there was only one CODEC per card. This made it clear
which CODEC to use for the jack DAPM operations. But the multi-component
patchset added support for having multiple CODECs per card and with it the
API was updated to register jacks with a specific CODEC instance instead.
Subsequently DAPM support at the card level has been introduced, but the
snd_soc_jack_new() API has so remained unchanged.
This leaves us with the issue that the DAPM pins that are managed by the
jack detection logic usually are part of the card DAPM context but are
accessed through a CODEC DAPM context. Currently this works fine, but might
break in the future if we take a more hierarchical approach to DAPM
contexts.
Furthermore with componentization progressing systems that do not register
a snd_soc_codec might appear, while these system may still want to able to
register a jack.
This patch addresses these issues by adding a new function called
snd_soc_card_jack_new() that can be used to register jacks with the card
rather than a CODEC.
This new function is mostly identical to snd_soc_jack_new() except that it
additionally allows to directly specify the DAPM pins associated with the
jack. This was done since most users of snd_soc_jack_new() typically call
snd_soc_jack_add_pins() right after it, which is not necessary with the new
API and allows to reduce the amount of boiler plate code.
The old snd_soc_jack_new() is re-implemented as a wrapper around
snd_soc_card_jack_new().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch fix spelling typo found in alsa-driver-api.xml.
It is because this file is generated from comments in source files,
I have to fix source files.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revive snd_device_disconnect() again so that it can be called from the
individual driver. This time, HD-audio will need it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current helper functions, snd_soc_of_parse_audio_simple_widgets()
and snd_soc_of_parse_audio_routing(), set dapm_widgets and dapm_routes
without caring if they are already set by using build-in widgets and
routes in the card driver. So there could be one of them, build-in one
or Device Tree one, overrided by the other depending on which one was
assigned later.
This patch adds an extra pair of dapm_widgets and dapm_routes for DT
use only so as to prevent unexpected overriding.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA core with commit 257f8cce5d - "ALSA: pcm: Allow nonatomic trigger
operations" allows trigger ops to implemented as nonatomic. For ASoC, we can
specify this in dailinks and is updated while snd_pcm is created
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a helper to set pcm format directly from params
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
can be removed without breaking git-bisect now
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce more generic .get_time_info to retrieve
system timestamp and audio timestamp in single routine.
Backwards compatibility is preserved with same functionality
as with .wall_clock method (to be removed in following commits
to avoid breaking git bisect)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audio timestamps can be extracted from sample counters, wall clocks,
PHC clocks (Ethernet AVB), on-demand synchronized snapshots. This
patch provides the ability to report timestamping capabilities, select
timestamp types and retrieve timestamp accuracy, if supported.
Details can be found in Documentations/sound/alsa/timestamping.txt
This functionality is introduced by reclaiming the reserved_aligned
field introduced by commit9c7066aef4a5eb8e4063de28f06c508bf6f2963a
in snd_pcm_status to provide userspace with selection/query capabilities.
Additional driver_tstamp and audio_tstamp_accuracy fields are also added.
snd_pcm_mmap_status remains a read-only structure with only
the audio timestamp value accessible from user space. The selection
of audio timestamp type is done through snd_pcm_status only
This commit does not impact ABI and does not impact the default
behavior. By default audio timestamp is aligned with hw_pointer and
reports the DMA position. Backwards compatibility is handled by using
the HDAudio wall clock for playback and the hw_ptr for all other
cases.
For timestamp selection a new STATUS_EXT ioctl is introduced with
read/write parameters. Alsa-lib will be modified to make use of
STATUS_EXT.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The autoload lock became already superfluous due to the recent rework
of autoload code. Let's drop them now. This allows us to simplify a
few codes nicely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch moves the driver object initialization and allocation to
each driver's module init/exit code like other normal drivers. The
snd_seq_driver struct is now published in seq_device.h, and each
driver is responsible to define it with proper driver attributes
(name, probe and remove) with snd_seq_driver specific attributes as id
and argsize fields. The helper functions snd_seq_driver_register(),
snd_seq_driver_unregister() and module_snd_seq_driver() are used for
simplifying codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use const string pointer instead of copying the id string to each
object. Also drop the status and list fields of snd_seq_device struct
that are no longer used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've used the old house-made code for binding the sequencer device
and driver. This can be far better implemented with the standard
bus nowadays.
This patch refactors the whole sequencer binding code with the bus
/sys/bus/snd_seq. The devices appear as id-card-device on this bus
and are bound with the drivers corresponding to the given id like the
former implementation. The module autoload is also kept like before.
There is no change in API functions by this patch, and almost all
transitions are kept inside seq_device.c. The proc file output will
change slightly but kept compatible as much as possible.
Further integration works will follow in later patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use generic snapshot of trigger_tstamp if low-level driver or
hardware can get a more precise value for better audio/system time
synchronization.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nothing too exciting here yet, a small optimization for DAPM from
Lars-Peter and a few small bits and pieces for drivers but nothing
that really stands out.
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Merge tag 'asoc-v3.19-rc2' into asoc-linus
ASoC: Updates for v3.20
Nothing too exciting here yet, a small optimization for DAPM from
Lars-Peter and a few small bits and pieces for drivers but nothing
that really stands out.
# gpg: Signature made Tue 30 Dec 2014 00:15:48 HKT using RSA key ID 5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: key AF88CD16: no public key for trusted key - skipped
# gpg: key AF88CD16 marked as ultimately trusted
# gpg: key 5621E907: no public key for trusted key - skipped
# gpg: key 5621E907 marked as ultimately trusted
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
For assigning sysfs entries for a card device from the driver,
introduce a new helper function, snd_card_add_dev_attr(). In this
way, we can avoid the possible race between the device registration
and the sysfs addition / removal.
The driver can pass a new attribute group to add freely. This has to
be called before snd_card_register().
Currently, up to two extra groups can be added. More than that, it'll
return an error.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More updates for v3.20:
- Lots of refactoring from Lars-Peter Clausen, moving drivers to more
data driven initialization and rationalizing a lot of DAPM usage.
- Much improved handling of CDCLK clocks on Samsung I2S controllers.
- Lots of driver specific cleanups and feature improvements.
- CODEC support for TI PCM514x and TLV320AIC3104 devices.
- Board support for Tegra systems with Realtek RT5677.
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Merge tag 'asoc-v3.20-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.20
More updates for v3.20:
- Lots of refactoring from Lars-Peter Clausen, moving drivers to more
data driven initialization and rationalizing a lot of DAPM usage.
- Much improved handling of CDCLK clocks on Samsung I2S controllers.
- Lots of driver specific cleanups and feature improvements.
- CODEC support for TI PCM514x and TLV320AIC3104 devices.
- Board support for Tegra systems with Realtek RT5677.
Conflicts:
sound/soc/intel/sst-mfld-platform-pcm.c
Instead of calling device_create_file() manually, assign the static
attribute group entries at the device registration. This simplifies
the error handling and avoids the possible races.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the device is no longer hidden but embedded into each component,
we no longer need snd_get_device(). Let's drop it and relevant codes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that all callers have been replaced with
snd_device_register_for_dev(), let's drop the obsolete device
registration code and concentrate only on the code handling struct
device directly. That said,
- remove the old snd_device_register(),
- rename snd_device_register_for_dev() with snd_device_register(),
- drop superfluous arguments from snd_device_register(),
- change snd_unregister_device() to pass the device pointer directly
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like previous patches, this one embeds the struct device into struct
snd_compr. As the dev field wasn't used beforehand, it's reused as
the new device struct.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like previous patches, this changes the device management for rawmidi,
embedding the struct device into struct snd_rawmidi. The required
change is more or less same as hwdep device.
The currently unused dev field is reused as the new embedded struct
field now.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like previous patches, at this time we embed the struct device into
PCM object. However, this needs a bit more caution: struct snd_pcm
doesn't own one device but two, for both playback and capture! Thus
not struct snd_pcm but struct snd_pcm_str object contains the device.
Along with this change, pcm->dev field is dropped for avoiding
confusion. It was meant to point to a non-standard parent. But,
since now we can touch each struct device directly, we can manipulate
the parent field easily there, too.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous patch, this one embeds the device object into hwdep
object. For a proper object lifecycle, it's freed in the release
callback.
This also allows us to create sysfs entries via passing to the groups
field of the device without explicit function calls. Since each
driver can see the device and touch its groups field directly, we
don't need to delegate in hwdep core any longer. So, remove the
groups field from snd_hwdep, and let the user (in this case only
hda_hwdep.c) modify the device groups.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch embeds a struct device for the control device into the card
object and avoid the device creation at registration time.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new helper function snd_device_initialize() to initialize
the device object for sound devices.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of open-coding the search over the control file loop, provide
a helper function for the preferred subdevice assigned to the current
process.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for the further work on embedding struct
device into each sound device instance. It changes
snd_register_device*() helpers to receive the device object directly
for skipping creating a device there.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Protect the call with a mutex, as this may be called in parallel
(either from the PCM rate change and the clock change).
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define snd_ak4114_suspend() and snd_ak4114_resume() functions to
handle PM properly, stopping and restarting the work at PM.
Currently only ice1712/juli.c deals with the PM and ak4114, so fix the
calls there appropriately.
The same PM functions are defined in ak4113.c, too, although they
aren't currently called yet (ice1712/quartet.c may be enhanced to
support PM later).
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When ak4114 work calls its callback and the callback invokes
ak4114_reinit(), it stalls due to flush_delayed_work(). For avoiding
this, control the reentrance by introducing a refcount. Also
flush_delayed_work() is replaced with cancel_delayed_work_sync().
The exactly same bug is present in ak4113.c and fixed as well.
Reported-by: Pavel Hofman <pavel.hofman@ivitera.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Pavel Hofman <pavel.hofman@ivitera.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add helper functions to allow drivers to specify several disjoint
ranges for a variable. In particular, there is a codec (PCM512x) that
has a hole in its supported range of rates, due to PLL and divider
restrictions.
This is like snd_pcm_hw_constraint_list(), but for ranges instead of
points.
Signed-off-by: Peter Rosin <peda@axentia.se>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
make the sta32x driver usable with device tree configs. Code is heavily based
on the sta350 driver.
Signed-off-by: Thomas Niederprüm <niederp@physik.uni-kl.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
The user-control put and get handlers as well as the tlv do not protect against
concurrent access from multiple threads. Since the state of the control is not
updated atomically it is possible that either two write operations or a write
and a read operation race against each other. Both can lead to arbitrary memory
disclosure. This patch introduces a new lock that protects user-controls from
concurrent access. Since applications typically access controls sequentially
than in parallel a single lock per card should be fine.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
do_posix_clock_monotonic_gettime() is a leftover from the initial
posix timer implementation which maps to ktime_get_ts().
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more updates from the last week of development, nothing too
exciting. Highlights include:
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers.
- New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677.
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Merge tag 'asoc-v3.16-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.16
A few more updates from the last week of development, nothing too
exciting. Highlights include:
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers.
- New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677.
This is useful if we have a pointer to a DAPM context and know that it is a
CODEC or platform DAPM context and want to get a pointer to the CODEC or
platform.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit f025d3b9c6 ("ASoC: jack: Add support for GPIO descriptor defined
jack pins") caused build error when CONFIG_GPIOLIB is not set:
sound/include/sound/soc.h:470: multiple definition of `snd_soc_jack_add_gpiods'
sound/soc/soc-core.o:sound/include/sound/soc.h:470: first defined here
make[2]: *** [sound/soc/snd-soc-core.o] Error 1
make[2]: *** Waiting for unfinished jobs....
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Fix this by marking snd_soc_jack_add_gpiods() as static inline in soc.h.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Clarify struct snd_soc_jack_gpio documentation for the idx and name fields.
Because name is passed as connection ID to gpiod_get_index() when using GPIO
descriptor defined jack pins it is not only used as a label in debugfs but
also as function name lookup in systems that support functions names for
GPIOs.
Clarify also idx since the index is within the function of the GPIO consumer
device and not within the device itself only.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make including the omap-pcm.h outside sound/soc/omap more convenient.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow jack GPIO pins be defined also using GPIO descriptor-based interface
in addition to legacy GPIO numbers. This is done by adding two new fields to
struct snd_soc_jack_gpio: idx and gpiod_dev.
Legacy GPIO numbers are used only when GPIO consumer device gpiod_dev is
NULL and otherwise idx is the descriptor index within the GPIO consumer
device.
New function snd_soc_jack_add_gpiods() is added for typical cases where all
GPIO descriptor jack pins belong to same GPIO consumer device. For other
cases the caller must set the gpiod_dev in struct snd_soc_jack_gpio before
calling snd_soc_jack_add_gpios().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch does basic GPIO descriptor conversion to soc-jack. Even the GPIOs
are still passed and requested using legacy GPIO numbers the driver
internals are converted to use GPIO descriptor API.
Motivation for this is to prepare soc-jack so that it will allow registering
jack GPIO pins using both GPIO descriptors and legacy GPIO numbers.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Lots of cleanup work going on in the core this release but very little
visible to external users except for the new drivers that have been
added.
- Support for specifying aux CODECs in DT.
- Removal of the deprecated mux and enum macros.
- More moves towards full componentisation.
- Removal of some unused I/O code.
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers.
- Several drivers exposed directly in Kconfig for use with simple-card.
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
ST STA350.
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Merge tag 'asoc-v3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.16
Lots of cleanup work going on in the core this release but very little
visible to external users except for the new drivers that have been
added.
- Support for specifying aux CODECs in DT.
- Removal of the deprecated mux and enum macros.
- More moves towards full componentisation.
- Removal of some unused I/O code.
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers.
- Several drivers exposed directly in Kconfig for use with simple-card.
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
ST STA350.
This patch adds DVC (Digital Volume Controller)
support which is member of CMD unit.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This was initially removed in commit 6423c1875 ("ASoC: Remove runtime field from
DAI"), but was, presumably by accident, brought back in commit f0fba2ad1 ("ASoC:
multi-component - ASoC Multi-Component Support"). But has never been
initialized to anything but NULL ever since. This commit removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit commit de9ba98b6d ("ASoC: dapm: Make widget power register settings more
flexible") added generic support for on_val/off_val in the DAPM core. With this
in place there is no need anymore for having a special event callback for
SND_SOC_DAPM_REG() widgets.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now, all platform is using new style rsnd_dai_platform_info.
Keeping compatibility is no longer needed.
We can cleanup code.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit d191bd8de8 ("ASoC: snd_soc_codec includes snd_soc_component") removed the
last user of the num_dai field. Also remove the field itself.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The global card list was removed in commit b19e6e7b7 ("ASoC: core: Use driver
core probe deferral"). The 'list' field of the snd_soc_card struct has been
unused since then. This patch removes the field.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support") added
a per card list that keeps track of all the DAIs that have been registered with
the card, but the list has never been used. This patch removes it again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Cirrus Logic Low Power Stereo I2C CODEC
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for RPDNEN, NSHHPEN, BRIDGOFF, CPWMEN and PNDLSL, and add DT
bindings to access them.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the Realtek ALC5645 codec driver. It is the base
version that because the jack detect function is not implemented to
it, the headphone and AMIC1 are not workable. We will fill up the
further functions later.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data
for DSP modules
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for specifying auxiliary codecs and
codec configuration via device tree phandles.
This change adds new fields to snd_soc_aux_dev and snd_soc_codec_conf
and adds support for the changes to SoC core methods.
Signed-off-by: Pavel Machek <pavel@ucw.cz>
Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Adds struct device_node **bitclkmaster and struct device_node **framemaster
function parameters. With the new syntax bitclock-master and frame-master
properties can explicitly indicate the dai-link bit-clock and frame masters
with a phandle. This patch also makes the minimal changes to simple-card
for it to work with the updated snd_soc_of_parse_daifmt(). Simple-card appears
to be the only user of snd_soc_of_parse_daifmt() for now.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
After moving the IO layer inside ASoC to the component level we can now easily
move the standard control helpers also to the component level. This allows to
reuse the same standard helper control implementations for other components.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We currently have two very similar IO abstractions in ASoC, one for CODECs, the
other for platforms. Moving this to the component level will allow us to unify
those two. It will also enable us to move the standard kcontrol helpers as well
as DAPM support to the component level.
The new component level abstraction layer is primarily build around regmap.
There is a per component pointer for the regmap instance for the underlying
device. There are four new function snd_soc_component_read(),
snd_soc_component_write(), snd_soc_component_update_bits() and
snd_soc_component_update_bits_async(). They have the same signature as their
regmap counter-part and will internally forward the call one-to-one to regmap.
If the component it not using regmap it will fallback to using the custom IO
callbacks. This is done to be able to support drivers that haven't been
converted to regmap yet, but it is expected that this will eventually be removed
in the future once all component drivers have been converted to regmap.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The CODEC's write callback can return a negative error code, make sure to pass
that on correctly.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify error handling and remove repetitive (and rarely executed) code
for unregistration by providing a devm_snd_soc_register_platform()
platform.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This platform data member is unused, so remove it.
Signed-off-by: Alexander Stein <alexanders83@web.de>
Acked-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This should obviously be snd_soc_platform * and not snd_soc_codec *
Fixes: f6272ff8a5 ("ASoC: Add snd_soc_kcontrol_platform() helper function")
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since there are no users left, we can remove the deprecated ENUM and MUX macros
which are just alias for other macros.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The TI STA350 is an integrated 2.1-channel power amplifier that is
controllable over I2C. This patch adds an ASoC driver for it.
At a glance, this chip is very similar to the STA320 for which a driver
already exists. In details, however, the register maps contain subtle
differences which made a whole new driver easier to write and maintain.
[daniel@zonque.org: cleanups, DT property rework, rebased on asoc-next]
Signed-off-by: Sven Brandau <brandau@gmx.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Change reg's type from unsigned short to unsigned int. So that we can use
32 bits reg value in snd_soc_update_bits.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is an increasing amount of code that is very similar between platforms,
CODECS and other components. Making platforms a component will allow us to
share this code. For now the patch just adds component and component_driver
fields to the platform and platform_driver structs and registers the platform as
a component. Followup patches will be used to consolidate code between the
different types of components.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_unregister_component() takes the parent device of the component as a
parameter and then looks up the component based on this. This is a problem if
multiple components are registered for the same parent device. Currently drivers
do not do this, but some drivers register a CPU DAI component and a platform for
the same parent device. This will become a problem once platforms are also made
components. To make sure that snd_soc_unregister_component() will not
accidentally unregister the platform in such a case only consider components
that were registered with snd_soc_register_component(). This is only meant as
short term stopgap solution to be able to continue componentisation. Long term
we'll need something different.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are no ASoC drivers left that use them and new drivers are supposed to use
regmap for this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For platform controls snd_kcontrol_chip() currently returns a pointer to the
platform that registered the control. With the upcoming consolidation of
platform and CODEC controls this will change. Prepare for this by introducing
the snd_soc_kcontrol_platform() helper function that will hide the
implementation details of how the platform for a control can be obtained. This
will allow us to change this easily in the future.
The patch also updates all platforms to use this new helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For CODEC controls snd_kcontrol_chip() currently returns a pointer to the
CODEC that registered the control. With the upcoming consolidation of
platform and CODEC controls this will change. Prepare for this by introducing
the snd_soc_kcontrol_codec() helper function that will hide the implementation
details of how the CODEC for a control can be obtained. This will allow us to
change this easily in the future.
The patch also updates all CODEC drivers to use the new helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add the default regmap I/O setting to snd_soc_register_codec() while
the CODEC is initialising, which will be called by CODEC driver device
probe(), and then we can make XXX_set_cache_io() go away entirely from each
CODEC ASoC probe.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Separate HW initialization from device creation.
This is needed for suspend/resume support.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more updates from last week - use of the tdm_slot mapping from
Xiubo plus a few smaller fixes and cleanups.
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Merge tag 'asoc-v3.15-4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Final updates for v3.15 merge window
A few more updates from last week - use of the tdm_slot mapping from
Xiubo plus a few smaller fixes and cleanups.
The 'of_' is not appropriate here for there hasn't any DT parsing.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A few more updates for the merge window:
- Fixes for the simple-card DAI format DT mess.
- A new driver for Cirrus cs42xx8 devices.
- DT support for a couple more devices.
- A revert of a previous buggy fix for soc-pcm, plus a few more fixes
and cleanups.
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Merge tag 'asoc-v3.15-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.15
A few more updates for the merge window:
- Fixes for the simple-card DAI format DT mess.
- A new driver for Cirrus cs42xx8 devices.
- DT support for a couple more devices.
- A revert of a previous buggy fix for soc-pcm, plus a few more fixes
and cleanups.
Add a helper function to cast back from a component struct to the CODEC struct
it is embedded in. This is useful in situations where we know that a certain
component is a CODEC and want to get access to some CODEC specific properties.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is mostly a few additional fixes from Lars-Peter, a new driver and
cleaning up a git failure with merging the Intel branch (combined with
an xargs failure to pay attention to error codes). The history lists a
bunch of additional commits for the branch but the content of those
commits is actually present already but not recorded in history due to
git failing. Unfortunately xargs is used in the merge script and it
doesn't do a good job of noticing errors from the commands it invokes.
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Merge tag 'asoc-v3.15-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.15
This is mostly a few additional fixes from Lars-Peter, a new driver and
cleaning up a git failure with merging the Intel branch (combined with
an xargs failure to pay attention to error codes). The history lists a
bunch of additional commits for the branch but the content of those
commits is actually present already but not recorded in history due to
git failing. Unfortunately xargs is used in the merge script and it
doesn't do a good job of noticing errors from the commands it invokes.
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
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Merge tag 'asoc-v3.15' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.15
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
Now that all users have been converted to regmap and the config.reg_bits
and config.val_bits can be setted by each user through regmap core API.
So these two params are redundant here.
Since the only control type that left is SND_SOC_REGMAP, so remove it. Drop
the control params and add struct regmap *regmap to simplify the code.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit 83fc3bc095.
sh-specific "CCR" and "CCR2" have been prefixed by "SH_" in commit
a5f6ea29f9 ('sh: prefix sh-specific "CCR" and
"CCR2" by "SH_"').
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that every DAI has a component we can track the DAIs on a per component
basis. This simplifies the DAI lookup when we are only interested in DAIs of a
specific component and also makes it possible to have multiple components with
the same parent device and also register DAIs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In preparation for componentization move the ignore_pmdown_time field from the
snd_soc_codec struct to the snd_soc_component struct. Set it to true for non
CODEC components for now.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no reason why active count tracking should only be done for CODECs but
not for other components. Moving the active count from the snd_soc_codec struct
to the snd_soc_component struct reduces the differences between CODECs and other
components and will eventually allow component to component DAI links (Which is
a prerequisite for converting CODECs to components).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Keep track of which component registered a DAI. We'll need this as
componentization progresses.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of directly checking the 'active' field of the CODEC struct add a new
helper function that will return either true or false depending on whether the
CODEC is active. This will make the migration to the component level easier.
The patch also updates all CODEC drivers that check the active attribute to use
the new helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have the same code that increments and decrements the active field of the
various PCM runtime components (all with the same bugs). Factor this out into
common helper functions.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For CODEC to CODEC links we should only immediately power down if both CODECs
are configured to ignore the power down delay. Factor the logic for this
into a helper function that can be used for both compressed and normal PCMs.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
R-Car sound has SCU unit which has SRC/CTU/MIX/DVC,
and current rsnd driver has scu.c and scu module.
Current scu.c has SRC support only.
My first concept was control these feature on scu.c
but, it become difficult and un-understandable now.
This patch rename scu to src
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rsnd driver is assuming that SCU/SRU ID is
same as SSIU/SSI ID, because Gen1 can't select it.
But, Gen2 can select it.
The SCU/SRU/SSIU/SSI pair depends on the platform.
This patch get correct SCU ID from platform info.
To keep compatible, it still assuming SCU ID = SSI ID
if platform doesn't have info
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
R-Car sound DAI consists from SSI/SCU/SSIU/SRU...
Current R-Car sound DAI is decided from these settings,
but it is intuitively unclear, and is not good design for DT support.
This patch adds new rsnd_dai_platform_info to solve this issue.
But now, many platform is using this driver without
rsnd_dai_platform_info.
So, this patch still supports DAI settings via SSI to keep compatible.
It will be removed in next Linux version.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We now no longer have any users of hw_read() in the kernel so remove the
code in order to prevent any new users being added. Users should be using
regmap.
Signed-off-by: Mark Brown <broonie@linaro.org>
The hda codecs all use this struct and, with an HDA platform driver,
will be able to be built without PCI.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MUXs and virtual MUXs are almost identical, the only difference is that for
virtual MUX there is no hardware backing register in which setting is stored.
This patch adds code, which is similar to what we already do for DAPM mixer
controls to support virtual mixer controls, to DAPM enum controls. The new code
will check if the enum does a hardware backing register and skip over reading
and writing to the register if it has not. This allows us to use the same code
path for both MUXs and virtual MUXs and a lot of nearly identical code can be
removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
With the upcoming consolidation of normal MUXs and virtual MUXs we need to be
able to distinguish between enums with and without a backing register at the
enum level. Use the same approach as used for virtual mixer controls by setting
the reg field of the enum to SND_SOC_NOPM for enums without a backing register.
This patch adds a set of helper macros that can be used to define such enums.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
MUXs and value MUXs are almost identical, the only difference is that a value
MUX uses a look-up table to map from the selected control item to a register
value, while MUXs use a direct mapping. This patch uses
snd_soc_enum_item_to_val() and snd_soc_enum_val_to_item(), which where earlier
introduced during the consolidation of enum and value enum controls, to hide
this difference. This allows us to use the same code path for both MUXs and
value MUXs and a lot of nearly duplicated code can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The implementations for enum and value enum controls are almost identical. The
only difference is that the value enum uses an additional look-up table to map
the control value to the register value, while the enum control uses a direct
mapping. Enums and value enums can easily be distinguished at runtime, for value
enums the values field of the snd_soc_enum struct contains the look-up table,
while for enums it is NULL. This patch adds two new small helper functions
called snd_soc_enum_item_to_val() and snd_soc_enum_val_to_item() which map
between register value and control item. If the items field of the snd_soc_enum
struct is NULL the function will do a direct mapping otherwise they'll use the
look-up table to do the mapping. Using these small helper functions it is
possible to use the same kcontrol handlers for both enums and value enums. The
functions are added a inline functions in soc.h so they can also be used by the
DAPM code to accomplish similar consolidation.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
A few code cleanups and optimizations. In addition, drop
snd_device_disconnect() that isn't used at all, and drop the return
values from snd_device_free*().
Another slight difference by this change is that now the device state
will become always SNDRV_DEV_REGISTERED no matter whether dev_register
ops is present or not. It's for better consistency. There should be
no impact for the current tree, as the state isn't checked.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Basically, the device type specifies the priority of the device to be
registered / freed, too. However, the priority value isn't well
utilized but only it's checked as a group. This results in
inconsistent register and free order (where each of them should be in
reversed direction).
This patch simplifies the device list management code by simply
inserting a list entry at creation time in an incremental order for
the priority value. Since we can just follow the link for register,
disconnect and free calls, we don't have to specify the group; so the
whole enum definitions are also simplified as well.
The visible change to outside is that the priorities of some object
types are revisited. For example, now the SNDRV_DEV_LOWLEVEL object
is registered before others (control, PCM, etc) and, in return,
released after others. Similarly, SNDRV_DEV_CODEC is in a lower
priority than SNDRV_DEV_BUS for ensuring the dependency.
Also, the unused SNDRV_DEV_TOPLEVEL, SNDRV_DEV_LOWLEVEL_PRE and
SNDRV_DEV_LOWLEVEL_NORMAL are removed as a cleanup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like PCM, allow hwdep to be assigned to a different parent device
than the card. It'll be used for the HD-audio codec device in the
later patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling each time device_create_file(), create the groups
of sysfs attribute files at once in a normal way. Add a new helper
function, snd_get_device(), to return the associated device object,
so that we can handle the sysfs addition locally.
Since the sysfs file addition is done differently now,
snd_add_device_sysfs_file() helper function is removed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For some CPU/CODEC DAI devices the slot information maybe needed. This
patch adds the slot information parsing for simple-card driver.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For some CPU/CODEC DAI devices the TDM slot infomation maybe needed. This
patch adds the slot parsing from DT supports.
TDM slot properties:
dai-tdm-slot-num : Number of slots in use.
dai-tdm-slot-width : Width in bits for each slot.
For instance:
dai-tdm-slot-num = <2>;
dai-tdm-slot-width = <8>;
And for each spcified driver, there could be one .of_xlate_tdm_slot_mask()
to specify a explicit mapping of the channels and the slots. If it's absent
the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the
tx and rx masks.
For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit
for an active slot as default, and the default active bits are at the LSB of
the masks.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now that all users have been converted to regmap we can eliminate the ASoC
level wrapper for I2C I/O reducing the amount of duplicated functionality.
Signed-off-by: Mark Brown <broonie@linaro.org>
'snd_rawmidi_transmit_reset()' and 'snd_rawmidi_receive_reset()' are declared
but not defined.
This state has been continue over 10 years. So let us remove them.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We will often call sync after several functions that require the DAPM
mutex to be held. Rather than release and immediately relock the mutex
provide an unlocked function for this situation.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acquiring the DAPM mutex is necessary before using several DAPM
functions and dereference is quite ugly. This patch provides a helper
function to simplify this.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_soc_dapm_xxxx_pin all require the dapm_mutex to be held when
they are called as they edit the dirty list, however very few of the
callers do so.
This patch adds unlocked versions of all the functions replacing the
existing implementations with one that holds the lock internally. We
also fix up the places where the lock was actually held on the caller
side.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Since these macros are supposed to be used for decalring const
objects, let's add the const modifier there.
The doubled const appearing in usages will be cleaned by later
patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The name "max" in struct soc_enum is rather confusing since it
actually takes the number of items. With "max", one might try to
assign (nitems - 1) value.
Rename the field to a more appropriate one, "items", which is also
used in struct snd_ctl_elem_info, too.
This patch also rewrites some code like "if (x > e->nitems - 1)" with
"if (x >= e->nitems)". Not only the latter improves the readability,
it also fixes a potential bug when e->items is zero.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
All ASoC CODEC drivers that use SPI have now been converted to use regmap
so we can delete SND_SOC_SPI, preventing any new users being added.
Signed-off-by: Mark Brown <broonie@linaro.org>
Using __bitwise and typedefs for the attributes of snd_device struct
isn't so useful, and rather it worsens the readability. Let's drop
them and use the straightforward enum.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() & co as much as possible. If not available (no device
assigned at the calling point), use pr_xxx() helpers instead.
For simplicity, introduce new helpers for pcm stream, pcm_err(), etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop the own refcount but use the standard device refcounting via
get_device() and put_device(). Introduce a new completion to snd_card
instead of the wait queue for syncing the last release, which is used
in snd_card_free().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As prepared in the previous patch, we are ready to create a device
struct for the card object in snd_card_create() now. This patch
changes the scheme from the old style to:
- embed a device struct for the card object into snd_card struct,
- initialize the card device in snd_card_create() (but not register),
- registration is done in snd_card_register() via device_add()
The actual card device is stored in card->card_dev. The card->dev
pointer is kept unchanged and pointing to the parent device as before
for compatibility reason.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a part of preliminary works for modernizing the ALSA device
structure. So far, we set card->dev at later point after the object
creation. Because of this, the core layer doesn't always know which
device is being handled before it's actually registered, and it makes
impossible to show the device in error messages, for example. The
first goal is to achieve a proper struct device initialization at the
very beginning of probing.
As a first step, this patch introduces snd_card_new() function (yes
there was the same named function in the very past), in order to
receive the parent device pointer from the very beginning.
snd_card_create() is marked as deprecated.
At this point, there is no functional change other than that. The
actual change of the device creation scheme will follow later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last argument, name, of snd_oss_register_device() is nowhere
referred in the function in the current code. Let's drop it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds snd_soc_of_parse_audio_simple_widgets() and supports
below style of widgets name on DT:
"template-wname", "user supplied wname"
For instance:
simple-audio-widgets =
"Microphone", "Microphone Jack",
"Line", "Line In Jack",
"Line", "Line Out Jack",
"Headphone", "Headphone Jack",
"Speaker", "Speaker External";
The "template-wname" currently includes: "Microphone", "Line", "Headphone"
and "Speaker".
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform data structure contains information which is used only by
the driver, and the driver allocates platform information fields which
are of no use.
Move the driver specific data to a new private structure and cleanup
the platform data structure.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a macro for signed mixer with two registers and tlv array.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some codecs use signed volume control representation with non standard
register sizes, e.g. 6 or 7 bit signed integers.
This patch adds generic signed register volume control logic to
soc-core. Instead of a fixed width signed register control, this
implementation uses a 'min' value and the signed bit location to translate
it to an absolute volume. Using the 'sign_bit' we can calculate a
correct mask for the register values and translate it back into signed
integers of standard size.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound Gen2 has SRC (= Sampling Rate Converter)
which needs 2 DMAC.
The data path image when you use SRC on Gen2 is
[mem] -> Audio-DMAC -> SRC -> Audio-DMAC-peri-peri -> SSIU -> SSI
This patch support SRC and DMAEnine.
It is tested on R-Car H2 Lager board
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas Chip is supporting multi pin sound,
but the HW setting is very difficult and confusable.
But driver is supporting it halfway.
Remove SYNC option at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This branch is reducing in size for every release since most board-related
changes have started happening in devicetrees now. Still, we have some things
going on here.
* Renesas platforms are still adding a bit more legacy device support, something
that should trail off shortly as they move to full DT.
* We group most defconfig updates into this branch out of old habits
* Removal of legacy OMAP2 platforms over to DT continues, and a handful of old
code is being removed here.
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Merge tag 'boards-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull ARM SoC board updates from Olof Johansson:
"This branch is reducing in size for every release since most
board-related changes have started happening in devicetrees now.
Still, we have some things going on here.
* Renesas platforms are still adding a bit more legacy device
support, something that should trail off shortly as they move to
full DT
* We group most defconfig updates into this branch out of old habits
* Removal of legacy OMAP2 platforms over to DT continues, and a
handful of old code is being removed here"
* tag 'boards-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (94 commits)
ARM: dts: OMAP2: fix interrupt number for rng
ARM: dts: Split omap3 pinmux core device
ARM: dts: Add omap specific pinctrl defines to use padconf addresses
ARM: bcm2835: bcm2835_defconfig updates
ARM: msm_defconfig: Enable restart driver
defconfig: msm_defconfig: Enable CONFIG_ARCH_MSM8974
ARM: msm: Add support for APQ8074 Dragonboard
ARM: exynos_defconfig: Enable S2MPS11 voltage regulator
ARM: tegra: Enable DRM panel support
ARM: shmobile: mackerel: Fix USBHS pinconf entry
ARM: shmobile: Let Koelsch multiplatform boot with Koelsch DTB
ARM: shmobile: Let Lager multiplatform boot with Lager DTB
ARM: shmobile: Remove non-multiplatform Koelsch reference support
ARM: shmobile: Remove non-multiplatform Lager reference support
ARM: shmobile: koelsch-reference: Instantiate clkdevs for SCIF and CMT
ARM: shmobile: lager-reference: Instantiate clkdevs for SCIF and CMT
ARM: shmobile: koelsch-reference: Remove duplicate CCF initialization
ARM: shmobile: lager-reference: Enable multiplaform kernel support
ARM: shmobile: armadillo: Set backlight enable GPIO
ARM: shmobile: Koelsch: add Ether support
...
Conflicts:
arch/arm/mach-omap2/omap_hwmod_2xxx_ipblock_data.c
Currently compressed audio streams are statically routed from the /dev
to the DAI link. Some DSPs can route compressed data to multiple BE DAIs
like they do for PCM data.
Add support to allow dynamically routed compressed streams using the existing
DPCM infrastructure. This patch adds special FE versions of the compressed ops
that work out the runtime routing.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC compressed code needs to call the internal DPCM APIs in order to
dynamically route compressed data to different DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A bit of special care is necessary when creating the intersection of two rate
masks. This comes from the special meaning of the SNDRV_PCM_RATE_CONTINUOUS and
SNDRV_PCM_RATE_KNOT bits, which needs special handling when intersecting two
rate masks. SNDRV_PCM_RATE_CONTINUOUS means the hardware supports all rates in a
specific interval. SNDRV_PCM_RATE_KNOT means the hardware supports a set of
discrete rates specified by a list constraint. For all other cases the supported
rates are specified directly in the rate mask.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Nowadays we have CMA for obtaining the contiguous memory pages
efficiently. Let's kill the old kludge for reserving the memory pages
for large buffers. It was rarely useful (only for preserving pages
among module reloading or a little help by an early boot scripting),
used only by a couple of drivers, and yet it gives too much ugliness
than its benefit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Connect the DAPM graph through each BE DAI link to the componnent(s) on the
other side of the BE DAI link. This allows the graph to be walked on
both sides of the link when graph changes are made.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Provide a quick way to tell if a DAI is a dummy DAI or a regular DAI.
This is for internal DAPM usage only and is used to determine whether to
insert a DAI link connection into the DAPM graph.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such
wont have set a minimum number of playback or capture channels required for BE
DAI registration (to establish supported stream directions).
Force machine drivers to explicitly set whether they support playback and capture
stream directions for every BE DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Not a lot going on framework wise, partly due to Christmas at least in
the case of the work I've been doing, but there's been quite a lot of
cleanup activity going on and the usual trickle of new drivers:
- Update to the generic DMA code to support deferred probe and managed
resources.
- New drivers for BCM2835 (used in Raspberry Pi), Tegra with MAX98090
and Analog Devices AXI I2S and S/PDIF controller IPs.
- Device tree support for the simple card, max98090 and cs42l52.
- Conversion of the Samsung drivers to native dmaengine, making them
multiplatform compatible and hopefully helping keep them more modern
and up to date.
- More regmap conversions, including a very welcome one for twl6040
from Peter Ujfalusi.
- A big overhaul of the DaVinci drivers also from Peter Ujfalusi.
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Merge tag 'asoc-v3.14' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.14
Not a lot going on framework wise, partly due to Christmas at least in
the case of the work I've been doing, but there's been quite a lot of
cleanup activity going on and the usual trickle of new drivers:
- Update to the generic DMA code to support deferred probe and managed
resources.
- New drivers for BCM2835 (used in Raspberry Pi), Tegra with MAX98090
and Analog Devices AXI I2S and S/PDIF controller IPs.
- Device tree support for the simple card, max98090 and cs42l52.
- Conversion of the Samsung drivers to native dmaengine, making them
multiplatform compatible and hopefully helping keep them more modern
and up to date.
- More regmap conversions, including a very welcome one for twl6040
from Peter Ujfalusi.
- A big overhaul of the DaVinci drivers also from Peter Ujfalusi.
This patch adds SRC support to Renesas sound driver.
SRC converts sampling rate between codec <-> cpu.
It needs special codec chip,
or very simple DA/AD converter to use it.
This patch was tested via ak4554 codec,
and supports Gen1 only at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add helpers for obtaining the width of a format directly from params
since this is expected to become a common operation in ASoC.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
* Global
- Kconfig: Mention Renesas ARM SoCs instead of SH-Mobile
* r7s72100 SoC (RZ/A1H) based Genmai Board
- Add Multiplatform support
- Add Reference DT
* r8a7791 (R-Car M2) based Koelsch board
- Add pinctrl_register_mappings() for Koelsch
- Hook up SW30-SW36 on Koelsch
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Add Multiplatform support
- Set .debounce_interval for GPIO keys
- Add SW2 to GPIO keys
- Add Led 6, 7 and 8 support
- Add reference DT
- Enable PFC/GPIO
* r8a7790 (R-Car H2) based Lager board
- Add gpio/fixed regulator for SDHI
- Add SPI FLASH support on QSPI
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Set .debounce_interval for GPIO keys
* r8a7778 (R-Car M1) based Bock-W board
- bockw: remove unused RSND_SSI_CLK_FROM_ADG
- Set .debounce_interval for GPIO keys
- Correct FPGA ioremap area
- Use regulator for MMCIF
* r8a7740 (R-Mobile A1) based Armadillo board
- Correct FSI address size
* sh7374 (SH-Mobile AP4) based Mackerel board
- Use pinconf API to configure pin pull-down
- clk_round_rate() can return a zero to indicate an error
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Merge tag 'renesas-boards-for-v3.14' of git://git.kernel.org/pub/scm/linux/kernel/git/horms/renesas into next/boards
From Simon Horman:
Renesas ARM based SoC board updates for v3.14
* Global
- Kconfig: Mention Renesas ARM SoCs instead of SH-Mobile
* r7s72100 SoC (RZ/A1H) based Genmai Board
- Add Multiplatform support
- Add Reference DT
* r8a7791 (R-Car M2) based Koelsch board
- Add pinctrl_register_mappings() for Koelsch
- Hook up SW30-SW36 on Koelsch
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Add Multiplatform support
- Set .debounce_interval for GPIO keys
- Add SW2 to GPIO keys
- Add Led 6, 7 and 8 support
- Add reference DT
- Enable PFC/GPIO
* r8a7790 (R-Car H2) based Lager board
- Add gpio/fixed regulator for SDHI
- Add SPI FLASH support on QSPI
- Mark GPIO keys as wake-up sources
- Use ->init_late()
- Set .debounce_interval for GPIO keys
* r8a7778 (R-Car M1) based Bock-W board
- bockw: remove unused RSND_SSI_CLK_FROM_ADG
- Set .debounce_interval for GPIO keys
- Correct FPGA ioremap area
- Use regulator for MMCIF
* r8a7740 (R-Mobile A1) based Armadillo board
- Correct FSI address size
* sh7374 (SH-Mobile AP4) based Mackerel board
- Use pinconf API to configure pin pull-down
- clk_round_rate() can return a zero to indicate an error
* tag 'renesas-boards-for-v3.14' of git://git.kernel.org/pub/scm/linux/kernel/git/horms/renesas: (75 commits)
ARM: shmobile: lager: add gpio/fixed regulator for SDHI
ARM: shmobile: bockw: remove unused RSND_SSI_CLK_FROM_ADG
ARM: shmobile: armadillo: fixup FSI address size
ARM: Kconfig: Mention Renesas ARM SoCs instead of SH-Mobile
ARM: shmobile: mackerel: Use pinconf API to configure pin pull-down
ARM: shmobile: Lager:add SPI FLASH support on QSPI
ARM: shmobile: mackerel: clk_round_rate() can return a zero to indicate an error
ARM: shmobile: Add pinctrl_register_mappings() for Koelsch
ARM: shmobile: Use ->init_late() on Lager
ARM: shmobile: Hook up SW30-SW36 on Koelsch
ARM: shmobile: koelsch: mark GPIO keys as wake-up sources
ARM: shmobile: Use ->init_late() on Koelsch
ARM: shmobile: lager: mark GPIO keys as wake-up sources
ARM: shmobile: r7s72100 Genmai Multiplatform
ARM: shmobile: r7s72100 Genmai DT reference C bits
ARM: shmobile: r7s72100 Genmai DT reference DTS bits
ARM: shmobile: Initial r8a7791 and Koelsch multiplatform support
ARM: shmobile: koelsch: set .debounce_interval
ARM: shmobile: lager: set .debounce_interval
ARM: shmobile: bockw: add pin pull-up setting for SDHI
...
Signed-off-by: Olof Johansson <olof@lixom.net>
spear_pcm_request_chan() is almost identical to
dmaengine_pcm_compat_request_channel(), with the exception that the
latter:
a) Assumes that the DAI DMA data is a struct snd_dmaengine_dai_dma_data
pointer rather than some custom type.
b) dma_data->filter_data rather than dma_data should be passed to
snd_dmaengine_pcm_request_channel() as the filter data.
Make minor changes to the SPEAr DAI drivers so that those two conditions
are met. This allows removal of the custom .compat_request_channel().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When running a 32bit kernel the hda_intel driver is still reporting
a 64bit dma_mask if the HW supports it.
From sound/pci/hda/hda_intel.c:
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(64));
else {
pci_set_dma_mask(pci, DMA_BIT_MASK(32));
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32));
}
which means when there is a call to dma_alloc_coherent from
snd_malloc_dev_pages a machine address bigger than 32bit can be returned.
This can be true in particular if running the 32bit kernel as a pv dom0
under the Xen Hypervisor or PAE on bare metal.
The problem is that when calling setup_bdle to program the BLE the
dma_addr_t returned from the dma_alloc_coherent is wrongly truncated
from snd_sgbuf_get_addr if running a 32bit kernel:
static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
size_t offset)
{
struct snd_sg_buf *sgbuf = dmab->private_data;
dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
addr &= PAGE_MASK;
return addr + offset % PAGE_SIZE;
}
where PAGE_MASK in a 32bit kernel is zeroing the upper 32bit af addr.
Without this patch the HW will fetch the 32bit truncated address,
which is not the one obtained from dma_alloc_coherent and will result
to a non working audio but can corrupt host memory at a random location.
The current patch apply to v3.13-rc3-74-g6c843f5
Signed-off-by: Stefano Panella <stefano.panella@citrix.com>
Reviewed-by: Frediano Ziglio <frediano.ziglio@citrix.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
92eba04e4b
(ASoC: rcar: remove RSND_SSI_CLK_FROM_ADG) removed
RSND_SSI_CLK_FROM_ADG, it is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Simon Horman <horms+renesas@verge.net.au>
Add fields to struct snd_dmaengine_pcm_config to allow custom:
- DMA channel names.
This is useful when the default "tx" and "rx" channel names don't
apply, for example if a HW module supports multiple channels, each
having different DMA channel names. This is the case with the FIFOs
in Tegra's AHUB. This new facility can replace
SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME.
- DMA device
This allows requesting DMA channels for a device other than the device
which is registering the "PCM" driver. This is quite unusual, but is
currently useful on Tegra. In much HW, and in Tegra20, each DAI HW
module contains its own FIFOs which DMA writes to. However, in Tegra30,
the DMA FIFOs were split out AHUB HW module, which then routes the data
through a cross-bar, and into the DAI HW modules. However, the current
ASoC driver structure does not expose this detail, and acts as if the
FIFOs are still part of the DAI HW modules. Consequently, the "PCM"
driver is registered with the DAI HW module, yet the DMA channels must
be looked up in the AHUB HW module's device tree node. This new config
field allows that to happen. Eventually, the Tegra drivers will be
reworked to fully expose the AHUB, and this config field can be
removed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since there are more HD-audio compatible codecs, move the definitions
of HD-audio verbs into common header location, include/sound, so that
it can be included cleanly from other drivers than HD-audio driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For many drivers using the generic dmaengine PCM driver one of the few (or the
only) things left to do in the drivers remove function is to unregister the PCM
device. This patch adds a resource managed version of snd_dmaengine_pcm_register()
which makes it possible to simplify the remove function as well as the error
path in the probe function for those drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
MICA/B Single-Ended input selection depends on mica/b config so lets
make the mixer controls for them only show for selected mic's
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch reworks the MICA an MICB config for single-ended or
differential and the selection of which MIC for the single config
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_jack_gpio stuff is currently enabled for CONFIG_GPIOLIB
explicitly with ifdef, and this causes build errors on some drivers
such as:
sound/soc/omap/rx51.c:220:33: error: array type has incomplete element type
Remove ifdef and provide dummy functions for CONFIG_GPIOLIB=n case
instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SND_SOC_DAPM_MUX() doesn't currently initialize the .mask field. This
results in the mux never affecting HW, since no bits are ever set or
cleared. Fix SND_SOC_DAPM_MUX() to use SND_SOC_DAPM_INIT_REG_VAL() to
set up the reg, shift, on_val, and off_val fields like almost all other
SND_SOC_xxx() macros. It looks like this was a "typo" in the fixed
commit linked below.
This makes the speakers on the Toshiba AC100 (PAZ00) laptop work again.
Fixes: de9ba98b6d ("ASoC: dapm: Make widget power register settings more flexible")
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: <stable@vger.kernel.org> # v3.12+
Some SoCs can only work in mono or stereo mode at one time. So if
we let them capture a mono stream while playing a stereo stream,
there might be a problem occur to one of these two streams: double
paced or slowed down.
In soc-pcm.c, we have soc_pcm_apply_symmetry() to apply the rate
symmetry. But we don't have one for channels.
Likewise, we can treat symmetric_rate as a solution for those SoCs
or CODECs which can not handle asymmetrical LRCLK. But it's also
impossible for them to handle asymmetrical BCLK. And accodring to
BCLK = LRCLK * channel number * slot size(fixed or sample bits),
sample bits might also be a problem if they are not using a fixed
slot size.
Thus, this patch applys symmetry for channels and sample bits.
Meanwhile, there might be a race between two substreams if starting
simultaneously. Previously, we only added warning to compalin but
still using conservative way to let it carry on. However, this patch
rejects the second stream with any unmatched parameter to make sure
the first existing stream won't be broken.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The drain and drain_notify callback were blocked by low level driver
until the draining was complete. Due to this being invoked with big
fat mutex held, others ops like reading timestamp, calling pause, drop
were blocked.
So to fix this we add a new snd_compr_drain_notify() API. This would
be required to be invoked by low level driver when drain or partial
drain has been completed by the DSP. Thus we make the drain and
partial_drain callback as non blocking and driver returns immediately
after notifying DSP. The waiting is done while releasing the lock so
that other ops can go ahead.
[ The commit 917f4b5cba was wrongly applied from the preliminary
patch. This commit corrects to the final version.
Sorry for inconvenience! -- tiwai ]
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
CC: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few final updates for v3.13, all driver updates apart from some DPCM
and Coverity fixes which should have minor impact on practical systems.
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Merge tag 'asoc-v3.13-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.13
A few final updates for v3.13, all driver updates apart from some DPCM
and Coverity fixes which should have minor impact on practical systems.
The size of the register cache array is actually 6 instead of 7,
as it caches up to AK4114_REG_INT1_MASK. This resulted in unexpected
access out of array range, although most of them aren't so serious
(just reading one more byte on the stack at snd_ak4114_create()).
Also, the check of cache size was wrongly done by checking with
sizeof() instead of ARRAY_SIZE(). Fixed this together.
(And yes, hardcoded numbers are bad, but I keep the coding style as is
for making it clear what this patch actually does.)
Spotted by coverity among several CIDs, e.g. 711621.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds platform data support for a reset GPIO.
Also uses reset_gpio to toggle reset of the CODEC
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
- Further work on the dmaengine helpers, including support for
configuring the parameters for DMA by reading the capabilities of the
DMA controller which removes some guesswork and magic numbers fromm
drivers.
- A refresh of the documentation.
- Conversions of many drivers to direct regmap API usage in order to
allow the ASoC level register I/O code to be removed, this will
hopefully be completed by v3.14.
- Support for using async register I/O in DAPM, reducing the time taken
to implement power transitions on systems that support it.
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Merge tag 'asoc-v3.13' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.13
- Further work on the dmaengine helpers, including support for
configuring the parameters for DMA by reading the capabilities of the
DMA controller which removes some guesswork and magic numbers fromm
drivers.
- A refresh of the documentation.
- Conversions of many drivers to direct regmap API usage in order to
allow the ASoC level register I/O code to be removed, this will
hopefully be completed by v3.14.
- Support for using async register I/O in DAPM, reducing the time taken
to implement power transitions on systems that support it.
The drain and drain_notify callback were blocked by low level driver untill the
draining was complete. Due to this being invoked with big fat mutex held, others
ops like reading timestamp, calling pause, drop were blocked.
So to fix this we add a new snd_compr_drain_notify() API. This would be required
to be invoked by low level driver when drain or partial drain has been completed
by the DSP. Thus we make the drain and partial_drain callback as non blocking
and driver returns immediately after notifying DSP.
The waiting is done while relasing the lock so that other ops can go ahead.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
CC: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that we can't use gen_pool_*() functions on archs
without CONFIG_GENERIC_ALLOCATOR (resulting in missing symbols), since
linux/genalloc.h doesn't provide dummy functions for all. We'd be
able to fix linux/genalloc.h size, but I take an easier path for
now...
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices have more than just simple TX and RX DMA channels, for example
modern Samsung I2S IPs support a secondary transmit DMA stream which is
mixed into the primary stream during playback. Allow such devices to
specify the names of the channels to be requested in their dma_data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Now it's quite common that an SoC contains its on-chip internal RAM.
By using this RAM space for DMA buffer during audio playback/record,
we can shutdown the voltage for external RAM to save power.
So add new DEV type with iram malloc()/free() and accordingly modify
current default mmap() for the iram circumstance.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current SSI needs RSND_SSI_DEPENDENT flag to
decide dependent/independent mode.
And SCU needs RSND_SCU_USE_HPBIF flag
to decide HPBIF is enable/disable.
But these 2 means same things.
This patch adds new rsnd_scu_hpbif_is_enable()
function, and merges above methods.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow DMA data to be set at probe time for devices that can do that,
avoiding the need to do it every time we start a stream and supporting
non-DT dmaengine users using the helpers.
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently each platform making use the the generic dmaengine PCM driver still
needs to provide a custom snd_pcm_hardware struct which specifies the
capabilities of the DMA controller, e.g. the maximum period size that can be
supported. This patch adds code which uses the newly introduced
dma_get_slave_caps() API to query this information from the dmaengine driver.
The new code path will only be taken if the 'pcm_hardware' field of the
snd_dmaengine_pcm_config struct is NULL.
The patch also introduces a new 'fifo_size' field to the
snd_dmaengine_dai_dma_data struct which is used to initialize the
snd_pcm_hardware 'fifo_size' field and needs to be set by the DAI driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current snd_soc_of_get_dai_name() needs .of_xlate_dai_name()
callback on each component drivers.
But required behavior on almost all these drivers is
just returns its indexed driver's name.
This patch adds this feature as default behavior.
.of_xlate_dai_name() can overwrite it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for RST GPIO and Charge Pump Freq in platform data
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a comment to the trigger function in snd_soc_dai_ops struct about
possible command sequences.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rcar is using rsnd_is_gen1/gen2() to checking its
IP generation, but it needs data mask.
This patch fixes it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for virtual DAPM mixer controls. They are similar to
virtual DAPM enums. There is no hardware register backing the control, so
changing the control's value wont have any direct effect on the hardware. But it
still influences the DAPM graph by causing the path it sits on to be connected
or disconnected. This in turn can cause power changes for some of the widgets on
the DAPM graph, which will then modify the hardware state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ASoC sound driver requires CPU/CODEC drivers for probing,
and each CPU/CODEC has some DAI on it.
Then, "dai name matching" have been used to identify
CPU-CODEC DAI pair on ASoC.
But, the "dai port number matching" is now required from DeviceTree.
The solution of this issue is to replace
the dai port number into dai name.
Now, CPU/CODEC are based on struct snd_soc_component,
and it can care above as common issue.
This patch adds .of_xlate_dai_name callback interface
on struct snd_soc_component_driver,
and snd_soc_of_get_dai_name() which is using .of_xlate_dai_name.
Then, #sound-dai-cells which enables DAI specifier is required
on CPU/CODEC device tree properties.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Codec includes component by this patch,
and component moved to upside of codec
to avoid extra declaration.
Codec dai will be registered via component
by this patch.
Current component register function
is used for cpu, and it is using
dai/dais functions properly to keep
existing cpu dai name.
And now, it will be used from codec also.
But codec driver had been used dais function only
even though it was single dai.
This patch adds new flag which can selects
dai/dais function on component register
function to keep existing codec dai name.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some codec drivers when running in slave mode require that BCLK to sample rate ratio
is explicitly set by the machine driver as it may not be exactly rate * frame size.
Extend the DAI API by adding :-
int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify error handling and remove repetitive (and rarely executed) code
for unregistration by providing a devm_snd_soc_register() card.
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Since with the wider use of devres many drivers are now only calling
snd_soc_unregister_component() in their remove functions providing a
managed version will save a reasonable amount of code.
Signed-off-by: Mark Brown <broonie@linaro.org>
The only cache type left is the flat cache and new other cache types won't be
added since new drivers are supposed to use regmap directly for IO and caching.
This patch removes the snd_soc_cache_ops indirection that was added to support
multiple cache types and modifies the code to always use the flat cache
directly.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The reg_size field is calculated in snd_soc_register_codec() and then used
exactly once in snd_soc_flat_cache_init(). Since it is calculated based on other
fields from the codec struct just move the calculation to
snd_soc_flat_cache_init() and remove the 'reg_size' field from the codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
reg_def_copy was introduced in commit 3335ddca ("ASoC: soc-cache: Use
reg_def_copy instead of reg_cache_default") to keep a copy of the register
defaults around in case the register defaults where placed in the __devinitdata
section. With the __devinitdata section gone we effectivly keep the same data
around twice. This patch removes reg_def_copy and uses reg_cache_default
directly instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No users of snd_soc_bulk_write_raw() are left and new drivers are going to use
regmap directly for this, so the function can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No users of reg_access_defaults are left and new drivers are going to use regmap
for this, so support for it can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Pull media updates from Mauro Carvalho Chehab:
"This series contains:
- Exynos s5p-mfc driver got support for VP8 encoder
- Some SoC drivers gained support for asynchronous registration
(needed for DT)
- The RC subsystem gained support for RC activity LED;
- New drivers added: a video decoder(adv7842), a video encoder
(adv7511), a new GSPCA driver (stk1135) and support for Renesas
R-Car (vsp1)
- the first SDR kernel driver: mirics msi3101. Due to some troubles
with the driver, and because the API is still under discussion, it
will be merged at staging for 3.12. Need to rework on it
- usual new boards additions, fixes, cleanups and driver
improvements"
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (242 commits)
[media] cx88: Fix regression: CX88_AUDIO_WM8775 can't be 0
[media] exynos4-is: Fix entity unregistration on error path
[media] exynos-gsc: Register v4l2 device
[media] exynos4-is: Fix fimc-lite bayer formats
[media] em28xx: fix assignment of the eeprom data
[media] hdpvr: fix iteration over uninitialized lists in hdpvr_probe()
[media] usbtv: Throw corrupted frames away
[media] usbtv: Fix deinterlacing
[media] v4l2: added missing mutex.h include to v4l2-ctrls.h
[media] DocBook: upgrade media_api DocBook version to 4.2
[media] ml86v7667: fix compile warning: 'ret' set but not used
[media] s5p-g2d: Fix registration failure
[media] media: coda: Fix DT driver data pointer for i.MX27
[media] s5p-mfc: Fix input/output format reporting
[media] v4l: vsp1: Fix mutex double lock at streamon time
[media] v4l: vsp1: Add support for RT clock
[media] v4l: vsp1: Initialize media device bus_info field
[media] davinci: vpif_capture: fix error return code in vpif_probe()
[media] davinci: vpif_display: fix error return code in vpif_probe()
[media] MAINTAINERS: add entries for adv7511 and adv7842
...
Add 'playback_only' and 'capture_only' fields that can be used for specifying
that a dai_link has a unidirectional capability.
The motivation for this is for the cases of systems, such as Freescale MX28,
that has two unidirectional DAIs.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The sysfs_registered field was added to the snd_soc_codec struct in commit
f0fba2ad ("ASoC: multi-component - ASoC Multi-Component Support"), but has never
been used.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DAPM context struct has its own field where it stores the pointer to the
DAPM debugfs entry. The debugfs_dapm field in the snd_soc_platform and
snd_soc_codec structs are completely unused and can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The control_type field was used by the core to track which raw IO methods to
use, but when switching to regmap this was no longer necessary and so the last
user of the field was removed in commit be3ea3b9 ("ASoC: Use new register map
API for ASoC generic physical I/O"). The field is now completely unused and can
be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_dapm_new_widgets() works on the ASoC card as a whole not on a specific
DAPM context. The DAPM context that is passed as the parameter is only used to
look up the pointer to the card. This patch updates the signature of
snd_soc_dapm_new_widgets() to take the card directly.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
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Merge tag 'asoc-v3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.12
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
This patch adds generic ac97 reset functions using pincontrol and gpio
parsed from devicetree.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Direct calls to printk_limit() will emit log noise even when CONFIG_SND_DEBUG is not
defined. Add a wrapper macro around printk_limit() that is conditionally defined by
CONFIG_SND_DEBUG.
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move include/sound/tea575x-tuner.h to include/media/tea575x.h and update files that include it.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
Use snd_dmaengine_dai_dma_data for passing the dma parameters from
clients to the pxa pcm lib. This does no functional change, it's just an
intermedia step to migrate the pxa bits over to dmaengine.
The calculation of dcmd is a transition hack which will be removed again
in a later patch. It's just there to make the transition more readable.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_info_enum_ext() and snd_soc_info_enum_double() are almost identical. The
only difference is that snd_soc_info_enum_double() is also able to handle stereo
controls. Using snd_soc_info_enum double() instead of snd_soc_info_enum_ext()
for the SOC_ENUM_EXT control's info callback allows us to remove
snd_soc_info_enum_ext().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The SOC_SINGLE_EXT control has been using snd_soc_info_volsw() for its info
callback since commit 1c433fb ("[ALSA] soc - 0.13 ASoC headers"). The
snd_soc_info_volsw_ext() function has been unused ever since then, so remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds BUSIF support for R-Car sound DMAEngine transfer.
The sound data will be transferred via FIFO which can cover blank time
which will happen when DMA channel is switching.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds DMAEngine transfer on SSI.
But, it transfers sound data from memory to SSI directly
without using HPBIF at this time.
It will be updated soon
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rsnd driver is using struct rsnd_dai_platform_info
so that indicate sound DAI information (playback/capture SSI ID).
But, SSI settings were also required separately.
Thus, platform settings was very un-understandable.
This patch adds dai_id to SSI
settings, and removed rsnd_dai_platform_info.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some devices have the problem that if a internal audio signal source is disabled
the output of the source becomes undefined or goes to a undesired state (E.g.
DAC output goes to ground instead of VMID). In this case it is necessary, in
order to avoid unwanted clicks and pops, to disable any mixer input the signal
feeds into or to active a mute control along the path to the output. Often it is
still desirable to expose the same mixer input control to userspace, so cerain
paths can sill be disabled manually. This means we can not use conventional DAPM
to manage the mixer input control. This patch implements a method for letting
DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable
the control if the path on which the control sits becomes inactive. Userspace
will then only see a cached copy of the controls state. Once DAPM powers the
path up again it will sync the userspace setting with the hardware and give
control back to userspace.
To implement this a new widget type is introduced. One widget of this type will
be created for each DAPM kcontrol which has the auto-disable feature enabled.
For each path that is controlled by the kcontrol the widget will be connected to
the source of that path. The new widget type behaves like a supply widget,
which means it will power up if one of its sinks are powered up and will only
power down if all of its sinks are powered down. In order to only have the mixer
input enabled when the source signal is valid the new widget type will be
disabled before all other widget types and only be enabled after all other
widget types.
E.g. consider the following simplified example. A DAC is connected to a mixer
and the mixer has a control to enable or disable the signal from the DAC.
+-------+
+-----+ | |
| DAC |-----[Ctrl]-| Mixer |
+-----+ : | |
| : +-------+
| :
+-------------+
| Ctrl widget |
+-------------+
If the control has the auto-disable feature enabled we'll create a widget for
the control. This widget is connected to the DAC as it is the source for the
mixer input. If the DAC powers up the control widget powers up and if the DAC
powers down the control widget is powered down. As long as the control widget
is powered down the hardware input control is kept disabled and if it is enabled
userspace can freely change the control's state.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently the DAPM code is limited to only setting or clearing a single bit in a
register to power a widget up or down. This patch extends the DAPM code to be
more flexible in that regard and allow widgets to use arbitrary values to be
used to put a widget in either on or off state.
Since the snd_soc_dapm_widget struct already contains a on_val and off_val field
no additional fields need to be added and in fact the invert field can even be
removed. Also the generated code is slightly smaller.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently we store for each path which control (if any at all) is associated
with that control. But we are only ever interested in the reverse relationship,
i.e. we want to know all the paths a certain control is associated with. This is
currently implemented by always iterating over all paths. This patch updates the
code to keep a list for each control which contains all the paths that are
associated with that control. This improves the run time of e.g.
soc_dapm_mixer_update_power() and soc_dapm_mux_update_power() from O(n) (with n
being the number of paths for the card) to O(1).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The 'value' field is really per control and not per widget. Currently it is only
used for virtual MUXes, which only have one control per widget. So in that case
there is not so much of a difference between whether it is stored per widget or
per control. Moving the 'value' field from the widget to the control will allow
us to use it also for cases where we have more than one control per widget. E.g.
for mixers with multiple input controls.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We use the same 3 lines to get the CODEC for a kcontrol in a quite a few places.
This patch puts them into a common helper function. Having this encapsulated in
a helper function will also make it more easier to eventually change the data
layout of the kcontrol's private data.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The update field of a DAPM context is only assigned while the card's dapm_mutex
is locked, the field is also cleared again while the mutex is stil locked. So
there will only ever be one DAPM context at a time with a non-NULL update field.
So it is safe to move the update field from the DAPM context struct to the card
struct. Doing so will allow further cleanups in this area.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is useful for drivers who want to grab a pointer to
snd_kcontrol outside of the kcontrol callbacks.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
As 1st protype, this patch adds SSI feature on this driver.
But, it is PIO sound playback support only at this point.
The DMA transfer, and capture feature will be supported in the future
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
This patch adds ADG feature which controls sound clock
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
This patch adds SCU feature on this driver.
But, it defines SCU style only, does nothing at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuit is different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2)
(Actually, there are many difference in Generation1 chips)
The main difference between Gen1 and Gen2 are
1) register offset, 2) data path
In order to control Gen1/Gen2 by same method,
this patch adds gen.c.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas R-Car series sound circuit consists of SSI and its peripheral.
But this peripheral circuits are different between
R-Car Generation1 (E1/M1/H1) and Generation2 (E2/M2/H2).
(Actually, there are many difference in Generation1 chips)
Basically, for the future, Renesas R-Car series will use
Gen2 style sound circuit, but driver should care Gen1 also.
The main differences between Gen1 and Gen2 peripheral
are 1) register offset, 2) data path.
This patch adds basic (core) feature for R-Car
series sound driver as prototype
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move HW initialization to separate function to allow using the code without
the v4l parts. This is needed for use in the bttv driver.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
In order to avoid race conditions the assignment of dapm->update should happen
while card->dapm_mutex is being held. To allow CODEC drivers to run a register
update when using snd_soc_dapm_mux_update_power() or
snd_soc_dapm_mixer_update_power() add a update parameter to these two functions.
The update parameter will be assigned to dapm->update while card->dapm_mutex is
locked.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently when updating a control that is shared between multiple widgets the
whole power-up/power-down sequence is being run once for each widget. The
control register is updated during the first run, which means the CODEC internal
routing is also updated for all widgets during this first run. The input and
output paths for each widgets are only updated though during the respective run
for that widget. This leads to a slight inconsistency between the CODEC's
internal state and ASoC's state, which causes non optimal behavior in regard to
click and pop avoidance.
E.g. consider the following setup where two MUXs share the same control.
+------+
A1 ------| |
| MUX1 |----- C1
B1 ------| |
+------+
|
control ---+
|
+------+
A2 ------| |
| MUX2 |----- C2
B2 ------| |
+------+
If the control is updated to switch the MUXs from input A to input B with the
current code the power-up/power-down sequence will look like this:
Run soc_dapm_mux_update_power for MUX1
Power-down A1
Update MUXing
Power-up B1
Run soc_dapm_mux_update_power for MUX2
Power-down A2
(Update MUXing)
Power-up B2
Note that the second 'Update Muxing' is a no-op, since the register was already
updated.
While the preferred order for avoiding pops and clicks should be:
Run soc_dapm_mux_update_power for control
Power-down A1
Power-down A2
Update MUXing
Power-up B1
Power-up B2
This patch changes the behavior to the later by running the updates for all
widgets that the control is attached to at the same time.
The new code is also a bit simpler since callers of
soc_dapm_{mux,muxer}_update_power don't have to loop over each widget anymore
and neither do we need to keep track for which of the kcontrol's widgets the
current update is.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
soc_dpcm_runtime_update() operates on a ASoC card as a whole. Currently it takes
a snd_soc_dapm_widget as its only parameter though. The widget is then used to
look up the card and is otherwise unused. This patch changes the function to
take a pointer to the card directly. This makes it possible to to call
soc_dpcm_runtime_update() for updates which are not related to one specific
widget.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some more fixes and enhancements, and also a bunch of refectoring for
AC'97 support which enables more than one AC'97 controller driver to be
built in.
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Merge tag 'asoc-v3.11-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.11
Some more fixes and enhancements, and also a bunch of refectoring for
AC'97 support which enables more than one AC'97 controller driver to be
built in.
Currently we can only have a single platform built in with AC'97 support
due to the use of a global variable to provide the bus operations. Fix
this by making that variable a pointer and having the bus drivers set the
operations prior to registering.
This is not a particularly good or nice approach but it avoids blocking
multiplatform and a real fix involves fixing the fairly deep problems
with AC'97 support - we should be converting it to a real bus.
Acked-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Introduce a new helper function, snd_ctl_sync_vmaster(), which updates
the slave put callbacks forcibly as well as calling the hook. This
will be used in the upcoming patch in HD-audio codec driver for
toggling the mute in vmaster slaves.
Along with the new function, the old snd_ctl_sync_vmaster_hook() is
replaced as a macro calling with the argument hook_only=true.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not a big release subsystem wise, the main changes have been some nice
improvements on the driver side:
- Lots of cleanups and fixes for Blackfin, SGTL5000 and UX500.
- Generalisation of the Bluetooth and HDMI stub drivers.
- New CODEC drivers for SSM2518 and RT5640.
- New machine driver for Tegra CPUs with RT5640.
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Merge tag 'asoc-v3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.11
Not a big release subsystem wise, the main changes have been some nice
improvements on the driver side:
- Lots of cleanups and fixes for Blackfin, SGTL5000 and UX500.
- Generalisation of the Bluetooth and HDMI stub drivers.
- New CODEC drivers for SSM2518 and RT5640.
- New machine driver for Tegra CPUs with RT5640.
* for-linus: (635 commits)
ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310
ALSA: hda - Fix pin configurations for MacBook Air 4,2
ALSA: usb-audio: work around Android accessory firmware bug
ALSA: hda - Headset mic support for three more machines
Linux 3.10-rc6
smp.h: Use local_irq_{save,restore}() in !SMP version of on_each_cpu().
powerpc: Fix missing/delayed calls to irq_work
powerpc: Fix emulation of illegal instructions on PowerNV platform
powerpc: Fix stack overflow crash in resume_kernel when ftracing
snd_pcm_link(): fix a leak...
use can_lookup() instead of direct checks of ->i_op->lookup
move exit_task_namespaces() outside of exit_notify()
fput: task_work_add() can fail if the caller has passed exit_task_work()
xfs: don't shutdown log recovery on validation errors
xfs: ensure btree root split sets blkno correctly
xfs: fix implicit padding in directory and attr CRC formats
xfs: don't emit v5 superblock warnings on write
mei: me: clear interrupts on the resume path
mei: nfc: fix nfc device freeing
mei: init: Flush scheduled work before resetting the device
...
Since commit 85762e71 ("ASoC: dapm: Implement mixer control sharing") the
long_name field of the snd_soc_dapm_path struct is unused. All of the name
handling now happens entirely in dapm_create_or_share_mixmux_kcontrol(). So we
can remove the long_name field from the snd_soc_dapm_path struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the ALC5640 codec driver.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Even though they are virtual widgets DAI widgets still get counted for the
DAPM context power management so we can't just use the active state to
check if they should be powered as they may not be part of a complete path.
Instead split them into input and output widgets and do the same power
checks as we perform on AIFs.
Reported-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently ALSA supports up to 32 card instances when the dynamic minor
is used. While 32 cards are usually big enough for normal use cases,
there are sometimes weird requirements with more card support.
Actually, this limitation, 32, comes from the index option, where you
can pass the bit mask to assign the card. Other than that, we can
actually give more cards up to the minor number limits (currently 256,
which can be extended more, too).
This patch adds a new Kconfig to specify the max card numbers, and
changes a few places to accept more than 32 cards.
The only incompatibility with high card numbers would be the handling
of index option. The index option can be still used to pass the
bitmask for card assignments, but this works only up to 32 slots.
More than 32, no bitmask style option is available but only a single
slot can be specified via index option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ops field of the snd_pcm_substream struct is never modified inside the ALSA
core. Making it const allows drivers to declare their snd_pcm_ops struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices may benefit from being able to start some parts of the widget
power up/down sequence earlier on in the sequence than the point at which
the final power state is committed. Support these by providing events which
are called before any power state changes are done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
Revert "ALSA: hda - Don't set up active streams twice"
ALSA: Add comment for control TLV API
ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
ALSA: HDA: Fix Oops caused by dereference NULL pointer
ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()
ALSA: mips/hal2: Remove redundant platform_set_drvdata()
ALSA: hda - Fix 3.9 regression of EAPD init on Conexant codecs
sound: Fix make allmodconfig on MIPS
ALSA: hda - Fix system panic when DMA > 40 bits for Nvidia audio controllers
ALSA: atmel: Remove redundant platform_set_drvdata()
ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.
ASoC: wm8994: missing break in wm8994_aif3_hw_params()
ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format
ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare
Userspace is not meant to have to handle all strange dB ranges,
so add a specification comment.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone mic
and headset mic support, jack_modes hint consolidation, proper beep
attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone
mic and headset mic support, jack_modes hint consolidation, proper
beep attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
ALSA: pcm_format_to_bits strong-typed conversion
ALSA: compress: fix the states to check for allowing read
ALSA: hda - Move Thinkpad X220 to use auto parser
ALSA: USB: adjust for changed 3.8 USB API
ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
sound: oss/dmabuf: use dma_map_single
ALSA: ali5451: use mdelay instead of large udelay constants
ALSA: hda - Add the support for ALC286 codec
ALSA: usb-audio: USB quirk for Yamaha THR10C
ALSA: usb-audio: USB quirk for Yamaha THR5A
ALSA: usb-audio: USB quirk for Yamaha THR10
ALSA: usb-audio: Fix autopm error during probing
ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
ALSA: sound kconfig typo
ALSA: emu10k1: Fix dock firmware loading
ASoC: ux500: forward declare msp_i2s_platform_data
ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
...
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.
Change such conversions to use this function and silence sparse
warnings.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more fixes, nothing too major though the DMA changes fix modular
builds.
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Merge tag 'asoc-v3.10-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
A few more fixes, nothing too major though the DMA changes fix modular
builds.
Fix typo in printk and comments within various drivers.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The commit [b209c4df: ALSA: emu10k1: cache emu1010 firmware] broke the
firmware loading of the dock, just (mistakenly) ignoring a different
firmware for docks on some models. This patch revives them again.
Bugzilla: https://bugs.archlinux.org/task/34865
Reported-and-tested-by: Tobias Powalowski <tobias.powalowski@googlemail.com>
Cc: <stable@vger.kernel.org> [v3.8+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some platforms which are half-duplex share the same DMA channel between the
playback and capture stream. Add support for this to the generic dmaengine PCM
driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch reworks the writes to use cumulative values thus making the
app_pointer unecessary and removing it.
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The buffer passed to the copy callback should not be const because the
copy callback can be used for capture and playback.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
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Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window. These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
Use the generic dmaengine PCM driver instead of a custom implemention. There is
a minor functional change, the ux500 PCM driver did not preallocate the audio
buffer, while the generic dmaengine PCM driver will do this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital
DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.
The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).
DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:
configured hardware
176.4KHz 352.8kHz 705.6KHz <---- sample rate
8-bit 2.8MHz 5.6MHz
16-bit 2.8Mhz 5.6MHz 11.2MHz
`-----------------------------'
actual DSD sample rates
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unfortunately there are still quite a few platforms with a dmaengine driver
which do not support reporting the number of bytes left to transfer. If we want
to support these platforms in the generic dmaengine PCM driver we have.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for platforms which don't use devicetree yet or have to optionally
support a non-devicetree way to request the DMA channel. The patch adds the
compat_request_channel and compat_filter_fn callbacks to the
snd_dmaengine_pcm_config struct. If the compat_request_channel is implemented it
will be used to request the DMA channel. If not dma_request_channel with
compat_filter_fn as the filter function will be used to request the channel.
The patch also exports the snd_dmaengine_pcm_request_chan() function, since
compat platforms will want to use it to request their DMA channel.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a generic dmaengine PCM driver. It builds on top of the
dmaengine PCM library and adds the missing pieces like DMA channel management,
buffer management and channel configuration. It will be able to replace the
majority of the existing platform specific dmaengine based PCM drivers.
Devicetree is used to map the DMA channels to the PCM device.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_{add,remove}_platform are similar to snd_soc_register_platform and
snd_soc_unregister_platform with the difference that they won't allocate and
free the snd_soc_platform structure.
Also add snd_soc_lookup_platform which looks up a platform by the device it has
been registered for.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Refactor the dmaengine PCM library to allow the DMA channel to be requested
before opening a PCM substream. snd_dmaengine_pcm_open() now expects a DMA
channel instead of a filter function and filter parameter as its parameters.
snd_dmaengine_pcm_close() is updated to not release the DMA channel. This allows
a dmaengine based PCM driver to request its channels before the substream is
opened.
The patch also introduces two new functions, snd_dmaengine_pcm_open_request_chan()
and snd_dmaengine_pcm_close_release_chan(), which have the same signature and
behaviour of the old snd_dmaengine_pcm_{open,close}() and internally use the new
variants of these functions. All users of snd_dmaengine_pcm_{open,close}() are
updated to use snd_dmaengine_pcm_open_request_chan() and
snd_dmaengine_pcm_close_release_chan().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
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Merge tag 'asoc-v3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
This patch adds a common DMA data struct which can be used by DAI drivers to
communicate their DMA configuration requirements to the DMA pcm driver. Having
a common data structure for this allows us to implement common functions on top
of them, which can be used by multiple platforms.
This patch also introduces a new function to initialize certain fields of a
dma_slave_config struct from the common DAI DMA data struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does not modify a platform driver's compr_ops structure. Making it
const allows ASoC platform drivers to declare their snd_compr_ops struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does not modify a platform driver's ops structure. Making it const
allows ASoC platform drivers to declare their snd_pcm_ops struct as const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does no not modify the driver of a platform. Making it const
allows ASoC platform drivers to declare the snd_soc_platform_driver struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All drivers are using snd_soc_register_component()
instead of snd_soc_register_dai[s]()
snd_soc_[un]register_dai[s]() are no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These functions were initially added to be able to support some oddball dma
drivers, but all users have been updated to deal with the situation without the
help of snd_dmaengine_pcm_{set,get}_data, so these two functions can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Source files shouldn't have the executable bit set.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds .name member on snd_soc_component_driver.
But this patch doesn't care about whether cmpnt_drv was NULL,
and/or its name was NULL in snd_soc_register_component()
at this point.
Because, it is easy to switch over to
snd_soc_register_component() from snd_soc_register_dais()
if it doesn't care cmpnt_drv was NULL.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC has register function for platform/codec/dai/card,
but doesn't have for cpu.
It often produces confusion and fault on ASoC.
As result of ASoC community discussion,
we consider new struct snd_soc_component for CPU/CODEC,
and will switch over to use it.
This patch adds very basic struct snd_soc_component,
and register function for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
script/kernel-doc reports the following type of warnings (when run in verbose
mode):
Warning(sound/core/init.c:152): No description found for return value of
'snd_card_create'
To fix that:
- add missing descriptions of function return values
- use "Return:" sections to describe those return values
Along the way:
- complete some descriptions
- fix some typos
Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Having snd_BUG_ON() only evaluate its conditional when CONFIG_SND_DEBUG
is set leads to frequent bugs, since other similar macros in the kernel
have different behavior. Let's make snd_BUG_ON() act like those macros
so it will stop being accidentally misused.
Signed-off-by: Christine Spang <christine.spang@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a driver for TI's TA5086 6-channel PWM processor.
This chip has a very unusual register layout, specifically because the
registers are of unequal size, and multi-byte registers require bulk
writes to take effect. Regmap does not support these kind of mappings.
Currently, the driver does not touch any of the registers >= 0x20, so
it doesn't matter, because the register map is mapped to an 8-bit array.
In case more features will be added in the future that require access
to higher registers, the entire regmap H/W I/O routines have to be
open-coded.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 497098be ("ASoC: dapm: Remove bodges for no-widget CODECs") removed the
last user of the n_widgets field. Currently it is incremented for each widget
added, but the value is never used, so we can remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This field was added in commit 2e72f8e ("ASoC: New enum type: value_enum"), but
has never been used since. Considering that the soc_enum struct is usually
shared between all instances of a CODEC, it also doesn't make much sense to have
a pointer to DAPM specific data in it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tegra only supports, and always enables, device tree. Remove all runtime
checks for DT support from the driver.
This allows removal of the hard-coded Harmony ASoC mapping table, since
Harmony only boots with DT now.
All board-specific configuration now comes from device tree, so there is
no need to have a platform_data structure. Rework the driver to parse the
device tree directly into struct tegra_wm8903.
Also some slight re-ordering of probe() so that the code more closely
resembles other drivers for easier comparison. Inparticular, the GPIO DT
parsing and initial programming are moved together for each GPIO.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This branch contains changes for OMAP that came in late during the release
staging, close to when the merge window opened.
It contains, among other things:
- OMAP PM fixes and some patches for audio device integration
- OMAP clock fixes related to common clock conversion
- A set of patches cleaning up WFI entry and blocking.
- A set of fixes and IP block support for PM on TI AM33xx SoCs (Beaglebone, etc)
- A set of smaller fixes and cleanups around AM33xx restart and revision
detection, as well as removal of some dead code (CONFIG_32K_TIMER_HZ)
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Merge tag 'late-omap' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull ARM SoC late OMAP changes from Olof Johansson:
"This branch contains changes for OMAP that came in late during the
release staging, close to when the merge window opened.
It contains, among other things:
- OMAP PM fixes and some patches for audio device integration
- OMAP clock fixes related to common clock conversion
- A set of patches cleaning up WFI entry and blocking.
- A set of fixes and IP block support for PM on TI AM33xx SoCs
(Beaglebone, etc)
- A set of smaller fixes and cleanups around AM33xx restart and
revision detection, as well as removal of some dead code
(CONFIG_32K_TIMER_HZ)"
* tag 'late-omap' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (34 commits)
ARM: omap2: include linux/errno.h in hwmod_reset
ARM: OMAP2+: fix some omap_device_build() calls that aren't compiled by default
ARM: OMAP4: hwmod data: Enable AESS hwmod device
ARM: OMAP4: hwmod data: Update AESS data with memory bank area
ARM: OMAP4+: AESS: enable internal auto-gating during initial setup
ASoC: TI AESS: add autogating-enable function, callable from architecture code
ARM: OMAP2+: hwmod: add enable_preprogram hook
ARM: OMAP4: clock data: Add missing clkdm association for dpll_usb
ARM: OMAP2+: PM: Fix the dt return condition in pm_late_init()
ARM: OMAP2: am33xx-hwmod: Fix "register offset NULL check" bug
ARM: OMAP2+: AM33xx: hwmod: add missing HWMOD_NO_IDLEST flags
ARM: OMAP: AM33xx hwmod: Add parent-child relationship for PWM subsystem
ARM: OMAP: AM33xx hwmod: Corrects PWM subsystem HWMOD entries
ARM: DTS: AM33XX: Add nodes for OCMC RAM and WKUP-M3
ARM: OMAP2+: AM33XX: Update the hardreset API
ARM: OMAP2+: AM33XX: hwmod: Update the WKUP-M3 hwmod with reset status bit
ARM: OMAP2+: AM33XX: hwmod: Fixup cpgmac0 hwmod entry
ARM: OMAP2+: AM33XX: hwmod: Update TPTC0 hwmod with the right flags
ARM: OMAP2+: AM33XX: hwmod: Register OCMC RAM hwmod
ARM: OMAP2+: AM33XX: CM/PRM: Use __ASSEMBLER__ macros in header files
...
Currently if a path loops back on itself we correctly skip over it to
avoid going into an infinite loop but this causes us to ignore the need
to power up the path as we don't count the loop for the purposes of
counting inputs and outputs. This means that internal loopbacks within a
device that have powered devices on them won't be powered up.
Fix this by treating any path that is currently in the process of being
recursed as having a single input or output so that it is counted for
the purposes of power decisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
A few more updates from the past week - a new driver from Dialog and
some small fixes and tweaks.
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Merge tag 'asoc-3.9-updates' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.9
A few more updates from the past week - a new driver from Dialog and
some small fixes and tweaks.
This patch adds support for the Dialog DA7213 audio codec.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
this add new API for sound compress to support gapless playback.
As noted in Documentation change, we add API to send metadata of encoder and
padding delay to DSP. Also add API for indicating EOF and switching to
subsequent track
Also bump the compress API version
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A fairly quiet release for ASoC:
- Support for a wider range of hardware in the compressed stream code.
- The ability to mute capture streams as well as playback streams while
inactive.
- DT support for AK4642, FSI, Samsung I2S and WM8962.
- AC'97 support for Tegra.
- New driver for max98090, replacing the stub which was there.
Due to dependencies we've also got support for asynchronous I/O in regmap
and DTification of DMA support for Samsung platforms (used only by the
I2S driver and SPI) merged here as well.
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Merge tag 'asoc-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.9
A fairly quiet release for ASoC:
- Support for a wider range of hardware in the compressed stream code.
- The ability to mute capture streams as well as playback streams while
inactive.
- DT support for AK4642, FSI, Samsung I2S and WM8962.
- AC'97 support for Tegra.
- New driver for max98090, replacing the stub which was there.
Due to dependencies we've also got support for asynchronous I/O in regmap
and DTification of DMA support for Samsung platforms (used only by the
I2S driver and SPI) merged here as well.
Add a basic header file for the TI AESS IP block, located in the OMAP4
Audio Back-End subsystem.
Currently, this header file only contains a function to enable the
AESS internal clock auto-gating. This will be used by a subsequent
patch to ensure that the AESS won't block the entire chip
low-power-idle mode. We wish to be able to place the AESS into idle
even when no AESS driver has been compiled in.
Signed-off-by: Paul Walmsley <paul@pwsan.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Péter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Tony Lindgren <tony@atomide.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.
The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.
Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
This patch completes the replacement of the existing max98090 driver,
by installing a more complete driver.
Signed-off-by: Jerry Wong <jerry.wong@maximintegrated.com>
Tested-by: Matthew Mowdy <matthew.mowdy@maximintegrated.com>
Reviewed-by: Ralph Birt <ralph.birt@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert MicBias widgets to supply widget.
On tlv320aic3x, Mic bias power on/off shares the same register bits
with output mic bias voltage. So, when power on mic bias, we need
reclaim it to voltage value.
Provide a new platform data so that the micbias voltage can be sent
according to board requirement. Now since tlv320aic3x codec driver
is DT aware, update dt files and functions to handle this new
"micbias-vg" platform data.
Because of sharing of bits, when enabling the micbias, voltage also
needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD
macro to create an event to handle this.
Since micbias is converted to supply widget, updated machine drivers as
well.
This change is runtime tested on da850-evm with audio loopback
(arecord|aplay) for confirmation.
Signed-off-by: Hebbar Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current soc-dai.h defines SND_SOC_DAIFMT_GATED as (2 << 4),
but gated clock should be default settings (= 0).
This patch fixup SND_SOC_DAIFMT_GATED as (0 << 4).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some audio drivers are calling snd_dma_continuous_data(GFP_KERNEL)
which makes "sparse" give a warning:
$ make C=2 M=sound/usb modules
...
sound/usb/6fire/pcm.c:625:25: warning: cast from restricted gfp_t
sound/usb/caiaq/audio.c:845:41: warning: cast from restricted gfp_t
sound/usb/usx2y/usbusx2yaudio.c:997:54: warning: cast from restricted gfp_t
sound/usb/usx2y/usbusx2yaudio.c:1001:54: warning: cast from restricted gfp_t
sound/usb/usx2y/usx2yhwdeppcm.c:774:54: warning: cast from restricted gfp_t
sound/usb/usx2y/usx2yhwdeppcm.c:778:54: warning: cast from restricted gfp_t
Add __force to the cast to silence the warning.
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds snd_soc_of_parse_daifmt() and supports below style on DT.
[prefix]format = "i2c";
[prefix]clock-gating = "continuous";
[prefix]bitclock-inversion;
[prefix]bitclock-master;
[prefix]frame-master;
Each driver can use specific [prefix]
(ex simple-card,cpu,dai,format = xxx;)
This sample will be
SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CONT |
SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]
For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros. This should have the same effect but shut up warnings like
above.
But since we had already put ifdefs, changing to inline functions
would trigger compile errors. So, such ifdefs is removed in this
patch.
In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too. For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.
Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current soc-dai.h defines SND_SOC_DAIFMT_NB_NF as (1 << 8),
but normal bit clock / normal frame should be
default settings (= 0).
This patch fixup SND_SOC_DAIFMT_NB_NF as (0 << 8).
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The core does not modify these fields, so they can be made const. This allows
drivers to declare their op tables as const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current simple-card driver calls asoc_simple_card_dai_init()
if platform had a asoc_simple_card_dai_init pointer.
And, this initialization function works only
when platform has an applicable initial value for each dai settings.
And basically, almost all sound card requires certain initialization.
This means that almost all platform has initialization settings,
and driver do nothing if it doesn't have settings.
And additionally, current simple-card supports sysclk settings but it was
only for codec. In order to abolish deviation between cpu and codec,
and in order to simplify processing,
this patch adds asoc_simple_dai, and removed pointless
struct asoc_simple_dai_init_info which was trigger of
calling asoc_simple_card_dai_init().
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All MXS users have been converted to device tree and the board files have been
removed.
No need to keep platform data in the driver.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Empty files can get deleted by the patch program, so remove empty Kbuild
files and their links from the parent Kbuilds.
Signed-off-by: David Howells <dhowells@redhat.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
FSI driver's flag usage was changed/removed by
3449f5fab8
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
ab6f6d8521
(ASoC: fsi: add master clock control functions)
And unused flags had been removed on FSI driver,
but the definition had been kept to avoid compile error.
It is possible to cleanup sh_fsi.h now.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
3449f5fab8
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
added clock inversion support via snd_soc_dai_set_fmt().
Thus, this patch removed SH_FSI_xxx_INV and fsi_get_info()
from fsi driver, and modified platform settings to use new style.
Then, it cleaned up meaningless settings from platform.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Simon Horman <horms+renesas@verge.net.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ab6f6d8521
(ASoC: fsi: add master clock control functions)
added driver level clock control functions.
And now, platform depended .set_rate() is no longer needed.
This patch removed unnecessary .set_rate() platform callback support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The CS4271 requires its LRCLK and MCLK to be stable before its RESET
line is de-asserted. That also means that clocks cannot be changed
without putting the chip back into hardware reset, which also requires
a complete re-initialization of all registers.
One (undocumented) workaround is to assert and de-assert the PDN bit
in the MODE2 register.
This patch adds a new flag to both the DT bindings as well as to the
platform data to enable that workaround.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since we are now using the clock API integration to manage MCLK we can now
use clk_get_rate() to determine if we need to divide MCLK without relying
on platform data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Although we've had macros defining double _RANGE controls for a while now
they've not actually been backed up properly by the implementation, it's
treated everything as mono. Fix that by implementing the handling in the
stereo controls, ensuring that the mono controls don't mistakenly get
treated as stereo.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Nothing terribly exciting here, just small localised changes.
As well as fixes there are a couple of Cirrus changes and one devm_
change which were in prior to the merge window but got missed from the
original pull to Takashi.
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Merge tag 'asoc-3.8p1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.8
Nothing terribly exciting here, just small localised changes.
As well as fixes there are a couple of Cirrus changes and one devm_
change which were in prior to the merge window but got missed from the
original pull to Takashi.
pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.
In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.
One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE
Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Very quiet release for ASoC really:
- Standardisation of the logging.
- DT and dmaengine support for Atmel.
- Support for Wolfson ADSP cores.
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090.
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Merge tag 'asoc-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.8
Very quiet release for ASoC really:
- Standardisation of the logging.
- DT and dmaengine support for Atmel.
- Support for Wolfson ADSP cores.
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090.
Make the flag in the pdata of type bool to fix a sparse warning.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Yet again like previous two commits, drop the old hwdep user-space
firmware code from vx driver (snd-vxpocket and snd-vx222).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes master clock selection
independent from platform information pointer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver required set_rate() platform callback function
to set audio clock if it was master mode,
because it seemed that CPG/FSI-DIV clocks calculation depend on
platform/board/cpu.
But it was calculable regardless of platform.
This patch supports audio clock calculation method,
but the sampling rate under 32kHz is not supported at this point.
Old type set_rate() is still supported now,
but it will be deleted on next version
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the possibility to specify a gpio through platform data
so that a HW reset can be issued to the codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.
The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA did not provide any direct means to infer the audio time for A/V
sync and system/audio time correlations (eg. PulseAudio).
Applications had to track the number of samples read/written and
add/subtract the number of samples queued in the ring buffer. This
accounting led to small errors, typically several samples, due to the
two-step process. Computing the audio time in the kernel is more
direct, as all the information is available in the same routines.
Also add new .audio_wallclock routine to enable fine-grain synchronization
between monotonic system time and audio hardware time.
Using the wallclock, if supported in hardware, allows for a
much better sub-microsecond precision and a common drift tracking for
all devices sharing the same wall clock (master clock).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep track of boundary crossing when hw_ptr
exceeds boundary limit and wraps-around. This
will help keep track of total number
of frames played/received at the kernel level
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CS4271 has a feature to sync its analog mute flags, so one mute
circuitry can be used for both channels.
Give users access to this feature with a new DT property and a flag in
the platform data.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: David Howells <dhowells@redhat.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Michael Kerrisk <mtk.manpages@gmail.com>
Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: Dave Jones <davej@redhat.com>
This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support of
D3 clock-stop. Also changing the power_save option in sysfs kicks
off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in HD-audio
are continued cleanups and standardization for the generic auto
parser and bug fixes (HBR, device-specific fixups), in addition to
the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode.
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support
of D3 clock-stop. Also changing the power_save option in sysfs
kicks off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in
HD-audio are continued cleanups and standardization for the generic
auto parser and bug fixes (HBR, device-specific fixups), in
addition to the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode."
Fix up various arm soc header file reorg conflicts.
* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
ALSA: hda - Add new codec ALC283 ALC290 support
ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
ALSA: hda - fix indices on boost volume on Conexant
ALSA: aloop - add locking to timer access
ALSA: hda - Fix hang caused by race during suspend.
sound: Remove unnecessary semicolon
ALSA: hda/realtek - Fix detection of ALC271X codec
ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
ALSA: hda - make a generic unsol event handler
ASoC: codecs: Add DA9055 codec driver
ASoC: eukrea-tlv320: Convert it to platform driver
ALSA: ASoC: add DT bindings for CS4271
ASoC: wm_hubs: Ensure volume updates are handled during class W startup
ASoC: wm5110: Adding missing volume update bits
ASoC: wm5110: Add OUT3R support
ASoC: wm5110: Add AEC loopback support
ASoC: wm5110: Rename EPOUT to HPOUT3
ASoC: arizona: Add more clock rates
ASoC: arizona: Add more DSP options for mixer input muxes
...
Pull media updates from Mauro Carvalho Chehab:
"The first part of the media updates for Kernel 3.7.
This series contain:
- A major tree renaming patch series: now, drivers are organized
internally by their used bus, instead of by V4L2 and/or DVB API,
providing a cleaner driver location for hybrid drivers that
implement both APIs, and allowing to cleanup the Kconfig items and
make them more intuitive for the end user;
- Media Kernel developers are typically very lazy with their duties
of keeping the MAINTAINERS entries for their drivers updated. As
now the tree is more organized, we're doing an effort to add/update
those entries for the drivers that aren't currently orphan;
- Several DVB USB drivers got moved to a new DVB USB v2 core; the new
core fixes several bugs (as the existing one that got bitroted).
Now, suspend/resume finally started to work fine (at least with
some devices - we should expect more work with regards to it);
- added multistream support for DVB-T2, and unified the API for
DVB-S2 and ISDB-S. Backward binary support is preserved;
- as usual, a few new drivers, some V4L2 core improvements and lots
of drivers improvements and fixes.
There are some points to notice on this series:
1) you should expect a trivial merge conflict on your tree, with the
removal of Documentation/feature-removal-schedule.txt: this series
would be adding two additional entries there. I opted to not
rebase it due to this recent change;
2) With regards to the PCTV 520e udev-related breakage, I opted to
fix it in a way that the patches can be backported to 3.5 even
without your firmware fix patch. This way, Greg doesn't need to
rush backporting your patch (as there are still the firmware cache
and firmware path customization issues to be addressed there).
I'll send later a patch (likely after the end of the merge window)
reverting the rest of the DRX-K async firmware request, fully
restoring its original behaviour to allow media drivers to
initialize everything serialized as before for 3.7 and upper.
3) I'm planning to work on this weekend to test the DMABUF patches
for V4L2. The patches are on my queue for several Kernel cycles,
but, up to now, there is/was no way to test the series locally.
I have some concerns about this particular changeset with regards
to security issues, and with regards to the replacement of the old
VIDIOC_OVERLAY ioctl's that is broken on modern systems, due to
GPU drivers change. The Overlay API allows direct PCI2PCI
transfers from a media capture card into the GPU framebuffer, but
its API is crappy. Also, the only existing X11 driver that
implements it requires a XV extension that is not available
anymore on modern drivers. The DMABUF can do the same thing, but
with it is promising to be a properly-designed API. If I can
successfully test this series and be happy with it, I should be
asking you to pull them next week."
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (717 commits)
em28xx: regression fix: use DRX-K sync firmware requests on em28xx
drxk: allow loading firmware synchrousnously
em28xx: Make all em28xx extensions to be initialized asynchronously
[media] tda18271: properly report read errors in tda18271_get_id
[media] tda18271: delay IR & RF calibration until init() if delay_cal is set
[media] MAINTAINERS: add Michael Krufky as tda827x maintainer
[media] MAINTAINERS: add Michael Krufky as tda8290 maintainer
[media] MAINTAINERS: add Michael Krufky as cxusb maintainer
[media] MAINTAINERS: add Michael Krufky as lg2160 maintainer
[media] MAINTAINERS: add Michael Krufky as lgdt3305 maintainer
[media] MAINTAINERS: add Michael Krufky as mxl111sf maintainer
[media] MAINTAINERS: add Michael Krufky as mxl5007t maintainer
[media] MAINTAINERS: add Michael Krufky as tda18271 maintainer
[media] s5p-tv: Report only multi-plane capabilities in vidioc_querycap
[media] s5p-mfc: Fix misplaced return statement in s5p_mfc_suspend()
[media] exynos-gsc: Add missing static storage class specifiers
[media] exynos-gsc: Remove <linux/version.h> header file inclusion
[media] s5p-fimc: Fix incorrect condition in fimc_lite_reqbufs()
[media] s5p-tv: Fix potential NULL pointer dereference error
[media] s5k6aa: Fix possible NULL pointer dereference
...
A couple more updates for 3.7, enhancements to the ux500 and wm2000
drivers, a new driver for DA9055 and the support for regulator bypass
mode. With the exception of the DA9055 this has all had a chance to
soak in -next (the driver was added on Friday so should be in -next
today).
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Additional updates for v3.7
A couple more updates for 3.7, enhancements to the ux500 and wm2000
drivers, a new driver for DA9055 and the support for regulator bypass
mode. With the exception of the DA9055 this has all had a chance to
soak in -next (the driver was added on Friday so should be in -next
today).
Convert #include "..." to #include <path/...> in kernel system headers.
Signed-off-by: David Howells <dhowells@redhat.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: Dave Jones <davej@redhat.com>
This patch adds support for Dialog semiconductor's DA9055 audio codec.
This has been tested on DA9055 EVB with Samsung SMDK6410 board.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <david.chen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow regulators managed via DAPM to make use of the bypass support that
has recently been added to the regulator API by setting a flag
SND_SOC_DAPM_REGULATOR_BYPASS. When this flag is set the regulator will
be put into bypass mode before being disabled, allowing the regulator to
fall into bypass mode if it can't be disabled due to other users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Passing struct snd_dma_buffer pointer instead, so that they work no
matter whether real SG buffer is used or not.
This is a preliminary work for the HD-audio DSP loader code.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.7
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
The 'dres' field (discharge resistance for headphone outputs) is no longer
used in the driver, so remove it.
It was used in the original version of the driver when entering standby
from off, but we stopped using it when we switched from having a single
startup sequence to having separate cap and capless sequences.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the LRCLK is shared and the WM8960 is clock master then we should
enable the LRCM bit to tell the device that it should drive LRCLK when
either ADC or DAC is enabled rather than separately driving the two
LRCLKs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For ENUM controls the bitmask is calculated based on the number of items.
Currently this is done each time the control is accessed. And while the
performance impact of this should be negligible we can easily do better. The
roundup_pow_of_two macro performs the same calculation which is currently done
manually, but it is also possible to use this macro with compile time constants
and so it can be used to initialize static data. So we can use it to initialize
the mask field of a ENUM control during its declaration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for tuning AM (on devices with the necessary additional
hardware components), and advertise the available bands using the new
VIDIOC_ENUM_FREQ_BANDS ioctl.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
For following the standard, define more channel map positions and
shuffle the items a bit:
- As both PulseAudio and gstreamer define MONO channel position
explicitly, we should follow that, too. The mono streams point to
this channel position unless they are explicitly assigned to certain
channel positions.
- Top-front-* and Top-rear-* positions are added, carried from
PulseAudio's definitions.
- Move NA and MONO definitions at the top of table right after
UNKNOWN, since these are more abstract in comparison with other
practical positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will be used to enable additional control of the regulators.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
There is already a set of channel position definitions in alsa-lib
mixer.h, and it'd be more practical to keep the same order for the
PCM channel map, too. The value is shifted with 1 to keep zero for
UNKNOWN.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC650 has a channel swap option between surround and CLFE channels,
so we need to tweak the channel maps dynamically depending on the
register bit.
Now struct snd_ac97 can contain chmap pointers for playback and
capture. The driver may store these and let ac97 driver changing the
channel mapping dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements the basic data types for the standard channel
mapping API handling.
- The definitions of the channel positions and the new TLV types are
added in sound/asound.h and sound/tlv.h, so that they can be
referred from user-space.
- Introduced a new helper function snd_pcm_add_chmap_ctls() to create
control elements representing the channel maps for each PCM
(sub)stream.
- Some standard pre-defined channel maps are provided for
convenience.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since bypass paths aren't part of DAPM streams and we may not have any
DAPM streams there may not be anything that triggers a DAPM sync for
them. Mark all input and output widgets as dirty and then sync to do so
at the end of suspend and resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The only user was removed over two years ago in commit a6c65736 ("ASoC: Remove
current PGA control handling").
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the Tegra+WM8903 ASoC platform data header out of
arch/arm/mach-tegra, as a pre-requisite of single zImage.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the main ALSA version number from the kernel ALSA driver.
The ALSA driver package release diverges from the upstream. This may
confuse users to see the same ALSA version for many kernel releases
and this version lost it's original purpose and connection.
The "ioctl" APIs have own version numbers, so the user space may check
for specific API changes only.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The WM0010 is a compact digital signal processor that has been
highly optimised for low-power audio applications. Extensive memory
resources and core optimisation allow the device to manage all audio
processing algorithms efficiently and autonomously, while the host
processor sleeps or performs other tasks.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sometimes the analogue circuitry connected to the microphone needs some
time to settle after power up. Allow systems to configure this delay in
the platform data, the driver will then insert the required delay during
power up of paths that involve the microphone.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Here we update the asoc structures to add compress stream definations
First the struct snd_soc_dai_driver adds a new member to indicate if the dai is
compressed or pcm. Next we add a new structre the struct snd_soc_compr_ops in
the struct snd_soc_dai_link. This is to be used for machine driver to perform
any opertaions required for setting up compressed audio streams
next is the compressed data operations, they are added using struct
snd_compr_ops in the struct snd_soc_platform_driver.
Signed-off-by: Namarta Kohli <namartax.kohli@intel.com>
Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context. The biggest batch are for the newly
added Arizona drivers.
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Additional updates for 3.6
A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context. The biggest batch are for the newly
added Arizona drivers.
Implement suspend/resume support for AD1816 chips.
Tested with Terratec SoundSystem Base-1.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
struct snd_card_ad1816a is only set but the values are never used then.
Removing it allows struct snd_card's private_data to be used for
struct snd_ad1816a, simplifying the code.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix kernel-doc warning in <sound/pcm.h> and add function name to make
the kernel-doc notation complete.
Warning(include/sound/pcm.h:1081): No description found for parameter 'substream'
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move snd_legacy_find_free_ioport() function back to initval.h as it is used
by two drivers.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes that have been found recently.
Most of the commits are regression fixes in HD-audio and some other
random drivers.
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Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes that have been found recently. Most of
the commits are regression fixes in HD-audio and some other random
drivers."
* tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: snd-usb: fix clock source validity index
ALSA: hda - Fix mute-LED GPIO initialization for IDT codecs
ALSA: hda - Add descriptions for missing IDT 92HD83x models
ALSA: hda - Fix polarity of mute LED on HP Mini 210
ALSA: es1688 - freeup resources on init failure
ALSA: hda - Workaround for silent output on VAIO Z with ALC889
ALSA: hda - Fix WARNING from HDMI/DP parser
ALSA: hda - Detach from converter at closing in patch_hdmi.c
ALSA: hda - Fix mute-LED GPIO setup for HP Mini 210
ALSA: mpu401: Fix missing initialization of irq field
ALSA: hda - Fix invalid D3 of headphone DAC on VT202x codecs
Pull second set of media updates from Mauro Carvalho Chehab:
- radio API: add support to work with radio frequency bands
- new AM/FM radio drivers: radio-shark, radio-shark2
- new Remote Controller USB driver: iguanair
- conversion of several drivers to the v4l2 core control framework
- new board additions at existing drivers
- the remaining (and vast majority of the patches) are due to
drivers/DocBook fixes/cleanups.
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (154 commits)
[media] radio-tea5777: use library for 64bits div
[media] tlg2300: Declare MODULE_FIRMWARE usage
[media] lgs8gxx: Declare MODULE_FIRMWARE usage
[media] xc5000: Add MODULE_FIRMWARE statements
[media] s2255drv: Add MODULE_FIRMWARE statement
[media] dib8000: move dereference after check for NULL
[media] Documentation: Update cardlists
[media] bttv: add support for Aposonic W-DVR
[media] cx25821: Remove bad strcpy to read-only char*
[media] pms.c: remove duplicated include
[media] smiapp-core.c: remove duplicated include
[media] via-camera: pass correct format settings to sensor
[media] rtl2832.c: minor cleanup
[media] Add support for the IguanaWorks USB IR Transceiver
[media] Minor cleanups for MCE USB
[media] drivers/media/dvb/siano/smscoreapi.c: use list_for_each_entry
[media] Use a named union in struct v4l2_ioctl_info
[media] mceusb: Add Twisted Melon USB IDs
[media] staging/media/solo6x10: use module_pci_driver macro
[media] staging/media/dt3155v4l: use module_pci_driver macro
...
Conflicts:
Documentation/feature-removal-schedule.txt
Some devices which use the tea575x tuner chip don't allow direct control
over the IO pins, and thus cannot mute the audio output.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Some devices which use the tea575x tuner chip don't allow bit banging the
lines, instead they offer a method to directly set / get the contents of the
25 bit shift-register in the chip. Notably the Griffin radioSHARK USB radio
receiver does this.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
This merges the changes for converting to new PM ops for platform
and some other drivers.
Also move some header files to local places from the public
include/sound.
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for 3.6
This has been a pretty quiet release - very little activity in framework
terms, mostly just a few new drivers and updates:
- Added the ability to add and remove DAPM paths dynamically, mostly for
reparenting on clock changes.
- New machine drivers for Marvell Brownstone, ST-Ericsson Ux500
reference platform and ttc-dkp.
- New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP,
Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF
- New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI
Isabelle and Wolfson Microelectronics WM5102 and WM5110
Add a DECLARE_TLV_DB_RANGE() macro so that dB range information
can be specified without having to count the items manually for
TLV_DB_RANGE_HEAD().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the DECLARE_TLV_CONTAINER() macro to allow having static
TLVs containing more than one item.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add helper macros with a little bit of preprocessor magic to
automatically compute the length of a TLV item. This lets us avoid
having to compute this by hand, and will allow to use items that do
not use a fixed length.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we're now relying on DAPM for things like enabling clocks when we
reparent the clocks for widgets we need to either use conditional routes
(which are expensive) or remove routes at runtime. Add a route removal
API to support this use case.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
They aren't modified by the core so the drivers can declare them const.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a bit clean up of public sound header directory.
Some header files in include/sound aren't really necessary to be
located there but can be moved to their local directories gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.
Since we change vx222, vx_core and vxpocket have to be converted,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull media fixes from Mauro Carvalho Chehab.
Trivial conflict due to new USB HID ID's being added next to each other
(Baanto vs Axentia).
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (44 commits)
[media] smia: Fix compile failures
[media] Fix VIDIOC_DQEVENT docbook entry
[media] s5p-fimc: Fix control creation function
[media] s5p-mfc: Fix checkpatch error in s5p_mfc_shm.h file
[media] s5p-mfc: Fix setting controls
[media] v4l/s5p-mfc: added image size align in VIDIOC_TRY_FMT
[media] v4l/s5p-mfc: corrected encoder v4l control definitions
[media] v4l: mem2mem_testdev: Fix race conditions in driver
[media] s5p-mfc: Bug fix of timestamp/timecode copy mechanism
[media] cxd2820r: Fix an incorrect modulation type bitmask
[media] em28xx: Show a warning if the board does not support remote controls
[media] em28xx: Add remote control support for Terratec's Cinergy HTC Stick HD
[media] USB: Staging: media: lirc: initialize spinlocks before usage
[media] Revert "[media] media: mx2_camera: Fix mbus format handling"
[media] bw-qcam: driver and pixfmt documentation fixes
[media] cx88: fix firmware load on big-endian systems
[media] cx18: support big-endian systems
[media] ivtv: fix support for big-endian systems
[media] tuner-core: return the frequency range of the correct tuner
[media] v4l2-dev.c: fix g_parm regression in determine_valid_ioctls()
...
The code handles this fine already, we just need new macros in the header
for drivers to create the controls.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch implements the spdif IN driver for ST peripheral
Signed-off-by: Vipin Kumar <vipin.kumar@st.com>
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add support for the SPEAr ASoC pcm layer in ASoC
framework. The pcm layer uses common snd_dmaengine framework.
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add support for synopsys I2S controller as per the ASoC
framework.
Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the sound dmaengine pcm helper functions implement the pcm_pointer
callback by trying to count the number of elapsed periods. This is done by
advancing the stream position in the dmaengine callback by one period.
Unfortunately there is no guarantee that the callback will be called for each
elapsed period. It may be possible that under high system load it is only called
once for multiple elapsed periods. This patch addresses the issue by
implementing support for querying the current stream position directly from the
dmaengine driver. Since not all dmaengine drivers support reporting the stream
position yet the old period counting implementation is kept for now.
Furthermore the new mechanism allows to report the stream position with a
sub-period granularity, given that the dmaengine driver supports this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the sound dmaengine pcm helper functions implement the pcm_pointer
callback by trying to count the number of elapsed periods. This is done by
advancing the stream position in the dmaengine callback by one period.
Unfortunately there is no guarantee that the callback will be called for each
elapsed period. It may be possible that under high system load it is only called
once for multiple elapsed periods. This patch renames the current implementation
and documents its shortcomings and that it should not be used anymore in new
drivers.
The next patch will introduce a new snd_dmaengine_pcm_pointer which will be
implemented based on querying the current stream position from the dma device.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is essentially the reverse of snd_pcm_rate_to_rate_bit().
This is generally useful as the Compress API uses the rate bit
directly and it helps to be able to map back to the actual sample
rate.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Before this patch the owner field of the /dev/radio# device fops was set to
the snd-tea575x-tuner module itself. Meaning that the module which was using
it could be rmmod-ed while the device is open, and then BAD things happen.
I know, as I found out the hard way :)
Note that there is no need to also somehow increase the refcount of the
snd-tea575x-tuner module itself, since any drivers using it will have
symbolic references to it.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Control type added for cases where a specific range of values
within a register are required for control.
Added convenience macros:
SOC_SINGLE_RANGE
SOC_SINGLE_RANGE_TLV
Added accessor implementations:
snd_soc_info_volsw_range
snd_soc_put_volsw_range
snd_soc_get_volsw_range
Signed-off-by: Michal Hajduk <Michal.Hajduk@diasemi.com>
Signed-off-by: Adam Thomson <Adam.Thomson@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Prior to this patch, the CPU side of a DAI link was specified using a
single name. Often, this was the result of calling dev_name() on the
device providing the DAI, but in the case of a CPU DAI driver that
provided multiple DAIs, it needed to mix together both the device name
and some device-relative name, in order to form a single globally unique
name.
However, the CODEC side of the DAI link was specified using separate
fields for device (name or OF node) and device-relative DAI name.
This patch allows the CPU side of a DAI link to be specified in the same
way as the CODEC side, separating concepts of device and device-relative
DAI name.
I believe this will be important in multi-codec and/or dynamic PCM
scenarios, where a single CPU driver provides multiple DAIs, while also
booting using device tree, with accompanying desire not to hard-code the
CPU side device's name into the original .cpu_dai_name field.
Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link()
would now be identical. However, two things prevent that at present:
1) The need to save rtd->codec for the CODEC side, which means we have
to search for the CODEC explicitly, and not just the CODEC side DAI.
2) Since we know the CODEC side DAI is part of a codec, and not just
a standalone DAI, it's slightly more efficient to convert .codec_name/
.codec_of_node into a codec first, and then compare each DAI's .codec
field, since this avoids strcmp() on each DAI's CODEC's name within
the loop.
However, the two loops are essentially semantically equivalent.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds a supply-widget variant for connection to the clock-framework.
This widget-type corresponds to the variant for regulators.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds a function getting the stream-name as a string for
a specific stream.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These are all new code, they've been in -next already so should be OK
for merge this time round. I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.
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Merge tag 'asoc-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Last minute updates
These are all new code, they've been in -next already so should be OK
for merge this time round. I'd been planning to send a pull request
today after they'd had a bit of exposure there to make sure breakage
didn't propagate into your tree.
SupherH FSI2 can use special data transfer,
but it depends on CPU-FSI2 connection style.
We can use 16bit data stream mode if it was valid connection,
and it is required for 16bit data DMA transfer / SPDIF sound output.
We can use 24bit data transfer if it was invalid connection.
We can select connection type if CPU is SH7372,
and it is always valid connection if latest SuperH.
This patch adds new bus_option and fsi_bus_setup()
for supporting these feature.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Devices with many DAIs are becoming more and more common, and generally
the more modern devices have consistent register layouts between DAIs.
Rather than have drivers open code lookups based on the DAI ID or cause
uglification in UI by having register addresses for IDs provide a base
address field they can use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Georgi Vlaev <joe@nucleusys.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's
necessary to allow some flexability wrt to PCM operations here so that we
can define a bespoke DPCM trigger() PCM operation for such HW.
A bespoke DPCM trigger() allows exact ordering and timing of component
triggering by allowing a component driver to manage the final enable
and disable configurations without adding extra complexity to other
component drivers. e.g. The McPDM DAI and ABE are tightly coupled on
OMAP4 so we have a bespoke trigger to manage the trigger to improve
performance and reduce complexity when triggering new McPDM BEs.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some component drivers will need to be able to look up their
DAI link substream and RTD data. Provide a mechanism for this.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows DPCM to dynamically alter the FE to BE PCM links
at runtime based on mixer setting updates. DAPM is looked up after
every mixer update and we perform a DPCM runtime update if the
mixer has a change of value.
This patchs adds/changes the following :-
o Adds DPCM runtime update core.
o Changes soc_dapm_mixer_update_power() and soc_dapm_mux_update_power()
to return if a change has occured rather than 0. No other users check
atm.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debugFS files for DPCM link management information.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Dynamic PCM core allows digital audio data to be dynamically
routed between different ALSA PCMs and DAI links on SoC CPUs with
on chip DSP devices. e.g. audio data could be played on pcm:0,0 and
routed to any (or all) SoC DAI links.
Dynamic PCM introduces the concept of Front End (FE) PCMs and Back
End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that
they can dynamically route digital audio data to any supported BE
PCM. A BE PCM has no ALSA device, but represents a DAI link and it's
substream and audio HW parameters.
e.g. pcm:0,0 routing digital data to 2 external codecs.
FE pcm:0,0 ----> BE (McBSP.0) ----> CODEC 0
+--> BE (McPDM.0) ----> CODEC 1
e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec.
FE pcm:0,0 ---
+--> BE (McBSP.0) ----> CODEC
FE pcm:0,1 ---
The digital audio routing is controlled by the usual ALSA method
of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the
routing based upon the mixer settings and configures the BE PCMs
based on routing and the FE HW params.
DPCM is designed so that most ASoC component drivers will need no
modification at all. It's intended that existing CODEC, DAI and
platform drivers can be used in DPCM based audio devices without
any changes. However, there will be some cases where minor changes
are required (e.g. for very tightly coupled HW) and there are
helpers to support this too.
Somethimes the HW params of a FE and BE do not match or are
incompatible, so in these cases the machine driver can reconfigure
any hw_params and make any DSP perform sample rate / format conversion.
This patch adds the core DPCM code and contains :-
o The FE and BE PCM operations.
o FE and BE DAI link support.
o FE and BE PCM creation.
o BE support API.
o BE and FE link management.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added support for a control that strobes a bit in
a register to high then back to low (or the inverse).
This is typically useful for hardware that requires
strobing a singe bit to trigger some functionality
and where exposing the bit in a normal single control
would require the user to first manually set then
again unset the bit again for the strobe to trigger.
Added convenience macro.
SOC_SINGLE_STROBE
Added accessor implementations.
snd_soc_get_strobe
snd_soc_put_strobe
Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added control type that can span multiple consecutive codec registers
forming a single signed value in a MSB/LSB manner.
The control dynamically adjusts to the register word size configured
in driver.
Added convenience macro.
SOC_SINGLE_XR_SX
Added accessor implementations.
snd_soc_info_xr_sx
snd_soc_get_xr_sx
snd_soc_put_xr_sx
Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.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>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Along with the IEC-60958 channel status word, CEA-861 Audio InfoFrames
are used in HDMI and DisplayPort to describe the parameters of the audio
stream. Hence, drivers for such devices may use these definitions to, for
instance, fill a CEA-861 data structure and pass it to a display driver
to configure an IP.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASoC: Merge tag 'v3.4-rc3' into for-3.5
Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting
annoyingly with the new development that's going on for Tegra so merge
it up to resolve those conflicts.
Conflicts:
sound/soc/soc-core.c
sound/soc/tegra/tegra_i2s.c
sound/soc/tegra/tegra_spdif.c
Rather than having the user half start a stream but avoid any DMA to
trigger data flow on links which don't pass through the CPU create a
DAPM route between the two DAI widgets using a hw_params configuration
provided by the machine driver with the new 'params' member of the
dai_link struct. If no configuration is provided in the dai_link then
use the old style even for CODEC<->CODEC links to avoid breaking
systems.
This greatly simplifies the userspace usage of such links, making them
as simple as analogue connections with the stream configuration being
completely transparent to them.
This is achieved by defining a new dai_link widget type which is created
when CODECs are linked and triggering the configuration of the link via
the normal PCM operations from there. It is expected that the bias
level callbacks will be used for clock configuration.
Currently only the DAI format, rate and channel count can be configured
and currently the only DAI operations which can be called are hw_params
and digital_mute(). This corresponds well to the majority of CODEC
drivers which only use other callbacks for constraint setting but there
is obviously much room for extension here. We can't simply call
hw_params() on startup as things like the system clocking configuration
may change at runtime and in future it will be desirable to offer some
configurability of the link parameters.
At present we are also restricted to a single DAPM link for the entire
DAI. Once we have better support for channel mapping it would also be
desirable to extend this feature so that we can propagate per-channel
power state over the link.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch uses simple-card driver instead of fsi-ak4642 on each board.
To select AK4642 driver, each boards select it on Kconfig.
This patch removes fsi-ak4642 driver which is no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current ASoC requires card.c file to each platforms in order to
specifies its CPU and Codecs pair.
But the differences between these were only value/strings of setting.
In order to reduce duplicate driver, this patch adds generic/simple-card.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also remove two warnings when CONFIG_SND_DEBUG is not set:
sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Supports larger register maps, not using unsigned ints for the full 32
bit as we rely on checking for negative registers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle.
Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros.
Add single register macro : SOC_SINGLE_SX_TLV.
Use snd_soc_info_volsw for .info
Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double.
kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros.
The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet.
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are no users any more and new drivers should be using supply widgets
which fully replace it anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Zeng Zhaoming <zengzm.kernel@gmail.com>
This change adds the logic to support using the jack detect mechanism built
in to the codec to detect both when a jack was inserted and what type of
jack is present.
This change also supports the use of an external mechanism for headphone
detection. If this mechanism exists, when the max98095_jack_detect function
is called, the hp_jack is simply passed NULL.
This change supports both simple headphones, powered headphones, microphones
and headsets with both headphones and a mic.
Signed-off-by: Rhyland Klein <rklein@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In version 3.4 the driver core acquired probe deferral which is a core way
of doing essentially the same thing as ASoC has been doing since forever
to make sure that all the devices needed to make up the card are present
without needing open coding in the subsystem.
Make basic use of this probe deferral mechanism for the cards, removing the
need to handle partially instantiated cards. We should be able to remove
even more code than this, though some of the checks we're currently doing
should stay since they're about things like suppressing unneeded DAPM runs
rather than deferring probes.
In order to avoid robustness issues with our teardown paths (which do need
quite a bit of TLC) add a check for aux_devs prior to attempting to set
things up, this means that we've got a reasonable idea that everything will
be there before we start. As with the removal of partial instantiation
support more work will be needed to make this work neatly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Currently operations on jack reporting take the CODEC mutex both to protect
the current jack status and also to protect the DAPM run which is triggered
on status updates. Since the addition of a DAPM-specific lock we no longer
need to worry about locking DAPM as it has its own finer grained lock so
create a per jack lock to take care of the jack status.
This is both cleaner where the jack isn't specifically associated with a
CODEC and clearer as it's much more obvious what the lock is protecting.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently DAPM widgets use the private data for their regulator.
Add a regulator * for widgets to use instead of private data.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename SND_SOC_DAPM_CLASS_PCM to SND_SOC_DAPM_CLASS_RUNTIME to
better match the usage and align with card mutex too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change SND_SOC_CARD_CLASS_PCM to SND_SOC_CARD_CLASS_RUNTIME to better
describe all uses for this mutex subclass and align with DAPM too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently stream events are only perfomed on codec stream widgets only.
There is now a need to be able to perform stream events on platform
widgets too.
e.g. we have the ABE platform driver with several DAI links
to dummy codecs. We need to be able to perform stream events on any
of the dummy codec DAI links.
This patch also removes the snd_soc_dai * parameter since it's already
contained within the rtd * parameter.
Finally makle stream event return void since no one checks it anyway.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add platform driver support for CPU DAI DAPM widgets.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It has now become necessary to use a DAPM mutex instead of the codec
mutex to lock the DAPM operations. This is due to the recent multi
component support and forth coming Dynamic PCM updates.
Currently we lock DAPM operations with the codec mutex of the calling
RTD context. However, DAPM operations can span the whole card context
and all components.
This patch updates the DAPM operations that use the codec mutex to
now use the DAPM mutex PCM subclass for all DAPM ops.
We also add a mutex subclass for DAPM init and PCM operations.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the first part of a change that is intended to improve
ASoC locking protection for DAPM and PCM operations.
This part of the series adds a mutex class for the soc_card mutex. The
SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the
SND_SOC_CARD_CLASS_PCM class is used for the forth coming Dynamic
PCM operations. The new mutex classes are required otherwise we will see a false
positive mutex deadlock warning between the card initialisation and the PCM
operations (something that would never deadlock in real life).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
"[RFC PATCH 0/2] audit of linux/device.h users in include/*"
https://lkml.org/lkml/2012/3/4/159
--
Nearly every subsystem has some kind of header with a proto like:
void foo(struct device *dev);
and yet there is no reason for most of these guys to care about the
sub fields within the device struct. This allows us to significantly
reduce the scope of headers including headers. For this instance, a
reduction of about 40% is achieved by replacing the include with the
simple fact that the device is some kind of a struct.
Unlike the much larger module.h cleanup, this one is simply two
commits. One to fix the implicit <linux/device.h> users, and then
one to delete the device.h includes from the linux/include/ dir
wherever possible.
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Merge tag 'device-for-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux
Pull <linux/device.h> avoidance patches from Paul Gortmaker:
"Nearly every subsystem has some kind of header with a proto like:
void foo(struct device *dev);
and yet there is no reason for most of these guys to care about the
sub fields within the device struct. This allows us to significantly
reduce the scope of headers including headers. For this instance, a
reduction of about 40% is achieved by replacing the include with the
simple fact that the device is some kind of a struct.
Unlike the much larger module.h cleanup, this one is simply two
commits. One to fix the implicit <linux/device.h> users, and then one
to delete the device.h includes from the linux/include/ dir wherever
possible."
* tag 'device-for-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux:
device.h: audit and cleanup users in main include dir
device.h: cleanup users outside of linux/include (C files)
Merge second batch of patches from Andrew Morton:
- various misc things
- core kernel changes to prctl, exit, exec, init, etc.
- kernel/watchdog.c updates
- get_maintainer
- MAINTAINERS
- the backlight driver queue
- core bitops code cleanups
- the led driver queue
- some core prio_tree work
- checkpatch udpates
- largeish crc32 update
- a new poll() feature for the v4l guys
- the rtc driver queue
- fatfs
- ptrace
- signals
- kmod/usermodehelper updates
- coredump
- procfs updates
* emailed from Andrew Morton <akpm@linux-foundation.org>: (141 commits)
seq_file: add seq_set_overflow(), seq_overflow()
proc-ns: use d_set_d_op() API to set dentry ops in proc_ns_instantiate().
procfs: speed up /proc/pid/stat, statm
procfs: add num_to_str() to speed up /proc/stat
proc: speed up /proc/stat handling
fs/proc/kcore.c: make get_sparsemem_vmemmap_info() static
coredump: add VM_NODUMP, MADV_NODUMP, MADV_CLEAR_NODUMP
coredump: remove VM_ALWAYSDUMP flag
kmod: make __request_module() killable
kmod: introduce call_modprobe() helper
usermodehelper: ____call_usermodehelper() doesn't need do_exit()
usermodehelper: kill umh_wait, renumber UMH_* constants
usermodehelper: implement UMH_KILLABLE
usermodehelper: introduce umh_complete(sub_info)
usermodehelper: use UMH_WAIT_PROC consistently
signal: zap_pid_ns_processes: s/SEND_SIG_NOINFO/SEND_SIG_FORCED/
signal: oom_kill_task: use SEND_SIG_FORCED instead of force_sig()
signal: cosmetic, s/from_ancestor_ns/force/ in prepare_signal() paths
signal: give SEND_SIG_FORCED more power to beat SIGNAL_UNKILLABLE
Hexagon: use set_current_blocked() and block_sigmask()
...
This addresses some header check warnings. DRM headers which include
"drm.h" have been excluded, as they indirectly include types.h.
Signed-off-by: Bobby Powers <bobbypowers@gmail.com>
Cc: Chris Ball <cjb@laptop.org>
Cc: Dave Airlie <airlied@linux.ie>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Pull media updates from Mauro Carvalho Chehab:
- V4L2 API additions to better support JPEG compression control
- media API additions to properly support MPEG decoders
- V4L2 API additions for image crop/scaling
- a few other V4L2 API DocBook fixes/improvements
- two new DVB frontend drivers: m88rs2000 and rtl2830
- two new DVB drivers: az6007 and rtl28xxu
- a framework for ISA drivers, that removed lots of common code found
at the ISA radio drivers
- a new FM transmitter driver (radio-keene)
- a GPIO-based IR receiver driver
- a new sensor driver: mt9m032
- some new video drivers: adv7183, blackfin, mx2_emmaprp, sii9234_drv,
vs6624
- several new board additions, driver fixes, improvements and cleanups.
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (295 commits)
[media] update CARDLIST.em28xx
[media] partially reverts changeset fa5527c
[media] stb0899: fix the limits for signal strength values
[media] em28xx: support for 2304:0242 PCTV QuatroStick (510e)
[media] em28xx: support for 2013:0251 PCTV QuatroStick nano (520e)
[media] -EINVAL -> -ENOTTY
[media] gspca - sn9c20x: Cleanup source
[media] gspca - sn9c20x: Simplify register write for capture start/stop
[media] gspca - sn9c20x: Add automatic JPEG compression mechanism
[media] gspca - sn9c20x: Greater delay in case of sensor no response
[media] gspca - sn9c20x: Optimize the code of write sequences
[media] gspca - sn9c20x: Add the JPEG compression quality control
[media] gspca - sn9c20x: Add a delay after Omnivision sensor reset
[media] gspca - sn9c20x: Propagate USB errors to higher level
[media] gspca - sn9c20x: Use the new video control mechanism
[media] gspca - sn9c20x: Fix loss of frame start
[media] gspca - zc3xx: Lack of register 08 value for sensor cs2102k
[media] gspca - ov534_9: Add brightness to OmniVision 5621 sensor
[media] gspca - zc3xx: Add V4L2_CID_JPEG_COMPRESSION_QUALITY control support
[media] pvrusb2: fix 7MHz & 8MHz DVB-T tuner support for HVR1900 rev D1F5
...
* tag 'v3.3': (1646 commits)
Linux 3.3
Don't limit non-nested epoll paths
netfilter: ctnetlink: fix race between delete and timeout expiration
ipv6: Don't dev_hold(dev) in ip6_mc_find_dev_rcu.
nilfs2: fix NULL pointer dereference in nilfs_load_super_block()
nilfs2: clamp ns_r_segments_percentage to [1, 99]
afs: Remote abort can cause BUG in rxrpc code
afs: Read of file returns EBADMSG
C6X: remove dead code from entry.S
wimax/i2400m: fix erroneous NETDEV_TX_BUSY use
net/hyperv: fix erroneous NETDEV_TX_BUSY use
net/usbnet: reserve headroom on rx skbs
bnx2x: fix memory leak in bnx2x_init_firmware()
bnx2x: fix a crash on corrupt firmware file
sch_sfq: revert dont put new flow at the end of flows
ipv6: fix icmp6_dst_alloc()
MAINTAINERS: Add Serge as maintainer of capabilities
drivers/video/backlight/s6e63m0.c: fix corruption storing gamma mode
MAINTAINERS: add entry for exynos mipi display drivers
MAINTAINERS: fix link to Gustavo Padovans tree
...
The tea575x-tuner module has been updated to use the latest V4L2 framework
functionality. This also required changes in the drivers that rely on it.
The tea575x changes are:
- The drivers must provide a v4l2_device struct to the tea module.
- The radio_nr module parameter must be part of the actual radio driver,
and not of the tea module.
- Changed the frequency range to the normal 76-108 MHz range instead of
50-150.
- Add hardware frequency seek support.
- Fix broken rxsubchans/audmode handling.
- The application can now select between stereo and mono.
- Support polling for control events.
- Add V4L2 priority handling.
And radio-sf16fmr2.c now uses the isa bus kernel framework.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Thanks-to: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
The <linux/device.h> header includes a lot of stuff, and
it in turn gets a lot of use just for the basic "struct device"
which appears so often.
Clean up the users as follows:
1) For those headers only needing "struct device" as a pointer
in fcn args, replace the include with exactly that.
2) For headers not really using anything from device.h, simply
delete the include altogether.
3) For headers relying on getting device.h implicitly before
being included themselves, now explicitly include device.h
4) For files in which doing #1 or #2 uncovers an implicit
dependency on some other header, fix by explicitly adding
the required header(s).
Any C files that were implicitly relying on device.h to be
present have already been dealt with in advance.
Total removals from #1 and #2: 51. Total additions coming
from #3: 9. Total other implicit dependencies from #4: 7.
As of 3.3-rc1, there were 110, so a net removal of 42 gives
about a 38% reduction in device.h presence in include/*
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
Allows the constraint lists to be declared const by drivers which seems
reasonable; there's plenty of other constification we could do if we were
being complete but this was easy and quick.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a hook to vmaster control to be called at each time
when the master value is changed. It'd be handy for an additional
mute LED control following the Master switch, for example.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mutex support for platform IO operations. e.g. can be used
for platform DAPM widget IO ops.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There's now core code which falls back to global CODEC operations for
DAI calls that needs to be able to tell if it's dealing with a CPU or
CODEC DAI and given the small number of DAIs in a typical system and
overall memory usage pattern saving a pointer per DAI is really not
worth the effort.
Reported-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch adds a set of functions which are intended to be used when
implementing a dmaengine based sound PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is intended to facilitate the merge of the two jack detection
mechanisms.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Chip designers frequently include things like the enable and disable
controls for algorithms in the register blocks which also hold the
coefficients. Since it's desirable to split out the enable/disable
control from userspace the plain SND_SOC_BYTES() isn't optimal for
these devices.
Add a SND_SOC_BYTES_MASK() which allows a bitmask from the first word
of the block to be excluded from the control. This supports the needs
of devices I've looked at and lets us have a reasonably simple API.
Further controls can be added in future if that's needed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow devices to export blocks of registers to the application layer,
intended for use for reading and writing coefficient data which can't
usefully be worked with by the kernel at runtime (for example, due to
requiring complex and expensive calculations or being the results of
callibration procedures). Currently drivers are using platform data to
provide configurations for coefficient blocks which isn't at all
convenient for runtime management or configuration development.
Currently only devices using regmap are supported, an error will be
generated for any attempt to work with a byte control on a non-regmap
device. There's no fundamental block to other devices so support could
be added if required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch supports DMAEngine to FSI driver.
It supports only Tx case at this point.
If platform/cpu doesn't support DMAEngine, FSI driver will
use PIO transfer.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to allow us to do smarter things with DAI links create DAPM
widgets which directly represent the DAIs in the DAPM graph. These are
automatically created from the DAIs as we probe the card with references
held in both directions between the widget and the DAI.
The widgets are not made available for direct instantiation by drivers,
they are created automatically from the DAIs. Drivers should be updated
to create stream routes using DAPM maps rather than by annotating AIF
and DAC widgets with streams.
In order to ease transition to this model from existing drivers we
automatically create DAPM routes between the DAI widgets and the existing
stream widgets which are started and stopped by the DAI widgets, though
the old stream handling mechanism is still in place. This also has the
nice effect of removing non-DAPM devices as any device with a DAI
acquires a widget automatically which will allow future simplifications
to the core DAPM logic.
The intention is that in future the AIF and DAI widgets will gain the
ability to interact such that we are able to manage activity on
individual channels independantly rather than powering up and down the
entire AIF as we do currently.
Currently we only generate these for CODECs, mostly as I have no systems
with non-CODEC DAPM to integrate with. It should be a simple matter of
programming to add the additional hookup for these.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Neater and avoids warnings when used in other places where const strings
are desired.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In order to allow us to do something smarter than iterate through widgets
doing strcmp() to work out what to power up for stream events change the
interface used to generate them to be based on the combination of a DAI
and a stream direction rather than just a simple string identifying the
stream.
At some point we'll probably want a set of channels too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Everything now uses snd_soc_dapm_new_controls() instead so we don't need
to make it part of the external API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
We never modify it and this lets us use a const string as the name without
warnings.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow platform widgets to be visible in debugfs like codec widgets.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
.. the number of the half-beast?
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Merge tag 'v3.3-rc3' as we've got several bugfixes in there which are
colliding annoyingly with development.
Linux 3.3-rc3
.. the number of the half-beast?
Conflicts:
sound/soc/codecs/wm5100.c
sound/soc/codecs/wm8994.c
This is usually not a use case dependant flag anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The new ASoC dynamic PCM core needs to create PCMs and substreams that are
for use by internal ASoC drivers only and not visible to userspace for
direct IO. These new PCMs are similar to regular PCMs expect they have no
device nodes or procfs entries. The ASoC component drivers use them in exactly
the same way as regular PCMs for PCM and DAI operations.
The intention is that a dynamic PCM based driver will register both regular
PCMs and internal PCMs. The regular PCMs will be used for all IO with userspace
however the internal PCMs will be used by the driver to route digital audio
through numerous back end DAI links (with potentially a DSP providing different
hw_params, DAI formats based on the regular front end PCM params) to devices
like CODECs, MODEMs, Bluetooth, FM, DMICs, etc
This patch adds a new snd_pcm_new_internal() API call to create the internal PCM
without device nodes or procfs. It also adds adds a new internal flag to snd_pcm.
[fixed minor coding-style issues by tiwai]
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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 is useful with complex DAPM Mixer operations
where we need to do additional work in addition to setting a few mixer register
bits.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC can only add kcontrols using codec and platform component device
handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for
SoC card machine drivers too. This allows the kcontrol to have a direct handle to
the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily
get it's private data.
This change makes snd_soc_add_controls() static and wraps it in the folowing
calls (card and dai are new) :-
snd_soc_add_card_controls()
snd_soc_add_codec_controls()
snd_soc_add_dai_controls()
snd_soc_add_platform_controls()
This patch also does a lot of small mechanical changes in individual codec drivers
to replace snd_soc_add_controls() with snd_soc_add_codec_controls().
It also updates the McBSP DAI driver to use snd_soc_add_dai_controls().
Finally, it updates the existing machine drivers that register controls to either :-
1) Use snd_soc_add_card_controls() where no direct codec control is required.
2) Use snd_soc_add_codec_controls() where there is direct codec control.
In the case of 1) above we also update the machine drivers to get the correct
component data pointers from the kcontrol (rather than getting the machine pointer
via the codec pointer).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI got each PortA/B parameter by porta_flags/portb_flags from platform.
And .set_rate function was shared for PortA/B.
This structure was not readable and not flexible.
This patch adds sh_fsi_port_info, and its own settings was added on each platform.
it is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a driver supporting the volume control and the mute pin. Shdn pin
and DAPM are not taken care of yet.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes below build warning when CONFIG_PCI is not set.
CC sound/sound_core.o
In file included from sound/sound_core.c:15:
include/sound/core.h:454: warning: 'struct pci_dev' declared inside parameter list
include/sound/core.h:454: warning: its scope is only this definition or declaration, which is probably not what you want
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM2200 is a low power mobile CODEC with enhanced Wolfson myZone
Ambient Noise Cancellation (ANC) intended for mobile telephony
applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modern devices allow systems to enable and disable individual supplies on
the device, allowing additional power saving by switching off regulators
which power portions of the device which are not currently in use. Add a
new SND_SOC_DAPM_REGULATOR_SUPPLY widget type factoring out the code for
managing such widgets from individual drivers.
The widget name will be used as the supply name when requesting the
regulator from the regulator API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When the hardware is configured with one or both of the IN4 inputs used
for DC measurement (with no DC blocking capacitor connected) then we can
improve power consumption slightly in idle modes by applying a register
write sequence. Provide platform data to enable this, implemented using
a regmap patch.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a driver is using regmap directly ensure that we're coherent with
non-ASoC register updates by using the regmap API directly to do our
read/modify/write cycles. This will bypass the ASoC cache but drivers
using regmap directly should not be using the ASoC cache.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Most devices accept data in formats that don't correspond directly to
their internal format. ALSA allows us to set a msbits constraint which
tells userspace about this in case it finds it useful (for example, in
order to avoid wasting effort dithering bits that will be ignored when
raising the sample size of data) so provide a mechanism for drivers to
specify the number of bits that are actually significant on a DAI and
add the appropriate constraints along with all the others.
This is done slightly awkwardly as the constraint is specified per sample
size - we loop over every possible sample size, including ones that the
device doesn't support and including ones that have fewer bits than are
actually used, but this is harmless as the upper layers do the right thing
in these cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
ALSA: usb-audio: add Yamaha MOX6/MOX8 support
ALSA: virtuoso: add S/PDIF input support for all Xonars
ALSA: ice1724 - Support for ooAoo SQ210a
ALSA: ice1724 - Allow card info based on model only
ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
ALSA: hdspm - Provide unique driver id based on card serial
ASoC: Dynamically allocate the rtd device for a non-empty release()
ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
ALSA: hda/cirrus - support for iMac12,2 model
ASoC: cx20442: add bias control over a platform provided regulator
ALSA: usb-audio - Avoid flood of frame-active debug messages
ALSA: snd-usb-us122l: Delete calls to preempt_disable
mfd: Put WM8994 into cache only mode when suspending
...
Fix up trivial conflicts in:
- arch/arm/mach-s3c64xx/mach-crag6410.c:
renamed speyside_wm8962 to tobermory, added littlemill right
next to it
- drivers/base/regmap/{regcache.c,regmap.c}:
duplicate diff that had already come in with other changes in
the regmap tree
The device model needs a release() function so it can free devices when
they become dereferenced. Do that for rtds.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Again, a lot of platforms have changes in here: pxa, samsung, omap,
at91, imx, ...
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Merge tag 'drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Driver specific changes
Again, a lot of platforms have changes in here: pxa, samsung, omap,
at91, imx, ...
* tag 'drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (54 commits)
ARM: sa1100: clean up of the clock support
ARM: pxa: add dummy clock for sa1100-rtc
RTC: sa1100: support sa1100, pxa and mmp soc families
RTC: sa1100: remove redundant code of setting alarm
RTC: sa1100: Clean out ost register
Input: zylonite-wm97xx - replace IRQ_GPIO() with gpio_to_irq()
pcmcia: pxa: replace IRQ_GPIO() with gpio_to_irq()
ARM: EXYNOS: Modified files for SPI consolidation work
ARM: S5P64X0: Enable SDHCI support
ARM: S5P64X0: Add lookup of sdhci-s3c clocks using generic names
ARM: S5P64X0: Add HSMMC setup for host Controller
ARM: EXYNOS: Add USB OHCI support to ORIGEN board
USB: Add Samsung Exynos OHCI diver
ARM: EXYNOS: Add USB OHCI support to SMDKV310 board
ARM: EXYNOS: Add USB OHCI device
net: macb: fix build break with !CONFIG_OF
i2c: tegra: Support DVC controller in device tree
i2c: tegra: Add __devinit/exit to probe/remove
net/at91_ether: use gpio_is_valid for phy IRQ line
ARM: at91/net: add macb ethernet controller in 9g45/9g20 DT
...
Export compress_offload.h and compress_params.h for userland to use
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the header files for ioctl definitions and header file for
driver APIs for lower level device drivers to use
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds the various definations used to define the encoder
and decoder parameters
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the minor numbers 2 and 3 for audio compressed offload devices.
Also add support for these devices in core
Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ensure that everything is seeing the same declaration by moving it to
a header file rather than putting the declaration in soc-core.c
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
DAI link endpoints and platform (DMA) devices are currently specified
by name. When instantiating sound cards from device tree, it may be more
convenient to refer to these devices by phandle in the device tree, and
for code to describe DAI links using the "struct device_node *"
("of_node") those phandles map to.
This change adds new fields to snd_soc_dai_link which can "name" devices
using of_node, enhances soc_bind_dai_link() to allow binding based on
of_node, and enhances snd_soc_register_card() to ensure that illegal
combinations of name and of_node are not used.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implement snd_soc_of_parse_audio_routing(), a utility function that can
parses a simple DAPM route table from device tree.The machine driver
specifies the DT property to use, since this is binding-specific.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implement snd_soc_of_parse_card_name(), a utility function that sets a
card's name from device tree. The machine driver specifies the DT
property to use, since this is binding-specific.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
wm8903_platform_data.gpio_cfg[] was intended to be interpreted as follows:
0: Don't touch this GPIO's configuration register
1..7fff: Write that value to the GPIO's configuration register
8000: Write zero to the GPIO's configuration register
other: Undefined (invalid)
The rationale is that platform data is usually global data, and a value of
zero means that the field wasn't explicitly set to anything (e.g. because
the field was new to the pdata type, and existing users weren't update to
initialize it) and hence the value zero should be ignored. 0x8000 is an
explicit way to get 0 in the register.
The code worked this way until commit 7cfe561 "ASoC: wm8903: Expose GPIOs
through gpiolib", where the behaviour was changed due to my lack of
awareness of the above rationale.
This patch reverts to the intended behaviour, and updates all in-tree users
to use the correct scheme. This also makes WM8903 consistent with other
devices that use a similar scheme.
WM8903_GPIO_NO_CONFIG is also renamed to WM8903_GPIO_CONFIG_ZERO so that
its name accurately reflects its purpose.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Cc: Olof Johansson <olof@lixom.net>
Cc: Colin Cross <ccross@android.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The existence of this parameter is purely historical. None of the CODEC drivers
uses it and we always pass in the same value anyway, so it should be safe to
remove it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A signal generator behaves as an input would but is not considered for
any of the special behaviour associated with external input pins. This
is especially useful when automatically working out not connected widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Add master_mode and master_id in platfrom_data since it's board
specific and board knows it.
Then we can remove the function pointer in platfrom_data to make
the driver more devicetree friendly.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Current fsi-ak4642 was using id_entry name in order to specify
FSI port and ak464x codec.
But it was no sense, no flexibility.
Platform can specify FSI/ak464x pair by this patch.
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A card is fully routed if the DAPM route table describes all connections on
the board.
When a card is fully routed, some operations can be automated by the ASoC
core. The first, and currently only, such operation is described below, and
implemented by this patch.
Codecs often have a large number of external pins, and not all of these pins
will be connected on all board designs. Some machine drivers therefore call
snd_soc_dapm_nc_pin() for all the unused pins, in order to tell the ASoC core
never to activate them.
However, when a card is fully routed, the information needed to derive the
set of unused pins is present in card->dapm_routes. In this case, have
the ASoC core automatically call snd_soc_dapm_nc_pin() for each unused
codec pin.
This has been tested with soc/tegra/tegra_wm8903.c and soc/tegra/trimslice.c.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now move the helper function for creating and reporting the jack-detection
to the common place. The driver that needs this functionality should
select CONFIG_SND_KCTL_JACK kconfig.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sta32x resets and loses all configuration during ESD test.
Work around by polling the CONFA register once a second
and restore all coeffcients and registers when CONFA
changes unexpectedly.
Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a structure for platform specific configuration and use it,
thereby removing a few FIXMEs which marked hard-coded values.
Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are no current users and new drivers ought to be using the regmap
API and its cache implementation directly so just delete the ASoC copy.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My usual technique for finding definitions is to search for "name {"
which breaks with the extra space.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'modsplit-Oct31_2011' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux: (230 commits)
Revert "tracing: Include module.h in define_trace.h"
irq: don't put module.h into irq.h for tracking irqgen modules.
bluetooth: macroize two small inlines to avoid module.h
ip_vs.h: fix implicit use of module_get/module_put from module.h
nf_conntrack.h: fix up fallout from implicit moduleparam.h presence
include: replace linux/module.h with "struct module" wherever possible
include: convert various register fcns to macros to avoid include chaining
crypto.h: remove unused crypto_tfm_alg_modname() inline
uwb.h: fix implicit use of asm/page.h for PAGE_SIZE
pm_runtime.h: explicitly requires notifier.h
linux/dmaengine.h: fix implicit use of bitmap.h and asm/page.h
miscdevice.h: fix up implicit use of lists and types
stop_machine.h: fix implicit use of smp.h for smp_processor_id
of: fix implicit use of errno.h in include/linux/of.h
of_platform.h: delete needless include <linux/module.h>
acpi: remove module.h include from platform/aclinux.h
miscdevice.h: delete unnecessary inclusion of module.h
device_cgroup.h: delete needless include <linux/module.h>
net: sch_generic remove redundant use of <linux/module.h>
net: inet_timewait_sock doesnt need <linux/module.h>
...
Fix up trivial conflicts (other header files, and removal of the ab3550 mfd driver) in
- drivers/media/dvb/frontends/dibx000_common.c
- drivers/media/video/{mt9m111.c,ov6650.c}
- drivers/mfd/ab3550-core.c
- include/linux/dmaengine.h
The <linux/module.h> pretty much brings in the kitchen sink along
with it, so it should be avoided wherever reasonably possible in
terms of being included from other commonly used <linux/something.h>
files, as it results in a measureable increase on compile times.
The worst culprit was probably device.h since it is used everywhere.
This file also had an implicit dependency/usage of mutex.h which was
masked by module.h, and is also fixed here at the same time.
There are over a dozen other headers that simply declare the
struct instead of pulling in the whole file, so follow their lead
and simply make it a few more.
Most of the implicit dependencies on module.h being present by
these headers pulling it in have been now weeded out, so we can
finally make this change with hopefully minimal breakage.
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
ALSA: hda - Keep EAPD turned on for old Conexant chips
ALSA: hda/realtek - Fix missing volume controls with ALC260
ASoC: wm8940: Properly set codec->dapm.bias_level
ALSA: hda - Fix pin-config for ASUS W90V
ALSA: hda - Fix surround/CLFE headphone and speaker pins order
ALSA: hda - Fix typo
ALSA: Update the sound git tree URL
ALSA: HDA: Add new revision for ALC662
ASoC: max98095: Convert codec->hw_write to snd_soc_write
ASoC: keep pointer to resource so it can be freed
ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
ASoC: da7210: Add support for line out and DAC
ASoC: da7210: Add support for DAPM
ALSA: hda/realtek - Fix DAC assignments of multiple speakers
ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
ASoC: Set sgtl5000->ldo in ldo_regulator_register
ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
...
With this flag, each dai_link in machine driver can choose
to ignore pmdown_time during DAPM shut down sequence.
If the ignore_pmdown_time is set, the DAPM for corresponding DAI
will be executed immediately.
Signed-off-by: Ramesh Babu K V <ramesh.babu@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With this flag codec drivers can indicate that it is desired
to ignore the pmdown_time for DAPM shutdown sequence when
playback stream is stopped.
The DAPM sequence will be executed without delay in this case.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We really should be doing this in the core, not in a driver...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
The number of connected input and output endpoints for a given widgets
can't change during a DAPM run so there is no need to redo the recursion
through branches of the tree we've already visited. Doing this on one of
my test systems gives an improvement of:
Power Path Neighbour
Before: 63 607 731
After: 63 141 181
which scales up well as more widgets are involved in paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handling of user control elements was implemented for all types except
ENUMERATED. This type will be needed for the device-specific mixers of
upcoming FireWire drivers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By accident few places still uses the _2r calls from
the core.
This is a quick fix, the drivers using the old callbacks
going to be changed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We do not have users for snd_soc_put_volsw_2r anymore.
It can be removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the put_volsw/put_volsw_2r in one function.
To avoid build breakage in twl6040 keep the
snd_soc_put_volsw_2r as define, and map it snd_soc_put_volsw.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the get_volsw/get_volsw_2r in one function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle the info_volsw/info_volsw_2r in one function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SOC_SINGLE/DOUBLE_VALUE is used for mixer controls, where the
bits are within one register.
Assign .rreg to be the same as .reg for these types.
With this change we can tell if the mixer in question:
is mono:
mc->reg == mc->rreg && mc->shift == mc->rshift
is stereo, within single register:
mc->reg == mc->rreg && mc->shift != mc->rshift
is stereo, in two registers:
mc->reg != mc->rreg
The patch provide a small inline function to query, if the mixer
is stereo, or mono.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Similar to Line Out, these constants form the base for future
patches enabling input jack reporting for Line in jacks.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some widgets will get power_check() run on them more than once during a
DAPM run, most commonly due to supply widgets checking to see if their
consumers are powered up. It's wasteful to do this so cache the result
of power_check() during a run. For one system I tested this on I got an
improvement of:
Power Path Neighbour
Before: 106 970 1186
After: 69 727 905
from this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to reduce the number of DAPM power checks we run keep a list of
widgets which have been changed since the last DAPM run and iterate over
that rather than the full widget list. Whenever we change the power state
for a widget we add all the source and sink widgets it has to the dirty
list, ensuring that all widgets in the path are checked.
This covers more widgets than we need to as some of the neighbour widgets
won't be connected but it's simpler as a first step. On one system I tried
this gave:
Power Path Neighbour
Before: 207 1939 2461
After: 114 1066 1327
which seems useful.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the new macro we can remove duplicated code
for the SOC_DOUBLE_R type of controls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the new macro we can remove duplicated code
for the SOC_DOUBLE type of controls.
We can also remap the SOC_SINGLE_VALUE macro to
SOC_DOUBLE_VALUE
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Export the default mmap function, snd_pcm_lib_default_mmap().
The upcoming non-snooping support in HD-audio driver will use this
to override the mmap method.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For almost all machines the DAI format is a constant, always set to the
same thing. This means that not only should we normally set it on init
rather than in hw_params() (where it has been for historical reasons) we
should also allow users to configure this by setting a variable in the
dai_link structure. The combination of these two will make many machine
drivers even more data driven.
Implement a new dai_fmt field in the dai_link doing just that. Since 0 is
a valid value for many format flags and we need to be able to tell if the
field is actually set also add one to all the values used to configure
formats.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM1250 EV1 has some GPIOs which can be used to control the behaviour
at runtime. Request them all if supplied and add a set_bias_level()
function to start and stop the clocks.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The number of times we look at a potentially connected neighbour is just
as important as the number of times we actually recurse into looking at
that neighbour so also collect that statistic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since commit [e58aa3d2: genirq: Run irq handlers with interrupts disabled],
We run all interrupt handlers with interrupts disabled
and we even check and yell when an interrupt handler
returns with interrupts enabled (see commit [b738a50a:
genirq: Warn when handler enables interrupts]).
So now this flag is a NOOP and can be removed.
Signed-off-by: Yong Zhang <yong.zhang0@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model_id is no longer needed within the platform_data
for the TPA driver since the model of TPA specified
with the device name (tpa6130a2/tpa6140a2).
Also update rx51 (the only affected user) to use the device name rather
than platform data.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The orginal code does not cover the case that one DAI such as codec
may be shared between other two DAIs(CPU).
When do symmetry checking, altough the codec DAI requires symmetry,
the two CPU DAIs may still be configured to run on different rates.
We change to check each DAI's state separately instead of only checking
the dai link to prevent this issue.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the longest standing areas for improvement in ASoC has been the
DAPM algorithm - it repeats the same checks many times whenever it is run
and makes no effort to limit the areas of the graph it checks meaning we
do an awful lot of walks over the full graph. This has never mattered too
much as the size of the graph has generally been small in relation to the
size of the devices supported and the speed of CPUs but it is annoying.
In preparation for work on improving this insert a trace point after the
graph walk has been done. This gives us specific timing information for
the walk, and in order to give quantifiable (non-benchmark) numbers also
count every time we check a link or check the power for a widget and report
those numbers. Substantial changes in the algorithm may require tweaks to
the stats but they should be useful for simpler things.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM5100 is a highly integrated low power audio subsystem with advanced
digital signal processing capabilities including effects, speech clarity
enhancement and active noise cancellation. This initial driver provides
support for basic audio paths, further patches will provide more
complete functionality.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Add a helper function to allow drivers to disable hardware resampling
when the application has specified the SNDRV_PCM_HW_PARAMS_NORESAMPLE
flag.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
1. add different clkmux mode handling
SAIF can use two instances to implement full duplex (playback &
recording) and record saif may work on EXTMASTER mode which is
using other saif's BITCLK&LRCLK.
The clkmux mode could be set in pdata->init() in mach-specific code.
For generic saif driver, it only needs to know who is his master
and the master id is also provided in mach-specific code.
2. support playback and capture simutaneously however the sample
rates can not be different due to hw limitation.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Acked-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The semantics of snd_mpu401_uart_new()'s interrupt parameters are
somewhat counterintuitive: To prevent the function from allocating its
own interrupt, either the irq number must be invalid, or the irq_flags
parameter must be zero. At the same time, the irq parameter being
invalid specifies that the mpu401 code has to work without an interrupt
allocated by the caller. This implies that, if there is an interrupt
and it is allocated by the caller, the irq parameter must be set to
a valid-looking number which then isn't actually used.
With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value,
which forces us to handle the parameters differently.
This patch introduces a new flag MPU401_INFO_IRQ_HOOK for when the
device interrupt is handled by the caller, and makes the allocation of
the interrupt to depend only on the irq parameter. As suggested by
Takashi, the irq_flags parameter was dropped because, when used, it had
the constant value IRQF_DISABLED.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly to PLLs/FLLs some modern CODECs provide selectable system clock
sources. When the clock is the clock for a DAI we do not usually need to
identify which clock is being configured so can use clk_id for the source
clock but with CODEC wide system clocks we will need to specify both the
clock being configured and the source.
Add a source argument to the CODEC driver set_sysclk() operation to
reflect this. As this operation is not as widely used as the DAI
set_sysclk() operation the change is not very invasive. We probably
ought to go and make the same alternation for DAIs at some point.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Devices that need this exist; obviously the newer regmap defaults
mechanism will deal with this more happily.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Misc fixes to improve code readability:
* rename struct pm_qos_request_list to struct pm_qos_request,
* rename pm_qos_req parameter to req in internal code,
consistenly use req in the API parameters,
* update the in-kernel API callers to the new parameters names,
* rename of fields names (requests, list, node, constraints)
Signed-off-by: Jean Pihet <j-pihet@ti.com>
Acked-by: markgross <markgross@thegnar.org>
Reviewed-by: Kevin Hilman <khilman@ti.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
The PM QoS implementation files are better named
kernel/power/qos.c and include/linux/pm_qos.h.
The PM QoS support is compiled under the CONFIG_PM option.
Signed-off-by: Jean Pihet <j-pihet@ti.com>
Acked-by: markgross <markgross@thegnar.org>
Reviewed-by: Kevin Hilman <khilman@ti.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
If devices can unconditionally support idle_bias_off let them flag it in
their driver structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch adds support for the Analog Devices ADAU1373 audio codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My gmail account got disabled and I'm not going to reopen it.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow drivers to set up their own regmap API structures. This is mainly
useful with MFDs where the core driver will have set up regmap at the
minute, though it may make sense to push the existing regmap setup out
of the core into the drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Remove all the ASoC specific physical I/O code and replace it with calls
into the regmap API. The bulk write code can only be used safely if all
regmap calls are locked with the CODEC lock, we need to add bulk support
to the regmap API or replace the code with an open coded loop (though
currently it has no users...).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
For marketing reasons the part will be called WM8996. In order to avoid
user confusion rename the driver to reflect this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Kukjin Kim <kgene.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@ti.com>
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (430 commits)
[media] ir-mce_kbd-decoder: include module.h for its facilities
[media] ov5642: include module.h for its facilities
[media] em28xx: Fix DVB-C maxsize for em2884
[media] tda18271c2dd: Fix saw filter configuration for DVB-C @6MHz
[media] v4l: mt9v032: Fix Bayer pattern
[media] V4L: mt9m111: rewrite set_pixfmt
[media] V4L: mt9m111: fix missing return value check mt9m111_reg_clear
[media] V4L: initial driver for ov5642 CMOS sensor
[media] V4L: sh_mobile_ceu_camera: fix Oops when USERPTR mapping fails
[media] V4L: soc-camera: remove soc-camera bus and devices on it
[media] V4L: soc-camera: un-export the soc-camera bus
[media] V4L: sh_mobile_csi2: switch away from using the soc-camera bus notifier
[media] V4L: add media bus configuration subdev operations
[media] V4L: soc-camera: group struct field initialisations together
[media] V4L: soc-camera: remove now unused soc-camera specific PM hooks
[media] V4L: pxa-camera: switch to using standard PM hooks
[media] NetUP Dual DVB-T/C CI RF: force card hardware revision by module param
[media] Don't OOPS if videobuf_dvb_get_frontend return NULL
[media] NetUP Dual DVB-T/C CI RF: load firmware according card revision
[media] omap3isp: Support configurable HS/VS polarities
...
Fix up conflicts:
- arch/arm/mach-omap2/board-rx51-peripherals.c:
cleanup regulator supply definitions in mach-omap2
vs
OMAP3: RX-51: define vdds_csib regulator supply
- drivers/staging/tm6000/tm6000-alsa.c (trivial)
Change locking to allow tea575x-radio device to be opened multiple times.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Convert tea575x-tuner to use the new V4L2 control framework. Also add
ext_init() callback that can be used by a card driver for additional
initialization right before registering the video device (for SF16-FMR2).
Also embed struct video_device to struct snd_tea575x to simplify the code.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Move the macros depending on snd_mask_min() and co out of pcm.h into
pcm_params.h. Otherwise using some params_*() macros will give comiple
errors without inclusion of pcm_params.h.
Also use hw_param_interval_c() and hw_param_mask_c() for const pointer.
Reported-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for Dynamic PCM (AKA 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
after completion of stream events.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (43 commits)
fs: Merge split strings
treewide: fix potentially dangerous trailing ';' in #defined values/expressions
uwb: Fix misspelling of neighbourhood in comment
net, netfilter: Remove redundant goto in ebt_ulog_packet
trivial: don't touch files that are removed in the staging tree
lib/vsprintf: replace link to Draft by final RFC number
doc: Kconfig: `to be' -> `be'
doc: Kconfig: Typo: square -> squared
doc: Konfig: Documentation/power/{pm => apm-acpi}.txt
drivers/net: static should be at beginning of declaration
drivers/media: static should be at beginning of declaration
drivers/i2c: static should be at beginning of declaration
XTENSA: static should be at beginning of declaration
SH: static should be at beginning of declaration
MIPS: static should be at beginning of declaration
ARM: static should be at beginning of declaration
rcu: treewide: Do not use rcu_read_lock_held when calling rcu_dereference_check
Update my e-mail address
PCIe ASPM: forcedly -> forcibly
gma500: push through device driver tree
...
Fix up trivial conflicts:
- arch/arm/mach-ep93xx/dma-m2p.c (deleted)
- drivers/gpio/gpio-ep93xx.c (renamed and context nearby)
- drivers/net/r8169.c (just context changes)
Formatting a PCM name is useful for module debug too.
Add snd_prefix when making function public.
[minor coding-style fixes by tiwai]
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All these are instances of
#define NAME value;
or
#define NAME(params_opt) value;
These of course fail to build when used in contexts like
if(foo $OP NAME)
while(bar $OP NAME)
and may silently generate the wrong code in contexts such as
foo = NAME + 1; /* foo = value; + 1; */
bar = NAME - 1; /* bar = value; - 1; */
baz = NAME & quux; /* baz = value; & quux; */
Reported on comp.lang.c,
Message-ID: <ab0d55fe-25e5-482b-811e-c475aa6065c3@c29g2000yqd.googlegroups.com>
Initial analysis of the dangers provided by Keith Thompson in that thread.
There are many more instances of more complicated macros having unnecessary
trailing semicolons, but this pile seems to be all of the cases of simple
values suffering from the problem. (Thus things that are likely to be found
in one of the contexts above, more complicated ones aren't.)
Signed-off-by: Phil Carmody <ext-phil.2.carmody@nokia.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Add a convenience macro for external enumerated widgets.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform probe to register platform kcontrols and DAPM just like
the CODEC probe().
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform driver widgets to perform any IO required for DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for Dynamic PCM (AKA DSP) support.
Allow platform drivers to register kcontrols.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for ASoC Dynamic PCM (AKA DSP) support.
Allow platform driver to perform IO. Intended for platform DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of ioctl definition in sound/sb16_csp.h contains the data size
over 8kB, and this causes build errors on architectures like MIPS,
which define _IOC_SIZEBITS=13.
For avoiding this build errors but keeping the compatibility, manually
expand with _IOC() instead of using _IOW() for the problematic ioctl.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kill tasklet usage in rawmidi core code. Use workq for the event callback
instead of tasklet (which is used only in core/seq/seq_midi.c).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will be removed in -next so let's drop it from mainline as soon as
we can in order to minimise surprises.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Normally DAPM will power up any connected audio path. This is not ideal
for sidetone paths as with sidetone paths the audio path is not wanted in
itself, it is only desired if the two paths it provides a sidetone between
are both active. If the sidetone path causes a power up then it can be
hard to minimise pops as we first power up either the sidetone or the main
output path and then power the other, with the second power up potentially
introducing a DC offset.
Address this by introducing the concept of a weak path. If a path is marked
as weak then DAPM will ignore that path when walking the graph, though all
the relevant controls are still available to the application layer to allow
these paths to be configured.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).
The new ASoC Dynamic PCM core allows 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 routes
audio in the analog domain). The Dynamic PCM 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 PCM hw_params()
could configure multiple backend DAI hw_params() with similar or different
hw parameters at the same time.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some ASoC components depend on other ASoC components to provide clocks and
power resources in order to probe() and vice versa for remove().
Allow components to be ordered so that components can be probed() and removed()
in sequences that conform to their dependencies.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently pcm_new() passes in 3 arguments :- card, pcm and DAI.
Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains
card, pcm and DAI along with other members too that are useful too.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
A driver for the AK4641 codec used in iPAQ hx4700 and Glofiish M800
among others.
Signed-off-by: Harald Welte <laforge@gnumonks.org>
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide real card and bus_info instead of hardcoded values.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
struct snd_card *card is present in struct snd_tea575x but never used.
Remove it.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
freq_fixup is a constant, no need to hold it in struct snd_tea575x and set in
each driver.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow ASoC machine drivers to register a driver name
and a longname. This allows user space to determine
the flavour of machine driver.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Implement generic read/write functions to access TEA575x tuners. They're now
implemented 4 times (once in es1968 and 3 times in fm801).
This also allows mute to work on all cards.
Also improve tuner detection/initialization.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change will allow multiple widgets to be affected by the same
control. For example, a single register bit that controls separate muxes
in both the L and R audio paths.
This change updates the code that handles relevant controls to be able
to iterate over a list of affected widgets. Note that only the put
functions need significant modification to implement the iteration; the
get functions do not need to iterate, nor unify the results, since all
affected widgets reference the same kcontrol.
When creating the list of widgets, always create a 1-sized list, since
the control sharing is not implemented in this change.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Future changes will need reference to the kcontrol created for a given
kcontrol_new. Store the created kcontrol values now.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A future change will modify struct snd_soc_dapm_widget to store the
actual kcontrol pointers for each kcontrol_new in a field named
kcontrols. Rename the existing kcontrols field to enable this.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the creation of the DAPM debugfs directory to snd_soc_dapm_debugfs_init
instead of having the same duplicated code in both codec and card DAPM setup.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The enum texts are supposed to be const char * const []. Without the
second const, it gets compile warnings like
sound/soc/codecs/max98095.c:607:2: warning: initialization discards qualifiers from pointer target type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the equalizer and biquad filter controls.
Signed-off-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
replace the tab with spaces,
make it align with other paragraphs
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Those should not be modified (and are not) by the core code, so make them const.
This also makes them consistent with the same members of snd_soc_codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8915 is an ultra low power mobile CODEC designed for smartphones,
featuring a mixture of digital and analogue I/O with flexible mixing
options and advanced low power accessory detection functionality in a
compact package.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow CODEC and card drivers to point to an array of controls from their
driver structure rather than explicitly calling snd_soc_add_controls().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the MAX98095 CODEC driver.
Signed-off-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide the top level ASoC core functions for indicating whether
a given register is readable or writable.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using struct snd_soc_reg_access for the read/write/vol attributes
of the registers, we provide callbacks that automatically determine whether
a given register is readable/writable or volatile.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is mainly used by the soc-cache code to easily determine the
currently used underlying serial bus. Set SND_SOC_CUSTOM to 1 so we
can distinguish it if it is not initialized or set.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As it has become more common to have to write firmware or similar
large chunks of data to the hardware, add a function to perform
raw bulk writes that bypass the cache. This only handles volatile
registers as we should avoid getting out of sync with the actual
cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's a big no-no to use pgprot_noncached() when mmap'ing such buffers
into userspace since they are mapped cachable in kernel space.
This can cause all sort of interesting things ranging from to garbled
sound to lockups on various architectures. I've observed that usb-audio
is broken on powerpc 4xx for example because of that.
Also remove the now unused snd_pcm_lib_mmap_noncached(). It's
an arch business to know when to use uncached mappings, there's
already hacks for MIPS inside snd_pcm_default_mmap() and other
archs are supposed to use dma_mmap_coherent().
(See my separate patch that adds dma_mmap_coherent() to powerpc)
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
CC: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This stops code that handles widgets generically from attempting to access
registers for these widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
Add a function to dynamically replace a given control. If the
control does not already exist, a third parameter is used to determine
whether to actually add that control. This is useful in cases where
downloadable firmware at runtime can add or replace existing controls.
A separate patch needs to be made to allow ALSA Mixer to render the
replaced controls on the fly.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve tea575x-tuner with various good things from radio-maestro:
- extend frequency range to 50-150MHz
- fix querycap(): card name, CAP_RADIO
- improve g_tuner(): CAP_STEREO, stereo and tuned indication
- improve g_frequency(): tuner index checking and reading frequency from HW
- improve s_frequency(): tuner index and type checking
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new API function snd_ctl_activate_id() for activate / inactivate
the control element dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently will ignore prefixes when creating DAPM controls. Since currently
all control creation goes through snd_soc_cnew() we can fix this by factoring
the prefixing into that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When multi component systems use DAIless amplifiers which require clocking
configuration it is at best hard to use the current clocking API as this
requires a DAI even though the device may not even have one. Address this
by adding set_sysclk() and set_pll() operations and APIs for CODECs.
In order to avoid issues with devices which could be used either with or
without DAIs make the DAI variants call through to their CODEC counterparts
if there is no DAI specific operation. Converting over entirely would create
problems for multi-DAI devices which offer per-DAI clocking setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow a slight simplification of CODEC drivers by allowing DAPM routes and
widgets to be provided in a table. They will be instantiated at the end of
CODEC probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This patch adds support for tlv320aic3205 and tlv320aic3254 codecs.
It doesn't include miniDSP support for aic3254.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is run after the DAPM widgets and routes are added, allowing setup
of things like jacks using the routes. The main card probe() is run before
anything else so can't be used for this purpose.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
These will be added after all devices are registered and allow most DAI
init functions in machine drivers to be replaced by simple data.
Regular controls are not supported as the registration function still
works in terms of CODECs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This means that rather than adding the board specific DAPM widgets to a
random CODEC DAPM context they can be added to the card itself which is
a bit cleaner. Previously there only was one DAPM context and it was
tied to the single supported CODEC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM9081 IRQ output can be either active high or active low and can
support either CMOS or open drain modes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some systems wish to use jacks as wake sources. Provide a wake flag in the
GPIO configuration which causes the driver to enable the IRQ as a wake
source.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Use AC97 macros (sometimes already existing, or newly added)
instead of error-prone repetition of open-coded values.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the core code where sparse complains. In most cases, this means
just adding annotations to confirm that we indeed want to do the dirty
things we're doing.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* There is no hysteresis enable field in the current datasheet.
* Mic detection threshold field is only 2 bits wide.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds soc-jack support for adding voltage zones and for
detecting jack type
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move Chip Select control out of the CODEC code for CS4271.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Reviewed-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide driver data for cards within the card structure. To simplify the
implementation of the PM operations we don't use the struct device driver
data as this is used by the core to retrieve the card in callbacks from
the device model and PM core.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allows drivers to distinguish which subsequence is being notified when
they get called back.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Could just as well live in sysfs but sysfs doesn't have the simple
value export helpers debugfs does.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow hookup of cards registered directly with the core to the PM
operations by exporting the device power management operations to
modules, also exporting the default PM operations since it is
expected that most cards will end up using exactly the same setup.
Note that the callbacks require that the driver data for the card be
the snd_soc_card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In order to support cards instantiated without using soc-audio remove
the use of the platform device in the card probe() and remove() ops.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The platform device for the card is tied closely to the soc-audio
implementation which we're currently trying to remove in favour of
allowing cards to have their own devices. Begin removing it by
replacing it with the card in the suspend and resume callbacks we
give to cards, also taking the opportunity to remove the legacy
suspend types which are currently hard coded anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
As requested by Takashi and Jaroslav, these arrays should not be in the
header file.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Incorporate changes by Florian Faber into hdspm.c. Code taken from
http://wiki.linuxproaudio.org/index.php/Driver:hdspe
Heavily reworked to mostly comply with the coding standard (whitespace
fixes, line width, C++ style comments)
The code was tested and confirmed to be working on RME RayDAT.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current format selection of FSI-codecs depended on platform information for FSI,
and chip default settings for codecs. It is not understandable/formal method.
This patch modify FSI and FSI-codecs to use snd_soc_dai_set_fmt.
But FSI can use I2S/PCM and SPDIF format today.
It can be selected to I2S/PCM by snd_soc_dai_set_fmt, but can not select SPDIF.
So, this patch change FSI platform information to have DAI/SPDIF mode.
If platform selects DAI mode (default),
FSI-codecs can select I2S/PCM by snd_soc_dai_set_fmt,
and if it is SPDIF mode, FSI become SPDIF format.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have zero users for PGA controls and the core support for them was
removed a while ago so no point in cut'n'pasting them into new macros,
even if it's too much hassle to update the existing ones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
We generally refer to registers as unsigned ints (including in the
underlying CODEC driver operation).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch add snd_soc_dai_ops :: set_fmt to FSI driver and
select master/slave clock mode by snd_soc_dai_set_fmt on
fsi-xxx.c instead of platform infomation code.
This patch remove fsi_is_master function which is no longer needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added support for CS4271 codec to ASoC.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also, update platform_data GPIO handling to have an explicit "don't
touch this pin" option.
Add #defines for the GPIO pin functions.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is primarily needed to avoid writing back to the cache
whenever we are syncing the cache with the hardware. This gives a
performance benefit especially for large register maps.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many modern devices have features such as DC servos which take time to start.
Currently these are handled by per-widget events but this makes it difficult
to paralleise operations on multiple widgets, meaning delays can end up
being needlessly serialised. By providing a callback to drivers when all
widgets of a given type have been handled during a DAPM sequence the core
allows drivers to start operations separately and wait for them to complete
much more simply.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
With larger devices there may be many widgets of the same type in series
in an audio path. Allow drivers to specify an additional level of ordering
within each widget type by adding a subsequence number to widgets and then
splitting operations on widgets so that widgets of the same type but
different sequence numbers are processed separately. A typical example
would be a supply widget which requires that another widget be enabled
to provide power or clocking.
SND_SOC_DAPM_PGA_S() and SND_SOC_DAPM_SUPPLY_S() macros are provided
allowing this to be used with PGAs and supplies as these are the most
commonly affected widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The machine driver can't register the card directly and need to do this thru
soc-audio device creation
This patch allows the register and unregister card to be directly called by
machine drivers
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <priya.harsha@intel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the soc_probe initializes the card hence it does the card list
initialzation. But if machines directly register the card they would need to
do these steps, so putting them as inline would save lot of code
This patch adds an inline to do list initialzation
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Harsha Priya <harsha.priya@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that all calls to readable_register()/volatile_register() go via
the snd_soc_codec function pointers.
If the default register access table has been given but no functions
for handling readable()/volatile() registers, use the default ones provided
by soc-cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For common scenarios, device drivers can provide a table of all the
registers that are at least either readable/writable/volatile. The idea
is that if a register lookup fails, all of its read/write/vol members
will be zero and will be treated as default. This also reduces the
size of the register access array.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Simplify the use of reg_size, by calculating it once and storing it in
the codec structure for later reference. The value of reg_size is
reg_cache_size * reg_word_size.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Everything else is using snd_soc_ so we should use it here too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
A couple Tegra ASoC drivers will create debugfs entries. Mark requested
these by under debugfs/asoc/ not just debugfs/. To enable this, export
the dentry representing debugfs/asoc/.
Also, rename debugfs_root -> asoc_debugfs_root now it's exported to
prevent potential symbol name clashes.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce the helper function snd_ctl_enum_info() to fill out the
elem_info fields for an enumerated control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This new type is a virtual version of snd_soc_dapm_mux. It is used
when a backing register value is not necessary for deciding which
input path to connect. A simple virtual enumeration control e.g.
SOC_DAPM_ENUM_VIRT() can be exposed to userspace which will be used
to choose which path to connect.
The snd_soc_dapm_virt_mux type ensures that during the initial
path setup, the first (which is also the default) input path will
be connected.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Attempt to minimise audible effects from mixer and mux updates by
implementing the actual register changes between powering down widgets
that have become unused and powering up widgets that are newly used.
This means that we're making the change with the minimum set of widgets
powered, that the input path is connected when we're powering up widgets
(so things like DC offset correction can run with their signal active)
and that we bring things down to cold before switching away. Since
hardware tends to be designed for the power on/off case more than for
dynamic reconfiguration this should minimise pops and clicks during
reconfiguration while active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Power change event like stream start/stop or kcontrol change in a
cross-device path originates from one device but codec bias and widget power
changes must be populated to another devices on that path as well.
This patch modifies the dapm_power_widgets so that all the widgets on a
sound card are checked for a power change, not just those that are specific
to originating device. Also bias management is extended to check all the
devices. Only exception in bias management are widgetless codecs whose bias
state is changed only if power change is originating from that context.
DAPM context test is added to dapm_seq_run to take care of if power sequence
extends to an another device which requires separate register writes.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling widgets from DAPM context is required when extending the ASoC
core to cross-device paths. Even the list of widgets are now kept in
struct snd_soc_card, the widget listing in sysfs and debugs remain sorted
per device.
This patch makes possible to build cross-device paths but does not extend
yet the DAPM to handle codec bias and widget power changes of an another
device.
Cross-device paths are registered by listing the widgets from device A in
a map for device B. In case of conflicting widget names between the devices,
a uniform name prefix is needed to separate them. See commit ead9b91
"ASoC: Add optional name_prefix for kcontrol, widget and route names" for
help.
An example below shows a path that connects MONO out of A into Line In of B:
static const struct snd_soc_dapm_route mapA[] = {
{"MONO", NULL, "DAC"},
};
static const struct snd_soc_dapm_route mapB[] = {
{"Line In", NULL, "MONO"},
};
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling DAPM paths from DAPM context is a first prerequisite when
extending ASoC core to cross-device paths. This patch is almost a nullop and
does not allow to construct cross-device setup but the path clean-up part in
dapm_free_widgets is prepared to remove cross-device paths between a device
being removed and others.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are no users of these and it's not clear what they would do given
the mono flow modelling which DAPM does. If need arises we can add them
again.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In some cases it was not possible to follow the appropiate power
ON/OFF sequence like in cases where the PGA needs to be enabled
before the driver and disabled before the PGA for pop reduction.
Add a widget to support output driver (speaker, haptic, vibra, etc)
drivers where power ON/OFF ordering is important.
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes some legacy structure definitions which are not using
in current ASoC drivers.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added an optional name member to snd_soc_cache_ops to enable more
sensible diagnostic messages during cache init, exit and sync.
Remove redundant newline in source code.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the machine driver can only do bias level configuration before
the CODEC bias level is brought up. This means that the machine cannot do
any configuration which depends on the CODEC bias level being maintained.
Provide a post-CODEC callback which allows the machine driver to do things
like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow the CODEC driver structure to be marked const by making all
the APIs that use it do so.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch allows machine drivers to override the compression type
provided by the codec driver.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure to use codec->reg_def_copy instead of codec_drv->reg_cache_default
wherever necessary. This change is necessary because in the next patch we
move the cache initialization code outside snd_soc_register_codec() and by that
time any data marked as __devinitconst such as the original reg_cache_default
array might have already been freed by the kernel.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The snd_soc_codec_conf struct now holds codec specific configuration
information.
A new configuration option has been added to allow machine drivers to
override the compression type set by the codec driver.
In the absence of providing an snd_soc_codec_conf struct or when providing
one but not setting the compress_type member to anything, the one supplied
by the codec driver will be used instead. In all other cases the one
set in the snd_soc_codec_conf struct takes effect.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that the base value of compress_type starts at 1 so that
we know whether the machine driver has provided a compress_type
for overriding the codec supplied one.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to keep a copy of the compress_type supplied by the codec driver
so that we can override it if necessary with whatever the machine driver
has provided us with. The reason for not modifying the codec->driver
struct directly is that ideally we'd like to keep it const.
Adjust the code in soc-cache and soc-core to make use of the compress_type
member in the snd_soc_codec struct.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We shouldn't be assigning to the driver structure (which really ought
to be const, further patch to follow) though there's unlikely to be any
actual problem except in the unlikely case that two devices with the
same driver but different bus types appear in the same system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Neither drivers nor the core should be fiddling with the actual ops
structure at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added.
Signed-off-by: Florian Faber <faberman@linuxproaudio.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This makes possible to register auxiliary dailess codecs in a machine
driver. Term dailess is used here for amplifiers and codecs without DAI or
DAI being unused.
Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs
are probed after initializing the DAI links. There are no major differences
between DAI link codecs and dailess codecs in ASoC core point of view. DAPM
handles them equally and sysfs and debugfs directories for dailess codecs
are similar except the pmdown_time node is not created.
Only suspend and resume functions are modified to traverse all probed codecs
instead of DAI link codecs.
Example below shows a dailess codec registration.
struct snd_soc_aux_dev foo_aux_dev[] = {
{
.name = "Amp",
.codec_name = "codec.2",
.init = foo_init2,
},
};
static struct snd_soc_card card = {
...
.aux_dev = foo_aux_dev,
.num_aux_devs = ARRAY_SIZE(foo_aux_dev),
};
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current AP4 FSI set_rate function used bogus clock process
which didn't care enable/disable and clk->usecound.
To solve this issue, this patch also modify FSI driver to call
set_rate with enough options.
This patch modify it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
If CONFIG_SND_DYNAMIC_MINORS is used, assign /dev/snd/seq and
/dev/snd/timer the usual static minors, and export specific
module aliases to generate udev module on-demand loading
instructions:
$ cat /lib/modules/2.6.33.4-smp/modules.devname
# Device nodes to trigger on-demand module loading.
microcode cpu/microcode c10:184
fuse fuse c10:229
ppp_generic ppp c108:0
tun net/tun c10:200
uinput uinput c10:223
dm_mod mapper/control c10:236
snd_timer snd/timer c116:33
snd_seq snd/seq c116:1
The last two lines instruct udev to create device nodes, even
when the modules are not loaded at that time.
As soon as userspace accesses any of these nodes, the in-kernel
module-loader will load the module, and the device can be used.
The header file minor calculation needed to be simplified to
make __stringify() (supports only two indirections) in
the MODULE_ALIAS macro work.
This is part of systemd's effort to get rid of unconditional
module load instructions and needless init scripts.
Cc: Lennart Poettering <lennart@poettering.net>
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no need anymore to include soc.h in soc-dapm.h and soc-dai.h as
drivers are converted to include only soc.h.
Thanks to Lars-Peter Clausen <lars@metafoo.de> for pointing out the issue.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows to disable period interrupts which are
not needed when the application relies on a system timer
to wake-up and refill the ring buffer. The behavior of
the driver is left unchanged, and interrupts are only
disabled if the application requests this configuration.
The behavior in case of underruns is slightly different,
instead of being detected during the period interrupts the
underruns are detected when the application calls
snd_pcm_update_avail, which in turns forces a refresh of the
hw pointer and shows the buffer is empty.
More specifically this patch makes a lot of sense when
PulseAudio relies on timer-based scheduling to access audio
devices such as HDAudio or Intel SST. Disabling interrupts
removes two unwanted wake-ups due to period elapsed events
in low-power playback modes. It also simplifies PulseAudio
voice modules used for speech calls.
To quote Lennart "This patch looks very interesting and
desirable. This is something have long been waiting for."
Support for this in hardware drivers is optional.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a need to prefix codec kcontrol, widget and internal route names in
an ASoC machine that has multiple codecs with conflicting names. The name
collision would occur when codec drivers try to registering kcontrols with
the same name or when building audio paths.
This patch introduces optional prefix_map into struct snd_soc_card. With it
machine drivers can specify a unique name prefix to each codec that have
conflicting names with anothers. Prefix to codec is matched with codec
name.
Following example illustrates a machine that has two same codec instances.
Name collision from kcontrol registration is avoided by specifying a name
prefix "foo" for the second codec. As the codec widget names are prefixed
then second audio map for that codec shows a prefixed widget name.
static const struct snd_soc_dapm_route map0[] = {
{"Spk", NULL, "MONO"},
};
static const struct snd_soc_dapm_route map1[] = {
{"Vibra", NULL, "foo MONO"},
};
static struct snd_soc_prefix_map codec_prefix[] = {
{
.dev_name = "codec.2",
.name_prefix = "foo",
},
};
static struct snd_soc_card card = {
...
.prefix_map = codec_prefix,
.num_prefixes = ARRAY_SIZE(codec_prefix),
};
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for rbtree compression when storing the
register cache. It does this by not adding any uninitialized registers
(those whose value is 0). If any of those registers is written
with a nonzero value they get added into the rbtree.
Consider a sample device with a large sparse register map. The
register indices are between [0, 0x31ff]. An array of 12800 registers
is thus created each of which is 2 bytes. This results in a 25kB
region. This array normally lives outside soc-core, normally in the
driver itself. The original soc-core code would kmemdup this region
resulting in 50kB total memory. When using the rbtree compression
technique and __devinitconst on the original array the figures are
as follows. For this typical device, you might have 100 initialized
registers, that is registers that are nonzero by default. We build
an rbtree with 100 nodes, each of which is 24 bytes. This results
in ~2kB of memory. Assuming that the target arch can freeup the
memory used by the initial __devinitconst array, we end up using
about ~2kB bytes of actual memory. The memory footprint will increase
as uninitialized registers get written and thus new nodes created in
the rbtree. In practice, most of those registers are never changed.
If the target arch can't freeup the __devinitconst array, we end up
using a total of ~27kB. The difference between the rbtree and the LZO
caching techniques, is that if using the LZO technique the size of
the cache will increase slower as more uninitialized registers get
changed.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for LZO compression when storing the register
cache. The initial register defaults cache is marked as __devinitconst
and the only change required for a driver to use LZO compression is
to set the compress_type member in codec->driver to SND_SOC_LZO_COMPRESSION.
For a typical device whose register map would normally occupy 25kB or 50kB
by using the LZO compression technique, one can get down to ~5-7kB. There
might be a performance penalty associated with each individual read/write
due to decompressing/compressing the underlying cache, however that should not
be noticeable. These memory benefits depend on whether the target architecture
can get rid of the memory occupied by the original register defaults cache
which is marked as __devinitconst. Nevertheless there will be some memory
gain even if the target architecture can't get rid of the original register
map, this should be around ~30-32kB instead of 50kB.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch introduces the new caching API and migrates the
old caching interface into the new one. The flat register caching
technique does not use compression at all and it is equivalent to
the old caching technique. One can still access codec->reg_cache
directly but this is not advised as that will not be portable
across different caching strategies.
None of the existing drivers need to be changed to adapt to this
caching technique. There should be no noticeable overhead associated
with using the new caching API.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on discussion the dapm_pop_time in debugsfs should be per card rather
than per device. Single pop time value for entire card is cleaner when the
DAPM sequencing is extended to cross-device paths.
debugfs/asoc/{card->name}/{codec dir}/dapm_pop_time
->
debugfs/asoc/{card->name}/dapm_pop_time
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There will be need to have sound card specific debugfs entries. This patch
introduces a new debugfs/asoc/{card->name}/ directory but does not add yet
any entries there.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Decoupling Dynamic Audio Power Management (DAPM) from codec devices is
required when developing ASoC further. Such as for other ASoC components to
have DAPM widgets or when extending DAPM to handle cross-device paths.
This patch decouples DAPM related variables from struct snd_soc_codec and
moves them to new struct snd_soc_dapm_context that is used to encapsulate
DAPM context of a device. ASoC core and API of DAPM functions are modified
to use DAPM context instead of codec.
This patch does not change current functionality and a large part of changes
come because of structure and internal API changes.
Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some
minor core changes, codecs and machine driver conversions from
Jarkko Nikula <jhnikula@gmail.com>.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Manuel Lauss <manuel.lauss@googlemail.com>
Cc: Mike Frysinger <vapier.adi@gmail.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Cc: Kevin Hilman <khilman@deeprootsystems.com>
Cc: Ryan Mallon <ryan@bluewatersys.com>
Cc: Timur Tabi <timur@freescale.com>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Lars-Peter Clausen <lars@metafoo.de>
Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org>
Cc: Wan ZongShun <mcuos.com@gmail.com>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Jassi Brar <jassi.brar@samsung.com>
Cc: Daniel Gloeckner <dg@emlix.com>
Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Facilitating adding trace type stuff. For a first pass add some dev_dbg()
statements into them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (365 commits)
ALSA: hda - Disable sticky PCM stream assignment for AD codecs
ALSA: usb - Creative USB X-Fi volume knob support
ALSA: ca0106: Use card specific dac id for mute controls.
ALSA: ca0106: Allow different sound cards to use different SPI channel mappings.
ALSA: ca0106: Create a nice spot for mapping channels to dacs.
ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence.
ALSA: ca0106: Pull out dac powering routine into separate function.
ALSA: ca0106 - add Sound Blaster 5.1vx info.
ASoC: tlv320dac33: Use usleep_range for delays
ALSA: usb-audio: add Novation Launchpad support
ALSA: hda - Add workarounds for CT-IBG controllers
ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs
ASoC: tpa6130a2: Error handling for broken chip
ASoC: max98088: Staticise m98088_eq_band
ASoC: soc-core: Fix codec->name memory leak
ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066
ALSA: hda - Add some workarounds for Creative IBG
ALSA: hda - Fix wrong SPDIF NID assignment for CA0110
ALSA: hda - Fix codec rename rules for ALC662-compatible codecs
ALSA: hda - Add alc_init_jacks() call to other codecs
...
This patch removes the old CONFIG_SYSFS_DEPRECATED_V2 config option,
but it keeps the logic around to handle block devices in the old manner
as some people like to run new kernel versions on old (pre 2007/2008)
distros.
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Cc: Jens Axboe <axboe@kernel.dk>
Cc: Stephen Hemminger <shemminger@vyatta.com>
Cc: "Eric W. Biederman" <ebiederm@xmission.com>
Cc: Alan Stern <stern@rowland.harvard.edu>
Cc: "James E.J. Bottomley" <James.Bottomley@suse.de>
Cc: Andrew Morton <akpm@linux-foundation.org>
Cc: Alexey Kuznetsov <kuznet@ms2.inr.ac.ru>
Cc: Randy Dunlap <randy.dunlap@oracle.com>
Cc: Tejun Heo <tj@kernel.org>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: Peter Zijlstra <a.p.zijlstra@chello.nl>
Cc: David Howells <dhowells@redhat.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
This patch is adding support for alc562[123] codecs. It's based
on the source code available in HP source code and other places.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit f6765502f8 and adds
the missing include file.
Signed-off-by: Peter Hsiang <Peter.Hsiang@maxim-ic.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CCR is defined in emu10k1, but SuperH is defined too.
If user use this driver with SuperH, it becomes a double definition.
Signed-off-by: Nobuhiro Iwamatsu <nobuhiro.iwamatsu.yj@renesas.com>
Cc: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If we compile the ASoC code with PM disabled, we hit stuff like:
sound/soc/soc-dapm.c: In function 'snd_soc_dapm_suspend_check':
sound/soc/soc-dapm.c:440: warning: unused variable 'codec'
So tweak the stub macro to avoid these issues.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With generic AC97 ASoC glue driver (codec/ac97.c), we get following warning when
the device is registered (slightly stripped the backtrace):
kobject (c5a863e8): tried to init an initialized object, something is seriously
wrong.
[<c00254fc>] (unwind_backtrace+0x0/0xec)
[<c014fad0>] (kobject_init+0x38/0x70)
[<c0171e94>] (device_initialize+0x20/0x70)
[<c017267c>] (device_register+0xc/0x18)
[<bf20db70>] (snd_soc_instantiate_cards+0x924/0xacc [snd_soc_core])
[<bf20e0d0>] (snd_soc_register_platform+0x16c/0x198 [snd_soc_core])
[<c0175304>] (platform_drv_probe+0x18/0x1c)
[<c0174454>] (driver_probe_device+0xb0/0x16c)
[<c017456c>] (__driver_attach+0x5c/0x7c)
[<c0173cec>] (bus_for_each_dev+0x48/0x78)
[<c0173600>] (bus_add_driver+0x98/0x214)
[<c0174834>] (driver_register+0xa4/0x130)
[<c001f410>] (do_one_initcall+0xd0/0x1a4)
[<c0062ddc>] (sys_init_module+0x12b0/0x1454)
This happens because the generic AC97 glue driver creates its codec->ac97 via
calling snd_ac97_mixer(). snd_ac97_mixer() provides own version of
snd_device.register which handles the device registration when
snd_card_register() is called.
To avoid registering the AC97 device twice, we add a new flag to the
snd_soc_codec: ac97_created which tells whether the AC97 device was created by
SoC subsystem.
Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than block the workqueue by sleeping to do the debounce use delayed
work to implement the debounce time. This should also means that we extend
the debounce time on each new bounce, potentially allowing shorter debounce
times for clean insertions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8962 features five GPIOs, add support for controlling their output
state via gpiolib.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add the widget for MICBIAS power control and allow configuration of the
microphone bias setup via the platform data for the WM8962. When
microphone status signals are brought out to GPIO this should be
sufficient to enable microphone detection.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide an initial hookup for interrupts on the WM8962. Currently we simply
report error status via log messages if an IRQ is provided for the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some devices have more flexible microphone detection and can detect
a wider range of buttons.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Swapping the bias level enumeration is only meant to help debugging. It is
easier if number 0 means bias off and bigger number means bigger bias level.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current code in pcm_lib.c do all checks using only the position
in the ring buffer. Unfortunately, where the interrupts gets delayed or
merged into one, we need another timing source to check when the
buffer size boundary overlaps to avoid the wrong updating of the
ring buffer pointers.
This code uses jiffies to check the right time window without any
performance impact.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With some hardware combinations, the PCM interrupts are acknowledged
before the period boundary from the emu10k1 chip. The midlevel PCM code
gets confused and the playback stream is interrupted.
It seems that the interrupt processing shift by 2 samples is enough
to fix this issue. This default value does not harm other,
non-affected hardware.
More information: Kernel bugzilla bug#16300
[A copmile warning fixed by tiwai]
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fairly simple conflicts, the most serious ones are the i.MX ones which I
suspect now need another rename.
Conflicts:
arch/arm/mach-mx2/clock_imx27.c
arch/arm/mach-mx2/devices.c
arch/arm/mach-omap2/board-rx51-peripherals.c
arch/arm/mach-omap2/board-zoom2.c
sound/soc/fsl/mpc5200_dma.c
sound/soc/fsl/mpc5200_dma.h
sound/soc/fsl/mpc8610_hpcd.c
sound/soc/pxa/spitz.c
unifdef-y and header-y has same semantic.
So there is no need to have both.
Drop the unifdef-y variant and sort all lines again
Signed-off-by: Sam Ravnborg <sam@ravnborg.org>
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits)
ALSA: hda - Add pin-fix for HP dc5750
ALSA: als4000: Fix potentially invalid DMA mode setup
ALSA: als4000: enable burst mode
ALSA: hda - Fix initial capsrc selection in patch_alc269()
ASoC: TWL4030: Capture route runtime DAPM ordering fix
ALSA: hda - Add PC-beep whitelist for an Intel board
ALSA: hda - More relax for pending period handling
ALSA: hda - Define AC_FMT_* constants
ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
ALSA: hda - Add support for HDMI HBR passthrough
ALSA: hda - Set Stream Type in Stream Format according to AES0
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
ASoC: wm9081: fix resource reclaim in wm9081_register error path
ASoC: wm8978: fix a memory leak if a wm8978_register fail
ASoC: wm8974: fix a memory leak if another WM8974 is registered
ASoC: wm8961: fix resource reclaim in wm8961_register error path
ASoC: wm8955: fix resource reclaim in wm8955_register error path
ASoC: wm8940: fix a memory leak if wm8940_register return error
ASoC: wm8904: fix resource reclaim in wm8904_register error path
...
The WM8962 is a low power, high performance stereo CODEC designed for
portable digital audio applications.
This initial driver release supports the key audio paths of the WM8962.
Extended functionality, such as microphone detection, digital microphones
and the advanced DSP signal enhancements provided by the device are not
yet supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
All current users of pm_qos_add_request() have the ability to supply
the memory required by the pm_qos routines, so make them do this and
eliminate the kmalloc() with pm_qos_add_request(). This has the
double benefit of making the call never fail and allowing it to be
called from atomic context.
Signed-off-by: James Bottomley <James.Bottomley@suse.de>
Signed-off-by: mark gross <markgross@thegnar.org>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
Specified ID is necessary, when some codecs are used with FSI.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.
This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no necessity that each bit in this area has the meaning.
This patch modify it to sequence number
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When using poll() to wait for the next period -- or avail_min samples --
one gets a consistent delay for each system call that is usually just a
little short of the selected period time. However, When using
snd_pcm_read/write(), one gets a jittery delay that alternates between
less than a millisecond and approximately two period times. This is
caused by snd_pcm_lib_{read,write}1() transferring any available samples
to the user's buffer and adjusting the application pointer prior to
sleeping to the end of the current period. When the next period
interrupt occurs, there is then less than avail_min samples remaining to
be transferred in the period, so we end up sleeping until a second
period occurs.
This is solved by using runtime->twake as the number of samples needed
for a wakeup in addition to selecting the proper wait queue to wake in
snd_pcm_update_state(). This requires twake to be non-zero when used
by snd_pcm_lib_{read,write}1() even if avail_min is zero.
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems codecs need to configure some registers before and after
powering down some of their part. As a convenience add a macro for that.
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This defines the 24bps and 40bps (8khz sample rate) G.723 codec
formats. They are going to be used once I submit the driver for
an mpeg4/g723 compression card.
I've updated the signed value to -1 as per Takashi's comments
since these are non-linear formats.
Signed-off-by: Ben Collins <bcollins@bluecherry.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits)
ALSA: hda: Storage class should be before const qualifier
ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
ASoC: TWL6040: Enable earphone path in codec
ASoC: SDP4430: Add support for Earphone speaker
ASoC: SDP4430: Add sdp4430 machine driver
ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
ALSA: sound/pci/asihpi: Use kzalloc
ALSA: hdmi - dont fail on extra nodes
ALSA: intelhdmi - add id for the CougarPoint chipset
ALSA: intelhdmi - user friendly codec name
ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
ALSA: asihpi: incorrect range check
ALSA: asihpi: testing the wrong variable
ALSA: es1688: add pedantic range checks
ARM: McBSP: Add support for omap4 in McBSP driver
ARM: McBSP: Fix request for irq in OMAP4
OMAP: McBSP: Add 32-bit mode support
...
This patch is adding a new control which has the following capabilities:
- tlv
- variable data size (for instance, 7 ou 8 bit)
- double mixer
- data range centered around 0
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the register for the volume needs invert, than the inversion
need to be done from the chip maximum, and not from the platform
dependent limit.
Introduce soc_mixer_control.platform_max value, which initially
equals to chip maximum.
The snd_soc_limit_volume function only modify the platform_max,
all volsw_info call returns this as well.
The .max value holds the chip default (maximum), and it is used
for the inversion, if it is needed.
Additional check in the volsw_info call has been added to check
the validity of the platform_max in case, when custom macros
used by codec drivers are not initializing it correctly.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch changes the string based list management to a handle base
implementation to help with the hot path use of pm-qos, it also renames
much of the API to use "request" as opposed to "requirement" that was
used in the initial implementation. I did this because request more
accurately represents what it actually does.
Also, I added a string based ABI for users wanting to use a string
interface. So if the user writes 0xDDDDDDDD formatted hex it will be
accepted by the interface. (someone asked me for it and I don't think
it hurts anything.)
This patch updates some documentation input I got from Randy.
Signed-off-by: markgross <mgross@linux.intel.com>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
As well as allowing DAPM pins to be marked as ignoring suspend allow DAI
links to be similarly marked. This is primarily intended for digital
links between CODECs and non-CPU devices such as basebands in mobile
phones and will suppress all suspend calls for the DAI link. It is
likely that this will need to be revisited if used with devices which
are part of the SoC CPU.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some devices can usefully run audio while the Linux system is suspended.
One of the most common examples is smartphone systems, which are normally
designed to allow audio to be run between the baseband and the CODEC
without passing through the CPU and so can suspend the CPU when on a
voice call for additional power savings.
Support such systems by providing an API snd_soc_dapm_ignore_suspend().
This can be used to mark DAPM endpoints as not being sensitive to
system suspend. When the system is being suspended paths between
endpoints which are marked as ignoring suspend will be kept active.
Both source and sink must be marked, and there must already be an
active path between the two endpoints prior to suspend.
When paths are active over suspend the bias management will hold the
device bias in the ON state. This is used to avoid suspending the
CODEC while it is still in use.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We now manage suspend within the main power analysis rather than by
flipping the state of widgets.
Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ESS ES968 chip is nothing more then a PnP companion
for a non-PnP audio chip. It was paired with non-PnP ESS' chips:
ES688 and ES1688. The ESS' audio chips are handled by the es1688
driver in native mode. The PnP cards are handled by the ES968
driver in SB compatible mode.
Move the ES968 chip handling to the es1688 driver so the driver
can handle both PnP and non-PnP cards. The es968 is removed.
Also, a new PnP id is added for the card I acquired (the change
was tested on this card).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocate the snd_es1688 during the snd_card allocation.
This allows to remove the card pointer from the snd_es1688 structure.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 6f3991152f.
Since core has now support for limiting the volume on controls this
patch is not needed. Furthermore, this patch actually prevents the core
to set new volume on the TPA.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the core to limit the maximum volume on an
existing control.
The function will modify the soc_mixer_control.max value
of the given control.
The new value must be lower than the original one (chip maximum)
If there is a need for limiting a gain on a given control,
than machine drivers can do the following in their
snd_soc_dai_link.init function:
snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21);
This will modify the original 31 (chip maximum) to 21, so user
space will not be able to set the gain higher than this.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for platform dependent gain limiting on the
tpa6130a2 (and tpa6140a2) Headset amplifier.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Handle the reset GPIO within the codec driver in order to follow
the startup protocol for the tlv320aic3x codecs.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM9090 is a high performance low power audio subsystem, including
headphone and class D speaker drivers.
Note that this driver is a standalone CODEC driver and so is only
immediately suitable for use with the WM9090 as a standalone sound card
taking line inputs, or with a DAC with no software control. The pending
ASoC multi-CODEC support will expand the range of systems that can use
the driver, or system-specific adaptations can be made.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds support for Philips UDA1345 CODEC. The CODEC has only
volume control, de-emphasis, mute, DC filtering and power control features.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the features of the multi CODEC work is that it embeds a struct
device in the CODEC to provide diagnostics via a sysfs class rather than
via the device tree, at which point it's much better to use the struct
device private data rather than having two places to store it. Provide
an accessor function to allow this change to be made more easily, and
update all the CODEC drivers are updated.
To ensure use of the accessor the private data structure member is
renamed, meaning that if code developed with older an older core that
still uses private_data is merged it will fail to build.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
[Note that this is a backported version for 2.6.34.
Upstream commit is fd23b7dee]
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We actually pass an array of 7 chars not 5.
This silences a smatch warning.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some systems provide both mechanical and electrical detection of jack
status changes. On such systems power savings can be achieved by only
enabling the electrical detection methods when physical insertion has
been detected.
Begin supporting such systems by providing a notifier for jack status
changes which can be used to trigger any reconfiguration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some devices provide support for detection of a small number of
buttons on their jacks. One common implementation provides a single
button, implemented by shorting the microphone to ground and detected
along with microphone presence detection by detecting varying current
draws on the microphone bias signal.
Provide support for up to three buttons via the jack interface. These
default to reporting BTN_n but an API is provided to allow these to
be remapped to other keys by the machine driver where it knows what
the keys are. More keys can be added with ease if required.
This is only intended to support simple accessory button designs. If
the interface is limiting then either creating a child device for the
accessory or accessing the input device in the jack directly is
recommended.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8904 allows microphone detection signals to be brought out as
alternate functions of the GPIO signals which can be detected using
interrupt inputs on the CPU. Allow this to be configured using
platform data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide platform data allowing the configuration of the GPIO pins
on the WM8904 to be selected, allowing alternate functions to be
enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Support use of the WM8903 IRQ for reporting of microphone presence
and short detection.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently used to detect completion of the write sequencer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide support for WM8903 microphone presence and short detection
using the GPIOs to route out a logic signal suitable for handling
using snd_soc_jack_add_gpios() on the processor GPIOs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow users to pass in a default configuration for the GPIOs of
the WM8903 as platform data. This allows configuration of the pin
muxing of the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow pins to be forced on regardless of their power state. This is
intended for use with microphone bias supplies which need to be
enabled in order to support microphone detection - in systems without
appropriate hardware leaving the microphone unbiased when not in use
saves power.
The force done at power check time in order to avoid disrupting other
power detection logic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Platform data option for the codec to keep the BCLK clock
continuously running in FIFO modes (codec master).
OMAP3 McBSP when in slave mode needs continuous BCLK running
on the serial bus in order to operate correctly.
Since in FIFO mode the DAC33 can also shut down the BCLK clock
and enable it only when it is needed, let the platforms decide
if the CPU side needs the BCLK running or not.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The flag is no longer used in the code so it just wastes a bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8960 headphone outputs can be run in capless mode with OUT3
used to drive a pseudo ground for the headphone drivers. In this
mode the mono mixer is not used, the mixer should be turned on
in concert with the headphone output drivers and the device bias
levels are managed differently.
Also tweak the existing bias management to remove the use of active
discharge while we're at it since that's often audible.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Avoids machine files having to peer into sound/soc which is a bit
rude and icky.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The delay callback can be used by the core to query the delay
on the dai caused by FIFO or delay in the platform side.
In case if both CPU and CODEC dai has FIFO the delay reported
by each will be added to form the full delay on the chain.
If none of the dai has FIFO, than the delay will be kept as
zero.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two
or more dai_links we need to log the number of active users of the dai.
For that, we change semantics of the snd_soc_dai.active flag from indicator
to reference counter.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order for having snd_soc_dais shared among two or more dai_links,
remove the relatively global runtime field from the struct snd_soc_dai
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Passing pointer to relevant dai_link provides easier reach to the
ASoC tree in suspend/resume of snd_soc_platform. It also provides
direct access to the dai at the other end of the dai_link.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make the pmdown_time a per-card setting rather than a global one,
initialised before the card initialisation runs. This allows cards
to override the default setting if it makes sense to do so (for
example, due to an unavoidable pop).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM2000 is a low power, high quality handset receiver speaker
driver with Wolfson myZone™ Ambient Noise Cancellation (ANC). It
provides enhanced voice communication quality in a noisy environment
if the handset acoustics are designed appropriately.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add a bit to the CODEC structure indicating if a cache sync is required.
By default this will be set if a cache only write is done to a soc-cache
register cache. This allows us to avoid syncing the cache back after
using cache only writes if there were no changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently the soc-cache code will always write to the device, meaning
that we need the device to be powered and active at pretty much all
times the system is active. Allowing cache only writes lays some
groundwork for future enhancements to allow devices to be put into a
full off state when the audio subsystem is idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr
update functions" commit:
"It is possible for the status/delay ioctls to be called when the sound
card's pointer register alreay shows a position at the beginning of the
new period, but immediately before the interrupt is actually executed.
(This happens regularly on a SMP machine with mplayer.) When that
happens, the code thinks that the position must be at least one period
ahead of the current position and drops an entire buffer of data."
Return back the hw_ptr_interrupt variable. The last interrupt pointer
is always computed from the latest hw_ptr instead of tracking it
separately (in this case all hw_ptr checks and modifications might
influence also hw_ptr_interrupt and it is difficult to keep it
consistent).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Several shortcuts for popular uses of some of SOC_ENUM_* and
SOC_VALUE_ENUM_* macros.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many macros from include/sound/soc-dapm.h take an array and a number of
elements in it as arguments, whereas most users use static arrays and use
"x, ARRAY_SIZE(x)" as arguments. This patch adds simplified versions of
those macros, calling ARRAY_SIZE() internally.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.oc.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC always maintains the bias of the CODEC while the system
is active. With older mobile CODECs this is required since the outputs
are referenced to a non-zero voltage and enabling or disabling this
voltage without audible pops or clicks in the output takes too long to
do when starting or stopping audio.
As a result of features such as ground referenced outputs and class D
speaker drivers current generation devices are able to power on and off
much more quickly without these system level issues so provide a new
flag idle_bias_off in snd_soc_codec which will cause the core to turn
off the CODEC bias. The distinction between STANDBY and OFF is still
maintained. This is partly for consistency but also allows for
potential future extensions such as per-machine overrides or deferring
the bias removal.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O"
commit. New sleeping queue is introduced to separate user space and kernel
space wake_ups. runtime->nowake is renamed to twake (transfer wake).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add possibility to configure the burst mode BCLK divider through platform
data structure.
The BCLK divider changes the actual speed of the serial bus in burst mode,
which is faster than the sampling frequency of the running stream.
In this way platforms can experiment with the optimal burst speed without
the need to modify the codec driver itself.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd
need non-cached behavior more or less, even for the intermediate ring-
buffers.
Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback
that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new function to look up a quirk entry with the given PCI SSID
instead of a pci device pointer. This can be used when the searched ID
is overridden for debugging or such a purpose.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
tpa6140a2 uses different names for the regulators.
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As noted by pl bossart <bossart.nospam@gmail.com>, the PCM I/O routines
(snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls
until all samples are not processed.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them.
The main change is hw_ptr_interrupt variable removal to simplify code
logic. This variable can be computed directly from hw_ptr.
Ensure that updated hw_ptr is not lower than previous one (it was possible
with old code in some obscure situations when interrupt was delayed or
the lowlevel driver returns wrong ring buffer position value).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
In some debug cases, it might be usefull to see previous ring buffer
positions to determine position problems from the lowlevel drivers.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This is one of Sound Blaster Pro compatible chipsets which is supported
by Linux OSS driver and was missing native supoort for ALSA.
The Jazz16 audio codec is Crystal CS4216 which is capable
of playback and recording up to 48 kHz stereo.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8955 is a low power, high quality stereo DAC with integrated
headphone and loudspeaker amplifiers, designed to reduce external
component requirements in portable digital audio applications. This is
an initial driver implementing support for the majority of the
functionality in the device, currently OUT3 is not supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
There are now five copies of the code to allocate a PCM buffer using
vmalloc(). Add a sixth in the core so that the others can be removed.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8904 is a high performance ultra-low power stereo CODEC
optimised for portable audio applications, with features including
a class W amplifier, FLL with free running mode, Mobile ReTune and
ground referenced headphone and line outputs.
Support for some features, most particularly the digital microphone
interface, is not yet present.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The ACI mixer is used to control the radio FM module
installed on the Miro PCM20 sound card. Expose ACI mixer
outside the sound card driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the miro.h header to the include/sound directory. It can
be used in the Miro PCM20 radio driver (v4l).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update control names to be more closer to their meaning.
Change the "Mono" name to the "Beep" as this line is usually
used to forward the PC beeper signal to sound card's output.
Update names for both cs423x and wss.
Clean up cs4235 controls according to the cs4235 doc. Rename
some of the cs4235 controls to be consistent with the cs4236's
ones.
Also, delete one misnamed cs4231 register define.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack_status_check callback function is the interface to check the
status of the jack. Some target provides the method to distinguish what
is the jack inserted - headphone jack, microphone jack, tvout jack, etc,
so we can implement it using the jack_status_check function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices need to calculate the bit clock rate desired to
work out the clock configuration required for the device.
Provide utility functions to do this using both hw_params
structures and raw numbers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Record the pid of the task that opened a RawMIDI substream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Record the pid of the task that opened a PCM substream. For sound
cards with hardware mixing, this allows determining which process
is associated with a specific substream's volume control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of storing the PID number, take a reference to the task's pid
structure. This protects against duplicates due to PID overflows, and
using pid_vnr() ensures that the PID returned by snd_ctl_elem_info() is
correct as seen from the current namespace.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We do not need to save the ID of the process that locked a control
because that information is already available in the owner's file data.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The cs4236 was two step detection with call to the snd_wss_free()
between two steps. The snd_wss_free() did not free a sound device
created in the snd_wss_create(). This caused an OOPS during module
removal as the same sound device was released twice. The same OOPS
happened if the cs4236 module loading failed.
Fix this by adapting the snd_cs4236_create() to correctly work with
chips less capable then cs4236. The snd_cs4236_create() behaves the
same as the snd_wss_create() if the chip is less capable than the cs4236.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a port of the sound/oss/sh_dac_audio.c driver.
The driver uses an on-chip 8-bit D/A converter, which has a speaker connected
to one of its channels, found in several ancient HP machines.
For interrupts it uses a high-resolution timer (hrtimer).
Tested on SH7709 based hp6xx (HP Jornada 680/690 and HP Palmtop 620lx/660lx).
Also, since OSS Emulation works, the old OSS sound/oss/sh_dac_audio.c driver
would be obsolete soon, and it could be removed.
Signed-off-by: Rafael Ignacio Zurita <rizurita@yahoo.com>
Acked-by: Paul Mundt <lethal@linux-sh.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_init_card() is always called as the last part of the CODEC probe
function so we can factor it out into the core card setup rather than
have each CODEC replicate the code to do the initialiastation. This will
be required to support multiple CODECs per card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.
TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.
The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.
Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).
b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.
The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.
The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"
From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':
{"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
{"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},
Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:
debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
/dapm_pop_time
/dapm/{widgets}
With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The conversion solves the problem that firmware size was set to 64KB
while non PnP cards have 128KB firmware files.
An additional firmware initialization code has been moved from the OSS
driver.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add DAI format definition for PDM interfaces.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* complete support for ak4113
* based on code for ak4114 and ak4117
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* complete support for ak4620
* codec regs listed in proc for all codecs/chips
* adding total regs for each codec
* fixing nb. of steps in input attenuation controls
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* the previous version had a typo - values of AK4114_OPS10-12 were
identical with AK4114_OPS00-02
* Since no cards actually use this feature, the bug was not identified earlier
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds an interface to set the relationship between audio
channel number and slot number. The interface should be really useful
because audio channel n doesn't always use slot n in all platforms. And
for some devices, the relationship even can change with sound mode
switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* topic/tlv-minmax:
ALSA: usb-audio - Correct bogus volume dB information
ALSA: usb-audio - Use the new TLV_DB_MINMAX type
ALSA: Add new TLV types for dBwith min/max
Some chips with complex internal supply (particularly clocking)
arragements may have multiple options for some of the supply
connections. Since these don't affect user-visible audio routing
the expectation would be that they would be managed automatically
by one of the drivers.
Support these users by allowing routes to have a connected function
which is queried before the connectedness of the path is checked as
normal. Currently this is only done for supplies, other widgets
could be supported but are not currently since the expectation for
them is that audio routing will be under the control of userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add appropriate const prefix to char * arguments in proc helper functions.
Also fixed the caller side to be proper const pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Re-export snd_pcm_format_name() function to be used outside the PCM core.
As a first example, usbaudio is changed to use it now again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The struct snd_monitor_file is used locally only in sound/core/init.c,
thus it should be moved there from the public sound/core.h.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More and more devices feature PLLs and FLLs with the ability to select
between multiple input clocks. In order to better support these devices
a new argument, source, has been added to the set_pll() configuration
API. Using set_clkdiv() is often difficult due to the need to stop the
PLL/FLL before any reconfiguration can be done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debug module option to snd core.
This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value. debug=0 means no debug messsages.
As default, it's set to the verbose level 2.
Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently when built with DEBUG DAPM will dump information about
the power state decisions it is taking for each widget to dmesg.
This isn't an ideal way of getting the information - it requires
a kernel build to turn it on and off and for large hub CODECs the
volume of information is so large as to be illegible. When the
output goes to the console it can also cause a noticable impact
on performance simply to print it out.
Improve the situation by adding a dapm directory to our debugfs
tree containing a file per widget with the same information in
it. This still requires a decision to build with debugfs support
but is easier to navigate and much less intrusive.
In addition to the previously displayed information active streams
are also shown in these files.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver is very simple.
It support playback only now.
This patch is tested by ms7724se board.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently DAPM interfaces with the audio streams to and from the
processor at the DAC and ADC widgets. As the digital capabilities
of parts increases this is becoming a less and less able to meet
the needs of parts.
To meet the needs of these devices create new widgets interfacing
with the TDM bus but not integrated into any other functionality.
Audio can then be routed to and from these widgets using existing
routing widgets.
A slot number is provided in the definition but this is currently
not used yet. This is intended to support devices which can use
more than one TDM slot on a single interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for passing platform data to ac97 bus devices
from PXA2xx-AC97 driver..
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the interval timer to be programmed with its full 96 kHz
precision.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Extend set_tdm_slot to allow the user to arbitrarily set the frame width
and active TX/RX slots.
Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c
still doesn't handle the slot_width override.
While being there, correct an incorrect use of SlotsPerFrm(7) use in
bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ).
(this series is meant for Mark's for-2.6.32 branch)
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of this refactoring the type of the CODEC hw_read operation
is changed to match the regular read operation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.
Initially just use this to factor out hw_write_t for I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This helps CODECs with sparse register maps work better with the
register cache display interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows passing platform_data to devices attached to AC97 bus
(like touchscreens, battery measurement chips ...).
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for
double controls.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a macro for double controls with special callback function and
TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for double
controls.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 is a highly integrated ultra-low power hi-fi CODEC designed
for portable devices such as multimedia phones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using SG-buffers with dma_alloc_coherent() is often very inefficient
on non-coherent architectures because a tracking record could be
allocated in addition for each dma_alloc_coherent() call.
Instead, simply disable SG-buffers but just allocate normal continuous
buffers on non-supported (currently all but x86) architectures.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A lot of CODECs share the same register data formats and therefore
replicate the code to manage access to and caching of the register
map. In order to reduce code duplication centralised versions of
this code will be introduced with drivers able to configure the use
of the common code by calling the new snd_soc_codec_set_cache_io()
API call during startup.
As an initial user the 7 bit address/9 bit data format used by many
Wolfson devices is supported for write only CODECs and the drivers
with straightforward register cache implementations are converted to
use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a volatile_register() operation to the CODEC structure providing a
standard operation to query if a register is volatile. This will be used
to factor out the register cache I/O operations for the CODECs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that the audio subsystem is powered down cleanly when the system
shuts down by providing a shutdown operation. This ensures that all the
components have been returned to an off state cleanly which should avoid
audio issues from partially charged capacitors or noise on digital inputs
if the system is restarted quickly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Ben Dooks <ben-linux@fluff.org>
Add new types for TLV dB scale specified with min/max values instead
of min/step since the resolution can't match always with the one
a device provides. For example, usb audio devices give 1/256 dB
resolution while ALSA TLV is based on 1/100 dB resolution.
The new min/max types have less problems because the possible
rounding error happens only at min/max.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch mostly follows commit 5998102b90
"ASoC: Refactor WM8731 device registration" to make UDA1380 use standard
device instantiation. Similarly, the I2C device registration temporarily
moves into the magician machine driver before it will find its final
resting place in the board file.
At the same time, platform specific configuration is moved to platform data
and common power/reset GPIO handling moves into the codec driver.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that ASoC subdevices can be regular devices they can have normal
suspend and resume calls from their buses. However, suspending them
individually is not desirable since this can lead to problems such as
pops and clicks from devices being suspended with their signals being
amplified or clocks being stopped suddenly.
This will be resolved by having the normal device model suspend and
resume calls call into ASoC which will suspend the entire card while any
of its components are suspended. At present this is not yet implemented
but in order to aid the transition of drivers to the standard device
model this patch adds API calls for the notifications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* topic/pcm-jiffies-check:
ALSA: pcm - A helper function to compose PCM stream name for debug prints
ALSA: pcm - Fix update of runtime->hw_ptr_interrupt
ALSA: pcm - Fix a typo in hw_ptr update check
ALSA: PCM midlevel: lower jiffies check margin using runtime->delay value
ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed
ALSA: PCM midlevel: introduce mask for xrun_debug() macro
ALSA: PCM midlevel: improve fifo_size handling
* topic/asoc: (135 commits)
ASoC: Apostrophe patrol
ASoC: codec tlv320aic23 fix bogus divide by 0 message
ASoC: fix NULL pointer dereference in soc_suspend()
ASoC: Fix build error in twl4030.c
ASoC: SSM2602: assign last substream to the master when shutting down
ASoC: Blackfin: document how anomaly 05000250 is handled
ASoC: Blackfin: set the transfer size according the ac97_frame size
ASoC: SSM2602: remove unsupported sample rates
ASoC: TWL4030: Check the interface format for 4 channel mode
ASoC: TWL4030: Use reg_cache in twl4030_init_chip
ASoC: Initialise dev for the dummy S/PDIF DAI
ASoC: Add dummy S/PDIF codec support
ASoC: correct print specifiers for unsigneds
ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
ASoC: Switch FSL SSI DAI over to symmetric_rates
ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
ASoC: Fabric bindings for STAC9766 on the Efika
ASoC: Support for AC97 on Phytec pmc030 base board.
ASoC: AC97 driver for mpc5200
ASoC: Main rewite of the mpc5200 audio DMA code
...
They are now only accessed within dapm_power_widgets() so can be local
to that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the remaining unsigned shorts with unsigned ints.
Tested with pcap2 codec (25 bits registers).
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce snd_card_set_id() function to allow lowlevel drivers to set
default identification name for card slot. The function checks also
for identification name collisions and tries to create unique name.
Also, the snd_card_create() function is simplified, because this new
function is used. As bonus, proper name collision checks are evaluated
at the card create time.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the fifo_size assignment to hw->ioctl callback to allow lowlevel
drivers overwrite the default behaviour.
fifo_size is in frames not bytes as specified in asound.h and alsa-lib's
documentation, but most hardware have fixed byte based FIFOs. Introduce
internal SNDRV_PCM_INFO_FIFO_IN_FRAMES.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM9081 is designed to provide high power output at low distortion
levels in space-constrained portable applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Should be no impact on the generated code but it helps the compiler
print clearer messages.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than managing the bias level of the system based on if there is
an active audio stream manage it based on there being an active DAPM
widget. This simplifies the code a little, moving the power handling
into one place, and improves audio performance for bypass paths when no
playbacks or captures are active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAPM has always applied any changes to the power state of widgets as soon
as it has determined that they are required. Instead of doing this store
all the changes that are required on lists of widgets to power up and
down, then iterate over those lists and apply the changes. This changes
the sequence in which changes are implemented, doing all power downs
before power ups and always using the up/down sequences (previously they
were only used when changes were due to DAC/ADC power events). The error
handling is also changed so that we continue attempting to power widgets
if some changes fail.
The main benefit of this is to allow future changes to do optimisations
over the whole power sequence and to reduce the number of walks of the
widget graph required to check the power status of widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added runtime->delay field to adjust the delayed samples for snd_pcm_delay().
Typically a hardware FIFO length is stored in this field, so that the
extra delay between hwptr and applptr can be computed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DAI structure has two pointers to the codec, one in the body of the
DAI and one in a union for a parent pointer. Drop the parent pointer
version.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The defines for TDM and synchronous clocks are not used - they are
mostly a legacy of the automatic clocking configuration. TDM will
require configuration of the number of timeslots and which ones to use
so can't be fit into the DAI format and synchronous mode is handled by
symmetric_rates (and needs to be done by constraints rather than when
the DAI format is being configured).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The AC97 wire format is completely fixed so CODECs don't have any choice
about the formats they accept but controllers accept a variety of data
formats and render them down onto the bus. Have a shared define so all
the CODEC drivers will interoperate with any of our controller drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a macro for double controls with special callback functions.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many modern CODECs have shared resources on chip which must be enabled
for portions of the chip to work but which can be disabled at other times
in order to achieve power savings. Examples of such resources include
power supplies and some internal clocks.
Since these widgets are dependencies for the audio path but do not carry
audio signals they require slightly different handling to most widgets -
they do not contribute to the audio path and so should not be counted as
either inputs or outputs during path walks.
Cases where one supply provides a supply for another will require
additional work. There is also room for more optimisation of the graph
walking to avoid repeated checks for the same thing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having switch statements at point of use make the DAPM
power check a member of the widget structure and set it when we
instantiate the widget.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently there are two possible platform datas for the PXA AC97 driver:
one supported by the generic AC97 driver only which provides callbacks
to allow board-specific configuration at stream startup and teardown,
and another for pxa2xx-ac97-lib which allows configuration of the reset
GPIO for PXA2xx CPUs.
Obviously this won't actually work when using the generic AC97 driver
since the drivers will attempt to parse the platform data in both
formats. Fix this by merging the two structures.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Cc: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Added private_data and private_free fields to struct snd_jack so that
the caller can assign the data. It'll be helpful for avoiding the
double-free of the jack instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some drivers like Intel8x0 or Intel HDA are broken for some hardware variants.
This patch adds more strict buffer position checks based on jiffies when
internal hw_ptr is updated. Enable xrun_debug to see mangling of wrong
positions.
As a side effect, the hw_ptr interrupt update routine might do slightly better
job when many interrupts are lost.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.
A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Impact: cleanup
The earlier patch 'make most exported headers use strict integer
types' accidentally includes <linux/types.h> both from the common and
from the kernel-only parts.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: H. Peter Anvin <hpa@linux.intel.com>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
This takes care of all files that have only a small number
of non-strict integer type uses.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Cc: Mauro Carvalho Chehab <mchehab@redhat.com>
Cc: David Airlie <airlied@linux.ie>
Cc: Arnaldo Carvalho de Melo <acme@ghostprotocols.net>
Cc: YOSHIFUJI Hideaki <yoshfuji@linux-ipv6.org>
Cc: netdev@vger.kernel.org
Cc: linux-ppp@vger.kernel.org
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: David Woodhouse <dwmw2@infradead.org>
Signed-off-by: H. Peter Anvin <hpa@linux.intel.com>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
A number of standard posix types are used in exported headers, which
is not allowed if __STRICT_KERNEL_NAMES is defined. In order to
get rid of the non-__STRICT_KERNEL_NAMES part and to make sane headers
the default, we have to change them all to safe types.
There are also still some leftovers in reiserfs_fs.h, elfcore.h
and coda.h, but these files have not compiled in user space for
a long time.
This leaves out the various integer types ({u_,u,}int{8,16,32,64}_t),
which we take care of separately.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Cc: David Airlie <airlied@linux.ie>
Cc: Arnaldo Carvalho de Melo <acme@ghostprotocols.net>
Cc: YOSHIFUJI Hideaki <yoshfuji@linux-ipv6.org>
Cc: netdev@vger.kernel.org
Cc: linux-ppp@vger.kernel.org
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: David Woodhouse <dwmw2@infradead.org>
Signed-off-by: H. Peter Anvin <hpa@linux.intel.com>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got
merged into mainline. Since there's no way to use it on any of
supported machines (iPaq h3100 or h3600), better drop it for now.
It can be reimplemented later using ASoC infrastructure (there's
already a driver for uda1341 codec in mainline, so only CPU and machine
parts need to be written).
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Cc: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the PXA27x series allow 2 gpios to reset the ac97 bus,
allow through platform data configuration the definition of
the correct gpio which will reset the AC97 bus.
This comes from a silicon defect on the PXA27x series, where
the gpio must be manually controlled in warm reset cases.
Signed-off-by: Robert Jarzmik <rjarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line. Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use define instead of enum for ioctl definitions since strace can't
parse ioctls defined via enum properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the standard linked list for snd_monitor_file management.
Also, move the list deletion of shutdown_list element into
snd_disconnect_release() (for simplification).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_ctl_add_slave_uncached() function to add a slave element
with volatile controls. The values of normal slave elements are
supposed to be cachable, i.e. they are changed only via the put
callbacks. OTOH, when a slave element is volatile and its values may
be changed by other reason (e.g. hardware status change), the values
will get inconsistent.
The new function allows the slave elements with volatile changes.
When the slave is tied with this call, the native get callback is
issued at each time so that the values are always updated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The timer callbacks are called in the protected status by the lock
of the timer instance, so there is no need for an extra lock in the
PCM substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.
The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.
Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.
All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On some systems it is desirable for control for DAPM pins to be provided
to user space. This is the case with things like GSM modems which are
controlled primarily from user space, for example. Provide a helper which
exposes the state of a DAPM pin to user space for use in cases like this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
cs4232 and cs4236 driver merge to solve PnP BIOS detection.
Also, the patch adds recognition if the chip is cs4236b+
or earlier part. This unifies drivers for both cs4232
and cs4236+ chips. It allows to use the PnP BIOS
detection for the cs4236+ chips. Previously, only
the snd-cs4232 could be detected by the PnP BIOS.
The cs4232+ cards reports two separate PnP BIOS ids.
The patch adds search for the second id to find out
resources assigned to a control port.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds ALSA support for the AC97 controller found on Atmel
AVR32 devices.
Tested on ATSTK1006 + ATSTK1000 with a development board with a AC97
codec.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds ALSA support for the Audio Bistream DAC found on Atmel
AVR32 devices. The ABDAC is an Atmel IP which might show up on AT91
devices in the future, hence making a generic driver which can be
utilized by AT91 arch if needed.
Datasheet describing the ABDAC peripheral is available in the AT32AP7000
datasheet, http://www.atmel.com/dyn/products/datasheets.asp?family_id=682
Tested on ATSTK1006 + ATSTK1000 with a class D amplifier stage.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Impact: cleanup
snd_pcm_new takes a char *id argument, although it is not modifying
the string. it can therefore be declared as const char *id.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fix the following 'make headers_check' warning:
usr/include/sound/hdsp.h:33: found __[us]{8,16,32,64} type without #include <linux/types.h>
Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Introduced snd_card_create() function as a replacement of snd_card_new().
The new function returns a negative error code so that the probe callback
can return the proper error code, while snd_card_new() can give only NULL
check.
The old snd_card_new() is still provided as an inline function but with
__deprecated attribute. It'll be removed soon later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many codec drivers were implementing cookie-cutter copies of the function
that adds kcontrols to the codec.
This patch moves this code to a common function snd_soc_add_controls() in
soc-core.c and updates all drivers using copies of this function to use the
new common version.
[Edited to raise priority of error log message and document parameters.
-- broonie]
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a jack reporting interface to ASoC. This wraps the ALSA
core jack detection functionality and provides integration with DAPM to
automatically update the power state of pins based on the jack state.
Since embedded platforms can have multiple detecton methods used for a
single jack (eg, separate microphone and headphone detection) the report
function allows specification of which bits are being updated on a given
report.
The expected usage is that machine drivers will create jack objects and
then configure jack detection methods to update that jack.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Merge the recently introduced soc_value_enum structure to the soc_enum.
The value based enums are still handled separately from the normal enum types,
but with the merge some of the newly introduced functions can be removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch allows you to define the mixer paths as having the same name as the
paths they represent.
This is required to support codecs such as the wm9705 neatly without extra
controls in the alsa mixer.
Signed-off-by: Ian Molton <ian@mnementh.co.uk>
Add support for reporting new jack types SND_JACK_VIDEOOUT and
SND_JACK_AVOUT (a combination of LINEOUT and VIDEOOUT) to the jack
reporting API.
Also add the corresponding SW_VIDEOOUT_INSERT switch to the input system
header.
Signed-off-by: Jani Nikula <ext-jani.1.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch introduces a new enum type.
In this enum type each enumerated items referred with a value.
This new enum type can handle enums encoded in bitfield, or any other
weird ways. twl4030 codec has several mux selection register, where the
input/output mux is coded in a bitfield. With the normal enum type this type
of mux can not be handled in a clean way.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use a lookup table rather than explicit code to map input subsystem jack
types into ASoC ones, implemented as suggested by Takashi Iwai.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce a struct v4l2_file_operations for v4l2 drivers.
Remove the unnecessary inode argument.
Move compat32 handling (and llseek) into the v4l2-dev core: this is now
handled in the v4l2 core and no longer in the drivers themselves.
Note that this changeset reverts an earlier patch that changed the return
type of__video_ioctl2 from int to long. This change will be reinstated
later in a much improved version.
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
We'd like to use the High Pass Filter and V_REFOUT bitshift values elsewhere,
so stick them into a ac97_codec.h.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another part of the backporting of Liam's ASoC v2 work. Using this is
more complicated than the other registration types since currently the
codec is instantiated during the probe of the ASoC device so we can't
currently readily wait for the codec to register.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems support both mechanical and electrical jack detection,
allowing them to report that a jack is physically present but does
not have any functioning connections. Add a new jack type for these,
allowing user space to report faulty connections.
Thanks to Guillem Jover for the suggestion.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC v2 allows platform drivers to instantiate independantly of the
overall ASoC card. This API allows drivers to notify the core when
they are registered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add API calls to register and unregister DAIs with the core. Currently
these APIs are ineffective. Since multiple DAIs for a given device are
a common case bulk variants are provided.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows cards, codecs and platforms to instantiate separately,
with the overall ASoC device only being instantiated once all the
required components have registered. As part of backporting Liam's work
introduce an initial version of the card registration functions. At
present these do nothing active and are internal only, they will be
exposed to machine drivers after further backporting. Adding this now
allows the datastructures used for dynamic card instantiation to be
built up gradually.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is in preparation for the removal of struct snd_soc_device.
The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card. Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC card initialisation is completed by a function called
snd_soc_register_card(). As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.
This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 factors most of the contents of soc.h out into separate headers,
including soc-dai.h for the DAI. Factor the existing DAI API out into
this file in order to prepare for backporting of the ASoC v2 DAI API.
Also backport some of Liam's improvements to the documentation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change coding style to be more acceptable by checkpatch.pl.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change snd_BUG_ON() to evaluate the given condition, at least, in syntax
for avoiding compile warnings such as unused variables. The compiler
should optimize out the condition evaluation in the real code, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/pcxhr/pcxhr_core.c: In function 'pcxhr_set_pipe_cmd_params':
sound/pci/pcxhr/pcxhr_core.c:700: warning: statement with no effect
sound/pci/pcxhr/pcxhr_core.c:706: warning: statement with no effect
sound/pci/pcxhr/pcxhr_core.c:710: warning: statement with no effect
Due to
try to fix this, and be more conventional about the empty stubs.
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it. It was much more useful
when ASoC was out of tree.
Signed-off-by: Mark brown <broonie@opensource.wolfsonmicro.com>
i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When ASoC was converted to support full int width masks SOC_SINGLE_VALUE()
omitted the assignment of rshift, causing the control operatins to report
some mono controls as stereo. This happened to work some of the time due
to a confusion between shift and min in snd_soc_info_volsw().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch introduces support for reporting SW_LINEOUT_INSERT detection events
via the jack abstraction layer.
Also adds a SND_JACK_LINEOUT define to the input system header.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Cc: Dmitry Torokhov <dtor@mail.ru>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This was marked as deprecated in 2.6.27 and all users except for
playpaq_wm8510 fixed in that release.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (313 commits)
V4L/DVB (9186): Added support for Prof 7300 DVB-S/S2 cards
V4L/DVB (9185): S2API: Ensure we have a reasonable ROLLOFF default
V4L/DVB (9184): cx24116: Change the default SNR units back to percentage by default.
V4L/DVB (9183): S2API: Return error of the caller provides 0 commands.
V4L/DVB (9182): S2API: Added support for DTV_HIERARCHY
V4L/DVB (9181): S2API: Add support fot DTV_GUARD_INTERVAL and DTV_TRANSMISSION_MODE
V4L/DVB (9180): S2API: Added support for DTV_CODE_RATE_HP/LP
V4L/DVB (9179): S2API: frontend.h cleanup
V4L/DVB (9178): cx24116: Add module parameter to return SNR as ESNO.
V4L/DVB (9177): S2API: Change _8PSK / _16APSK to PSK_8 and APSK_16
V4L/DVB (9176): Add support for DvbWorld USB cards with STV0288 demodulator.
V4L/DVB (9175): Remove NULL pointer in stb6000 driver.
V4L/DVB (9174): Allow custom inittab for ST STV0288 demodulator.
V4L/DVB (9173): S2API: Remove the hardcoded command limit during validation
V4L/DVB (9172): S2API: Bugfix related to DVB-S / DVB-S2 tuning for the legacy API.
V4L/DVB (9171): S2API: Stop an OOPS if illegal commands are dumped in S2API.
V4L/DVB (9170): cx24116: Sanity checking to data input via S2API to the cx24116 demod.
V4L/DVB (9169): uvcvideo: Support two new Bison Electronics webcams.
V4L/DVB (9168): Add support for MSI TV@nywhere Plus remote
V4L/DVB: v4l2-dev: remove duplicated #include
...
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
ALSA: ASoC codec: remove unused #include <version.h>
ALSA: ASoC: update email address for Liam Girdwood
ALSA: hda: corrected invalid mixer values
ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
ALSA: ASoC: Add destination and source port for DMA on OMAP1
ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
ALSA: ASoC: Fix build of GTA01 audio driver
ALSA: ASoC: Add widgets before setting endpoints on GTA01
ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
ALSA: ASoC: Make TLV320AIC26 user-visible
ALSA: ASoC - clean up Kconfig for TLV320AIC2
ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
ALSA: ASoC: Implement WM8510 bias level control
ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
ALSA: ASoC: Add WM8510 SPI support
ALSA: ASoC: Add WM8753 SPI support
...
Add a new API call snd_soc_dapm_nc_pin() which allows machine drivers to
mark pins as being permanently disabled. At present this is identical
to snd_soc_dapm_disable_pin() except in terms of improving the internal
documentation of machine drivers that use it. The intention is that in
future it will be extended to provide additional features such as hiding
controls that are only relevant to paths using the disconnected pin.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the video_exclusive_open/release functionality into the
driver itself.
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Increase the card components[] (and thus snd_card_info.components[],
too) array size from 80 to 128 chars so that more strings can be
stored. The 80 chars aren't enough for more than 2 HD-audio codecs,
and this hits an ugly snd_BUG() as reported by Wu Fegguang for HP
2230s.
The control protocol number is increased to 2.0.6 as well, in case
it matters.
Reported-by: Wu Fengguang <wfg@linux.intel.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
We have some arithmetic operations against snd_pcm_hw_param_t, thus
bitwise isn't correct for it. Better to remove the flag to shut up
sparse warnings.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASoC and non-ASoC drivers for PCM DMA on PXA share lots of common code.
Move it to pxa2xx-lib.
[Fixed some checkpatch warnings -- broonie]
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASoC and non-ASoC drivers for ACLINK on PXA share lot's of common code.
Move all common code into separate module snd-pxa2xx-lib.
[Fixed handing of SND_AC97_CODEC in Kconfig and some checkpatch warnings
-- broonie]
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Empty files remained likely due to wrong patching.
Remove them now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- more register naming work
- finally figured out that weird CR register stuff
(and did I mention that I hate _really_ undecipherable open-coded values?)
- fix handling of IRQ sharing in interrupt handler
(hopefully properly, otherwise I'd be grateful to hear your
pedantic comments ;)
- add handy SPECS_PAGE references wherever useful
- comments, cleanup
- add me as module author
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most hardwares have limited buffer-descriptor table length. This
also restricts the max buffer size of the sound driver.
For example, snd-hda-intel has 1MB buffer size limit, and this is
because it can have at most 256 BDL entries. For supporting larger
buffers, we need to allocate larger pages even for sg-buffers.
This patch changes the sgbuf allocation code to try to allocate
larger pages first. At each head of the allocated pages, the
number of allocated pages is stored in the lowest bits of the
corresponding entry of the table addr field. This change isn't
visible as long as the driver uses snd_sgbuf_get_addr() helper.
Also, the patch adds a new function, snd_pcm_sgbuf_get_chunk_size().
This returns the size of the chunk on continuous pages starting at
the given position offset. If the chunk reaches to a non-continuous
page, it returns the size to the boundary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Clean up SG-buffer helper functions and macros. Helpers take substream
as arguments now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in other places, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kill snd_assert() in sound/core/*, either removed or replaced with
if () with snd_BUG_ON().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Introduced snd_BUG_ON() macro as a replacement of snd_assert() macro.
snd_assert() is pretty ugly as it has the control flow in its argument.
OTOH, snd_BUG_ON() behaves like a normal conditional, thus it's much
easier to read the flow.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss detection code and kill the ad1848 library.
The library is fully assimilated into the new wss library.
This required reworking of the AD1848 family code
so the code is changed to correctly detect chips from
the AD1848 and CS4231 families.
I have tested it on following cards:
Gallant SC-6600 (codec: AD1848, driver: snd-sc6600)
SoundScape VIVO/90 (codec: AD1845, driver: snd-sscape)
SG Waverider (codec: CS4231A, driver: Rene Herman's snd-galaxy)
Opti930 (codec: built-in - CS4231 compatible, driver: snd-opti93x)
Opti931 (codec: built-in - CS4231 compatible, driver: snd-opti93x)
Gallant SC-70P (chip/codec: CS4237B, driver: snd-cs4236)
Audio Plus 3D (chip/codec: CMI8330A, driver: snd-cmi8330)
Dell Latitude CP (chip/codec: cs4236, driver snd-cs4232)
Sound playback and recording works on all these cards.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss pcm code and kill the ad1848 pcm code.
The AD1848 chip is much slower than CS4231 chips
so the waiting loop was increased 100x (10x is not
enough).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss mixer code and kill the ad1848 mixer code.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use CS4231P instead of AD1848P (kill the AD1848P).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the wss macros instead of ad1848 ones.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use wss constants for mode.
Move ad1848 hardware constants to the wss.h.
Move mixer tlv macros into the ad1848_lib.c from the ad1848.h.
Drop the MODE_RUNNING spurious IRQ guard on AD1848 as it doesn not seem
to be needed.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The snd_wss is superset of the snd_ad1848 so kill
the latter and replace it with the snd_wss.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rename functions and structures from the former
cs4321_lib to names more corresponding with the
new name: wss_lib.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rename file include/sound/cs4231.h
into include/sound/wss.h
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine
to have more than 8 PCM devices per card, except one place - the
SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate
devices > 7. This patch fixes the issue, changing the devices list
organisation.
Instead of adding new device to the tail, the list is now kept always
ordered (by card number, then device number). Thus, during enumeration,
it is easy to discover the fact that there is no more given card's
devices.
Additionally the device field of struct snd_pcm had to be changed to int,
as its "unsignednity" caused a lot of problems when comparing it to
potentially negative signed values. (-1 is 0xffffffff or even more then ;-)
Signed-off-by: Pawel Moll <pawel.moll@st.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a new US122L usb-audio driver. This driver works together with a
dedicated alsa-lib plugin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This reverts commit fb3d6f2b77bdec75d45aa9d4464287ed87927866.
New, updated patch with same subject replaces this commit.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Updated IEC958 consumer status channel definitions according
to the third edition of IEC60958-3 spec.
Signed-off-by: Pawel Moll <pawel.moll@st.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When compiled with CONFIG_SND_DYNAMIC_MINORS the ALSA core is fine
to have more than 8 PCM devices per card, except one place - the
SNDRV_CTL_IOCTL_PCM_NEXT_DEVICE ioctl, which will not enumerate
devices > 7. This patch fixes the issue, changing the devices list
organisation.
Instead of adding new device to the tail, the list is now kept always
ordered (by card number, then device number). Thus, during enumeration,
it is easy to discover the fact that there is no more given card's
devices. The same limit was present in OSS emulation code. It has
been fixed as well.
Additionally the device field of struct snd_pcm is now int, instead of
unsigned int, as there is no obvious reason for keeping it unsigned.
This caused a lot of problems with comparing this value with other
(almost always signed) variables. There is just one more place where
device number is unsigned - in struct snd_pcm_info, which should be
also sorted out in future.
Signed-off-by: Pawel MOLL <pawel.moll@st.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASOC: convert use of uint to unsigned int
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The OpenFirmware API headers don't build on all platforms so ensure
that they are not included unless they are being used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Simple utility layer for creating ASoC machine instances based on data
in the OpenFirmware device tree. OF aware platform drivers and codec
drivers register themselves with this framework and the framework
automatically instantiates a machine driver. At the moment, the driver
is not very capable and it is expected to be extended as more features
are needed for specifying the configuration in the device tree.
This is most likely temporary glue code to work around limitations in
the ASoC v1 framework. When v2 is merged, most of this driver will
need to be reworked.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most of the ASoC controls refer to the maximum value that can be set for
a control as mask but there is no actual requirement for all bits to be
set at the highest possible value making the name mask misleading.
Change the code to use max instead.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Convert bitfields in ASoC into full int width. This is a
simple mechanical conversion. Two places in the DAPM code
were fixed to properly use mask.
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Some codecs have unusual features in their register maps such as very
large registers representing arrays of coefficients. Support these
codecs in the register cache sysfs file by allowing them to provide a
function register_display() overriding the default output for register
contents.
Also ensure that we don't overflow PAGE_SIZE while writing out the
register dump.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Currently very few systems provide information about jack status to user
space, even though many have hardware facilities to do detection. Those
systems that do use an input device with the existing SW_HEADPHONE_INSERT
switch type to do so, often independently of ALSA.
This patch introduces a standard method for representing jacks to user
space into ALSA. It allows drivers to register jacks for a sound card with
the input subsystem, binding the input device to the card to help user
space associate the input devices with their sound cards. The created
input devices are named in the form "card longname jack" where jack is
provided by the driver when allocating a jack. By default the parent for
the input device is the sound card but this can be overridden by the
card driver.
The existing user space API with SW_HEADPHONE_INSERT is preserved.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
dapm_reg_event() is used by devices using SND_SOC_DAPM_REG() so needs to
be exported to support building them as modules and prototyped to avoid
sparse warnings and potential build issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new ALSA driver for the audio device found inside
most of the SGI O2 workstation. The hardware uses a SGI custom chip,
which feeds a AD codec chip.
Signed-off-by: Thomas Bogendoerfer <tsbogend@alpha.franken.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
A bunch of things in alsa depend on CONFIG_KMOD,
use CONFIG_MODULES instead where the dependency
is needed at all.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds several functions for DAI control and config
and replaces the current method of calling function pointers within
the DAI struct.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch series merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai in preparation for further
ASoC v2 patches.
This merger removes duplication in both DAI structures and simplifies
the API for other users.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This generic register modifier widget is for updating multiple codec
register bits at once when the widget changes its power state.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This reverts commit 36b34d2437104f323e09d7c6af6451d3c0b9c0cd.
From: Al Viro <viro@ZenIV.linux.org.uk>
WIW, *all* this stuff is not bitwise at all. For crying out loud, half
of these types are routinely used as array indices and loop variables...
If anything, we want a different set of allowed operations - subtraction
between elements of type (yielding integer), addition/subtraction of
integer types not bigger than ours (yielding our type), comparisons,
assignments (=, +=, -=, passing to function as argument, return from
function, initializers) and second/third arguments in ?:. With 0 *not*
being allowed as a constant of such type.
It's not bitwise; we may use the same infrastructure in sparse, but it
should be a separate class of types (__attribute__((affine))).
dma_addr_t is another candidate for the same treatment, but there we'll
need helpers for conversions to hw-acceptable form (dma_to_le32(), etc.)
and gradual conversion of drivers.
ALSA ones and pm mess are absolutely straightforward cases, though.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Fully half of all alsa sparse warnings are from snd_pcm_hw_param_t degrading
to integer type, this goes a long way towards eliminating them.
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On OpenMoko soc-audio resume is taking 700ms of the whole resume time of
1.3s, dominated by writes to the codec over I2C. This patch shunts the
resume guts into a workqueue which then is done asynchronously.
The "card" is locked using the ALSA power state APIs as suggested by
Mark Brown.
[Added fix for race with resume to suspend and fixed a couple of nits
from checkpatch -- broonie.]
Signed-off-by: Andy Green <andy@openmoko.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
snd_ctl_elem_read() and snd_ctl_elem_write() are no longer used by
any other drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mike Montour <mail@mmontour.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This allows per-DAI initialisation to be done by the CPU DAI drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch adds support for WSS compatible Opti93x
codec to the cs4231-lib.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Tested-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
On Audigy2 Platinum, the Analog/Digital mixer switch is inverted.
https://bugzilla.novell.com/show_bug.cgi?id=396204
The patch adds a simple workaround.
There might be another device requiring a similar fix, too (or fix for
audigy2 generically), but right now I fix only the known broken one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SOC_DOUBLE_S8_TLV control type was originally implemented in the
UDA1380 driver by Philipp Zabel and was moved into the core by me.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace CONFIG_SND_DEBUG_DETECT with CONFIG_SND_DEBUG_VERBOSE to
represent its meaning more better. This config isn't provided only
for the detection but for more verbose debug prints in general.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes CVS keywords that weren't updated for a long time
from comments.
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the ASoC core configures the bias levels in the system using
a callback on codecs and machines called 'dapm_event', passing it PCI
style power levels as SNDRV_CTL_POWER_ constants. This is more obscure
than it needs to be and has caused confusion to driver authors,
especially given that DAPM is also performing power management.
Address this by renaming the callback function to 'set_bias_level' and
using constants explicitly representing the off, standby, pre-on and on
states which DAPM transitions through.
Also unexport the API for setting bias level: there are currently no
in-tree users of this API other than the core itself and it is likely
that the core would need to be extended to cater for any users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The CPU and codec DAI operations differ only in the presence of the
digital mute operation for the codec so they may as well be the same
type.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
ASoC codecs and machine drivers that use DAPM routes all cut'n'paste a
loop iterating over a null terminated array of routes. Factor out this
into a bulk registration function, improving the error reporting for
most users, and deprecate the old API to help out of tree users pick up
the changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Graeme Gregory <graeme@openmoko.org>
Cc: Frank Mandarino <fmandarino@endrelia.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Most SoC drivers cut'n'paste a loop iterating over an array to register
their DAPM controls. Provide a function they can call instead.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Graeme Gregory <graeme@openmoko.org>
Cc: Frank Mandarino <fmandarino@endrelia.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This adds a hook to read the power state of a DAPM widget, I use this
in the gta02 driver to expose certain DAPM widgets in the mixer for
ease of audio routing.
Signed-off-by: Graeme Gregory <graeme@openmoko.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I suspect that snd_ctl_boolean_mono should have been
snd_ctl_boolean_mono_info instead. This fixes the build for magician.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_minor_info_oss_* is an function returning int _or_ comment,
depending on config parameters. That is truly evil, fix it.
Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two sentences seem to be spliced into one in comment, fix that and fix
english. Also fix codingstyle.
Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* removing the hack with NON_AKM ak4xxx type
* support for card-specific flags in ak4114_stats
* definition of the flags for corresponding cards
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added definition for byte 4 of SPDIF channel status, according to
second edition of IEC 60958-3 (consumer) spec.
Signed-off-by: Pawel MOLL <pawel.moll@st.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the codes for virtual master controls to sound core part so that
not only hda-intel drivers can use it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of 8 channel sound for codecs that are known to work.
So far, only ALC850 is marked as a 8ch-support codec.
This fix is a modified version of the patch on ALSA BTS#2097 by
Martin Ellis:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2097
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper ifdef's to the patch loading code moved from the old instr
layer so that opl3 driver can be compiled without the sequencer support.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This patch improves E-Mu 1616(M) cardbus support. It adds definitions of the
new Microdock and 1010 cardbus registers (thanks again for descriptions
James) and improves mixer for this card. Now you can use S/PDIF and ADAT on
Mirodock and also use headpohone output on host cardbus card as another
independent output.
Signed-off-by: Ctirad Fertr <c.fertr@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This is improvement of the early support of the FM-only cards where the
fm801 chip represents the PCI to tuner bridge.
The tuner initialization isn't included the mute on as well as mute support
via V4L request. Proposed patch should fix this at least for 64-PCR model.
Signed-off-by: Andy Shevchenko <andy@smile.org.ua>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Change semantics for SNDRV_PCM_TSTAMP_MMAP. Doing timestamping only in
the interrupt handler might cause that hw_ptr is not related to actual
timestamp. With this change, grab timestamp at every hw_ptr update to
have always valid timestamp + ring buffer position pair.
With this change, SNDRV_PCM_TSTAMP_MMAP was renamed to
SNDRV_PCM_TSTAMP_ENABLE. It's no regression (I think).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This fixes a bug whereby PCMs were not being suspended when the rest of the
audio subsystem was suspended.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a device level dapm event so that both the machine and codec are informed
when dapm events occur.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This header file exists only for some hacks to adapt alsa-driver
tree. It's useless for building in the kernel. Let's move a few
lines in it to sound/core.h and remove it.
With this patch, sound/driver.h isn't removed but has just a single
compile warning to include it. This should be really killed in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The 'tick' in PCM is set (again) via sw_params. And, nobody uses
this feature at all except for a command line option of aplay.
(This is literally 'nobody', as I checked alsa-lib API calls in all
programs in major distros.)
Above all, if we need finer wake-ups for the position update, it's
basically an issue that the driver should solve, not tuned by each
application.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The xfer_align sw_params parameter has never been used in a sane manner,
and no one understands what this does exactly. The current
implementation looks also buggy because it allows write of shorter size
than xfer_align. So, if you do partial writes, the write isn't actually
aligned at all.
Removing this parameter will make some pcm_lib_* code more readable
(and less buggy).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch removes the indirect control access to the control elements.
The indirect access has never been used and is even broken on 32bit
ioctl wrapper. Let's clean it up.
The pointers still remain in snd_ctl_elem_* structs just to make sure
that the struct size won't change. Once after checking the size
consistency, we can get rid of them, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
info_oss: move prototype of snd_card_info_read_oss to info.h
Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
We need an accurate and continuous (monotonic) time sources to do
accurate synchronization among more timing sources. This patch allows
to enable monotonic timestamps for ALSA PCM devices and enables monotonic
timestamps for ALSA timer devices.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* support for switching rate in STAC9460 - using set_rate_val of the akm
infrastructure
* listing all STAC9460 registers in proc
* disabling mpu401 device for Prodigy192 - otherwise the currently
flawed mpu401 code hangs kernel when opening the midi device
* removing old unused commented-out code
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
remove dead config symbols from sound code
Signed-off-by: Jiri Olsa <olsajiri@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Updated the forgotten SNDRV_HWDEP_IFACE_LAST to point the really last member.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Remove sequencer instrument layer from the tree.
This mechanism hasn't been used much with the actual devices. The only
reasonable user was OPL3 loader, and now it was rewritten to use hwdep
instead. So, let's remove the rest of rotten codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the exclusive access lock in hwdep instead of the own one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Use the hwdep device for loading OPL2/3 patch data instead of the
messy sequencer instrument layer.
Due to this change, the sbiload program should be updated, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
open and close operations are called only from pcm layer
and mutexed there with pcm->open_mutex.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Since the last patch made the ENTER_UART command optional, the
enter_uart option and its corresponding flag have become superfluous.
The uart_enter option remains for backward compatibility but just prints
a warning when used.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
These patches enable some YMF743 controls (Tone/3D/IEC958) that won't
be detected with the current version of ALSA.
The first one contains only cosmetic changes to share a few
YMF753-specific symbols with YMF743.
Signed-off-by: Keita Maehara <maehara@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The volatile prefix is just useless there. Let's kill them, and then
gcc will be happier, too.
sound/acore/pcm.c:867: warning: passing argument 1 of ‘__constant_c_and_count_memset’ discards qualifiers from pointer target type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed cast messes in pcm.h.
include/sound/pcm.h: In function ‘hw_param_interval_c’:
include/sound/pcm.h:800: warning: passing argument 1 of ‘hw_param_interval’ discards qualifiers from pointer target type
Simply redefine the inline functions again for const pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Consistent variable naming is a good thing, but let's be a little less
sneaky about enforcing it... ;-/
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch splits the cs4231.h file into two parts:
- cs4231-regs.h which contain register constants and macros
- cs4231.h which includes the above and contain rest of the definitions
This will allow to share register definitions between x86 ISA cs4231
and SPARC cs4231.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add a snd_pcm_rate_to_rate_bit() function to factor out common code used
by several drivers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Merge the rates[] arrays from pcm_misc.c and pcm_native.c because they
are both the same.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a hwdep interface for each codec (enabled per kconfig).
This interface can be used for reading/writing HD-audio verbs
and other purposes as future extensions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix codes to follow more to the standard kernel coding style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The mode change / recalibration doesn't work always with opl3sa2 devices,
e.g. the first time it's played back. The patch fixes the problem.
Signed-off-by: Paul Vojta <vojta@math.berkeley.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clean up codes using the new common snd_ctl_boolean_*_info() callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added helper functions for frequenty used callbacks:
snd_ctl_boolean_mono_info() and snd_ctl_boolean_stereo_info()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Notebook.
Description: The .device=0x0008 chips have new, but different EMU32 in/out
channels. Driver updated to make use of these EMU32 channels.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed PM resume of cs46xx devices. It now restores properly the DSP
image and kick-off the DSP.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* adding 8 more 32-bit capture channels (total of 16) for emu1010 cards
* adding some code comments and card details description
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add support for Cyrix/NatSemi Geode SC5530 (VSA1).
The driver is snd-cs5530.
Signed-off-by Ash Willis <ashwillis@programmer.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the support of mute for front channels of M-Audio
Revolution 7.1 (the DAC AK4381 features a mute bit).
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch fixes a bug whereby AC97 bus device data was being clobbered
when AC97 codecs using the generic ac97_codec.c driver were being
registered. Codecs that didn't use the generic driver were unaffected
(e.g. WM9712, WM9713).
Changes:-
o Add new AC97 codec class for custom (or need bus dev registration)
AC97 codecs.
o Only register/deregister this custom codec device with the AC97 bus.
The generic AC97 driver already does this for generic codec devices.
This may be related to bug #3038 :-
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3038
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
* correct register for 'IEC958 Non-PCM Bitstream', 'IEC958 DTS Bitstream'
to use AK4114_REG_RCS0
* correct check for control name: if (strstr(kctl->id.name, 'Playback'))
* correct check: if (!chip->init) in snd_ak4114_external_rate
* added PCM control 'IEC958 PPL Lock Status'
Signed-off-by: Pavel Hofman <dustin@seznam.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed a typo in AK4114_DIF2 bit definition. This may fix some
problems for Audiophile 192 and Juli boards.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added MPU401_INFO_UART_ONLY bitflag to avoid issueing UART_ENTER command
at opening streams. Some devices support only UART mode and give errors
to UART_ENTER.
A new module option, uart_enter, is added to snd-mpu401 driver.
For UART-only devices, set uart_enter=0.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a new macro snd_pcm_group_for_each_entry() just for code cleanup.
Old macros, snd_pcm_group_for_each() and snd_pcm_group_substream_entry(),
are removed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Many struct file_operations in the kernel can be "const". Marking them const
moves these to the .rodata section, which avoids false sharing with potential
dirty data. In addition it'll catch accidental writes at compile time to
these shared resources.
Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The fix for sysfs breakage with CONFIG_SYSFS_DEPRECATED was flown
away by the conflicted merge of the ALSA git tree. The patch below
fixes it again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch updates the API's to include the new DAI configuration and
clocking architecture.
Changes:-
o Removed DAI automatic matching and capabilities structure (struct
snd_soc_dai_mode) and macros.
o Added DAI operations for codec and CPU interfaces.
o Removed config_sysclk() function and struct snd_soc_clock_info. No
longer needed as clocking is now configured manually in the machine
drivers. Also removed other clocking data from structures.
o Updated version to 0.13
o Added shift to SOC_SINGLE_EXT kcontrol macro.
Signed-off-by: Graeme Gregory <gg@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Mark TLV data as 'const'
Signed-of-by: Philipp Matthias Hahn <pmhahn@pmhahn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Make data passed to ak4xxx_create 'const'.
Signed-of-by: Philipp Matthias Hahn <pmhahn@pmhahn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix ABI for older ld10k1. When no EMU10K1_PVERSION ioctl is issued,
the driver accepts ioctls with the old struct size without TLV information.
Also, changed the struct field to make the conversion easier from the
old to the new structs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Previously, ac97_codec.c was coded to support AD1986 and AD1986A
CODECs using code written for the AD1985 CODEC. This allowed the
LINE_OUT and HEADPHONE jacks to function properly, however register
differences between the CODECs prevented line and microphone inputs
from functioning.
Specifically, this patch fixes issues with the following mixer
controls: 'V_REFOUT', 'Spread Front to Surround and Center/LFE',
'Exchange Front/Surround', 'Surround Jack Mode', and 'Channel Mode'.
This patch removes the undocumented AD1888 control
'High Pass Filter Enable' and adds the new control
'Exchange Mic/Line In'.
Signed-off-by: Randy Cushman <rcushman_linux@earthlink.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix the changes realted to delayed_work in soc/codecs/wm8750.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Use global workqueue for simplicity instead of own workqueue.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
From: Andrew Morton <akpm@osdl.org>
I converted the workqueues to per-device while I was there. It seems
strange to create a new kernel thread (on each CPU!) and to then only
have a single global work to ever be queued upon it.
Plus without this, I'd have to use the _NAR stuff, gawd help me.
Does that workqueue really need to be per-cpu?
Does that workqueue really need to exist? Why not use keventd?
Cc: Takashi Iwai <tiwai@suse.de>
Cc: David Howells <dhowells@redhat.com>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Implement functionallity in order to fixe ALSA bug#2058.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Following patch will make the driver to use the 44.1kHz SRC automatically
if the pcm source is 44.1kHz signed 16bit stereo.
The SRC is available in YMF754 only.
Signed-off-by: Teru KAMOGASHIRA <teru@sodan.ecc.u-tokyo.ac.jp>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added a helper function snd_pci_quirk_lookup()
to look up PCI SSID quirk list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Removed obsolete typedefs.h. It existes only for backward compatibility,
and now all codes should be free from such typedefs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Now that everyone uses snd_ctl_new1() and noone is using snd_ctl_new()
anymore, we can make it static.
Signed-off-by: Adrian Bunk <bunk@stusta.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Instead of using a somewhat algorithmic approach of initializing the
YSS225's registers, just use a simple series of port/value pairs.
This makes it easier to later replace or entirely remove the register
data blob.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Load the CSP programs using request_firmware(), if possible, instead of
using the built-in firmware blobs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Don't enable power-saving mode on drivers that don't support
it. The supporting drivers set AC97_SCAP_POWER_SAVE to scaps
at creation of ac97 instance.
Currently enable on the following drivers: intel8x0, intel8x0m,
atiixp, atiixp-modem, via82xx and via82xx-modem.
Also, a bit clean up of power-saving stuff:
- Don't create an own workq
- Remove superfluous ifdefs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds support for the DAI BCLK to be generated by multiplying
Rate * Channels * Word Size (RCW).
This now gives 3 options for BCLK clocking and synchronisation :-
1. BCLK = Rate * x
2. BCLK = MCLK / x
3. BCLK = Rate * Chn * Word Size. (New)
Changes:-
o Add support for RCW generation of BCLK
o Update Documentation to include RCW.
o Update DAI documentation for label = value DAI modes.
o Add RCW support to wm8731, wm8750 and pxa2xx-i2s drivers.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Load the DSP and controller microcode using request_firmware(), if
possible, instead of using the built-in firmware.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds the ASoC and DAPM headers.
Features:-
o Defines Digital Audio Interface (DAI) API
o Defines Codec, Platform and Machine API
o Defines Dynamic Audio Power Management API
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds a struct device pointer to struct snd_pcm in order to be
able to give it a different device than the card. It defaults to the card's
device, however, so it should behave identically for drivers not touching
the field.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds snd_register_device_for_dev taking a struct device
pointer to link the new device to and makes snd_register_device a simple
static inline wrapper around it.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Enable the analog loopback of the Revolution 5.1 card.
This patch adds support for the PT2258 volume controller and modifies
the Revolution 5.1 driver to make use of this facility. This allows
to control the analog loopback of the card.
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Enable capture from line-in and CD on the Revolution 5.1 card.
This patch adds support for switching between the 5 input channels of
the AK5365 ADC and modifies the Revolution 5.1 driver to make use of
this facility. Previously the capture channel was fixed to channel 0
(microphone on the Revolution 5.1 card).
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The recent change for a new sysfs tree with card* object breaks the
/sys/class/sound tree if CONFIG_SYSFS_DEPRECATED is enabled.
The device in each entry doesn't point the correct device object:
/sys/class/sound
...
|-- pcmC0D0c
| |-- dev
| |-- device -> ../../../class/sound/card0
| |-- pcm_class
| |-- power
| | `-- wakeup
| |-- subsystem -> ../../../class/sound
| `-- uevent
Also, this change breaks some drivers (like sound/arm/*) referring
card->dev directly to obtain the device object for memory handling.
This patch reverts the semantics of card->dev to the former version,
which points to a real device object. The card* object is stored in a
new card->card_dev field, instead. The device parent is chosen either
card->dev or card->card_dev according to CONFIG_SYSFS_DEPRECATED to
keep the tree compatibility.
Also, card* isn't created if CONFIG_SYSFS_DEPRECATED is enabled. The
reason of card* object is a root of all beloing devices, and it makes
little sense if each sound device points to the real device object
directly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Monty Montgomery <xiphmont@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
Fix the type of PCI revision to char from int and avoid invalid
assignment with pointer cast.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed the race among multiple threads accessing the OSS PCM
instance concurrently by simply introducing a mutex for protecting
a setup of the PCM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch fixes incorrect assignment of swap_rear,
which was broken since patch 'ymfpci - make rear channel swap optional'
It removes module_param rear_swap.
Signed-off-by: Glen Masgai <mimosius@gmx.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Conflicts:
drivers/ata/libata-scsi.c
include/linux/libata.h
Futher merge of Linus's head and compilation fixups.
Signed-Off-By: David Howells <dhowells@redhat.com>
Conflicts:
drivers/infiniband/core/iwcm.c
drivers/net/chelsio/cxgb2.c
drivers/net/wireless/bcm43xx/bcm43xx_main.c
drivers/net/wireless/prism54/islpci_eth.c
drivers/usb/core/hub.h
drivers/usb/input/hid-core.c
net/core/netpoll.c
Fix up merge failures with Linus's head and fix new compilation failures.
Signed-Off-By: David Howells <dhowells@redhat.com>
Converts from using struct "class_device" to "struct device" making
everything show up properly in /sys/devices/ with symlinks from the
/sys/class directory.
It also makes the struct sound_card to show up as a "real" device
where all the different sound class devices are placed as childs
and different card attribute files can hang off of. /sys/class/sound is
still a flat directory, but the symlink targets of all devices belonging
to the same card, point the the /sys/devices tree below the new card
device object.
Thanks to Kay for the updates to this patch.
Signed-off-by: Kay Sievers <kay.sievers@novell.com>
Acked-by: Jaroslav Kysela <perex@suse.cz>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
Alsa used to kmalloc one file->f_op per file per disconnecting snd_card.
This led to oopses sometimes when file->f_op was freed before __fput()
finished.
Patch adds a virtual device for disconnect: VDD.
VDD consists of:
LIST_HEAD(shutdown_files)
protected by DEFINE_SPINLOCK(shutdown_mutex)
static struct file_operations snd_shutdown_f_ops
and functions assigned to it
Additions to struct snd_monitor_file
to specify if instance is hidden by VDD or not.
A VDD's instance is
created in snd_card_disconnect() under the card->files_lock.
cleaned up in snd_card_file_remove() under the card->files_lock.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Maintain a per-CPU global "struct pt_regs *" variable which can be used instead
of passing regs around manually through all ~1800 interrupt handlers in the
Linux kernel.
The regs pointer is used in few places, but it potentially costs both stack
space and code to pass it around. On the FRV arch, removing the regs parameter
from all the genirq function results in a 20% speed up of the IRQ exit path
(ie: from leaving timer_interrupt() to leaving do_IRQ()).
Where appropriate, an arch may override the generic storage facility and do
something different with the variable. On FRV, for instance, the address is
maintained in GR28 at all times inside the kernel as part of general exception
handling.
Having looked over the code, it appears that the parameter may be handed down
through up to twenty or so layers of functions. Consider a USB character
device attached to a USB hub, attached to a USB controller that posts its
interrupts through a cascaded auxiliary interrupt controller. A character
device driver may want to pass regs to the sysrq handler through the input
layer which adds another few layers of parameter passing.
I've build this code with allyesconfig for x86_64 and i386. I've runtested the
main part of the code on FRV and i386, though I can't test most of the drivers.
I've also done partial conversion for powerpc and MIPS - these at least compile
with minimal configurations.
This will affect all archs. Mostly the changes should be relatively easy.
Take do_IRQ(), store the regs pointer at the beginning, saving the old one:
struct pt_regs *old_regs = set_irq_regs(regs);
And put the old one back at the end:
set_irq_regs(old_regs);
Don't pass regs through to generic_handle_irq() or __do_IRQ().
In timer_interrupt(), this sort of change will be necessary:
- update_process_times(user_mode(regs));
- profile_tick(CPU_PROFILING, regs);
+ update_process_times(user_mode(get_irq_regs()));
+ profile_tick(CPU_PROFILING);
I'd like to move update_process_times()'s use of get_irq_regs() into itself,
except that i386, alone of the archs, uses something other than user_mode().
Some notes on the interrupt handling in the drivers:
(*) input_dev() is now gone entirely. The regs pointer is no longer stored in
the input_dev struct.
(*) finish_unlinks() in drivers/usb/host/ohci-q.c needs checking. It does
something different depending on whether it's been supplied with a regs
pointer or not.
(*) Various IRQ handler function pointers have been moved to type
irq_handler_t.
Signed-Off-By: David Howells <dhowells@redhat.com>
(cherry picked from 1b16e7ac850969f38b375e511e3fa2f474a33867 commit)
Add maximum latency tracking to the ALSA subsystem for PCM playback. In
ALSA, the playback application controls the buffer size and thus indirectly
the period of latency that it can deal with. This patch uses 75% of the
total available latency as threshold to announce to the latency subsystem;
While 75% is a crude heuristic it's a quite reasonable one; the remaining
25% can be used for all driver processing for the next samples which is
also proportional to the size of the buffer.
With ogg123 a latency setting of about 4msec was seen (at 44Khz), while
with the "play" command a much longer maximum tolerable latency was seen.
Other, more multimedia oriented players as well as games, will have a lot
smaller buffers to allow better synchronization and those will actually get
into the latency domains where there is impact on the power management
rules.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
Remove IPGA volume controls and merge the IPGA range to ADC volume
controls. These two volumes are not really independent but connected
simply in different ranges 0-0x7f and 0x80-max. It doesn't make sense
to provide two controls.
Since both 0x7f and 0x80 specify 0dB, a hack is needed for IPGA range
to skip 0x80 (increment one) for such controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds a new attribute, pcm_class, to each PCM sysfs entry.
It's useful to detect what kind of PCM stream is, for example, HAL
can check whether it's a modem or not.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the definition of TLV dB range compound. It contains one or
more dB-range or linear-volume TLV entries with min/max ranges.
Used for volume controls with non-linear curves.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
- Clean up the code in AK4xxx-ADDA i2c code.
- Fix capture gain controls for AK5365
- Changed the static table for DAC/ADC mixer labels to use
structs
- Implemented TLV entries for each AK codec
The volumes in AK4524, AK4528 and AK5365 are corrected with
a table to be suitable for dB conversion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the definition of linear volume TLV type.
Some DSP chips and codecs (e.g. AK codec) use linear volume control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added the dB scale information to vxpocket and vx222 drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Add support for the AK5365 ADC.
Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
The flag to avoid 32bit-incompatible mmap for control/status records
should be outside the pcm substream instance since a substream can be
shared among multiple opens. Now it's flagged in pcm_file list that
is directly assigned to file->private_data.
Also, removed snd_pcm_add_file() and remove_file() functions and
substream.files field that are not really used in the code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Remove unused tlv_rw field from struct snd_kcontrol. The callback is
set in tlv.c field, instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Added CONFIG_SND_AC97_POWER_SAVE kernel config to enable the support
of aggressive AC97 power-saving mode. In this mode, the AC97
powerdown register bits are dynamically controlled at each open/close
of PCM streams.
The mode is activated via power_save option for snd-ac97-codec
driver. As default it's off. It can be turned on/off on the fly
via sysfs, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Orignally proposed by Sam Revitch <sam.revitch@gmail.com>.
Unregister device files at disconnection to avoid the futher accesses.
Also, the dev_unregister callback is removed and replaced with the
combination of disconnect + free.
A new function snd_card_free_when_closed() is introduced, which is
used in USB disconnect callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
- Add the linked list to each proc entry to enable a single-shot
disconnection (unregister)
- Deprecate snd_info_unregister(), use snd_info_free_entry()
- Removed NULL checks of snd_info_free_entry()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch implements a TLV mechanism to transfer an additional information
like dB scale to the user space. The types might be extended in future.
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fixed 'section mismatch' errors in ALSA PCI drivers:
- removed invalid __devinitdata from pci id tables
- fix/remove __devinit of functions called in suspend/resume
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Drop the snd_minor structure's name field that was just a helper for
devfs device deregistration.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Fix spaces, fold lines to fit 80 columns in ak4xxx-adda driver codes.
Split a long reset function to each codec routine just for better
readability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds stereo controls to revo cards by making the ak4xxx
driver mixers configurable from the card driver.
Signed-off-by: Jani Alinikula <janialinikula@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
This patch adds two mixer controls. The V_REFOUT enable is a
documented register that couples the microphone input lines
to the V_REFOUT DC source. The High Pass Filter enable in the
AC97_AD_TEST2 (0x5c) is an undocumented register provided by
Miller Puckette via Analog Devices that enables the AD codec
to apply a high pass filter to the input.
Signed-off-by: Jaya Kumar <jayakumar.alsa@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Suppress 'irq handler mismatch' messages at auto-probing of irqs
in ALSA ISA drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
In file included from sound/i2c/other/tea575x-tuner.c:30:
include/sound/tea575x-tuner.h:36: error: field 'vd' has incomplete type
include/sound/tea575x-tuner.h:37: error: field 'fops' has incomplete type
sound/i2c/other/tea575x-tuner.c:89: warning: 'struct inode' declared inside parameter list
sound/i2c/other/tea575x-tuner.c:89: warning: its scope is only this definition or declaration, which is probably not what you want
sound/i2c/other/tea575x-tuner.c: In function 'snd_tea575x_ioctl':
sound/i2c/other/tea575x-tuner.c:91: warning: implicit declaration of function 'video_devdata'
sound/i2c/other/tea575x-tuner.c:91: warning: initialization makes pointer from integer without a cast
sound/i2c/other/tea575x-tuner.c:92: warning: implicit declaration of function 'video_get_drvdata'
sound/i2c/other/tea575x-tuner.c:92: warning: initialization makes pointer from integer without a cast
sound/i2c/other/tea575x-tuner.c:96: warning: implicit declaration of function '_IOR'
sound/i2c/other/tea575x-tuner.c:96: error: syntax error before 'struct'
sound/i2c/other/tea575x-tuner.c:99: error: 'v' undeclared (first use in this function)
sound/i2c/other/tea575x-tuner.c:99: error: (Each undeclared identifier is reported only once
sound/i2c/other/tea575x-tuner.c:99: error: for each function it appears in.)
sound/i2c/other/tea575x-tuner.c:108: warning: implicit declaration of function 'copy_to_user'
sound/i2c/other/tea575x-tuner.c:112: warning: implicit declaration of function '_IOWR'
sound/i2c/other/tea575x-tuner.c:112: error: syntax error before 'struct'
sound/i2c/other/tea575x-tuner.c:115: warning: implicit declaration of function 'copy_from_user'
sound/i2c/other/tea575x-tuner.c: At top level:
sound/i2c/other/tea575x-tuner.c:129: error: syntax error before 'case'
sound/i2c/other/tea575x-tuner.c:146: warning: type defaults to 'int' in declaration of 'snd_tea575x_set_freq'
sound/i2c/other/tea575x-tuner.c:146: warning: parameter names (without types) in function declaration
sound/i2c/other/tea575x-tuner.c:146: error: conflicting types for 'snd_tea575x_set_freq'
sound/i2c/other/tea575x-tuner.c:62: error: previous definition of 'snd_tea575x_set_freq' was here
sound/i2c/other/tea575x-tuner.c:146: warning: data definition has no type or storage class
sound/i2c/other/tea575x-tuner.c:147: error: syntax error before 'return'
sound/i2c/other/tea575x-tuner.c:151: error: syntax error before '&' token
sound/i2c/other/tea575x-tuner.c:152: error: syntax error before '.' token
sound/i2c/other/tea575x-tuner.c:152: warning: type defaults to 'int' in declaration of 'strcpy'
sound/i2c/other/tea575x-tuner.c:152: warning: function declaration isn't a prototype
sound/i2c/other/tea575x-tuner.c:152: error: conflicting types for 'strcpy'
sound/i2c/other/tea575x-tuner.c:152: warning: data definition has no type or storage class
sound/i2c/other/tea575x-tuner.c: In function 'snd_tea575x_init':
sound/i2c/other/tea575x-tuner.c:194: warning: implicit declaration of function 'video_set_drvdata'
sound/i2c/other/tea575x-tuner.c:197: error: 'video_exclusive_open' undeclared (first use in this function)
sound/i2c/other/tea575x-tuner.c:198: error: 'video_exclusive_release' undeclared (first use in this function)
sound/i2c/other/tea575x-tuner.c:200: warning: implicit declaration of function 'video_register_device'
sound/i2c/other/tea575x-tuner.c:200: error: 'VFL_TYPE_RADIO' undeclared (first use in this function)
sound/i2c/other/tea575x-tuner.c: In function 'snd_tea575x_exit':
sound/i2c/other/tea575x-tuner.c:215: warning: implicit declaration of function 'video_unregister_device'
distcc[7333] ERROR: compile sound/i2c/other/tea575x-tuner.c on x/32 failed
make[1]: *** [sound/i2c/other/tea575x-tuner.o] Error 1
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
This patch by Rodolfo Giometti disables the AC97 AUX and VIDEO controls
on the WM9705 when the touchscreen is selected as the AUX and VIDEO
lines are shared with the touch controller.
Changes:-
o Added AC97_HAS_NO_AUX flag
o Test for AC97_HAS_NO_AUX flag in snd_ac97_mixer_build()
o Sets AC97_HAS_NO_VIDEO and AC97_HAS_NO_AUX in patch_wolfson05() when
WM9705 touch driver is selected.
Signed-off-by: Rodolfo Giometti <giometti@linux.it>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the 5th argument of snd_mpu401_uart_new() to bit flags
instead of a boolean. The argument takes bits that consist of
MPU401_INFO_XXX flags.
The callers that used the value 1 there are replaced with
MPU401_INFO_INTEGRATED.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed rwlock around snd_iprintf() in sound core part.
Replaced with mutex.
Also, make mutex and flags static variables with addition of
snd_card_locked() function (just for sound.c).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a get_port_info callback to the snd_rawmidi_global_ops structure to
allow the USB MIDI driver to supply information flags for the sequencer
ports created by seq_midi.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add four new information flags SNDRV_SEQ_PORT_TYPE_HARDWARE, _SOFTWARE,
_SYNTHESIZER, _PORT for sequencer ports. This makes it easier for apps
like Rosegarden to make policy decisions based on the port type.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Move mmap_count to snd_pcm_substream instead of runtime struct
so that multiplly opened substreams via O_APPEND can be handled
correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added O_APPEND flag support to PCM to enable shared substreams
among multiple processes. This mechanism is used by dmix and
dsnoop plugins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move EXPORT_SYMBOL()s to places adjacent to functions/variables.
Also move OSS-specific hw_params helper functions to pcm_oss.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds the Kbuild files listing the files which are to be installed by
the 'headers_install' make target, in generic directories.
Signed-off-by: David Woodhouse <dwmw2@infradead.org>
Fixed Oops at rmmod with CONFIG_SND_VERBOSE_PROCFS=n.
Add ifdef to struct fields for optimization and better compile
checks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modules: PCM Midlevel
This patch makes the needlessly global snd_pcm_format_name() static.
Signed-off-by: Adrian Bunk <bunk@stusta.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Fix possible race of referring the setup hook from the running PCM
- Fix memory leak in an error path of proc write
- Clean up the setup hook parser
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Clean up initialization and destruction of substream instance
Now snd_pcm_open_substream() alone does most initialization jobs.
Add pcm_release callback for cleaning up at snd_pcm_release_substream()
- Tidy up PCM oss code
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark the f_ops members of inodes as const, as well as fix the
ripple-through this causes by places that copy this f_ops and then "do
stuff" with it.
Signed-off-by: Arjan van de Ven <arjan@infradead.org>
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
OPL3_HW_OPL3_PC98 #define isn't used anywhere; previously in
sound/drivers/opl3/opl3_lib.c and sound/isa/cs423x/pc98.c, the latter of which
went away with the rest of PC98 subarch.
Signed-off-by: Arthur Othieno <apgo@patchbomb.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@osdl.org>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
