| Age | Commit message (Collapse) | Author |
|---|
|
Inform userspace that one channel of the internal mic has reversed
polarity, so it does not attempt to add both channels together and
end up with silence.
Cc: stable@vger.kernel.org
Reported-by: Andrzej Mendel <andrzej.mendel@gmail.com>
Alsa-info: http://www.alsa-project.org/db/?f=3088f82a0cf977855f92af9db8ad406c04f71efa
BugLink: https://bugs.launchpad.net/bugs/1529624
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This patch adds native DSD support for the Oppo HA-1. It uses a XMOS chipset
but they use their own vendor ID.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
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>
|
|
Cleanup of includes so that they are ordered alphabetically.
Signed-off-by: Martin Sperl <kernel@martin.sperl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
ACP SRAM banks gets turned on when ACP is powered on.
Not all banks are used for playback/capture. So, power on
required banks during audio device open and power off during
audio device close.
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>
|
|
genpd will power off/on ACP to manage runtime ACP PM. ACP runtime PM
hooks are added to get it deinitialized and initialized respectively,
after it is powered off/on.
When system goes to suspend when audio usecase is active, ACP will
be powered off through genpd. When it resumes, ACP needs to be
initialized and reconfigured.
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>
|
|
ACP IP has internal DMA controller with multiple channels which
can be programmed in cyclic/non cyclic manner. ACP can generate
interrupt upon completion of DMA transfer, if required.
The PCM driver provides the platform DMA component to ALSA core.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Reviewed-by: Alex Deucher <alexander.deucher@amd.com>
Reviewed-by: Murali Krishna Vemuri <murali-krishna.vemuri@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
These are register headers for the ACP (Audio CoProcessor) v2.2
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>
|
|
Mark some registers precious since their
reads have side effects (like clearing flags).
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
SACNT register should be marked volatile since
its WR and RD bits are cleared by SSI after
completing the relevant operation.
This unbreaks AC'97 register access.
Fixes: 05cf237972fe ("ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast")
Signed-off-by: Maciej S. Szmigiero <mail@maciej.szmigiero.name>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
This patch adds pcm capability to support Resume.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
On suspend the link register are lost so we need to reconfigure
them in resume. This patch adds the reconfiguration of the link
register in trigger resume.
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>
|
|
Use the DMA resume capability to resume the DMA position when
stream is suspended/resumed.
In suspend we save the position and when stream is resumed the stream needs
to be started from the position when the stream was suspended using the new
DMA resume capabilities
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>
|
|
In active suspend, any HDA interrupt should wake the system. When device
enters active suspend, we need to enable HDA controller interrupt as wake
source. Similarly disable HDA controller interrupt as wake source when
exiting active suspend.
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>
|
|
When device enters active suspend, we should turn off the links
as they are not in use. Similarly we need to bring back links
when we exit active suspend.
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>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Per HW recommendation, SW shall clear the CGCTL.MISCBDCGE and set
it back once data is transferred. So clear this when we get the
IPC and track using a driver flag, and set back on closure
Signed-off-by: Dharageswari.R <dharageswari.r@intel.com>
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
MISCBDCGE is a new register for Misc Backbone clock gate control
which is useful to control while resetting the link and ensuring
controller is in required state so add API to control it
HW recommends that we reset with CGCTL.MISCBDCGE disabled, so add
that while doing init chip and reset sequence.
Signed-off-by: Jayachandran B <jayachandran.b@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-intel
|
|
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>
|
|
Define soc_add_dai() as a wrapper to add a single DAI to a component.
It can be reused to register a DAI dynamically by topology.
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>
|
|
HW recommends 180us for worst case values for link power up
delay, so change the current delay value from 50 (150us) to 150
(450us)
Signed-off-by: Jayachandran B <jayachandran.b@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>
|
|
A stream is by default in coupled mode, in DSP operation we move
it to decoupled mode. On cleanup HW expects that we leave it back
to default state so couple the DMA on cleanup.
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>
|
|
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>
|
|
pcm1792a is compatible with pcm1795 and pcm1796 so it's
better to have them under the common name pcm179x
Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Signed-off-by: Michael Trimarchi <michael@amarulasolutions.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
The detection of direction for compress was only taking into account codec
capabilities and not CPU ones. Fix this by checking the CPU side capabilities
as well
Cc: <stable@vger.kernel.org>
Tested-by: Ashish Panwar <ashish.panwar@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Future platforms may have a different set of pins/converters.
So use lists to add pins and converters based on enumeration.
Also it may be required to connect any converter to any pin
dynamically as per different use cases (for example DP is
connected to pin 6 on skylake board). So this will help in
dynamically select and route.
Fix the dai map as well to use the pin/cvt from list. Not
enabling all dai maps for now.
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>
|
|
L/RINPUT1 can line to Left/Right Boost Mixer through boost switch.
If boost switch is open, there will be no voice when using L/RINPUT1.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
In normal operation, the left and right channel digital audio data is
converted to analogue in two separate DACs. There is a mono-mix mode
where the two audio channels are mixed together digitally and then
converted to analogue using only one DAC, while the other DAC is
switched off. The mono-mix signal can be selected to appear on both
analogue output channels. The mono mix is automatically attenuated by
6dB to prevent clipping.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
It is possible that some pin widget may return with no converter
connected. So don't throw error if none are found to be connected.
Instead print a warning and continue.
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>
|
|
Conflicts:
drivers/gpu/drm/i915/intel_display.c
sound/soc/intel/skylake/skl.h
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Last minute fixes for v4.4
A few final fixes for v4.4, the main one being the two patches to the
new Sky Lake drivers which fix a previous incorrect fix that went in
during an earlier -rc.
|
|
Data is read in blocks of up to one fragment is size from the circular
buffer on the DSP and is re-packed to remove the padding byte that
exists in the DSP memory map.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
Here support is added for responding to DSP IRQs that are used to
indicate data being available on the DSP. The idea is that we check the
amount of data available upon receipt of an IRQ and on subsequent calls
to the pointer callback we recheck once less than one fragment is
available (to avoid excessive SPI traffic), if there is truely less than
one fragment available we ack the last IRQ and wait for a new one.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
We don't want to use a bypassed write in wm5110_clear_pga_volume,
we might disable the DRE whilst the CODEC is powered down. A
normal regmap_write will always go to the hardware (when not on
cache_only) even if the written value matches the cache. As using
a normal write will still achieve the desired behaviour of bring
the cache and hardware in sync, this patch updates the function
to use a normal write, which avoids issues when the CODEC is
powered down.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
|
|
The bitwise OR has higher precedence than ?: so the val2 was always set
to 0x2.
Fixes: b4c83b171557 ('ASoC: rsnd: add Multi channel support')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
|
|
|
The nrpn_conv_table 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>
|
|
Some users have reported that their Dice based models generate ETIMEDOUT
when starting PCM playback. It means that current timeout (=100msec) is
not enough for their models to transfer notifications.
This commit expands the timeout up to 2 sec. As a result, in a worst case,
any operations to start AMDTP streams takes 2 sec or more. Then, in
userspace, snd_pcm_hw_params(), snd_pcm_prepare(), snd_pcm_recover(),
snd_rawmidi_open(), snd_seq_connect_from() and snd_seq_connect_to() may
take the time.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
In previous commit, card registration is processed under situation
with few bus reset. There's no need to add a workaround of transaction
re-initialization at timeout.
This commit purges the re-initialization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Some models based on ASIC for Dice II series (STD, CP) change their
hardware configurations after appearing on IEEE 1394 bus. This is due to
interactions of boot loader (RedBoot), firmwares (eCos) and vendor's
configurations. This causes current ALSA dice driver to get wrong
information about the hardware's capability because its probe function
runs just after detecting unit of the model.
As long as I investigated, it takes a bit time (less than 1 second) to load
the firmware after bootstrap. Just after loaded, the driver can get
information about the unit. Then the hardware is initialized according to
vendor's configurations. After, the got information becomes wrong.
Between bootstrap, firmware loading and post configuration, some bus resets
are observed.
This commit offloads most processing of probe function into workqueue and
schedules the workqueue after successive bus resets. This has an effect to
get correct hardware information and avoid involvement to bus reset storm.
For code simplicity, this change effects all of Dice-based models, i.e.
Dice II, Dice Jr., Dice Mini and Dice III.
I use a loose strategy to manage a race condition between the work and the
bus reset. This is due to a specification of dice transaction. When bus
reset occurs, registered address for the transaction is cleared. Drivers
must re-register their own address again. While, this operation is required
for the work because the work includes to wait for the transaction. This
commit uses no lock primitives for the race condition. Instead, checking
'registered' member of 'struct snd_dice' avoid executing the work again.
If sound card is not registered, the work can be scheduled again by bus
reset handler.
When .remove callback is executed, the sound card is going to be released.
The work should not be pending or executed in the releasing. This commit
uses cancel_delayed_work_sync() in .remove callback and wait till the
pending work finished. After .remove callback, .update callback is not
executed, therefore no works are scheduled again.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Before allocating an instance of sound card, ALSA dice driver checks
chip_ID_hi in Bus information block of Config ROM, then also checks
subaddresses. The former operation reads cache of Config ROM in Linux
FireWire subsystem, while the latter operation sends read transaction.
The latter can be merged into initialization of transaction system.
This commit splits these two operations to reduce needless transactions
in probe processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
|
|
|
|
|
|
commit 3c83ac23253c ("ASoC: hdac_hdmi: check error return") fixes
the static checker warning reported by Dan Carpenter:
sound/soc/codecs/hdac_hdmi.c:416 hdac_hdmi_parse_and_map_nid()
warn: unsigned 'hdac->num_nodes' is never less than zero.
But it doesn't fix the issue completely.
It's also a failure if no sub nodes found for an afg node. So modify
the return condition appropriately.
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>
|
|
By default the device latches data on the falling edge of the
BCLK in DSP mode, whereas the expectation for normal BCLK is to
latch on the rising edge. This updates the driver to invert the
BCLK configuration for DSP mode, to align with expected behaviour.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
|
