It seems that the code duplication was added at a merge operation.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patches for stubbing out the generic clock API still haven't been
applied so we need to either add ifdefs here or add a dependency until
someone decides to actually apply them.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This gets the registers set up as early as possible, mainly useful for the
GPIOs to ensure that they're in the correct mode.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is only one caller and this allows us to cleanly leave the CODEC
with the internal LDO powered down which is the default state we're
looking for and means that we can robustly disable the register cache
only when we either disable the LDO or power down the external
regulators.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Get it done as early as possible, it's neater and minimises the time the
pins aren't configured as requested.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Even though the WM8904 is able to use idle_bias_off during both probe
and resume we were needlessly leaving the device in standby mode. Instead
power the device down as soon as we've confirmed that we can talk to it
and don't manage the bias level at all over suspend and resume, the core
will take us down to our minimum power level.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Merge tag 'v3.5-rc4' into for-3.6
Linux 3.5-rc4 contains some bug fixes which overlap with new features.
For convenience, add "inv-dmic" model string for enabling the inverted
internal mic workaround to possible Realtek codecs, so far,
ALC882-variants, ALC262, ALC268, ALC269-variants, and ALC662-variants.
Also, the model strings for hardware inv-dmic workarounds,
"alc269-dmic" and "alc271-dmic", are added for ALC269(VA) and ALC271
codecs as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some laptops are equipped with ForteMedia digital mics that give the
differential input. With such devices, summing stereo streams into a
mono (like PulseAudio does) results in almost silence.
This patch provides a workaround for this bug by adding a new mixer
switch to turn on/off the right channel of digital mic, just like a
similar fix for Conexant codecs.
When the new switch "Inverted Internal Mic Capture Switch" is off and
the current input source is the digital mic, the right channel of the
recording stream is muted. When another input source is selected, the
right channel is restored.
Tested-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM5102 is a highly-integrated low-power audio system for smartphones,
tablets and other portable audio devices based on the Arizona platform.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Wolfson Arizona series of audio hub CODECs can share a large amount
of their driver code as the result of a common register map. This patch
adds some of this core support, providing a basis for the initial WM5102
audio driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We may as well print as many errors as we can in one go rather than
requiring developers to iterate through all their typos.
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>
After cancel_delayed_work_sync returns, the power down work either never
started (power_on == 1) or finished (power_on == 0). In the former case
there is no need to power up again.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo Ideapad S205 has an internal mic where the right channel
is phase inverted.
BugLink: https://bugs.launchpad.net/bugs/884652
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec ALC272X is a special codec for some Dell machines, and its
detection got broken in the recent kernel because SSID check (required
by ALC272X check) was moved to the later point. Now we need to move
this codec check to the right place, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the recent code, the value shown there is a tag number, and it's no
longer same as the pin nid. Correct the message to avoid confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VIA codecs seem not returning the event tag in the unsolicited events,
thus the current code relying on the tag value doesn't work.
Since simple_hdmi stuff has only a single pin, we can use simply
snd_hda_jack_set_dirty_all() to activate the pin-detection
independently from the tag value.
Tested-by: Annie Liu <AnnieLiu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The simple_hdmi stuff is designed only for a single pin and a single
converter (thus a single PCM stream), and no need for loops.
Let's flatten the code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The codes to initialize work struct or create a proc interface should
be called only once and never although it's called many times through
the init callback. Move that stuff into patch_generic_hdmi() so that
it's called only once.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This driver enables i2s mode support on blackfin bf6xx platform.
We reuse bf5xx-i2s-pcm.c as its i2s pcm driver because it's the same
for both bf5xx and bf6xx soc.
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The SPORT(Serial Port) module on bf6xx soc has a totally different ip
comparing to bf5xx soc. An individual SPORT module consists of two
independently configurable SPORT halves with identical functionality.
Each SPORT half can be configured for either transmitter or receiver.
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes the regression introduced by the commit d0b1252d for
refactoring simple_hdmi*(). The pin NID wasn't assigned correctly.
Reported-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes the last usage of linux/ac97_codec.h by
re-defining used AC97 registers.
We can't use sound/ac97_codec.h here, since it is an OSS driver.
Cc: Ralf Baechle <ralf@linux-mips.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Ezequiel Garcia <elezegarcia@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
add LTC (linear timecode) read function via proc interface to the pcxhr driver
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
fix a counter wrap to avoid resynchronization of stream positions every several
minutes. The resynchronization may create stream position jitter
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This webcam works mostly ok, exept with skype.
Skype sends lots of ctrl messages to dynamically ajust
record level. If for some reasons it pokes some error
every thing goes broken:
- first pulseaudio blocks sound for all apps
- then video is reseted
- then skype freez
dmesg has lots of messages like:
cannot set freq 16000 to ep 0x86"
Setting ignore_ctl_error=1 fixes this problem.
Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
if this cam is pluged in, pulse audio can't initiate capture
device.
dmesg has lots of messages like:
"cannot set freq 16000 to ep 0x86"
Setting ignore_ctl_error=1 fixes this problem.
Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch uses config() function to configure DMA
transmit options.
Signed-off-by: Boojin Kim <boojin.kim@samsung.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
Set the new capability flags in G_TUNER and return ENODATA if no channels
were found.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Acked-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
MED3931 card did not work (failed with "OPTI chip not found") because
snd-opti9xx gets mc_indir_index from pnp by adding 2 to the pnp-reported port.
It probably works for some cards but not for this one. Datasheet says that
the port is always at 0xe?e so just force the lowest nibble to be 0xe.
Also this card powers up with (ugly) 3D sound enabled. As there's no mixer
control for this, just disable it.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
This addresses an issue encountered when a pcm is opened while
transitioning to low power state (codec->power_on == 1 &&
codec->power_transition == -1). Add snd_pcm_power_up_d3wait to
hda_codec. This function is used to power up from azx_open as opposed
to snd_hda_power_up used from codec_exec_verb. When powering up from
azx_open, wait for pending power downs to complete, avoiding the power
up continuing in parallel with the power down on the work queue.
The specific issue seen was with the CS4210 codec, it powers off the ADC
and DAC nid in its suspend handler. If it is re-opened before the
~100ms power down process completes, the ADC and DAC nid are initialized
while powered down and audio is lost until another suspend/resume cycle.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a bogus sanity check for sync pipe in pcm.c. This flaw was
introduced during the streaming logic refactorization.
While at it, improve the error messages that are generated in such cases.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: <ben@b1c1l1.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All driver specific and fairly small. The pxa-ssp changes are larger
than I'd like but they're build failures and are pretty clear to
inspection.
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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: Updates for 3.5
All driver specific and fairly small. The pxa-ssp changes are larger
than I'd like but they're build failures and are pretty clear to
inspection.
There is a missing mutex_unlock() here. The cleanup path also has more
debug output.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add machine-driver for ST-Ericsson U8500 platform, including
support for the AB8500-codec.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ATI and Nvidia HDMI codecs have also the pin-detection capability,
so let's enable the jack-detecion for them, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The code refactoring using the same helper functions for sharing the
codes among ATI, Nvidia and VIA simple_hdmi* stuff. Except for that
spec->pcm_playback is no longer pointer, the functionality doesn't
change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the hotplug unsol event handling to simple_hdmi*().
It works on VIA VX900. If AMD or Nvidia chips support the
pin-detection similarly, it can be added easily, too.
Reported-by: Annie Liu <annieliu@viatech.com.cn>
Tested-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch activates the COMBO position_fix for recent Intel client chipsets.
COMBO mode is the recommended setting for Intel chipsets and eliminates HD
audio warnings in dmesg. This patch has been tested on Lynx Point, Panther
Point, and Cougar Pont.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PulseAudio sometimes have difficulties knowing that there is a
"Speaker" or "Internal Mic", if they have no individual volume
controls or selectors. As a result, only e g "Headphone" might
be created for a laptop, but no "Speaker".
To help out, create phantom jacks (that are always present,
at least for now) for "Speaker", "Internal Mic" etc, in case we
detect them.
The naming convention is e g "Speaker Phantom Jack".
In order not to pollute the /dev/input namespace with even more
devices, these are added to the kcontrols only, not the input devices.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the split to the auto-parser helper functions, the actual call
of init verbs was lost.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43366
Signed-off-by: Takashi Iwai <tiwai@suse.de>
soc_probe_dai_link() currently inter-mixes the probing of CODECs,
platforms, and DAIs. This can lead to problems such as a CODEC's DAI
being probed before the CODEC, if that DAI is used as the CPU-side of
a DAI link without any other of the CODEC's DAIs having been used as
the CODEC-side of any DAI link that was probed earlier.
To solve this, split soc_probe_dai_link() into soc_probe_link_components()
and soc_probe_link_dais(). The former is used to probe all CODECs and
platforms used by a card first, and then the latter is used to probe all
the DAIs and links later.
A similar change is made to soc_remove_dai_links().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This change simply factors out part of soc_remove_dai_link() into a
standalone function. This makes platform and CODEC removal much more
similar at the call-sites.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When a standalone CPU DAI (one not part of a CODEC) is probed, widgets
are created for it. Add a call to snd_soc_dapm_free() in order to clean
these up when the CPU DAI is removed.
In order for snd_soc_dapm_free() to work, the CPU DAI's DAPM context's
list member must be initialized, since snd_soc_dapm_free() removes that
from the list it's part of. Add it to the card's list of DAPM contexts.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the CPU-side of a DAI link is a CODEC rather than a standalone DAI,
the codec initialization will call try_module_get() and create the DAI
widgets. Ensure that this isn't duplicated when the CPU DAI itself is
probed, if the CPU DAI is part of a CODEC.
Note that this is not a complete fix on its own, since there's no
guarantee that the CODEC itself will be initialized - currently that only
happens if the CODEC is also used as the CODEC-side of a DAI link, and
that initialization may happen before or after the DAIs within the CODEC
are initialized. However, such a scenario doesn't necessarily currently
work, and I don't think this change alone makes it any worse. This is
fixed in a couple patches time.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
add ttc-dkb machine support for pxa910. It uses 88pm8607 as codec
dai, mmp-pcm as platform and pxa-ssp as cpu dai.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds Alsa audio platform driver for mmp brownstone machine
Signed-off-by: Zhangfei Gao <zhangfei.gao@marvell.com>
Signed-off-by: Leo Yan <leoy@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The SSPA is a configurable multi-channel audio serial (TDM) interface.
It's configurable at runtime to support up to 128 channels and the
number of bits per sample: 8, 12, 16, 20, 24 and 32 bits. It also
support stereo format: I2S, left-justified or right-justified.
Signed-off-by: Zhangfei Gao <zhangfei.gao@marvell.com>
Signed-off-by: Leo Yan <leoy@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
mmp-pcm handle audio dma based on soc-dmaengine
Support mmp and pxa910
Signed-off-by: Zhangfei Gao <zhangfei.gao@marvell.com>
Signed-off-by: Leo Yan <leoy@marvell.com>
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a leak if snd_soc_register_codec fails.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
GPLv2 -> GPL v2
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Acked-by: Linus Walleij <linus.walleij@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use clk_prepare/clk_unprepare as required by the generic clk framework.
Signed-off-by: Prashant Gaikwad <pgaikwad@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pin 0x1b was connected to the front panel connector, which according to
the HDA standard should contain a mic and a headphone. In this case,
the headphone was listed as "line out" by BIOS.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/993162
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
as the start can be called after stop again, we need to reset state
Signed-off-by: Namarta Kohli <namartax.kohli@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
during pause the core should maintain the status-quo on the device and pointers
and not wake up. If app needs it should call DROP explcitly.
Signed-off-by: Namarta Kohli <namartax.kohli@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If mic detect is left enabled and the WM8903 detects a status change,
the WM8903 driver will make a callback against the free()d jack, which
will cause a crash.
This problem can be triggered by fully initializing an audio card, then
removing and re-inserting the machine driver module.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also convert to use update_bits() while we're at it. No great need to do
this, it's just a bit neater to do as much as possible in the I2C probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
There's no reason to defer requesting of the interrupt until the CODEC
probe and doing so results in more work if we hit an error as we'll have
registered the CODEC with the core. It's neater to acquire as many of the
resources we'll need as we can in the bus probe function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
There's no urgent need for the interrupt handler to use the ASoC I/O
functions and it'll support a further move in where we request the
interrupt so call the regmap APIs directly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
Ensure that the device pins are configured as soon as possible by moving
the pin configration (including MICBIAS) into the I2C probe() function.
This had been done in the CODEC probe() function when we were relying on
the ASoC register I/O code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Stephen Warren <swarren@wwwdotorg.org>
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>
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>
My EU/US 5757 cannot tune below approx 86 Mhz, that is below that it
does not even generate the standard not tuned to anything radio noise anymore,
so clearly the 5757 cannot tune to the Japanese frequencies. This patch
assumes that likewise the 5759 cannot tune to the EU/US frequencies.
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>
The Tegra ASoC machine drivers only depend on Tegra architecture support
to compile, not specific board support. Remove Kconfig dependencies on
any particular board. This is required since Kconfig options for boards
are going away given the migration to device tree.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCI ID entries of Creative SoundCore3D HD-audio controllers should
be before the wildcard for vendor = Creative.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for Dialog DA732x audio codecs.
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>
They aren't marked as readable and the feature is never used so they'll
never get referenced.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we don't have control over the LDO but do have control over the other
regulators then we may end up trying to write to a powered off device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're using demv_regulator_bulk_get() so don't need to manually free and
this is in the CODEC driver not the I2C driver anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow mixer controls to be provided clearly in a table, to avoid
quantity of error checking at each use.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is patch supporting the CODECes of HD-Audio function of VIA GFX
cards which support HDMI.
For CODECes 0x9f80/0x9f81, which belong to VX900, since the hardware
is not fully compliant to HD-Audio 1.3, simple_i*() is adopted
temporarily.
Signed-off-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is patch supporting HD-Audio function of VIA GFX cards which
support HDMI.
Those are integrated graphics of chipsets VX900 and VX11 separately.
Signed-off-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add codec-driver for ST-Ericsson AB8500 mixed-signal ASIC.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We should be using the regmap API consistently for all the cache only
configuration and we should be going cache only before we power down
the supplies.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We no longer have a flat ASoC cache so can't peer directly into the array
any more but should instead use the register I/O functions to update the
cache.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org (v3.4)
pxa-ssp.c uses API like cpu_is_pxa3xx(), cpu_is_pxa2xx(), which is
defined under arch-pxa architecture, and drivers under mach-mmp
can't find it. so just use ssp->type to replace that API.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Acked-by: Haojian Zhuang <haojian.zhuang@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When VGA_SWITCHEROO support is enabled hda_intel initialises the HDMI
audio device on the current VGA device. When it's not enabled it only
initialises the HDMI device on the default VGA adaptor, this means
secondary cards get no audio support which is very unhelpful for
multi-seat!
With this patch, when SUPPORT_VGA_SWITCHEROO is disabled hda_intel
initialises all HDMI audio devices, not just the default VGA.
[minor optimizations by tiwai]
Signed-off-by: Steven Newbury <steve@snewbury.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When VGA-switcheroo is built in but unused on systems with multiple
graphics cards, the initializations of non-default graphics cards are
skipped and never enabled (because the switcheroo is activated only
when the controller supports). The current behavior is for avoiding
the system lockup by accessing the disabled GPU, but due to the recent
change in VGA-switcheroo, it determines the state simply by checking
with the default VGA device. This is the culprit.
Now with the new vga_switcheroo_get_client_state(), we can know the
initial state of the bound GPU, thus can determine the initial audio
client state more correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In 3.5 kernel, the endpoint is assigned dynamically for the
substreams, but the PCM assignment still checks the presence of the
endpoint pointer. This ended up in duplicated PCM substream creations
at probing time, resulting in kernel warnings like:
WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6()
Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2
Call Trace:
[<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c
[<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48
[<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b
[<ffffffff811292f0>] proc_register+0x169/0x1a6
[<ffffffff8112962e>] create_proc_entry+0x74/0x8c
[<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd]
[<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm]
[<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio]
[<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio]
[<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb
[<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio]
[<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio]
[<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio]
.....
This patch fixes the regression by checking the fixed endpoint number
for each substream instead of the endpoint pointer.
Reported-and-tested-by: Jamie Heilman <jamie@audible.transient.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
codec may reject power state transition requests(reporting PS-ERROR set),
in that case we re-issue a power state setting and check error bit again.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows the module to automatically load when instantiated from
device tree.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Stop open-coding the caching of the ctrl registers; instead, use
regmap_update_bits() to update parts of the register from different
places. The removal of the open-coded cache will allow controls to be
created which touch registers, which will be necessary if any of these
modules are converted to CODECs.
Get rid of tegra*_read/write; just call regmap_read/write directly.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the actual device of the I2S or SPDIF controller reporting the
problem. If a future change converts these controllers to be CODECs, then
there may be no pcm associated with the substream, so this change avoids
a crash.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
rtd->codec_dai->codec can be used instead.
This is a slight step along the way to not needing the rtd->codec field
any more.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Define the DAI format statically in the dai_link, rather than executing
code to set it each time the hw params are set.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is certainly required if the I2S and SPDIF controllers are converted
to be CODECs, and is probably good practice irrespective.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having varying timeouts depending on the transition always
use a 4s timeout. This provides better diagnostics for clocking errors
and ensures compatibility with current calibration firmwares.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We should set "isabelle_regmap" before using it. GCC complains.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We should only add source widgets to the input list.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure that the dpcm_get_be() only returns BE DAI links.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_hda_param_read() return value -1 means error, others are responses
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
if EPSS supported, transition from D3 state to D0 state in less
than 10ms
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add function to check whether power states supported by specific
codec node.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
add more power states information:
- reset status
- clock stop ok
- power states error
Output like:
Power: setting=D0, actual=D0, Error, Clock-stop-OK, Setting-reset
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove "Master Mono Playback Volume" and "Master Mono Playback Switch"
of ac97 mixer since au88x0 does no use "Master Mono Pin" of AC97 codec
even au88x0 support mono playback
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Volume updates may not be acted upon if there is no clock applied when
the volume update is written. Ensure this doesn't happen by writing out
registers with volume updates after we enable each of the clocks.
There are more registers updated than before as previously we were
relying on wm_hubs to set those for controls it manages.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Ensure that all the actions get taken at appropriate times by calling the
_PRE and _POST events for the aifNclk_ev functions explicitly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
The core fills in some blanks which makes it annoying to do the right thing
and constify the calls in the core.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This reverts commit 014e5b5670 since
PowerPC doesn't use clkdev and hasn't implemented devm_clk_get() itself.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With devm_ APIs regmap_exit() not needed, so remove regmap_exit().
Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande@ti.com>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: Support TI Isabelle Audio driver
The Isabelle Audio IC is a complete low power high fidelity CODEC with integrated
ADCs, DACs, decimation and interpolation filters, PLL, and power providers. This
device supports 2 analog and 2 digital microphone channels, a mono earpiece driver,
stereo class G headphone drivers with ultra low power and best SNR in the industry,
stereo Class D speaker drivers, and 2 high performance Line drivers.
The below patch is a basic driver code for TI Isabelle audio codec. The
functionalities like headset detection, etc., will be included incrementally
in the up-coming patches.
Signed-off-by: Vishwas A Deshpande <vishwas.a.deshpande@ti.com>
Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
version.h header file is no longer required.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
version.h header file is no longer needed.
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Richard Zhao <richard.zhao@freescale.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure we check the correct path for capture.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.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>
Less error prone and one less line of code in drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
ALSA mixers cannot classify this "Class-D Amplifier Gain" speaker output
gain control as a playback control. Fix this by changing the name as
"Class-D Playback Volume".
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
PIO handler is not good performance, but works on all platform.
So, switch to PIO handler if DMA handler was invalid case.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch used dmaengine helper functions instead of using hand setting.
And reduced local variables
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add platform-driver handling all DMA-activities.
Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.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>
In kernel 3.6, Seaboard will only be supported when booting using device
tree; the board files are being removed. Hence, remove the non-DT support
for Seaboard and derivatives Kaen and Aebl from the audio driver.
Harmony is the only remaining board supported by this driver when not
using DT. This support is currently scheduled for removal in 3.7.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By the time any widget callbacks could be called, if the GPIO ID they
will manipulate is valid, it must have already been requested, or the
card would have failed to probe or initialize. So, testing for GPIO
validity is equivalent to testing whether the GPIO was successfully
requested at this point in the code. Making this change will allow later
patches to remove the gpio_requested variable.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The headphone jack GPIOs are added/initialized in the DAI link's init()
method, and hence in theory may not always have been added before remove()
is called in some unusual cases. In order to prevent calling
snd_soc_jack_free_gpios() if snd_soc_jack_add_gpios() had not been, the
code kept track of the initialization state to avoid the free call when
necessary.
However, it appears that snd_soc_jack_free_gpios() is robust in the face
of being called without snd_soc_jack_add_gpios() first succeeding, so
there is little point manually tracking this information. Hence, remove
the tracking code. All other machine drivers already operate this way.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that deferred probe exists, we can parse device tree and request
GPIOs from probe(), rather than deferring this to the DAI link's init().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The headphone jack GPIOs are added/initialized in the DAI link's init()
method, and hence in theory may not always have been added before remove()
is called in some unusual cases. In order to prevent calling
snd_soc_jack_free_gpios() if snd_soc_jack_add_gpios() had not been, the
code kept track of the initialization state to avoid the free call when
necessary.
However, it appears that snd_soc_jack_free_gpios() is robust in the face
of being called without snd_soc_jack_add_gpios() first succeeding, so
there is little point manually tracking this information. Hence, remove
the tracking code. Almost all other machine drivers already operate this
way.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
By using this function, the driver no longer needs to explicitly free
the GPIOs. Hence, we can also remove the flags we use to track whether
we allocated these GPIOs.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that deferred probe exists, we can parse device tree and request
GPIOs from probe(), rather than deferring this to the DAI link's init().
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows the GPIOs to be available as soon as the I2C device has
probed, which in turn enables machine drivers to request the GPIOs in
their probe(), rather than deferring this to their ASoC machine init
function, i.e. after the whole sound card has been constructed, and
hence the WM8903 codec is available.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Small driver-specific updates.
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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: A few bug fixes for the merge window
Small driver-specific updates.
On PCI RME MADI cards, the PLL register does not contain the proper
value, so the calculated system_sample_rate is wrong. In this case, we
simply return the cached rate from struct hdspm.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The array of sample rates is reallocated every time when opening
the PCM device, but was freed only once when unplugging the device.
Reported-by: "Alexander E. Patrakov" <patrakov@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
FSI DMAEngine has to be stopped certainly at the start/stop time.
Without this patch, it will include noise on playback.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FSI driver is using dma_sync_single_xxx(),
but the dma area was not correct.
This patch fix it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DMA stream handler didn't care about master clock.
This patch fixes it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The new clock subsystem was merged in linux-3.4 without any users, this
now moves the first three platforms over to it: imx, mxs and spear.
The series also contains the changes for the clock subsystem itself,
since Mike preferred to have it together with the platforms that require
these changes, in order to avoid interdependencies and conflicts.
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Merge tag 'clock' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull arm-soc clock driver changes from Olof Johansson:
"The new clock subsystem was merged in linux-3.4 without any users,
this now moves the first three platforms over to it: imx, mxs and
spear.
The series also contains the changes for the clock subsystem itself,
since Mike preferred to have it together with the platforms that
require these changes, in order to avoid interdependencies and
conflicts."
Fix up trivial conflicts in arch/arm/mach-kirkwood/common.c (code
removed in one branch, added OF support in another) and
drivers/dma/imx-sdma.c (independent changes next to each other).
* tag 'clock' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (97 commits)
clk: Fix CLK_SET_RATE_GATE flag validation in clk_set_rate().
clk: Provide dummy clk_unregister()
SPEAr: Update defconfigs
SPEAr: Add SMI NOR partition info in dts files
SPEAr: Switch to common clock framework
SPEAr: Call clk_prepare() before calling clk_enable
SPEAr: clk: Add General Purpose Timer Synthesizer clock
SPEAr: clk: Add Fractional Synthesizer clock
SPEAr: clk: Add Auxiliary Synthesizer clock
SPEAr: clk: Add VCO-PLL Synthesizer clock
SPEAr: Add DT bindings for SPEAr's timer
ARM i.MX: remove now unused clock files
ARM: i.MX6: implement clocks using common clock framework
ARM i.MX35: implement clocks using common clock framework
ARM i.MX5: implement clocks using common clock framework
ARM: Kirkwood: Replace clock gating
ARM: Orion: Audio: Add clk/clkdev support
ARM: Orion: PCIE: Add support for clk
ARM: Orion: XOR: Add support for clk
ARM: Orion: CESA: Add support for clk
...
This is the second updates for 3.5-rc1. It's mainly for OMAP4 HDMI
updates and the device tree updates for OMAP, in addition to a couple
of PCM accuray improvement and Realtek ALC269VD codec support.
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Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound update from Takashi Iwai:
"This is the second updates for 3.5-rc1. It's mainly for OMAP4 HDMI
updates and the device tree updates for OMAP, in addition to a couple
of PCM accuray improvement and Realtek ALC269VD codec support."
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits)
ALSA: hda/realtek - Add new codec support for ALC269VD
ALSA: core: group read of pointer, tstamp and jiffies
ASoC: OMAP: HDMI: Rename sound card source file
ASoC: OMAP: HDMI: Make sound card naming more generic
ASoC: OMAP: HDMI: Make build config options more generic
ASoC: OMAP: HDMI: Expand capabilities of the HDMI DAI
ASoC: OMAP: HDMI: Improve how the display state is verified
ASoC: OMAP: HDMI: Expand configuration of hw_params
ASoC: OMAP: HDMI: Use the DSS audio interface
ASoC: OMAP: HDMI: Create a structure for private data of the CPU DAI
ASoC: OMAP: HDMI: Change error values in HDMI CPU DAI
ASoC: OMAP: HDMI: Update the platform device names
ASoC: omap-abe-twl6040: Introduce driver data for runtime parameters
ASoC: omap-abe-twl6040: Move Digital Mic widget into dapm table
ASoC: omap-abe-twl6040: Keep only one snd_soc_dai_link structure
ASoC: omap-dmic: Add device tree bindings
ASoC: omap-mcpdm: Add device tree bindings
ASoC: omap-mcbsp: buffer size constraint only applies to playback stream
ASoC: omap-mcbsp: Use the common interrupt line if supported by the SoC
ASoC: omap-mcbsp: Remove unused FRAME dma_op_mode
...
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Merge tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull VGA-switcheroo audio client support for HD-audio from Takashi Iwai.
This depended on the recent drm pull.
* tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - unlock on error in azx_interrupt()
ALSA: hda - Support VGA-switcheroo
ALSA: hda - Export snd_hda_lock_devices()
ALSA: hda - Check the dead HDMI audio controller by vga-switcheroo
- Fix mismatch between DMA mapping direction (was wrong) and DMA synchronization
direction (was correct) of isochronous reception buffers of userspace drivers
if vma-mapped for R/W access. For example, libdc1394 was affected.
- more consistent retry stategy in device discovery/ rediscovery, and improved
failure diagnostics
- various small cleanups, e.g. use SCSI layer's DMA mapping API in firewire-sbp2
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Merge tag 'firewire-updates' of git://git.kernel.org/pub/scm/linux/kernel/git/ieee1394/linux1394
Pull IEEE 1394 (FireWire) subsystem updates from Stefan Richter:
- Fix mismatch between DMA mapping direction (was wrong) and DMA
synchronization direction (was correct) of isochronous reception
buffers of userspace drivers if vma-mapped for R/W access. For
example, libdc1394 was affected.
- more consistent retry stategy in device discovery/ rediscovery, and
improved failure diagnostics
- various small cleanups, e.g. use SCSI layer's DMA mapping API in
firewire-sbp2
* tag 'firewire-updates' of git://git.kernel.org/pub/scm/linux/kernel/git/ieee1394/linux1394:
firewire: sbp2: document the absence of alignment requirements
firewire: sbp2: remove superfluous blk_queue_max_segment_size() call
firewire: sbp2: use scsi_dma_(un)map
firewire: sbp2: give correct DMA device to scsi framework
firewire: core: fw_device_refresh(): clean up error handling
firewire: core: log config rom reading errors
firewire: core: log error in case of failed bus manager lock
firewire: move rcode_string() to core
firewire: core: improve reread_config_rom() interface
firewire: core: wait for inaccessible devices after bus reset
firewire: ohci: omit spinlock IRQ flags where possible
firewire: ohci: correct signedness of a local variable
firewire: core: fix DMA mapping direction
firewire: use module_pci_driver
Pull media updates from Mauro Carvalho Chehab:
- some V4L2 API updates needed by embedded devices
- DVB API extensions for ATSC-MH delivery system, used in US for mobile
TV
- new tuners for fc0011/0012/0013 and tua9001
- a new dvb driver for af9033/9035
- a new ATSC-MH frontend (lg2160)
- new remote controller keymaps
- Removal of a few legacy webcam driver that got replaced by gspca on
several kernel versions ago
- a new driver for Exynos 4/5 webcams(s5pp fimc-lite)
- a new webcam sensor driver (smiapp)
- a new video input driver for embedded (sta2x1xx)
- several improvements, fixes, cleanups, etc inside the drivers.
Manually fix up conflicts due to err() -> dev_err() conversion in
drivers/staging/media/easycap/easycap_main.c
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (484 commits)
[media] saa7134-cards: Remove a PCI entry added by mistake
[media] radio-sf16fmi: add support for SF16-FMD
[media] rc-loopback: remove duplicate line
[media] patch for Asus My Cinema PS3-100 (1043:48cd)
[media] au0828: Move the Kconfig knob under V4L_USB_DRIVERS
[media] em28xx: simple comment fix
[media] [resend] radio-sf16fmr2: add PnP support for SF16-FMD2
[media] smiapp: Use v4l2_ctrl_new_int_menu() instead of v4l2_ctrl_new_custom()
[media] smiapp: Add support for 8-bit uncompressed formats
[media] smiapp: Allow generic quirk registers
[media] smiapp: Use non-binning limits if the binning limit is zero
[media] smiapp: Initialise rval in smiapp_read_nvm()
[media] smiapp: Round minimum pre_pll up rather than down in ip_clk_freq check
[media] smiapp: Use 8-bit reads only before identifying the sensor
[media] smiapp: Quirk for sensors that only do 8-bit reads
[media] smiapp: Pass struct sensor to register writing commands instead of i2c_client
[media] smiapp: Allow using external clock from the clock framework
[media] zl10353: change .read_snr() to report SNR as a 0.1 dB
[media] media: add support to gspca/pac7302.c for 093a:2627 (Genius FaceCam 300)
[media] m88rs2000 - only flip bit 2 on reg 0x70 on 16th try
...
This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten
for the better support of "implicit feedback". If anything about USB
got broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up immediately
at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital links
between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid the accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal routing
through components with tight sequencing and formatting constraints
within their internal paths or where there are multiple components
connected with CPU managed DMA controllers inside the SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like digital
basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124, Texas
Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be
sent slightly later as partly depending on the changes of DRM.
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Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten for
the better support of "implicit feedback". If anything about USB got
broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up
immediately at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital
links between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal
routing through components with tight sequencing and formatting
constraints within their internal paths or where there are multiple
components connected with CPU managed DMA controllers inside the
SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like
digital basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
Texas Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be sent
slightly later as partly depending on the changes of DRM."
Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree). Also some
fairly trivial context conflicts in the imx and mcx soc drivers.
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: snd-usb: fix stream info output in /proc
ALSA: pcm - Add proper state checks to snd_pcm_drain()
ALSA: sh: Fix up namespace collision in sh_dac_audio.
ALSA: hda/realtek - Fix unused variable compile warning
ASoC: sh: fsi: enable chip specific data transfer mode
ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
ASoC: sh: fsi: use same format for IN/OUT
ASoC: sh: fsi: add fsi_version() and removed meaningless version check
ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
ASoC: tegra: Add machine driver for WM8753 codec
ALSA: hda - Fix possible races of accesses to connection list array
ASoC: OMAP: HDMI: Introduce codec
ARM: mx31_3ds: Add sound support
ASoC: imx-mc13783 cleanup
mx31moboard: Add sound support
ASoC: mc13783 codec cleanups
ASoC: add imx-mc13783 sound support
ASoC: Add mc13783 codec
mfd: mc13xxx: add codec platform data
ASoC: don't flip master of DT-instantiated DAI links
...
Pull trivial updates from Jiri Kosina:
"As usual, it's mostly typo fixes, redundant code elimination and some
documentation updates."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (57 commits)
edac, mips: don't change code that has been removed in edac/mips tree
xtensa: Change mail addresses of Hannes Weiner and Oskar Schirmer
lib: Change mail address of Oskar Schirmer
net: Change mail address of Oskar Schirmer
arm/m68k: Change mail address of Sebastian Hess
i2c: Change mail address of Oskar Schirmer
net: Fix tcp_build_and_update_options comment in struct tcp_sock
atomic64_32.h: fix parameter naming mismatch
Kconfig: replace "--- help ---" with "---help---"
c2port: fix bogus Kconfig "default no"
edac: Fix spelling errors.
qla1280: Remove redundant NULL check before release_firmware() call
remoteproc: remove redundant NULL check before release_firmware()
qla2xxx: Remove redundant NULL check before release_firmware() call.
aic94xx: Get rid of redundant NULL check before release_firmware() call
tehuti: delete redundant NULL check before release_firmware()
qlogic: get rid of a redundant test for NULL before call to release_firmware()
bna: remove redundant NULL test before release_firmware()
tg3: remove redundant NULL test before release_firmware() call
typhoon: get rid of redundant conditional before all to release_firmware()
...
Group read of hw_ptr, tstamp and jiffies in a sequence
for better correlation. Previous code took timestamp at the
end, which could introduce delays between audio time and
system time.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With this, five platforms are moving to the relatively new pinctrl
subsystem for their pin management, replacing the older soc specific
in-kernel interfaces with common code.
There is quite a bit of net addition of code for each platform being
added to the pinctrl subsystem. but the payback comes later when adding
new boards can be done by only providing new device trees instead.
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Merge tag 'pinctrl' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc
Pull arm soc-specific pinctrl changes from Olof Johansson:
"With this, five platforms are moving to the relatively new pinctrl
subsystem for their pin management, replacing the older soc specific
in-kernel interfaces with common code.
There is quite a bit of net addition of code for each platform being
added to the pinctrl subsystem. But the payback comes later when
adding new boards can be done by only providing new device trees
instead."
Fix up trivial conflicts in arch/arm/mach-ux500/{Makefile,board-mop500.c}
* tag 'pinctrl' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (61 commits)
mtd: nand: gpmi: fix compile error caused by pinctrl call
ARM: PRIMA2: select PINCTRL and PINCTRL_SIRF in Kconfig
ARM: nomadik: enable PINCTRL_NOMADIK where needed
ARM: mxs: enable pinctrl support
video: mxsfb: adopt pinctrl support
ASoC: mxs-saif: adopt pinctrl support
i2c: mxs: adopt pinctrl support
mtd: nand: gpmi: adopt pinctrl support
mmc: mxs-mmc: adopt pinctrl support
serial: mxs-auart: adopt pinctrl support
serial: amba-pl011: adopt pinctrl support
spi/imx: adopt pinctrl support
i2c: imx: adopt pinctrl support
can: flexcan: adopt pinctrl support
net: fec: adopt pinctrl support
ARM: ux500: switch MSP to using pinctrl for pins
ARM: ux500: alter MSP registration to return a device pointer
ARM: ux500: switch to using pinctrl for uart0
ARM: ux500: delete custom pin control system
ARM: ux500: switch over to Nomadik pinctrl driver
...
Rename sound card driver source file to encompass not only OMAP4 but future
OMAP versions that feature HDMI.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Rename all the relevant structures, variables and functions to not
make specific reference to OMAP4. This is to make the driver encompass
future OMAP versions that feature HDMI and not only OMAP4. These
changes are only in naming. There are not functional changes.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Make Kconfig and Makefile more generic to encompass not only
OMAP4 but other OMAP processors featuring HDMI.
Also, relax the dependency list to depend only on any OMAP processor
supporting OMAP2_DSS and OMAP4_DSS_HDMI. As HDMI support for
future OMAP versions is added, the dependency list must change
accordingly.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
According to the HDMI specification, a source is permitted to
transmit L-PCM audio in the following sample rates: 32kHz, 44.1kHz,
48kHz, 88.2kHz, 96kHz, 176.4kHz or 192kHz.
It also supports up to 8 audio channels.
The sink may not necessarily support all these sample rates and
channels. However, as this CPU DAI describes the HDMI source, it
makes sense to include them. The limitation of capabilities as
supported by the sink should be done in the ASoC HDMI codec.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Before starting to play audio, we need to make sure that the
display is active and the current video mode supports audio. instead
of using the overlay manager in the machine driver, we use the DSS audio
interface's audio_supported function. As we already have a pointer to
the correct dssdev, we do not have to look for it every time audio is
to be played. Also, the CPU DAI startup function is called earlier
than the card hw_param function. Hence and we can detect the state of
the display earlier.
While there, add a error message if the constraint cannot be applied.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
squash to improve err
Signed-off-by: Liam Girdwood <lrg@ti.com>
Expand the configuration of the hw_params to include the IEC-60958
channel status word and the CEA-861 audio infoframe. The configuration
of such structures depends on the snd_pcm_hw_params received. A
omap_dss_audio is used to pass the configuration parameters to DSS.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Instead of accessing the HDMI IP directly, the CPU DAI driver takes
advantage of the audio interface provided by the DSS device driver.
The ASoC driver will link the DSS audio functionality with ALSA by
calling the appropriate DSS device driver functions at the relevant
moments. For this, three new DAI operations are added: trigger, prepare
and shutdown operations.
At the moment, it is assumed that only one HDMI display is available
in the system, as it is the case in OMAP4. However, in the future,
one DAI for each HDMI display should be provided.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Create a struct hdmi_priv to store the relevant data of the CPU DAI
driver. As more data is added to the driver, having all the data
in the same location eases its handling. At the moment, only the DMA
configuration parameters are included in the structure.
Also, the required memory is allocated using devm_kzalloc rather than
using a static global variable.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
When getting the needed resources fails, return -ENODEV. This is more
in line with other drivers do and it gives a more descriptive error.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
In order to utilize the new OMAP HDMI codec and the updated name of
the device of the CPU DAI, update the names at the drivers accordingly.
While there, also update the name of the machine driver to be more
generic and encompass more OMAP processors featuring HDMI and not
only OMAP4.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
In preparation to Device Tree support.
With DT booted kernel we can not rely on pdata which used to
hold information needed for the driver at runtime.
Use the card's driver data to hold these informations from now on.
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>
The needed change in routing will be done runtime for the non
twl6040 connected widgets, like the Digital microphone.
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>
There is no need to have two snd_soc_dai_link structure for the two setup
the machine driver supports.
We can just tell core to register only the first link if the DMIC link is
not in use on the device.
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>
Device tree support for OMAP4+ dmic cpu dai driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Device tree support for OMAP4+ McPDM cpu dai driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
In capture stream the buffer size does not need to be constrained to be
bigger then the McBSP FIFO.
In capture the FIFO content is taken out in period length burst, this
enusres that the FIFO is not going to overflow.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
