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This adds db gain information to M-Audio Fast Track Ultra (8R) devices.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename mixer_vol_tlv to snd_usb_mixer_vol_tlv and export it to make
it reuseable in mixer_quirks.c.
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().
[fixed minor checkpatch.pl warnings by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While we need to clean up unused single ended line outputs we don't want
to do this if the outputs are in differential mode.
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>
While reading through sound/soc/codecs/wm8994.c I noticed a fair
amount of trailing whitespace. This patch gets rid of it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Slightly larger than normal - the DAPM fix is a "this should always have
worked" type of thing which is very clear and should have no impact on
systems that don't need it. The WM8994 fix is driver specific but
pretty important for that driver.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: updates for 3.4
Slightly larger than normal - the DAPM fix is a "this should always have
worked" type of thing which is very clear and should have no impact on
systems that don't need it. The WM8994 fix is driver specific but
pretty important for that driver.
This ensures a clean startup of the channels, without this change some
use cases could result in issues in a small proportion of cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Merge tag 'mfd-for-linus-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6
Pull MFD fixes from Samuel Ortiz:
"We have 3 build fixes, a OMAP USB host PHY reset fix and the twl6040
conversion to an i2c driver. The latter may not sound like a fix but
the twl6040 MFD driver won't probe without it, triggering an OMAP4
audio regression."
* tag 'mfd-for-linus-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6:
mfd: Fix modular builds of rc5t583 regulator support
mfd: Fix asic3_gpio_to_irq
ARM: OMAP3: USB: Fix the EHCI ULPI PHY reset issue
mfd: Convert twl6040 to i2c driver, and separate it from twl core
mfd : Fix dbx500 compilation error
Drop some struct members and definitions that became obsolete during
the refactorization of the driver.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes for a few regressions of HD-audio, originated partly from 3.4
and partly 3.3.
The fixes for ThinkPad docking-station are for 3.3 kernels, thus they
are based on 3.3 then merged back to 3.4, so that they can be merged
to stable tree cleanly. The non-trivial merge conflicts are because
of this action.
In addition, a copule of trivial fixes for documentation and a long-
statnding issue in the listing of built-in sound driver at boot time.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Fixes for a few regressions of HD-audio, originated partly from 3.4
and partly 3.3.
The fixes for ThinkPad docking-station are for 3.3 kernels, thus they
are based on 3.3 then merged back to 3.4, so that they can be merged
to stable tree cleanly. The non-trivial merge conflicts are because
of this action.
In addition, a couple of trivial fixes for documentation and a long-
standing issue in the listing of built-in sound driver at boot time."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/conexant - Set up the missing docking-station pins
ALSA: hda/conexant - Don't set HP pin-control bit unconditionally
ALSA: workaround: change the timing of alsa_sound_last_init()
ALSA: hda/sigmatel - Fix inverted mute LED
ALSA: hda/realtek - Fix regression on Quanta/Gericom KN1
ALSA: fix core/vmaster.c kernel-doc warning
Some old codecs like ALC880 seem to give a bogus pin capability value 0
occasionally. This breaks the new sanity check in snd_hda_set_pin_ctl().
Skip the sanity checks in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper function to guess the default VREF pin control bits
for mic in. This can be used to set the pin control value safely
matching with the actual pin capabilities.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For setting the pin-control values more safely to match with the
actual pin capability bits, a copule of new helper functions,
snd_hda_set_pin_ctl() and snd_hda_set_pin_ctl_cache(), are
introduced. These are simple replacement of the codec verb write with
AC_VERB_SET_PIN_WIDGET but do more sanity checks and filter out
superfluous pin-control bits if they don't fit with the corresponding
pin capabilities.
Some codecs are screwed up or ignore the command when such a wrong bit
is set. These helpers will avoid such secret errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the reporter, external mic starts to work if the
laptop-dmic model is used. According to BIOS pin config, all
pins are consistent with the alc269vb_laptop_dmic fixup, except
for the external mic, which is not present.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/950490
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge fixes for Thinkpad docking-station regressions for 3.3 kernels
back to 3.4. These were committed in that branch to make the stable
merging easier.
Conflicts:
sound/pci/hda/patch_conexant.c
ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the
docking-station ports, but BIOS doesn't initialize for these pins.
Thus, like the former X200, we need to set up the pins manually in the
driver.
The odd part is that the same PCI SSID is used for X200 and T400, thus
we need to prepare individual fixup tables for cx5051 and others.
Bugzilla entries:
https://bugzilla.redhat.com/show_bug.cgi?id=808559https://bugzilla.redhat.com/show_bug.cgi?id=806217https://bugzilla.redhat.com/show_bug.cgi?id=810697
Reported-by: Josh Boyer <jwboyer@redhat.com>
Reported-by: Jens Taprogge <jens.taprogge@taprogge.org>
Tested-by: Jens Taprogge <jens.taprogge@taprogge.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some output pins on Conexant chips have no HP control bit, but the
auto-parser initializes these pins unconditionally with PIN_HP.
Check the pin-capability and avoid the HP bit if not supported.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Springbank module can support a range of sample rates, selected at
runtime via GPIO configuration. Allow these to be configured at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current alsa_sound_last_init() was called as __initcall().
So, on current ALSA, only devices that had been properly
registered at this point were shown.
So, it will show "No soundcards found" if driver requests
probe deferment. it's often misleading.
This patch delays the timing of alsa_sound_last_init()
as workaround.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviwed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While refactoring the mute-LED handling for HP laptops, I messed up
the polarity check in a wrong way. The red (or the mute-LED if any)
should appear in the muted state, corresponding to GPIO on.
Reported-by: Mikko Vinni <mmvinni@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Through the transition to the auto-parser, the support for
Quanta/Gericom KN1 got broken. There are two problems behind it:
- This machine doesn't like the default COEF setup for ALC260 we take
now as default
- BIOS doesn't set the pins correctly at all; especially the machine
uses only the pin 0x0f for both headphone and speaker
This patch adds the fixup as a workaround for these issues.
Reported-and-tested-by: Uros Vampl <mobile.leecher@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Connect the WM1250-EV1 baseband simulator on Littlemill systems up to
the CODEC AIF2 using the new CODEC<->CODEC link support, allowing a wider
range of use cases to be represented.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Leading up to the ->device_prep_slave_sg change in
185ecb5f4fd43911c35956d4cc7d94a1da30417f 'dmaengine: add context
parameter to prep_slave_sg and prep_dma_cyclic' a generic wrapper was
added in place to guard against the API change, though the fsi driver
wasn't updated in the process (presumably its dmaengine support hadn't
been merged yet at the time). This trivially switches over to the new
wrapper and gets it building again.
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix kernel-doc warning in sound/core/vmaster.c:
Warning(sound/core/vmaster.c:429): No description found for parameter 'private_data'
Signed-off-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is nothing audio-specific about the rcode_string() helper, so move
it from snd-firewire-lib into firewire-core to allow other code to use it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de> (fixed sound/firewire/cmp.c)
Since AIF3 shares clock signals with other audio interfaces in order to
ensure it doesn't drive undesirable clocks we need to tristate it. Rather
than forcing the machine driver to do so have the driver do this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch converts multiple if conditions in to single if with "&&"s.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently we increment the number of RTD's per card during the DAI link
bind. This can cause an incorrect RTD count when we cannot find a component
and defer the probe (and hence perform the DAI link bind for the card again).
Fix the count so that it is cleared before every card registration
and bind attempt.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current DA7210 driver does support PLL mode fully. It uses fixed
value of input master clock and PLL mode is enabled and disabled based
on the sampling frequency being used for playback or recording. It also
doesn't support Sample Rate Measurement feature of DA7210 hardware.
This patch adds full support for PLL and SRM. Basically following three
modes of operation are possible for DA7210 hardware,
(1) I2S SLAVE mode with PLL bypassed
(2) I2S SLAVE mode with PLL enabled
(3) I2S Master mode with PLL enabled
This patch adds support for all three modes. Also, in case of SLAVE mode
with PLL, it supports SRM (Sample Rate Measurement) feature of the chip.
Actually this patch was submitted earlier and received some review
comments, but after that the driver got update by other patches. Because
of that, I am considering this as new patch and not versioning it based
of previous patches. This version tries to take care of all review
comments received for earlier submissions.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: David Dajun Chen <dchen@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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
Fix the following build warning:
sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event':
sound/soc/soc-dapm.c:2913: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64'
'%llx' should be used with 'u64' type.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
We should always have a CODEC already there when registering a CODEC DAI
and for CODEC<->CODEC links a dai_link will have two CODECs so it's much
simpler to do things at registration time.
This results in a slight change in the error handling for failed CODEC
DAI registrations but practically speaking these are never supposed to
fail so there shouldn't be much issue. The change is that we don't fail
the overall CODEC registration if the DAI registration fails; this seems
more robust anyway as we may not need to use a given DAI in a particular
system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When two CODEC DAIs are linked directly to each other then if we give the
same master mode settings to both devices things won't work as either
neither will drive or they'll drive against each other. Flip the settings
for the DAI in the CPU slot of the DAI link.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order to allow CODEC<->CODEC links to function we will need to allow
DAPM paths to be created that pass through DAIs rather than only ones
that are source or sunk at the DAI.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This helps us ignore errors in callers if the operation failed due to not
being available as opposed to an error.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Complete the separation of the twl6040 from the twl core since
it is a separate chip, not part of the twl6030 PMIC.
Make the needed Kconfig changes for the depending drivers at the
same time to avoid breaking the kernel build (vibra, ASoC components).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
A few regression fixes for Realtek HD-audio codecs, mainly specific to
some laptop models.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull another round of sound fixes from Takashi Iwai:
"A few regression fixes for Realtek HD-audio codecs, mainly specific to
some laptop models."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace).
ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines
ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940G
ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co
ALSA: hda/realtek - Add a few ALC882 model strings back
The mixer units from the firmware are corrupt, and even where they
are valid they presents mono controls as L and R channels of
stereo.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some interfaces reference endpoints which do not exists. To
accomodate these, do not fail completely, but skip over them.
This allows the Electrix Ebox-44 with earlier firmware to be
detected and used for audio.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We added locking here but there were a couple error paths where we
forgot to drop the lock before returning.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
