Thanks to Troy Kisky <troy.kisky@boundarydevices.com> for noticing.
- DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2
- DSP_B has 0-bit data delay which corresponds to submode 1
- Currently driver sets them opposite so swap the submode setting
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- OMAP McBSP DAI driver claims to support DSP_A format which has 1-bit data
delay but configures link for 0-bit data delay which is in fact DSP_B
- Fix this by changing format from DSP_A to DSP_B
- Fix also TLV320AIC23 codec and OSK5912 machine drivers since the same
error is populated also there
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add check to determine if dinput_mux is set by any of patch_stac*() functions,
otherwise a invalid pointer my be referenced causing gibberish to mixer values.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing __devexit annotation to wm8350_codec_remove():
sound/soc/codecs/wm8350.c:1546: warning: 'wm8350_codec_remove' defined but not used
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sense DaVinci does not support true I2S mode and
we don't have to use the hack, use dsp_b mode instead
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the meaning of SND_SOC_DAIFMT_NB_NF to match that
used in the codec.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DaVinci does not support true I2S or right justified
mode so not all I2S codecs will work with it when the codec is
master. Document this limitation.
Add dsp_a, dsp_b mode options
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Minor, just move a block of code to make next patch clearer.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Just at little cleanup of davinci_i2s_set_dai_fmt
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Document the current polarity choices.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add constants with a value of 0 to show more explicitly
what is being requested.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The capture with 44.1kHz on ca0106 seems to cause loud noises on
later playbacks, which doesn't support 44.1kHz. A simple fix is to
disable 44.1kHz, as the "default" PCM with dsnoop is anyway only with
48kHz.
Reference: Novell bnc#447624
https://bugzilla.novell.com/show_bug.cgi?id=447624
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When no jack detection is available, the pins should be always
turned on since it can't be turned on/off dynamically via unsol
events.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There will be a Oops or frequent underrun messages when playing music with
omap soc driver, this is because a data region is incorretly sized, other data
region will be overwriten when writing to this data region.
Signed-off-by: Stanley Miao <stanley.miao@windriver.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added probe_only module option to hd-audio driver.
This option specifies whether the driver creates and initializes the
codec-parser after probing. When this option is set, the driver skips
the codec parsing and initialization but gives you proc and other
accesses. It's useful to see the initial codec state for debugging.
The default of this value is off, so the default behavior is as same
as before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the line_out has only one DAC and it's unique (i.e. not shared
by other outputs), assign a more reasonable and distinct mixer name
such as "Headphone" or "Speaker".
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current auto-configuration code has several problems especially
for the new IDT codecs, e.g. wrong assignment of pins and DACs or
coupled volume for speaker and headphone.
This patch is a fairly large rewrite of the auto-configuration code.
Some remaks
- mic_switch and line_switch contain NIDs instead of bool
- dac_list isn't fixed for IDT 92HD* codecs now, they are all probed
- extra HP and speakers are stored in extra_dacs[].
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit re-enabled hp_nid setup for IDT92HD73*, but
it's unneeded indeed for Dell laptops that have multiple headphones.
Setting the extra hp_nid results in a non-working "Headpohne" mixer
control. Thus hp_nid should be 0 for these dell models.
Also, the automatic addition of hp_nid should check whether it's
a dual-HP model or not. For dual-HPs, the pins are already checked
by the early workaround.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added "IEC958 PCM Stream" controls for the per-stream IEC958 status
bits. Using this instead of "IEC958 Default" is safer since the status
bits will be recovered to the default states after closing the PCM
stream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the call of snd_ctl_add() by replacing with snd_hda_ctl_add()
so that this mixer element can be tracked for re-configuration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The re-initializations of codec amp and verb caches are missing
at reconfig, which may cause Oops occasionally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the model without the jack-detection for some desktops that
have really no jack-detection. The recent driver caused regressions
regarding the sound output on such machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 07f455f779.
ALSA: hda: removed unneeded hp_nid references
Removed unneeded hp_nid references for 92hd73xx codec family.
This caused the silent output on some Intel desktops due to missing
routing of widget 0x0a and 0x0d.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8350 is an integrated audio and power management subsystem which
provides a single-chip solution for portable audio and multimedia systems.
The integrated audio CODEC provides all the necessary functions for
high-quality stereo recording and playback. Programmable on-chip
amplifiers allow for the direct connection of headphones and microphones
with a minimum of external components. A programmable low-noise bias
voltage is available to feed one or more electret microphones.
Additional audio features include programmable high-pass filter in the
ADC input path.
This driver was originally written by Liam Girdwood with further updates
from me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This should never happen and it's helpful to identify the specific control
that failed when it does happen.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than listing lots of architectures per line in Kconfig and
Makefile, causing merge conflicts all the time, have one per line
in alphabetical order.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace all tasklet_hi_schedule() callers with the normal
tasklet_schedule(). The former often causes troubles with
RT-kernels, and has actually no merit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove codec vendor names from the codec name strings.
The vendor name is already given from the vendor name table, so
displayed doubly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some desktops seems to have no HP/mic jack detection on the front panel,
which results in the silent output in the recent driver, because the
driver mutes the output (to save power) when no plug is detected.
This patch adds a new model that disables the jack-detection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes an inconsistency that became apparent when I
documented the fields of snd_ca0106_details. spi_dac is always
used in a 'boolean' sense, so this cleanup should make no difference.
[Actually, there is one place checking explicitly spi_dac == 1, so
this will change the behavior. But, supposing it's rather a typo,
I apply this clean-up patch -- tiwai]
Signed-off-by: Ben Stanley <Ben.Stanley@exemail.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi wrote an email [1] explaining the fields of snd_ca0106_details,
so I captured the information into the ca0106.h header file.
[1] http://article.gmane.org/gmane.linux.alsa.devel/56783/match=takashi+gpio_type
Signed-off-by: Ben Stanley <Ben.Stanley@exemail.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Include sound/core.h in sound_core.c so that sound_class is declared
before it is defined, avoiding it looking like it should be static.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Netwinder was using gpio_xxx names which could clash with the GPIO
layer. Add a 'nw_' prefix to ensure that these remain separate.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
A special start-up sequence is required to reduce the pop-noise of Class D
amplifier when enable hands-free on TWL4030.
Signed-off-by: Stanley.Miao <stanley.miao@windriver.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixed a compile warning below:
sound/isa/sb/sb8.c: In function ‘snd_sb8_probe’:
sound/isa/sb/sb8.c:104: warning: ‘err’ may be used uninitialized in this function
by setting the return value correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the compile warning regarding the unused function when built
with CONFIG_PM=n:
sound/pci/hda/hda_intel.c:1905: warning: ‘snd_hda_codecs_inuse’ defined but not used
snd_hda_codecs_inuse() is used only in the resume callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the registration of dais in s3c2443-ac97.c.
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_init':
sound/soc/s3c24xx/s3c2443-ac97.c:401: warning: passing argument 1 of 'snd_soc_register_dai' from incompatible pointer type
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_exit':
sound/soc/s3c24xx/s3c2443-ac97.c:407: warning: passing argument 1 of 'snd_soc_unregister_dai' from incompatible pointer type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver now registers the codec and DAI when probed as an I2C device.
Also convert the driver to use a single dynamic allocation to simplify
error handling.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Redo the instantiation of the WM8900 to do most of the initialisation
work when the I2C driver probes rather than when the ASoC device is
instantiated, registering the codec with the ASoC core when done.
Also move all dynamic allocations into a single kmalloc() to simplify
error handling and rename the I2C driver to make output more sensible.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The GPIO stuff for OLPC in cs5535audio_olpc.c is implemented only for
Geode-LX, and enabled only when CONFIG_MGEODE_LX=y. Without this
config option, the driver gets build errors.
This patch adds a workaround to make it dependent on CONFIG_MGEODE_LX.
Ideally, the OLPC-GPIO stuff should be implemented in a way
independent from CPU type selection...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add copyright info to _olpc.c
- minor layout fixes
- make Makefile more concise
- silence a warning
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Always turn off mic bias; the MIC LED should never come on when the
driver is first loaded.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This drops the AD1888 V_REFOUT control, and replaces it with a MIC Bias
Enable control. It also moves the MIC bias enabling into a separate
function.
Signed-off-by: Andres Salomon <dilinger@debian.org>
The OLPC has a privacy light hooked up in series with the microphone's
V_Ref bias. We want to activate the bias while we are capturing audio.
Signed-off-by: Chris Ball <cjb@laptop.org>
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Checking the HPF register is irrelevant; HPF is secondary to the AI mode.
Instead, check for Analog Input mode via GPIO.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously, we had two separate controls; there's no need to have AD1888's
HPF control, so drop it if we're on an OLPC machine. Also, as per Arjun's
request, rename OLPC's Analog Input Switch control to "DC Mode Enable".
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We shouldn't be touching V_REFOUT when we toggle HPF/analog input, so just
drop that code.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix an audible pop described in <http://dev.laptop.org/ticket/977>. Originally
based upon fixes by Mitch Bradley and Chris Ball.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Use basic infrastructure code; geode_gpio* (rather than indexed i/o
EC access), and do an OLPC machine check in olpc_quirk.
[dilinger@debian.org: don't return failure in olpc_quirks if !OLPC]
[dilinger@debian.org: drop the <B2 workarounds; those machines are EOL'd]
Signed-off-by: Jordan Crouse <jordan.crouse@amd.com>
Signed-off-by: Andres Salomon <dilinger@debian.org>
This is a 2nd cut at adding support for OLPC analog input.
Signed-off-by: Jaya Kumar <jayakumar.lkml@gmail.com>
Signed-off-by: Andres Salomon <dilinger@debian.org>
snd_cs5535audio_suspend and snd_cs5535audio_resume are only defined when
CONFIG_PM is set; make that clear in the header file.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As per <http://dev.laptop.org/ticket/1420>, we need to properly turn off
the PCM if we're closing the device in order to save power. This also
causes the MIC led to turn off properly.
Signed-off-by: Jaya Kumar <jayakumar.lkml@gmail.com>
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We'd like to use the High Pass Filter and V_REFOUT bitshift values elsewhere,
so stick them into a ac97_codec.h.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another part of the backporting of Liam's ASoC v2 work. Using this is
more complicated than the other registration types since currently the
codec is instantiated during the probe of the ASoC device so we can't
currently readily wait for the codec to register.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems support both mechanical and electrical jack detection,
allowing them to report that a jack is physically present but does
not have any functioning connections. Add a new jack type for these,
allowing user space to report faulty connections.
Thanks to Guillem Jover for the suggestion.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To avoid confusion the names for the DACs changed:
DACL1 -> DAC Left1
...
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mux switch related texts fits to on line, no need to wrap
them.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SND_SOC_DAPM_OUTPUT definition for carkitL/R was missing.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BUG() should be marked as not returning but for at least some
configurations (including some widely deployed compilers) that's either
not happening or being forgotten by the compiler. Add some extra return
statements to the affected paths.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The return value of pci_enable_device() must be checked even in resume
callback:
sound/pci/ca0106/ca0106_main.c:1779: warning: ignoring return value of ‘pci_enable_device’, declared with attribute warn_unused_result
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add snd_ prefix to avoid the conflict of symbols in omac-mcbsp.c:
sound/soc/omap/omap-mcbsp.c:503: error: static declaration of 'omap_mcbsp_init' follows non-static declaration
arch/arm/plat-omap/include/mach/mcbsp.h:373: error: previous declaration of 'omap_mcbsp_init' was here
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the function name of module init entry for twl4030.c, which
conflicted with the existing hardware init function:
sound/soc/codecs/twl4030.c:1278: error: conflicting types for 'twl4030_init'
sound/soc/codecs/twl4030.c:1187: error: previous definition of 'twl4030_init' was here
Also fixed the section type of init function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This makes use of the support for delayed DAI registration to allow the
WM8900 I2C device to be registered by general platform/architecture code
rather than as part of the ASoC device probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will allow codec drivers to be refactored to allow them to be
registered out of line with the ASoC device registration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the lists of platforms, platform DAIs and cards to check to see that
everything has registered. Since relationships are still specified by
direct references to the structures in the drivers and the drivers all
register everything at modprobe there should be no practical effect yet.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently this is done at module probe time since ASoC ties in codec
device probe to the instantiation of the entire ASoC device. Subsequent
patches will refactor the codec drivers to handle probing separately.
Note that the core does not yet use this information.
AC97 is special since the codec is controlled over the AC97 link but
we want to give the machine driver a chance to set up the system before
trying to instantiate since it may need to do configuration before the
AC97 link will operate
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is done at modprobe time, mirroring current behaviour, except for
mpc5200_psc_i2s where we do registration at the same time as we register
with soc-of-simple. Since the core currently ignores registration this
has no practical impact.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows platform drivers to instantiate independantly of the
overall ASoC card. This API allows drivers to notify the core when
they are registered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Register all platform DAIs with the core. In line with current behaviour
this is done at module probe time rather than when the devices are probed
(since currently that only happens as the entire ASoC card is registered
except for those drivers that currently implement some kind of hotplug).
Since the core currently ignores DAI registration this has no practical
effect.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add API calls to register and unregister DAIs with the core. Currently
these APIs are ineffective. Since multiple DAIs for a given device are
a common case bulk variants are provided.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC v2 allows cards, codecs and platforms to instantiate separately,
with the overall ASoC device only being instantiated once all the
required components have registered. As part of backporting Liam's work
introduce an initial version of the card registration functions. At
present these do nothing active and are internal only, they will be
exposed to machine drivers after further backporting. Adding this now
allows the datastructures used for dynamic card instantiation to be
built up gradually.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a separate gain control for the Headset output already.
Do not reset the gain to 0 dB at power up.
In power-down, there is no need to set the Headset output gain
to power-down mode, since if the CODECPDZ is in powered off this
setting has no effect.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Handsfree outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Carkit outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Headset outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the PreDrive outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds DAPM muxing, routing for the Earpiece output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all four APGA switch to DAPM routing and widgets.
Add user control for DA enable for all APGA as normal
control.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all four DACs to dapm_widgets with power switch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the license and misc comments at the beginning of the code.
Also, use ns_to_ktime() for simplification.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the restoration of the standard PCI configuration registers
in the snd_hda_intel driver to a ->resume_early() callback executed
with interrupts disabled, since doing that with interrupts enabled
may lead to problems in some cases.
This patch addresses the regression from 2.6.26 tracked as
http://bugzilla.kernel.org/show_bug.cgi?id=12121 .
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds basic support for OMAP3 Pandora.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current code overrides the event type on input pins always to
PWR_EVENT. Although this still works (PWR_EVENT and INSERT_EVENT
are handled samely), it'd be better to avoid such overrides.
Also, currently the unsol events are registered even for fixed pins
which will never raise the pin-detection event.
This patch fixes both issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pin-detection function used in patch_sigmatel.c shouldn't be specific
to HP pin because it's used for input pins in general, too.
This patch fixes the detection function, removes the HP check from it
and moves to stac92xx_hp_detect().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is in preparation for the removal of struct snd_soc_device.
The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reduce the command timeout to 0.5sec. Should be enough to allow a
working command interface but removes a RCU stall and slow resume on
some revisions where the AC97 revision detection stalls in resume.
Signed-off-by: Peter Gruber <nokos@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add aic3x_set_headset_detection() function to define the headset
detection mode for tlv32aic3x chips
- added aic3x_button_pressed()
- Read from the real-time registers in aic3x_headset_detected() to query
headset presence without an occured interrupt
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The TWL4030 codec device has two ADCs. Both of them can have
several inputs routed to them, but TRM says that only one source
can be selected for every ADC, even though every source has a
dedicated bit in the registers.
This patch adds input source controls. It modifies default register
values to have no inputs selected and ADCs disabled. When some
input is selected, control handlers enable apropriate input
amplifier and ADC. If a microphone is selected, bias power is
automatically enabled. When some input is deselected, unused
chip parts are disabled.
Microphone and line input recording tested on OMAP3 pandora board.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a sparse warning caused by the lack of a connection with the
prototype for ac97_bus_type.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card. Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Due to the conversion (drop) from 24bit in the DSP to 16bit in AC97,
the maximum capture level on Audigy seems lower than it could be.
This patch adds a workaround to enable the artificial capture boost
switch. When this switch is on, the whole analog capature level is
boost up. However, this results in the lower capture resolution.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The optimal change would be to move the AC97 register definitions into
the AC97 driver, unfortunately, the registers are shared between several
files. Move them into a dedicated regs-ac97.h first.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
All outputs have dedicated gain controls except the
HandsFree output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add Playback volume controls for all four DACs.
All four paths has three levels of volume controls:
Digital Fine gain, Digital Coarse gain, Analog gain.
The controls are named to reflect their connection to the DACs.
Per DAC volume can be performed, if needed:
amixer sset 'DAC1 Analog' 5,10
DACL1 analog gain to 5
DACR1 analog gain to 10
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The digital Capture gain control has a range:
0 to 31 dB in 1 dB steps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently ASoC card initialisation is completed by a function called
snd_soc_register_card(). As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the old-style trigger callback in s3c2443-ac97.c:
sound/soc/s3c24xx/s3c2443-ac97.c:378: warning: initialization from incompatible pointer type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrong shutdown callback type. Also removed the unused variables
there:
sound/soc/pxa/corgi.c: In function 'corgi_shutdown':
sound/soc/pxa/corgi.c:114: warning: unused variable 'codec'
sound/soc/pxa/corgi.c: At top level:
sound/soc/pxa/corgi.c:175: warning: initialization from incompatible pointer type
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 9171e5e6a2.
I can't reproduce the compile warnings any more. The warnings
might be some weird cross-compiling set up.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dependency on SND_SOC is already fulfilled in sound/soc/Kconfig,
thus no more need in Kconfig of each sub directory.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit e669dae614, since it
is incomplete, and clashes with fuller patches and the sparc 32/64
unification effort.
Requested-by: David Miller <davem@davemloft.net>
Acked-by: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The WM9715 is software compatible with the WM9711 and WM9712.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hide annoying uninitialized warnings:
sound/soc/codecs/wm8903.c:382: warning: ‘reg’ may be used uninitialized in this function
sound/soc/codecs/wm8903.c:383: warning: ‘shift’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- compatibility issue : change firmware filenames
the pcxhr driver version <= 1.0.18a does not work
with new firmware > 1.0.17. Keep the old firmware files
and add new firmware files with different names
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The power-state changes in patch_sigmatel.c are accessed via *_cached()
but they shouldn't be really cached. Fixed to the normal write.
Also, stac92hd71xx_suspend and resume are no longer necessary as the
power-state changes are handled properly in the common routine.
Removed these hacks now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp switch of SPDIF outputs have to be cached in the amp cache
instead of codec cache. Otherwise it conflicts with the IEC958
playback switch control in hda_codec.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'merge' of git://git.kernel.org/pub/scm/linux/kernel/git/paulus/powerpc:
powerpc: Fix system calls on Cell entered with XER.SO=1
powerpc/cell: Fix GDB watchpoints, again
powerpc/mpic: Don't reset affinity for secondary MPIC on boot
powerpc/cell/axon-msi: Retry on missing interrupt
powerpc: Fix boot freeze on machine with empty memory node
powerpc: Fix IRQ assignment for some PCIe devices
powerpc/spufs: Fix spinning in spufs_ps_fault on signal
powerpc/mpc832x_rdb: fix swapped ethernet ids
powerpc: Use generic PHY driver for Marvell 88E1111 PHY on GE Fanuc SBC610
powerpc/85xx: L2 cache size wrong in 8572DS dts
powerpc/virtex: Update defconfigs
powerpc/52xx: update defconfigs
xsysace: Fix driver to use resource_size_t instead of unsigned long
powerpc/virtex: fix various format/casting printk mismatches
powerpc/mpc5200: fix bestcomm Kconfig dependencies
powerpc/44x: Fix 460EX/460GT machine check handling
powerpc/40x: Limit allocable DRAM during early mapping
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Check model for Dell 92HD73xx laptops
ALSA: hda - mark Dell studio 1535 quirk
ALSA: hda - No 'Headphone as Line-out' swich without line-outs
ALSA: hda - Fix AFG power management on IDT 92HD* codecs
ALSA: hda - Fix caching of SPDIF status bits
ALSA: hda - Add a quirk for Dell Studio 15
ALSA: hda: Add STAC_DELL_M4_3 quirk
sound/sound_core: Fix sparse warnings
ALSA: hda: STAC_DELL_M6 EAPD
switch to __init for those; unlike powerpc sparc has no hotplug support
for that stuff and their ->probe() tends to call __init functions while
being declared __devinit.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Gateway notebooks have their ID inside codec vendor ID, not at PCI ID. Due to
that, model auto-detection were not possible with the standard seek method.
This is what is found at lspci -vnn:
00:14.2 Audio device [0403]: ATI Technologies Inc SB450 HDA Audio [1002:437b] (rev 01)
Subsystem: ATI Technologies Inc SB450 HDA Audio [1002:437b]
Yet, autodetection is possible, since the codec properly reflects the vendor at
the Subsystem ID:
$ cat /proc/asound/card0/codec#0 |head -4
Codec: SigmaTel STAC9250
Address: 0
Vendor Id: 0x83847634
Subsystem Id: 0x107b0367
This patch adds a new autodetection function that seeks for codec subsystem ID.
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The global functions in hda_codec.c and other core parts are only
for HD-audio codec and controller drivers. When the HD-audio driver
is built in kernel, all stuff have to be statically linked, thus
we don't need any exports.
This patch introduces a conditional macro to do export only
when needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a hook for proc outputs of codec-specific stuff.
Moved realtek-specific coeff output into patch_realtek.c as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moved power_save field initialization inside a proper ifdef
to fix a build error without CONFIG_SND_HDA_POWER_SAVE.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split the monolithc HD-audio driver into several pieces:
- snd-hda-intel HD-audio PCI controller driver; loaded via udev
- snd-hda-codec HD-audio codec bus driver
- snd-hda-codec-* Specific HD-audio codec drivers
When built as modules, snd-hda-codec (that is invoked by snd-hda-intel)
looks up the codec vendor ID and loads the corresponding codec module
automatically via request_module().
When built in a kernel, each codec drivers are statically hooked up
before probing the PCI.
This patch adds appropriate EXPORT_SYMBOL_GPL()'s and the module
information for each driver, and driver-linking codes between
codec-bus and codec drivers.
TODO:
- Avoid EXPORT_SYMBOL*() when built-in kernel
- Restore __devinit appropriately depending on the condition
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function snd_mixart_create creates a link between mgr and card that
allows snd_mixart_free to free card as well. But if snd_mixart_create
fails, then the link has not been created and card has to be freed explicitly.
The semantic match that finds the problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S,S1;
position p1,p2,p3;
expression E,E1;
type T,T1;
expression *ptr != NULL;
@@
(
if ((x@p1 = snd_card_new(...)) == NULL) S
|
x@p1 = snd_card_new(...);
)
... when != snd_card_free(...,(T)x,...)
when != if (...) { <+... snd_card_free(...,(T)x,...) ...+> }
when != true x == NULL || ...
when != x = E
when != E = (T)x
when any
(
if (x == NULL || ...) S1
|
if@p2 (...) {
... when != snd_card_free(...,(T1)x,...)
when != if (...) { <+... snd_card_free(...,(T1)x,...) ...+> }
when != x = E1
when != E1 = (T1)x
(
return \(0\|<+...x...+>\|ptr\);
|
return@p3 ...;
)
}
)
@ script:python @
p1 << r.p1;
p3 << r.p3;
@@
print "* file: %s snd_card_new: %s return: %s" % (p1[0].file,p1[0].line,p3[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reconfiguration of PCM affected all PCM streams on the bus, but
this this should be done rather only for the target codec.
This patch does the following:
- introduce bitmap indicating the PCM device usages on a hda_bus
- refactor the PCM build functions
- fix __devinit prefix in some fucntions
- add a proper ifdef around HDA-reconfig-specific functions
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move power_save option into hda_intel.c, and make a field in hda_bus,
instead of keeping module parameters in separate files.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor clean ups: move snd_hda_codecs_inuse() into hda_intel.c and
make static. Also, make snd_hda_query_supported_pcm() static
as it's used only in hda_codec.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd-usb-caiaq: clean up the control adding code by moving dulpicate code
to a function.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables more routing functions for tlv320aic3x codecs.
It is now possible to
- control the volume of the PGA bypass path for the HPL, HPR, HPLCOM
and HPRCOM outputs individually
- route right line1 input to the left ADC channel
- route left line1 input to the right ADC channel
- route right mic3 input to left DAC channel
- route left mic3 input to right DAC channel
- route left line1 input to right line1 output
- route right line1 input to left line1 output
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the callback mode HRTIMER_CB_IRQSAFE_UNLOCK, the start of the
stream with zero delay doesn't work. Since IRQSAFE mode is removed,
we have to change the pcsp start-up code.
This patch splits the callback function to two parts, the triggering
of the port and the calculation of the expire time, and the update of
the ALSA PCM core. The first part is called both from the trigger-start
and the hrtimer callback while the latter is handled only in the
hrtimer callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no argument named @state in snd_hda_resume,
remove its' comment.
Signed-off-by: Qinghuang Feng <qhfeng.kernel@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no argument named @clk_id in snd_soc_dai_set_fmt,
remove its' comment.
Signed-off-by: Qinghuang Feng <qhfeng.kernel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add ASoC support for TI SDP3430. It's based on Gumstix
Overo SoC code by Steve Sakoman.
Signed-off-by: Misael Lopez Cruz <mesak82@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes Kconfig dependency of TWL4030 audio codec driver
with TWL4030 core driver on both overo and omap2evm
boards
Signed-off-by: Arun KS <arunks@mistralsolutions.com>
Acked-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Patch adds support for mono audio links so that McBSP DAI can operate with
real mono codecs. In I2S, the signalling remains the same but only first
frame (left channel) is transmitting audio data and second frame having null
data. In DSP_A, only first frame is transmitted.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Prepare for upcoming McBSP DAI update adding support for mono links by
restricting number of channels to 2 in N810. This is due tlv320aic3x which
claims channels_min = 1 and playing pure mono audio over I2S would cause
it to be played only from left channel if both cpu and codec DAI's claim to
support mono.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Impact: cleanup, move all hrtimer processing into hardirq context
This is an attempt at removing some of the hrtimer complexity by
reducing the number of callback modes to 1.
This means that all hrtimer callback functions will be ran from HARD-irq
context.
I went through all the 30 odd hrtimer callback functions in the kernel
and saw only one that I'm not quite sure of, which is the one in
net/can/bcm.c - hence I'm CC-ing the folks responsible for that code.
Furthermore, the hrtimer core now calls callbacks directly with IRQs
disabled in case you try to enqueue an expired timer. If this timer is a
periodic timer (which should use hrtimer_forward() to advance its time)
then it might be possible to end up in an inf. recursive loop due to the
fact that hrtimer_forward() doesn't round up to the next timer
granularity, and therefore keeps on calling the callback - obviously
this needs a fix.
Aside from that, this seems to compile and actually boot on my dual core
test box - although I'm sure there are some bugs in, me not hitting any
makes me certain :-)
Signed-off-by: Peter Zijlstra <a.p.zijlstra@chello.nl>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
Check the model type instead of PCI SSID for detection of the mic types
on Dell laptops with IDT 92HD73xx codecs. In this way, a new laptop
can be tested via model module option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the quirk string for Dell studio 1535 (the product name wasn't
published at the time the patch was made).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT driver creates "Headphone as Line-Out" switch even if there
is no line-out pins on the machine. For devices only with headpohnes
and speaker-outs, this switch shouldn't be created.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add support for pcxhr stereo cards mixer controls
- adjust tlv db scales to real dBu values
- fix bug with monitoring volume control pcxhr_monitor_vol_put
- do some cleanup
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for pcxhr stereo cards and their firmware
- autorize sound cards without analog IO
- do some cleanup
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for pcxhr stereo cards
- minor bugfixes : period and buffer size consraints
- fix PLL register values
- do some clean up
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for pcxhr stereo cards
- do some clean up
Signed-off-by: Markus Bollinger <bollinger@digigram.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we need to handle many unsolicited events assigned to different
widgets, allocate the event dynamically using the existing events
array, and use the tag appropriately instead of combination of fixed
number and widget nid. (Note that widget nid can be over 4 bits!)
Also, replaced the call of unsol_event handler with a dedicated
function to be more readable.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AFG pin power-mapping isn't properly set for the fixed I/O pins
on IDT 92HD* codecs. This resulted in the low power mode after the
boot until any jack detection is executed, thus no output from the
speaker.
This patch fixes the power mapping for the fixed pins, and also fixes
the GPIO bits and digital I/O pin settings properly in stac92xx_ini().
Reference: Novell bnc#446025
https://bugzilla.novell.com/show_bug.cgi?id=446025
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SPDIF status bits controls are written via snd_hda_codec_write()
without caching. This causes a regression at resume that the bits
are lost.
Simply replacing it with the cached version fixes the problem.
Reference:
http://lkml.org/lkml/2008/11/24/324
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the ASoC resume has been punted to a workqueue for a release
cycle without attracting bug reports it should be safe to make the
log messages associated with it debug level, reducing noise and kernel
size in production configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Special handling is required for suspend and resume of AC97 codecs
due to the control path going over the data bus.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Medion MD96630 has ALC268 codec on slot#2 although it's not used
for any purpose. This codec conflicts with the primiary codec ALC888
on slot#0, and gives mixer errors.
This patch adds a corresponding entry to probe_mask blacklist.
Reference: Novell bnc#412528
https://bugzilla.novell.com/show_bug.cgi?id=412528
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of the gain controls in TWL (mostly those which are associated with
the outputs) are implemented in an interesting way:
0x0 : Power down (mute)
0x1 : 6dB
0x2 : 0 dB
0x3 : -6 dB
Inverting not going to help with these.
Custom volsw and volsw_2r get/put functions to handle these gains.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add CGAIN (Coarse gain control) to TWL4030 codec.
The range of the CGAIN is:
0 dB to 12 dB in 6 dB steps.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 FGAIN volume control has a range:
-62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Keep Soft-volume disabled for now, since if it is enabled
the FGAIN volume controls are not working in the current
configuration:
CODEC_MODE:OPT_MODE = 1
OPTION:ARXR2_EN = 1
OPTION:ARXL2_EN = 1
OPTION:ARXR1_EN = 0
OPTION:ARXL1_VRX_EN = 0
RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1)
RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1)
After the patch, FGAIN volume control works.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the bug 0004240: ALC888 - Intel HDA - Headphone Controlling.
It is made against the 2008-11-23 snapshot.
Added Realtek ALC888 model entry for the Fujitsu-Siemens Amilo Xa3530
laptop. It has 4 jacks: HP out, Mic-in, Line-in and Line-out/Side/SPDIF
(this one is on the laptop side, the other ones are on the rear).
Model detection works.
Headphone jack sense works now.
Front mic works now, was same as Acer Aspire 4930G.
Added channel mode from 2 to 8 channels.
In 2ch and 4ch modes, the front is also sent to the Line-out/side jack
for convenience instead of just muting the Line-out/side jack like other
models do.
When using the Mic-in jack as CLFE, the sound is very low (bug?). To
work it around, in 6ch mode the CLFE channel is duplicated to the
Line-out/side jack because this one has a better amp.
Cc: manu@frogged.de
Signed-off-by: Vincent Petry <PVince81@yahoo.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
STAC/IDT driver creates "Headphone as Line-Out" switch even if there
is no line-out pins on the machine. For devices only with headpohnes
and speaker-outs, this switch shouldn't be created.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix "defined but not used" build warning by moving eld_versoin_names[]
and cea_edid_version_names[] into hdmi_print_eld_info().
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DisplayPort is a digital display interface standard put forth by
the Video Electronics Standards Association (VESA). It defines a
new license-free, royalty-free, digital audio/video interconnect,
intended to be used primarily between a computer and its display monitor,
or a computer and a home-theater system.
- From Wikipedia, the free encyclopedia
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- rename ELD proc write routine to hdmi_write_eld_info()
- support modifying WMAPro's profile
Write to some ELD fields (monitor_name, manufacture_id, product_id,
eld_version, edid_version) are deliberately not supported, since that
won't correct wrong behaviors and only leads to confusions.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- make some messages more user friendly
- add message prefix "HDMI:" to indicate the problem's domain
(also easier to do `dmesg | grep HDMI` ;-)
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Print some CA selecting info, which could be valuable for debugging when
something goes wrong.
Signed-off-by: Wu Fengguang <wfg@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_print_pcm_rates() and snd_print_pcm_bits() are used by both
hda_proc.c and hda_eld.c, thus they have to be defined in the common
place.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the quirk string for Dell studio 1535 (the product name wasn't
published at the time the patch was made).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following sparse warning:
sound/pci/hda/patch_nvhdmi.c:161:25: warning: symbol
'snd_hda_preset_nvhdmi' was not declared. Should it be static?
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement support for the Marvell Zylonite PXA3xx reference platform,
supporting standard AC97 stereo and AUX interfaces together with the
auxiliary I2S interface of the WM9713.
The board has two options for the MCLK of the WM9713: either the standard
AC97 system clock can be used or the 13MHz CLK_POUT output of the PXA3xx
can be used, selected via SW15 on the board. Currently only the AC97
system clock is supported by this driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.
This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clean up our record of the active streams in shutdown(), fixing
subsequent failures of snd_pcm_hw_constraints_complete after closure of
a stream.
NOTE:
- The ssm2602 allows pairs of non-matching PB/REC rates.
- This is a fix for less evil:
The logic is flawed (e.g. the slave might startup before the
master's rate and sample_bits are set).
Signed-off-by: Karl Beldan <karl.beldan@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.
Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make the codec re-configuration feature selectable via Kconfig,
CONFIG_SND_HDA_RECONFIG.
Also mark it as experimental (as it really is).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack instances created in patch_sigmatel.c may be double-freed.
The device management code checks the invalid element, and thus there
is no real breakage, but it spews annoying warning messages.
But, we can't simply remove the release calls of these jack instances
because they have to be freed when the codec is re-configured.
Now, a new flag, bus->shutdown is introduced to indicate that the bus
is really being unloaded, i.e. the objects managed by the device
manager will be automatically deleted. We release these objects only
when this flag isn't set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
