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Move the definition of the "generic" IRQ in the process.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
This patch adds support for passing platform data to ac97 bus devices
from PXA2xx-AC97 driver..
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is one instance of McASP on DA850/OMAP-L138 SoC. This is
connected to TLV320AIC3106 codec for audio playback and capture.
This patch adds audio support on this platform. Some of the
structure prefix names which are common for DA830/OMAP-L137 EVM and
DA850/OMAP-L138 EVM have been renamed to da8xx from da830.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch adds a DAI format: Codec bit clock master and frame sync slave,
to the driver.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO
support. This FIFO provides additional data buffering. It also provides
tolerance to variation in host/DMA controller response times.
The read and write FIFO sizes are 256 bytes each. If FIFO is enabled,
the DMA events from McASP are sent to the FIFO which in turn sends DMA requests
to the host CPU according to the thresholds programmed.
More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=
sprufm1&fileType=pdf
This patch adds support for FIFO configuration. The platform data has a
version field which differentiates the McASP on different SoCs.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 analogue control is shared with other devices in the same
product line. Since this is a very substantial proportion of the
driver move the definitions of these controls into a new wm_hubs module
which allows them to be shared between the two.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A few improvements for IDT 92HD83xxx codec pareser:
- Remove unused / deprecated mixer-amp controls
- Handle d-mics as normal inputs since this codec has no separate
MUXes for analog and digital
- Don't create duplicated controls for capture volumes with Mux
capture volumes
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There has been an ad1836 driver in sound/blackfin based on traditional alsa.
The new driver is based on asoc. The architecture of ad1836 codec driver is
very much like ad1938.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Dynamically control and control only the needed output amplifier
muting/un-muting.
The original code was muting and un-muting the following output
amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time
regardless which pin is actually in use at the given moment.
Move these as separate PGA so only the needed amplifier will be touched.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the function dapm_dac_check_power() in
sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any
output widget as sink. And according to dapm_adc_check_power(), adc
power can't be on/off stand-alone without any input widget as source. So
we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC
to hope their power can be managed dynamically.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable line-out detection for IDT/STAC codecs only when speaker pins
exist. In some cases, the speaker itself is identified as line-out,
and this confuses the situation. Only the extra line-outs should do
auto-muting.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
the hdsp driver refuses to report any information via the proc
interface, if the io box is not connected. with this patch, the
content of the control and status registers is printed before the
iobox check.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With auto config model of alc268 realtek codec, it allows to select any
of possible available digital microphone inputs when only one is
available. For example, when only digital mic in nid 0x12 is available,
on second input source it will allow you to select unavailable digital
mic in nid 0x13. The problem is that selecting unavailable digital mic
creates a source of noise when recording (I'm not sure if this happens
on all machines with alc268 and only one digital mic input, but testing
on a quanta uw1 netbook a lot of noise is introduced in recording from
digital mic 0x12/first input source, when you select the unavailable
digital mic 0x13 for capture source 0x24 in the second input source in
mixer).
Then to avoid noise when recording from digital mic with auto model in
this case, prevent a digital mic input source to be selected if
microphone is not available.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move static codes to setup from init_hook for each model.
Also, use the common auto-mic selection helper for devices that support
auto-mic selection. They just need to set up ext_mic, int_mic and
auto_mic flag in the setup section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Store the TDM slot width then if it's set use that rather than the
sample size to calculate BCLK. Leave imposing constraints to the
core (which should do this but doesn't yet) or machine driver.
Also allow 0 TDM slots to be configure (for use when disabling TDM).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added setup hook to ALC preset struct to be called at in the parser
but not at each init callback.
This can be used for setting up the static pins, etc, while the
init hook should be used for updating the status again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Created a white-list to enable MSI since some devices require MSI
explicitly due to BIOS/ACPI problems. Simply using a quirk list.
As the first case, take HP Compaq CQ40.
Reference: Novell bnc#529971
https://bugzilla.novell.com/show_bug.cgi?id=529971
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix soc build errors when I2C is built as a loadable module:
(.text+0x5d26b): undefined reference to `i2c_master_send'
soc-cache.c:(.text+0x5d32d): undefined reference to `i2c_transfer'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some Realtek codecs don't provide the full connections for certain pins
from each ADC; e.g. ACL662/ALC272 gives only one of two digital-mic pins
for each ADC. Thus, depending on the digital mic pin, the ADC/MUX to be
used has to be chosen properly.
This patch adds the check of the connectivity of pins at auto-mic mode.
If no proper connectivity is found, auto_mic flag is turned off to be
sure.
Also the mux_idx is determined during this check so it won't be checked
in the unsol event any more.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition. This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).
This patch avoids the overriding by adding the proper checks.
Reference: Novell bnc#529467
https://bugzilla.novell.com/show_bug.cgi?id=529467
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the compile warning due to different integer types used in min():
sound/usb/usbaudio.c: In function 'init_substream_urbs':
sound/usb/usbaudio.c:1087: warning: comparison of distinct pointer types lacks a cast
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for automatic mic selection via plugging for
Realtek codecs (in auto-probing mode). The auto-mic mode is enabled
only when one internal mic and one external mic are present.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If any OSS support is enabled, regardless of built-in or module,
sound_core claims full OSS major number (that is, the old 0-255
region) to trap open attempts and request sound modules using custom
module aliases. This feature is redundant as chrdev already has such
mechanism. This preemptive claiming prevents alternative OSS
implementation.
The custom module aliases are scheduled to be removed and the previous
patch made soundcore emit the standard chrdev aliases too to help
transition.
This patch schedule the feature for removal in a year and makes it
optional so that developers and distros can try new things in the
meantime without rebuilding the kernel. The pre-claiming can be
turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel parameter
soundcore.preclaim_oss.
As this allows sound minors to be individually grabbed by other users,
this patch updates sound_insert_unit() such that if registering
individual device region fails, it tries the next available slot.
For details on removal plan, please read the entry added by this patch
in feature-removal-schedule.txt .
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Till now missing OSS devices emitted sound-slot/service-* module
alises instead of the standard char-major-* if a missing device number
is opened if soundcore is loaded. The custom module aliases don't
have any inherent benefit than backward compatibility.
sound-slot/service-* module aliases is scheduled to be removed and to
help the transition this patch makes soundcore emit the standard
module alises along with the custom ones.
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the 2.1.6 kernel, the output loop in midi_poll() was changed to
enable interrupts during the outputc() call. Unfortunately, the check
whether the device has accepted the current byte ("ok") was moved behind
the code that removes the byte from the output queue, so one byte would
be lost every time the hardware FIFO is full.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the interval timer to be programmed with its full 96 kHz
precision.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using more packets in one URB do avoid interrupts does not make sense
when we have a sync pipe whose packets generate interrupts more often.
Therefore, limit the URB size to the synchronization packet interval.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without the initialization of vmaster NID, the dB information got
confused for ALC269 codec.
Reference: Novell bnc#527361
https://bugzilla.novell.com/show_bug.cgi?id=527361
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
The previous auto-mic patch for STAC/IDT codecs causes the Oops on
machines without digital mic pins. This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for audio on DA830 EVM- here McASP1 is interfaced to
TLV320AIC3106 codec.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig):
CC [M] sound/soc/s3c24xx/s3c2443-ac97.o
sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX'
sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read':
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function)
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.)
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write':
sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function)
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch changes the line discipline name registered in include/linux/tty.h
and updates the ams-delta machine driver to use it.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dma setup code assumes that the buffer size is a multiple
of the period size.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dai is a parameter to the functions, so use it instead of
looking it up.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Simultaneous audio playback and capture on OMAP1510 can cause that second
stream is stalled if there is enough delay between startup of the audio
streams.
Current implementation of the omap_mcbsp_start is starting both transmitter
and receiver at the same time and it is called only for firstly started
audio stream from the OMAP McBSP based ASoC DAI driver.
Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is
missed if there is no DMA transfer set up at that time when the first word
after McBSP startup is transmitted. The problem hasn't noted before since
later OMAPs are using level sensitive DMA request lines.
Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by
allowing to start and stop individually McBSP transmitter and receiver
logics. Then call those functions individually for both audio playback
and capture streams. This ensures that DMA transfer is setup before
transmitter or receiver is started.
Thanks to Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> for detailed problem
analysis and Peter Ujfalusi <peter.ujfalusi@nokia.com> for info about DMA
request line behavior differences between the OMAP generations.
Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extend set_tdm_slot to allow the user to arbitrarily set the frame width
and active TX/RX slots.
Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c
still doesn't handle the slot_width override.
While being there, correct an incorrect use of SlotsPerFrm(7) use in
bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ).
(this series is meant for Mark's for-2.6.32 branch)
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
