Add a quirk model=acer-aspire for Acer Ferrari 5000 with ALC883 codec.
Note that model=auto doesn't work for this laptop because of broken BIOS
(that doesn't set the subsystem id properly).
Tested-by: Russ Dill <russ.dill@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace with the standard function calls to use caches for reading
the widget caps and pin caps.
hda_proc.c is still using the direct verbs to get raw values as
much as possible.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In patch_realtek.c, don't create empty or single-item "Input Source"
control elements that are simply superfluous.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check for the amp-output must be done for widget-caps rather than
pin-caps as implemented in the recent change... Simply a thinko.
Also, add the similar checks to all places that put output-amp mutes
in the initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_query_pin_caps() to read and cache pin-cap values
to avoid too frequently issuing the same verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't set amp-out values to pins without PINCAP_OUT capability,
which are usually assigned for digital mics on ALC663/ALC272.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch does two things:
Output Intel HDA Function Id in /proc/asound/cardX/codec#X
Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros
Before:
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x100100
After:
Function Id: 0x1
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x0100100
As report on the Kernel Bugzilla #12888
Signed-off-by: Pascal de Bruijn <pascal@unilogicnetworks.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the boundary checks a bit safer.
These caese are rare or theoretically won't happen, but nothing
bad to keep the checks safer...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the detection of digital-mic inputs on ALC663 / ALC272 codecs
in the auto-detection mode. The automatic mic switch via plugging
isn't implemented yet, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headset was declared previously as a Headphone widget connecting
HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver.
The capture path becomes invalid as the Headphone widget is not a
valid input endpoint.
Instead of that, the Headset is declared as separate Microphone
and Headphone widgets. Current patch modifies audio map:
- Headset Mic: HSMIC with bias
- Headset Stereophone: HSOL, HSOR
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add functions "Headset" and "Mic" to the control "Jack Function" for
activating and de-activating codec input pin LINE1L which is connected to
the mic pin of 4-pole Nokia AV connecter.
Note there is no mic bias voltage management here since bias is coming from
Nokia ASIC and driver for it is not in mainline.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
So far, the prepare callback is called multiple times, BDL entries
are reset and re-programmed at each time.
This patch adds the check to avoid the reset of BDL entries when the
same parameters are used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the hw_ptr_interrupt reaches the boundary, it must check whether
the hw_base was already lapped and corret the delta value appropriately.
Also, rebasing the hw_ptr needs a correction because buffer_size isn't
always aligned to period_size.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Always reset the invalind hw_ptr position returned by the pointer
callback. The behavior should be consitent independently from the
debug option.
Also, add the printk_ratelimit() check to avoid flooding debug
prints.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an AVDD supply as well, normally one or more of the other
upplies would be tied to it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The active discharge does not bring sufficient benefit to justify the
lengthy times involved so don't do that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Drop inbound packets that are smaller than expected. This has been
observed at the very beginning of the streaming transaction.
And when the hardware is in panic mode (which can only very rarely
happen in case of massive EMI chaos), mute the input channels.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Tested-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Limit the number of printed warnings to one in case of streaming errors.
printk() happens to be expensive, especially in code called as often as
here.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_free_sgbuf_pags(), vunmap() is called after releasing the SG
pages, and it causes errors on Xen as Xen manages the pages
differently. Although no significant errors have been reported on
the actual hardware, this order should be fixed other way round,
first vunmap() then free pages.
Cc: Jan Beulich <jbeulich@novell.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an omitted unlock in one snd_mixart_hw_params fail path. Fix it.
Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
s/mutex_lock/mutex_unlock/ on 2 fail paths in snd_pcm_oss_proc_write.
Probably a typo, lock should be unlocked when leaving the function.
Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the time of initialization, SNDRV_MIXER_OSS_PRESENT_PVOLUME bit is not
set for MIC (slot 7).
So, the same should not be checked when an application tries to do gain
control for audio recording devices.
Just check slot->present for SNDRV_MIXER_OSS_PRESENT_CVOLUME independently.
Verified with a simple application which opens /dev/dsp for recording and
/dev/mixer for volume control.
Have tested two usb audio mic devices.
Signed-off-by: Viral Mehta <viral.mehta@einfochips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If ratesp or formatsp values are zero, wrong values are passed to ALSA's
the PCM midlevel code. The bug is showed more later than expected.
Also, clean a bit the code.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The position-buffer on ATI controllers are unreliable as well as
on VIA chips, thus the same workaround for DMA position reading as
VIA is useful for ATI.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATI controllers (at least some SB0600 models) appear buggy to handle
64bit DMA. As a workaround, reset GCAP bit0 and let the driver to
use only 32bit DMA on these controllers.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got
merged into mainline. Since there's no way to use it on any of
supported machines (iPaq h3100 or h3600), better drop it for now.
It can be reimplemented later using ASoC infrastructure (there's
already a driver for uda1341 codec in mainline, so only CPU and machine
parts need to be written).
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Cc: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 14fa43f53f ("ASoC: Only
register AC97 bus if it's not done already") added a condition for
calling of soc_ac97_dev_register() but not added for calling of
soc_ac97_dev_unregister(). This patch adds same condition for
soc_ac97_dev_unregister(). Without this fix, kernel crashes when
unloading an asoc driver.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the OOPS during a opl3sa2 card suspend
and resume if the driver is loaded but the card
is not found.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most fasync implementations do something like:
return fasync_helper(...);
But fasync_helper() will return a positive value at times - a feature used
in at least one place. Thus, a number of other drivers do:
err = fasync_helper(...);
if (err < 0)
return err;
return 0;
In the interests of consistency and more concise code, it makes sense to
map positive return values onto zero where ->fasync() is called.
Cc: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
Traditionally, changes to struct file->f_flags have been done under BKL
protection, or with no protection at all. This patch causes all f_flags
changes after file open/creation time to be done under protection of
f_lock. This allows the removal of some BKL usage and fixes a number of
longstanding (if microscopic) races.
Reviewed-by: Christoph Hellwig <hch@lst.de>
Cc: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
The previous commit breaks the (digital-) beep on ALC662.
ALC662 has the connection index 0x05 while ALC662 and ALC272 have the
index 0x04 for the beep widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID.
Confirmed by testing on real hardware.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CC sound/soc/codecs/twl4030.o
sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer
sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type
sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops')
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With this patch the drivers do not set the vmixer volume anymore at startup
because it is actually the output volume of the voices and ALSA mandates
that the volume must be 0 by default.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a long standing bug in the drivers for cards with a vmixer because
I overlooked a detail in the c++ generic driver by echoaudio. Those cards
do not have a line-out volume control. It is a virtual control provided by
the generic driver. The bug is harmless because the DSP just ignores the
command to change the volume.
*NB:* It breaks alsa-tools/echomixer. A patch for it will follow.
This patch removes the line-out volume control from vmixer-equipped cards.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the PXA27x series allow 2 gpios to reset the ac97 bus,
allow through platform data configuration the definition of
the correct gpio which will reset the AC97 bus.
This comes from a silicon defect on the PXA27x series, where
the gpio must be manually controlled in warm reset cases.
Signed-off-by: Robert Jarzmik <rjarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Two issues are fixed here:
- I2S transmits the left frame with the clock low but I don't seem to
get LRCLK out without SFRMDLY being set so invert SFRMP and set a
delay.
- I2S has a clock cycle prior to the first data byte in each channel
so we need to delay the data by one cycle.
Tested-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This switches the pxa ssp port usage from network mode to PSP mode.
Removed some comments and checks for configured TDM channels.
A special case is added to support configuration where BCLK = 64fs. We
need to do some black magic in this case which doesn't look nice but
there is unfortunately no other option than that.
Diagnosed-by: Tim Ruetz <tim@caiaq.de>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move headset jack registration to the codec/machine specific
initialization. Having the jack registration in machine init
causes that the jack device gets initialized but not registered
since the sound card is registered before the jack. Moving jack
registration to device initialization will register the jack
device along with all other devices associated to the card when
the card is registed. As a consequence of jack device registered
properly, the jack is detected as an input device.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the power state of each widget before starting the initialization
work so that all verbs are executed properly.
Also, keep power-up during hwdep reconfiguration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The drivers are basically duplicating the same code over and over.
As snd_soc_cnew is going to be made static some time after the next
merge window, we might as well convert them now.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Refactor the WM8580 device registration to probe via standard I2C device
registration, registering the DAIs once the device has probed via I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the places where the 0x1d widget is used for Conexant 5047, fixing
mismatch introduced after changing the connection.
Signed-off-by: Gregorio Guidi <gregorio.guidi@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line. Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8400 is a highly integrated audio CODEC and power management unit
intended for mobile multimedia application. This driver supports the
primary audio CODEC features, including:
- 1W speaker driver
- Fully differential headphone output
- Up to 4 differential microphone inputs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Buildfix:
CC sound/soc/omap/osk5912.o
sound/soc/omap/osk5912.c: In function 'osk_soc_init':
sound/soc/omap/osk5912.c:189: error: implicit declaration of function 'clk_get_usecount'
make[3]: *** [sound/soc/omap/osk5912.o] Error 1
There's no such (standard) clock interface.
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In pxa_ssp_set_dai_fmt(), check whether there is anything to do at all.
If there would be but the SSP port is in use already, bail out.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clean up Conexant 5047 pareser code:
- Split mixer elements to separate arrays to reduce the duplicated
entires
- Fix mixer element names to the standard ones
- Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event
handler works fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the initial connections of output pins 0x13 and 0x1d for Conexant
5047 codec to point to the mixer amp properly.
Removed unneeded (doubly) verbs from arrays, also removed the unneeded
changing of widget 0x1c, which is now completely unused.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove superfluous verbs from cxt5047_toshiba_init_verbs[].
Also fix comments and minor coding style issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create "Capture Source" control dynamically for Conexant codecs.
If only one capture item is available, don't create such a control
since it's just useless.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of binding volumes, create a virtual master volume for Conexant
codecs. This allows separate HP and speaker volume controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated.
Code tested on HP HDX16 (not tested on HDX18 yet).
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added codec recognition of HP HDX platforms and added support of the
MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE
is needed to use event handling for mute control.
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will break any boards that don't register the AC97 controller
device due to using ASoC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the standard linked list for snd_monitor_file management.
Also, move the list deletion of shutdown_list element into
snd_disconnect_release() (for simplification).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_ctl_add_slave_uncached() function to add a slave element
with volatile controls. The values of normal slave elements are
supposed to be cachable, i.e. they are changed only via the put
callbacks. OTOH, when a slave element is volatile and its values may
be changed by other reason (e.g. hardware status change), the values
will get inconsistent.
The new function allows the slave elements with volatile changes.
When the slave is tied with this call, the native get callback is
issued at each time so that the values are always updated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pxa-regs.h and hardware.h are not intended for use directly in driver
code, remove those unnecessary references.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
1. Driver code where pxa_request_dma() is called will most likely
reference DMA registers as well, and it is really unnecessary
to include pxa-regs.h just for this. Move the definitions into
<mach/dma.h> and make relevant drivers include it instead of
<mach/pxa-regs.h>.
2. Introduce DMAC_REGS_VIRT as the virtual address base for these
DMA registers. This allows later processors to re-use the same
IP while registers may start at different I/O address.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
The timer callbacks are called in the protected status by the lock
of the timer instance, so there is no need for an extra lock in the
PCM substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up and improve snd_pcm_update_hw_ptr*() functions.
snd_pcm_update_hw_ptr() tries to detect the unexpected hwptr jumps
more strictly to avoid the position mess-up, which often results in
the bad quality I/O with pulseaudio.
The hw-ptr skip error messages are printed when xrun proc is set to
non-zero.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a driver for the SPI connected AK4104 S/PDIF transmitter
device. Its features are fairly simple, but as there is need to set up
certain bits in the IEC958 information, this better goes into a real
driver.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@sirena.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On the HT-Omega Claro (halo) sound cards, the headphone amplifier must
be enabled explicitly by setting a GPIO bit.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Removes numbers from the list of features/limitations and makes it
reflect recent changes to the code.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix headphone-detect regression with multiple HP jacks
ALSA: hda - Fix typos in slave controls in patch_sigmatel.c
Add support for true pause and unpause. Without this, mplayer will drop some
audio (less than one second, but still noticeable) when pausing playback.
Remove support for PM suspend and resume from the trigger function, since the
driver doesn't support PM anyway.
Optimize the delay after starting capture. Instead of delaying 1ms, the driver
now polls the hardware. The new delay is shorter by over 90% yet still
effective.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Upgrade the severity of some failure messages from debug level so
they're displayed by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent set of S3C64xx patches re-added a lot of uses of DBG() that
had previously been removed - revert this so the standard pr_debug()
macro is used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Rob Maris <maris.rob@vdi.de>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add headset jack detection for SDP3430 boards using SoC jack
reporting interface. Headset detection on SDP3430 board is
achieved through TWL4030 GPIO_2 pin.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a new device tree property for the SSI node: "fsl,ssi-asynchronous". If
defined, the SSI is programmed into asynchronous mode, otherwise it is
programmed into synchronous mode. In asynchronous mode, pin SRCK must be
connected to the same clock source as STFS, and pin SRFS must be connected to
the same signal as STFS. Asynchronous mode allows playback and capture to
use different sample sizes. It also technically allows different sample rates,
but the driver does not support that.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We now support the 64xx series as well as the 24xx series - make sure
people using Kconfig know this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_dapm_switch ends up ends up in dapm_new_mixer() (since a switch
is a special case of a mixer with only one input) but this wasn't
correctly handled in the code.
Also fix the coding style for the switch below while we're here.
Reported-by: Joonyoung Shim <dofmind@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A bit in PXA's SSCR0 register was erroneously named ADC but its name is
in fact ACS (audio clock select).
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enum type for selecting the desired ramp delay for the headset output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Select the relevant DMA implementation when the
sound driver is selected.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the initial code to support the S3C64XX I2S hardware using the
s3c-i2s-v2 core code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S3C2412 I2S (IIS) interface is replicated on further Samsung SoC
parts in a broadly compatible way, so split the common code out into
a core called s3c-i2s-v2.[ch] so that the newer SoCs such as the
S3C6410 can make use of it.
As such, all the original s3c2412 functions are currently being left
with their original names, and will be renamed later in the series.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the Jive's WM8750 codec attached via the S3C2412 IIS.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Assign DACs to HP and speaker before mic-in/line-in shared outputs.
This improves the usability as it results in more intuitive mixer
names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In stac92xx_auto_fill_dac_nids[], connect to the primary DAC if no
individual DAC is available for each pin. This ensures that the pin
works somehow at least.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create multiple "Headphone" and "Speaker" controls with non-zero index
numbers instead of "Headphone2", etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve the parser to pick up more intuitive control names for the
outputs judging from the pin type, instead of fixed names assigned
to channels.
Also, revive the multi-HP workaround since this change fixes the
problem with the multi-HP detection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent changes over the DAC detection mechanism in patch_sigmatel.c
breaks the HP detection on the machines with multiple HP jacks.
It's basically because of the workaround to support the multi-channel
output. Since the HP detection is more important feature, disable
the HP-swap workaroud temporarily.
Reference: Novell bnc#482052
https://bugzilla.novell.com/show_bug.cgi?id=482052
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Headphone Playback ..." appears twice in slave_vols[] and slave_sws[].
They should be "Headphone Playback2 ..."
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Noises can be heard on analog outputs of (some model of) Lenovo
Ideapad due to the hardware problem, and the only workaround right now
is to fix the sample rate to 44.1kHz.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The <mach/audio.h> file needs to be common to both ARCH_S3C2410 and
ARCH_S3C64XX as they share common driver code, so move it to <plat/audio.h>.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the IIS headers to their correct place.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ignore MIDI and PCM events in the interrupt handler until the device
gets initialized properly. Otherwise you may get kernel panic by the
access to uninitialized devices via hotplugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.
The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mute speaker outputs on headphone insertion for machines that use
3stack-hp model.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When setting WM8510_MCLKDIV the pll was turned off.
When setting pll frequency you got twice the expected freq, because
the code calculated with postscaler of 8, but the hardware divide by 4.
Signed-off-by: Jonas Andersson <jonas@microbit.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.
Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.
All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent update enabled the model=sony-assamd for all ALC262 with
PCI SSID 104d:90xx. But this includes the VAIO VGN-AR* that has the
primary codec of STAC92xx and the secondary ALC262 as a slave
digital-only codec. For this device, the model=auto must be chosen
to work properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the mic input of HP dv6736 with Conexant 5051 codec chip.
This laptop seems have no mic-switching per jack connection.
A new model hp-dv6736 is introduced to match with the h/w implementation.
Reference: Novell bnc#480753
https://bugzilla.novell.com/show_bug.cgi?id=480753
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only allow SND_SOC_DAIFMT_CBS_CFS for the playback DAI.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo
in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result
in exactly the same behaviour.
Now it is possible to use 16-bit single slot transfers in pxa-ssp, which
are needed for Magician to get two frame clock pulses per sample
(one for each channel).
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Tested-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the UDA1380's interpolator or decimator are set to be clocked from
the WSPLL (which syncs to the WSI signal), the DAI link must be running
to change the interpolator/decimator registers (which include volume
controls and digital mute setting).
* Queue work in the alsa PCM_START .trigger to flush registers
as soon as the link is running. This replaces the .prepare
and .digital_mute callbacks.
* Use the SILENCE override instead of MTM for muting and remove
its alsa control to avoid confusion.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's false positive, but annoying.
sound/pci/hda/hda_codec.c: In function ‘get_empty_pcm_device’:
sound/pci/hda/hda_codec.c:2772: warning: ‘dev’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Add probe_mask default for Toshiba laptop with ALC268
ALSA: hda - Add quirk for new HP xw series
ALSA: hda - Fix digital mic on dell-m4-1 and dell-m4-3
Allow more options to be set/reset via hwdep hint entry.
hp_detect, gpio_mask, gpio_dir, gpio_data, eapd_mask and eapd_switch
can be checked.
For example, to disable hp_detect on the fly,
# echo "hp_detect=0" > /sys/class/sound/hwC0D0/hints
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't create "Analog Loopback" controls as default since these controls
are usually more harmful than useful for normal users.
Only created when "loopback = yes" hint is given.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_get_hint() and snd_hda_get_bool_hint() helper functions
to retrieve a hint value.
Internally, the hint is stored in a pair of two strings, key and val.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This removes a misspelled comment and got rid of superfluous switch
case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When checking for the maximum queue length, we have to take into account
that MAX_QUEUE is measured in milliseconds (i.e., frames) while the unit
of urb_packs is whatever data packet interval the device uses (possibly
less than one frame when using high speed devices).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When storing the channel numbers used by a format, and if the device
happens to support 32 channels, the code would try to store 1<<32 in
a 32-bit value.
Since no valid format can have zero channels, we can use 1<<(channels-1)
instead of 1<<channels so that all the channel numbers that we test for
fit into 32 bits.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two simple fixes:
1. Use the same pointer for the free_irq() and
the request_irq() calls.
2. A short name of card is appended with '2' or '3'
character depending on a detected chip. Remove
the '2' character from the short name.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't return a fatal error to the driver but continue to probe when
any error occurs at creating PCM streams for each codec.
It's often non-fatal and keeping it would help debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the codec probe instead of returning the error to the driver
when any error occurs at creating the control elements.
The control element conflict can be non-fatal in many cases,
especially if it comes from the digital-only codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
