The second argument of the probe method points to the amba_id
structure, so it's better passed with the correct type. None of the
current in-tree drivers uses the pointer, so they have only been
checked for a clean compile.
Change suggested by Russell King.
Signed-off-by: Alessandro Rubini <rubini@unipv.it>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
The commit 5a641bcd63 changed the
printk format to '%lu', but the value passed seems to be dependent
on the architecture. On x86-64, I got a new warning now because an
int value is passed actaully.
As a workaround, just cast the value always to unsigned long.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ctxfi driver requires explicitly the 4k page size, and gives a
build error on architectures with non-4k pages.
As a workaround, just add the kconfig dependency on X86, which is
the only architecture ever tested.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handles
the headset ramp up and down sequences needed for the pop noise
removal.
With this patch the order of the internal components in the twl4030
codec is turned on and off in a correct order.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Restructuring the twl4030 codec's DAPM routing to be able to handle the power
sequences correctly.
The twl4030 codec internal implementation have this order:
DAC -> Analog PGA -> Mixer/Mux
While the ASoC framework expects the following order:
DAC -> Mixer -> Analog PGA
This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER and
adds two levels of mixer to handle the digital and analog loopback
functionality.
Now the analog loopback does not powers on any of the DACs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the integrated ACLC of the TXx9 family.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The digital beep tone is calculated in two different ways depending
on the codec chip. The standard one is using a divider, and another
one is a linear tone for IDT/STAC codecs. Currently, only the
latter type is used for all codecs, which resulted in a wrong tone
pitch.
This patch adds the calculation of the standard HD-audio type.
Also clean-up the fields in hda_beep struct.
Reference: bko#13162
http://bugzilla.kernel.org/show_bug.cgi?id=13162
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper prefix to each kernel message in hda_intel.c.
Also, avoid the unneeded prefix when CONFIG_SND_VERBOSE_PRINTK is used
together with snd_print*().
Reference: bko#13207
http://bugzilla.kernel.org/show_bug.cgi?id=13207
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Connect twl4030 voice DAI to McBSP3 in sdp3430 machine driver.
Voice DAI init function enables corresponding interface by
writting directly to VOICE_IF codec register.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a control for selecting the codec operation mode. TWL4030 codec
has two modes:
- Option 1. Audio only (4 audio DACs)
- Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC)
Control is restricted when a stream is ongoing, since codec's
operation mode cannot be changed on-the-fly.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AUXR is selected by bit 2 and not by bit 1 in the ANAMICR register.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move all the gpio functions out of <mach/hardware.h> as
this file is for defining the generic IO base addresses
for the kernel IO calls.
Make a new header <mach/gpio-fns.h> to take this and
include it via the chain from <linux/gpio.h> which is
what most of these files should be using (and will be
changed as soon as possible).
Note, this does make minor changes to some drivers but
should not mess up any pending merges.
CC: Richard Purdie <rpurdie@rpsys.net>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Rather than managing the bias level of the system based on if there is
an active audio stream manage it based on there being an active DAPM
widget. This simplifies the code a little, moving the power handling
into one place, and improves audio performance for bypass paths when no
playbacks or captures are active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DAPM has always applied any changes to the power state of widgets as soon
as it has determined that they are required. Instead of doing this store
all the changes that are required on lists of widgets to power up and
down, then iterate over those lists and apply the changes. This changes
the sequence in which changes are implemented, doing all power downs
before power ups and always using the up/down sequences (previously they
were only used when changes were due to DAC/ADC power events). The error
handling is also changed so that we continue attempting to power widgets
if some changes fail.
The main benefit of this is to allow future changes to do optimisations
over the whole power sequence and to reduce the number of walks of the
widget graph required to check the power status of widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for MacBook 3,1 sound by adding a model new
"mb31" with the appropriate init verbs, mixers and channel modes to
the ALC883 configuration. patch_alc882() and patch_alc883() are
modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A)
correctly.
Signed-off-by: Torben Schulz <public@letorbi.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI entries of Creative with HD-audio class can be the devices
with emu20k1/emu20k2 chips. These are supported better by snd-ctxfi
driver. With that driver, the device will mutate from HD-audio to
its native class.
This patch adds a simple ifdef to avoid the conflict of device probe
between snd-hda-intel and snd-ctxfi drivers. 1102:0009 seems still
OK to be added as it has no emu20kx chip, and is a pure HD-audio
device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable TWL4030 VTXL/VTXR and VRX digital filters for uplink
and downlink paths, respectively.
This patch also corrects voice 8/16kHz mode selection bit
(SEL_16K) of CODEC_MODE register.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eliminate direct accesses to the driver_data field.
cf 82ab13b26f15f49be45f15ccc96bfa0b81dfd015
The semantic patch that makes this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@@
struct device *dev;
expression E;
type T;
@@
- dev->driver_data = (T)E
+ dev_set_drvdata(dev, E)
@@
struct device *dev;
type T;
@@
- (T)dev->driver_data
+ dev_get_drvdata(dev)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split the name string in hda_codec struct to vendor_name and chip_name
strings to be stored directly from the preset name.
Since mostly only the chip name is referred in many patch_*.c, this
results in the reduction of many codes in the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this warning:
sound/parisc/harmony.c:938: warning: format '%lx' expects type 'long unsigned int',
but argument 2 has type 'resource_size_t'
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a mixer control for the STAC92XX boards to control the
DC bias of mic ports, allowing recording from both powered and
non-powered sources. It replaces the "Mic Output Switch" with "Mic Jack
Mode" to switch between Mic, Line In, and Line Out.
Signed-off-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use default to optimize the switch/case in magicial_playback_hw_params(),
which also fixes the compile warnings below:
sound/soc/pxa/magician.c:89: warning: 'acds' may be used uninitialized in this function
sound/soc/pxa/magician.c:89: warning: 'acps' may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a build fix, resyncing the DaVinci EVM ASoC board code
with the version in the DaVinci tree. That resync includes
support for the DM355 EVM, although that board isn't yet in
mainline.
(NOTE: also includes a bugfix to the platform_add_resources
call, recently sent by Chaithrika U S <chaithrika@ti.com> but
not yet merged into the DaVinci tree.)
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This resyncs the DaVinci I2S code with the version in the DaVinci
tree. The behavioral change uses updated clock interfaces which
recently merged to mainline. Two other changes include adding a
comment on the ASP/McBSP/McASP confusion, and dropping pdev->id in
order to support more boards than just the DM644x EVM.
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a buildfix for the DaVinci PCM code, resyncing it with
the version in the DaVinci tree. The notable change is using
current EDMA interfaces, which recently merged to mainline.
(The older interfaces never made it into mainline.)
NOTE: open issue, the DMA should be to/from SRAM; see chip
errata for more info. The artifacts are extremely easy to
hear on DM355 hardware (not yet supported in mainline), but
don't seem as audible on DM6446 hardwaare (which does have
mainline support).
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modifies sh_dac_audio in the following ways:
- use high resolution timer instead of TMU1
- fix cpu/dac.h include
- add future rewrite comment
Signed-off-by: Magnus Damm <damm@igel.co.jp>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
The WM9712 can be configured by resistor strapping GPIO4 to behave like
the WM9713 and default to leaving the AC97 link disabled after cold
reset until a warm reset occurs. In this configuration we need to issue
a warm reset after cold to bring the link up so do so. The warm reset
will be harmless on systems that don't need it.
[Changelog rewritten to document the reasoning. -- broonie]
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pxa2xx_i2s_resume is :
- unconditionnaly setting SACR0_ENB
- unsetting SACR0_ENB in saved SACR0 pxa_i2s.sacr0
fix these.
In pxa2xx_i2s_{resume,suspend}, save/restore registers even
when !dai->active.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
i2s_clk is 'put' for no reason in pxa2xx_i2s_shutdown.
Now we 'get' i2s_clk at probe and 'put' it at driver removal or when
probe fails.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- hw_params enables both RPL and REC functions each time : Enable the
appropriate function in pxa2xx_i2s_trigger.
- pxa2xx_i2s_shutdown disables i2s anytime one of RPL or REC function is
off : Turn it off only when both functions are off.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure we are in a know good state at end of probe :
Reset FIFO logic and registers, and make sure REC and RPL functions
along with FIFO service are disabled (SACR0_RST enables REC and RPL).
Resetting loses current settings so remove reset from stream startup.
Now reset occurs only at probe.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the codec SSID for MacBook Pro 5,1 as compatible as MP51.
However, the headphone auto-muting function doesn't work. So,
this is just a tentative solution.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix printk format warning:
sound/drivers/pcsp/pcsp_mixer.c:54: warning: format '%d' expects type 'int', but argument 3 has type 'long unsigned int'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The i2c_device_name() macro is used only once and doesn't have much
value, it hurts redability more than it helps. Get rid of it.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Resending the patch after fixing the minor issues.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With a postfix increment these variables are incremented beyond
CMDIF_TIMEOUT / MAX_WRITE_RETRY.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reset FIFO logic and registers, and make sure REC and RPL functions along
with FIFO service are disabled at probe.
Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The model=acer for ALC883/889 doesn't work well for the recent Acer
Aspire laptops. Since model=auto works better nowadays, it's safer
to use the default fallback instead of the Acer specific one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the hdsp driver is built in kernel, the corresponding firmware
files have to be built into the kernel as well (otherwise the boot
will hang up for fairly long time), but there is no way to control
it in Kconfig yet, unfortunately. Instead, show a comment for user
just to make sure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The inputs of the twl4030 codec can be mixed, so we will use the mixer
DAPM for the analog microphone registers(0x05, 0x06), but if we enable
more than one input at the same time, the input impedance of the input
amplifier will be reduced.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the use of the FIRMWARE_IN_KERNEL option with hdsp cards and
in-kernel driver.
Also corrected a typo in the comment.
Signed-off-by: Raphael Doursenaud <rdoursenaud@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the near future, the driver core is going to not allow direct access
to the driver_data pointer in struct device. Instead, the functions
dev_get_drvdata() and dev_set_drvdata() should be used. These functions
have been around since the beginning, so are backwards compatible with
all older kernel versions.
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add specific configuration for Samsung NC10 mini notebook. Internal
mic/speakers will be correctly muted when plugging in external ones.
Mixer controls are added for speakers, headphones and PC beep.
"Boost" is added for the microphones.
Signed-off-by: Chris Pockelé <chris.pockele.f1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Acked-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the card was not instantiated in snd_soc_instantiate_card, calling
soc-remove will crash because some of codec, cpu_dai and card .remove
methods are called twice.
Fix this by returning from soc_remove immediately.
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the missing descriptions and the model names for Realtek codecs
to the documentation and the config table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC uses the standard ALSA data format definitions to specify the wire
format used between the CPU and CODEC. Since the ALSA data formats all
include the endianess of the data but this information is not relevant
by the time the data has been encoded onto the serial link to the CODEC
this means that either all the CODEC drivers need to declare both big and
little endian variants or the core needs to fix up the format constraints
specified by CODEC drivers.
For now take the latter approach - this will need to be revisited if any
CODECs are endianness dependant.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC driver for AT91SAM9260-based AFEB9260 board
Signed-off-by: Sergey Lapin <slapin@ossfans.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Most of unsol handlers defined in patch_realtek.c can be classified to
two types, mute via amp of pins and mute via ctl bits of pins.
Thus there are a big room to generalize each implementation.
This patch creates two generic functions, alc_automute_amp() and
alc_automute_pin(). The latter is actually changed from the previous
alc_sku_automute(). Each caller needs to initialize hp_pins and
speaker_pins properly at own init_hook.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The speaker auto-muting per HP plugging for ALC262 HIPPO and compatible
devices is slightly buggy as the "Master" or "Front" mixer control can
still toggle the speaker output even if the headphone is plugged.
This patch fixes the issue, and clean up the hippo-related codes
together with fixes of some inconsistent mixer names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Copy-paste error: TWL4030_PRECKL_GAIN >> TWL4030_PRECKR_GAIN
It has not caused problems, since
TWL4030_PRECKL_GAIN == TWL4030_PRECKR_GAIN == 0x30
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
VIA VT1708S and VT1702 codecs can have two SPDIF outputs. One of them
should have been handled as the extra digital out, but it's not
properly accessed.
This patch fixes the handling of the secondary SPDIF on these codecs
with the slave dig-out as found in patch_sigmatel.c. This makes the
use of such a device easier (for normal users).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on Mac Mini Core2 Duo sets both INPUT and OUTPUT pinctl bits to
the line-in jack, and it confuses the driver as if it's a valid input.
This patch adds the check of OUTPUT bit so that the driver fixes the
invalid pin setup.
Tested-by: Tino Keitel <tino.keitel@gmx.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added a new flag, own_routing, to allow spec drivers to create own
routing controls. Also, the basic get/put calls are changed to be
external for later use by maya44 driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The set_rate might return error but the current code doesn't check it.
This patch adds a proper error check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the magic 0x80 value with a suitable macro definition.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With 5 bits and 1.5 dB per step, the DXS volume range is only 48 dB.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove most of the serial port parameters from the card longname string
because it was way too long and overflowed into the mixername string.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added runtime->delay field to adjust the delayed samples for snd_pcm_delay().
Typically a hardware FIFO length is stored in this field, so that the
extra delay between hwptr and applptr can be computed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the VIBRA output on TWL4030 codec.
The VIBRA output can be driven with audio data or with
local vibrator driver.
Add the needed DAPM elements and routes for the VIBRA output and
controls for the VIBRA driver configuration.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the ASoC side of the board support for the Crossbow
IMB400 daughter board.
Thanks to Crossbow for considerable assistance.
Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add voice digital loopback (sidetone) to the twl4030
driver. It mixes voice uplink attenuated (by sidetone gain) with
voice downlink when the codec is working in option2 (voice/audio
mode).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds voice downlink analog bypass switch. It follows
the same approach as in other analog bypass switches.
DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer',
that will also allow voice DAC to be powered in digital voice
loopback (sidetone).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mis-typing exist in dapm controller definitions and dapm route definitions,
so happen mis-matched error when snd_soc_dapm_add_routes().
Cc: stable@kernel.org
Signed-off-by: Jinyoung Park <parkjy@mtekvision.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
Add module parameter to enable or disable
joystick port (gameport) on the SC6600 and
later cards.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AC97 wire format is completely fixed so CODECs don't have any choice
about the formats they accept but controllers accept a variety of data
formats and render them down onto the bus. Have a shared define so all
the CODEC drivers will interoperate with any of our controller drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
calls
The S3C24XX DMA API channel configuration registers are being passed
values comprised of register values which makes it hard to move the
API to cover both the S3C24XX and S3C64XX.
These values can be calculated from knowing which device the channel
is connected to, so remove them from the two calls s3c2410_dma_config
and s3c2410_dma_devconfig.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
They aren't used by anything external and aren't prototyped; if any
users appear they can be exported again for them.
Also report what modes we have a problem with when we encounter invalid
mode configurations.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On startup we try to make sure that the port is quiesced but if the
port is already stopped then this will generate a warning about the
RX/TX mode configuration. Configure the mode before doing the teardown
to suppress these warnings.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The data format configuration for S3C64xx IISv2 is completely different
to that for S3C24xx. Instead of a single bit configuration in bit 0 of
IISMOD we have format selection in bits 13 and 14 and bit clock rate
selection in bits 1 and 2. While we're here add support for 24 bit
samples in S3C64xx.
At some point it may be desirable to expose the bit clock rate selection
to users but given the limited configuration options that may not be
required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This makes the interface usable with the s3c-iis-v2 rate calculator
and consistent with S3C2412.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The gain control for earpiece amplifier uses 0dB ~ 12dB according to the
TRM, but the present code is implemented to -6dB ~ 6dB.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The hardware devices with SNDRV_PCM_INFO_BATCH flag can't give the
precise current position. And such hardwares have often big FIFO
in addition to the ring buffer, and it screws up the jiffies check
in pcm_lib.c.
This patch adds a simple check of info flag so that the driver skips
the jiffies check in snd_pcm_period_elapsed() when BATCH flag is set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added SNDRV_PCM_INFO_BATCH flag to PCM info field of some drivers that
really don't give the precise pointer value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commmit dfed0ef9b3 was reverted
accidentally by the merge of auto-detection fix patch.
Fixed again now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable auto-muting in model=auto only for devices with HP and speakers.
For devices with HP and line-outs, don't enable the auto-muting.
Also, add a debug print to show the auto-mute feature.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to check only if the WM8350 is master and only when starting
the stream so if either is not true then we can skip the check.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds a new control named 'Master Playback Switch' for cs4270
codecs. It is implemented using the new SOC_DOUBLE_EXT macro to catch
the put function and store the information about manually set mute
controls from userspace. When a manual mute is set, we don't want the
soc core to un-mute the outputs.
Renamed cs4270_mute() to cs4270_dai_mute() to avoid confusion.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control modifies the MUTE register, hence the polarity must be
inverted.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-By: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Say what invalid values we're seeing when we see an invalid value and
ensure that errors are displayed by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's expected behaviour for the CODEC header to provide them but the
WM8350 doesn't due to having all the registers together under drivers/mfd.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the realtek auto-probing mode, the initialization of amp with
some magic COEF or EAPD verbs is applied only when the codec SSID
has valid values to satisfy the realtek's definition.
However, many devices don't provide in that way, thus the device
doesn't work as is.
This patch allows the same initialization code even if the SSID
doesn't pass the bit test. Also, alc_subsystem_id() is changed
just to check and define the type, so that it's called in the
parser, instead of the initializer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* fix/asoc:
ASoC: s3c-i2s-v2 needs to declare a license for modular builds
ASoC: remove non-existing referece to CONFIG_SND_SOC_CODEC_WM8991
ASoC: Fix WM8580 volume update handling for large register changes
ASoC: Fix offset of freqmode in WM8580 PLL configuration
Fixed the compile warning below by initializatin iomidi variable properly.
sound/pci/cmipci.c: In function ‘snd_cmipci_probe’:
sound/pci/cmipci.c:3017: warning: ‘iomidi’ may be used uninitialized in this function
Signed-off-by: Subrata Modak <subrata@linux.vnet.ibm.com>
Cc: linux-pci <linux-pci@atrey.karlin.mff.cuni.cz>
Cc: Balbir Singh <balbir@linux.vnet.ibm.com>
Cc: Sachin P Sant <sachinp@linux.vnet.ibm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reset the internal period position counter upon stream startup. This
fixes initial aplay underruns and problems related to latency picky
applications such as pulseaudio.
Bumped the version number to 1.3.14.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The S3C64xx IIS code had a number of problems with device registration.
The hardware has two IIS ports of which the driver supported only one
at once via a single exported DAI, attempting to identify the DAI to
use based on the dev->id of the ASoC platform device. As well as
limiting the driver to only supporting one IIS port at once this also
meant that the ID of the soc-audio device (or in future the card device)
had to match the IIS ID.
Fix both problems by converting the driver to register the DAIs based on
probing of platform devices registered by the arch/arm code, using those
platform devices to interact with the clock API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the four channel TDM mode
on Beagle board.
Depending on the channel count, the interface needs to be
configured differently (I2S for stereo DSP_A for four channels)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The last patch to handle newer cards like SC7000
broke initialization of the SC6000. Fix this.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add 4 channel support to omap-mcbsp.
This mode is going to be used by the twl4030 codec, when it
is configured in Option1 mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The original idea came from pHilipp, and this makes the code looks
more consistent.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The SSP DMA parameters can actually be easily generated at run-time since
they are almost similar except for the FIFO width and direction. Another
benefit is the re-use of information from 'struct ssp_device', like SSDR
physical FIFO address and DRCMR register index for both directions.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reviewed-by: pHilipp Zabel <philipp.zabel@gmail.com>
Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we have
to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headset
Left/Right, Carkit Left/Right) from mux to mixer.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current ad1884a-mobile model has a problem that the speaker output
doesn't work sometimes after boot or power-saving on some HP laptops.
It seems that the verbs accessing to the non-functional widgets cause
this problem.
This patch simplifies the init verbs for mobile model not to touch
unnecessary setups so that it avoids the speaker-mute problem.
Reference: Novell bnc#495668
https://bugzilla.novell.com/show_bug.cgi?id=495668
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Save a little extra power by enabling the DC servo offset correction
for the output channels only when the relevant channels are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modify the default startup sequence in the chip to set the DC servo
dither level for optimal performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CLK_DSP provides a master clock for the DAC and ADC related functionality
on the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many modern CODECs have shared resources on chip which must be enabled
for portions of the chip to work but which can be disabled at other times
in order to achieve power savings. Examples of such resources include
power supplies and some internal clocks.
Since these widgets are dependencies for the audio path but do not carry
audio signals they require slightly different handling to most widgets -
they do not contribute to the audio path and so should not be counted as
either inputs or outputs during path walks.
Cases where one supply provides a supply for another will require
additional work. There is also room for more optimisation of the graph
walking to avoid repeated checks for the same thing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add checking in hw_params and prepare to detect bufferless pcms(i.e. BT
<--> codec).
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Appearently, a big delay ~300ms is required before hw is settled and ready
to transfer samples on some hardware variants. Also, return back
"clocking to 48000Hz" message when something fails.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The current error-recovery scheme for the codec communication errors
doesn't work always well. Especially falling back to the
single-command mode causes the fatal problem on many systems.
In this patch, the problematic verb is re-issued again after the error
(even with polling mode) instead of the single-cmd mode. The
single-cmd mode will be used only when specified via the command
option explicitly, mainly just for testing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cache quries for PCM and STREAM parameters as well as ampcap and
pincap sharing the hash table. This will reduce the superfluous
access of the same codec verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Support ASUS F81Se F5Q P80 U20A U80 U50 UX50 for ALC269
- Support ASUS F70SL UX20 X58LE F50Z N80Vc N81Te N505Tp Vx3V N5051A
for ALC663
- Support DELL ZM1 for ALC272
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't unmute unneeded amps for input mixers of ALC662 & co.
It caused possible recording noises.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The legacy i2c binding model is going away soon, so convert the ppc
keywest sound driver to the new model or it will break.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The legacy i2c binding model is going away soon, so convert the AOA
codec drivers to the new model or they'll break.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Tested-by: Johannes Berg <johannes@sipsolutions.net>
Tested-by: Andreas Schwab <schwab@linux-m68k.org>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather than having switch statements at point of use make the DAPM
power check a member of the widget structure and set it when we
instantiate the widget.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This also switches us to using a switch statement for the widget type
in dapm_power_widget().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will form a basis for further power check refactoring: the overall
goal of these changes is to allow us to check power separately to
applying it, allowing improvements in the power sequencing algorithms.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add Voice DAI to support the PCM voice interface of the twl4030 codec.
The PCM voice interface can be used with 8-kHz(voice narrowband) or
16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono
TX or stereo TX.
The PCM voice interface has two modes
- PCM mode1 : This uses the normal FS polarity and the rising edge of
the clock signal.
- PCM mode2 : This uses the FS polarity inverted and the falling edge
of the clock signal.
If the system master clock is not 26MHz or the twl4030 codec mode is not
option2, the voice PCM interface is not available.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use it to clean up snd_us122l_card_used[].
Without patch unplugging of an US122L soundcard didn't reset the
corresponding element of snd_us122l_card_used[] to 0.
The (SNDRV_CARDS + 1)th plugging in did not result in creating the soundcard
device anymore.
Index values supplied with the modprobe command line were not used correctly
anymore after the first unplugging of an US122L.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing definition of max channels for CA0110, which resulted
in an error at opening PCM devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Firstly, this patch makes the palm27x asoc driver a little more sane. Also,
since all affected devices use GPIO95 as AC97_nRESET, this patch sets that
properly. Affected are PalmT5, TX and LifeDrive.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
* fix/hda:
ALSA: hda - Set function_id only on FG nodes
ALSA: hda - Add upper-limit of mixer amp for AD1884A-laptop model, too
ALSA: hda - Fix headphone-detection on some machines with STAC/IDT codecs
ALSA: hda_intel.c - Consolidate bitfields
I notice that the fixes were merged, minus one:
sound/soc/codecs/wm9705.c: At top level:
sound/soc/codecs/wm9705.c:445: warning: initialization from incompatible pointer type
so you might find this trivial patch useful.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
(Re)set function_id only from the value on FG nodes.
The current code overrides the value with the last widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The original implementation of the constraints were good against sane
applications.
If the opening sequence is:
stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the
constraints are set correctly for stream2.
But if the sequence is:
stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2
would receive constraint rate = 0, sample_bits = 0, since the stream1 has not
yet called hw_params...
The command to trigger this event:
gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false
This patch does some 'black magic' in order to always set the correct
constraints and sets it only when it is needed for the other stream.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My email address is going to expire soon so update it. Adding also
Peter Ujfalusi <peter.ujfalusi@nokia.com> as a second contact to OMAP core
drivers since I won't have anymore access to non-public OMAP documentation
in the future and Peter is working with these drivers as well.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Those macros are just screwed as soon as CONFIG_PXA25x is enabled.
This patch
- changes ssp_set_scr to take an ssp_dev pointer instead of ssp_device
- adds a corresponding ssp_get_scr function.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Within 2.6.30's mergewindow, struct urb's transfer_buffer_length has become
unsigned. This changed an "int > int" comparision to an "unsigned > int" one
in snd_usb_122l.
Fix this by using a local int variable instead of urb->transfer_buffer_length
in comparisions.
Shorten playback_prep_freqn() a bit and tweak error-paths in
usb_stream_prepare_playback().
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DSP_A mode is similar to the DSP_B, but the MSB is delayed with
one bclk (appears after the FS pulse and not under it).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use single-phase mode for the DSP mode and keep the dual phase
mode for the I2S mode.
The mono (1 channel) mode already used single phase mode,
now it is more cleaner. There is no need to configure the
second phase, when the single phase is used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using inverted FS polarity in OSK5912 must be an error since TLV320AIC23
do not have support for inverted polarities. This is mostly due the hassle
with the DSP formats in OMAP McBSP DAI and inversion on OMAP side probably
just made this configuration working at some point.
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>
The DSP format wasn't still correct in OMAP McBSP DAI even after the commit
bd25867a6c.
Thanks to Peter Ujfalusi <peter.ujfalusi@nokia.com> for noticing and being
part of the fix. Now the FS length definition is more clear by defining
it with FWID(0).
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix accidental change of <mach/regs-gpio.h> to
<plat/regs-gpio.h> in s3c2412-i2s.c
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the build error in s3c-i2s-v2.c caused by
a change to the snd_soc_dai ops field.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The definition of s3c_i2sv2_iis_calc_rate was never
renamed from s3c2412_iis_calc_rate, so rename this
to allow the build to work.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build errors in sound/soc/s3c24xx/jive_wm8750.c
from changes to ASoC.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pxa_ssp_set_dai_fmt() currently has an early exit if the desired format
equals the current configuration. This is correct behaviour unless this
function is called with a zero value parameter for the first time.
Zero is a valid value for this function, but the early exit is bogus in
this case.
Hence, set priv->dai_fmt to -1 in the beginning so we can configure the
port.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: pHilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When the headphone can have no unique DAC, the current code doesn't
check the HP-detection although it should. Put the hp-detection check
before the DAC check to fix this bug.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some limited volume controls (mostly simple attenuations) have only two
settings so the ASoC info functions misreport them as booleans. Since
we currently have no better information check for " Volume" in the
control name and always report any controls matching as being integer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Check the length to copy via strlen() beforehand to avoid the stack
corruption, or use strlcpy() to be safe in HD-audio codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode.
In the HD-audio mode, no multiple streams are supported by just it
behaves like a normal HD-audio device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Subject says it all. Briefly, use hp_only for another Dell Inspiron 8600.
Reference: Ubuntu #41015 (https://launchpad.net/bugs/41015)
Signed-off-by: Daniel T Chen <seven.steps@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While cleaning up quirks, I noticed that there is a duplicated quirk for
the SSID 0x103c0934. Looking back through the bug reports, I've concluded
that there is only one necessary quirk (hp_mute_led), so this patch
removes the conflicting one.
Reference: Ubuntu #44066 (https://launchpad.net/bugs/44066)
Signed-off-by: Daniel T Chen <seven.steps@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit fa00e046b4
added a new bitfield not adjacent to other
bitfields in the same struct. Moved the new one.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the key value generation for get/set amp verbs. The upper bits of
the parameter have to be combined with the verb value to be unique for
each direction/index of amp access.
This fixes the resume problem on some hardwares like Macbook after
the channel mode is changed.
Tested-by: Johannes Berg <johannes@sipsolutions.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* topic/memdup_user:
ALSA: sound/pci: use memdup_user()
ALSA: sound/usb: use memdup_user()
ALSA: sound/isa: use memdup_user()
ALSA: sound/core: use memdup_user()
* 'master' of git://git.alsa-project.org/alsa-kernel:
[ALSA] intel8x0: add one retry to the ac97_clock measurement routine
[ALSA] intel8x0: fix wrong conditions in ac97_clock measure routine
[ALSA] intel8x0: do not use zero value from PICB register
[ALSA] intel8x0: an attempt to make ac97_clock measurement more reliable
[ALSA] pcm-midlevel: Add more strict buffer position checks based on jiffies
[ALSA] hda_intel: fix unexpected ring buffer positions
Added the models for quirk bitmask 1734:110x and 1734:113x of
Fujitsu laptops.
This will fix the model detection for Amilo Xa3540.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that on some hardware platforms, the first measurement is wrong.
This patch adds second measurement to this case.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Currently there are two possible platform datas for the PXA AC97 driver:
one supported by the generic AC97 driver only which provides callbacks
to allow board-specific configuration at stream startup and teardown,
and another for pxa2xx-ac97-lib which allows configuration of the reset
GPIO for PXA2xx CPUs.
Obviously this won't actually work when using the generic AC97 driver
since the drivers will attempt to parse the platform data in both
formats. Fix this by merging the two structures.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Cc: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Don't call snd_jack_report at release of sigmatel and conexnat codecs
which results in Oops at unloading the module.
The Oops is triggered by the power-up sequence during the free due to
the pincfg restoration. Since the power-up sequence is involved with
the unsol handling, the jack reporting may be issued during that.
The Oops occurs with this jack reporting because the jack instances
have been already released but the codec doesn't do the proper
book-keeping.
This patch adds the book-keeping of jack instances to avoid the access
to bogus pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added private_data and private_free fields to struct snd_jack so that
the caller can assign the data. It'll be helpful for avoiding the
double-free of the jack instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for later cards based on CompuMedia ASC-9408 chipsets.
These cards were produced by Gallant.
This patch make the OSS aedsp16 driver redundant.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
prototype of a driver for the digigram lx6464es 64 channel ethersound
interface.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the second go through of the old DMA_nBIT_MASK macro,and there're not
so many of them left,so I put them into one patch.I hope this is the last round.
After this the definition of the old DMA_nBIT_MASK macro could be removed.
Signed-off-by: Yang Hongyang <yanghy@cn.fujitsu.com>
Cc: Russell King <rmk@arm.linux.org.uk>
Cc: Tony Lindgren <tony@atomide.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Greg KH <greg@kroah.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
It seems that the zero value from the PICB (position in current buffer)
register is not reliable. Use jiffies to correct returned value
from the ring buffer pointer callback.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- use monotonic posix clock to measure time
- try to avoid reading zero from PICB (position in current buffer) register
- show also measured samples
- when clock is near 41000 or 44100, use exactly these values
(they appears to be reference clocks for hardware manufacturers)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The WM8960 is a low power, high quality stereo codec designed for
portable digital audio applications.
Stereo class D speaker drivers provide 1W per channel into 8W loads.
Guaranteed low leakage, excellent PSRR and pop/click suppression
mechanisms enable direct battery connection for the speaker supply.
The device also integrates a complete microphone interface and a stereo
headphone driver. External component requirements are drastically
reduced as no separate microphone, speaker or headphone amplifiers are
required. Advanced on-chip digital signal processing performs automatic
level control for the microphone or line input.
Stereo 24-bit sigma-delta ADCs and DACs are used with low power
over-sampling digital interpolation and decimation filters and a
flexible digital audio interface.
The master clock can be input directly or generated internally by an
onboard PLL, supporting most commonly-used clocking schemes.
This driver was originally written by Liam Girdwood, with substantial
subsequent additions and updates for feature completeness and changes in
the ASoC framework from me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low)
SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low)
SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High)
SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High)
SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0).
This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and
DSP_B modes.
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This ensures that we sync with the DAPM powerdown sequencing properly
and don't need to bounce the power on the voice DAC so often.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is simple code motion, intended to support future refactoring of
the DAPM algorithms and (more immediately) the additon of events for
DACs and ADCs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some drivers like Intel8x0 or Intel HDA are broken for some hardware variants.
This patch adds more strict buffer position checks based on jiffies when
internal hw_ptr is updated. Enable xrun_debug to see mangling of wrong
positions.
As a side effect, the hw_ptr interrupt update routine might do slightly better
job when many interrupts are lost.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I found two issues with ICH7-M (it should be related to other HDA chipsets
as well):
- the ring buffer position is not reset when stream restarts (after xrun) -
solved by moving azx_stream_reset() call from open() to prepare() callback
and reset posbuf to zero (it might be filled with hw later than position()
callback is called)
- irq_ignore flag should be set also when ring buffer memory area is not
changed in prepare() callback - this patch replaces irq_ignore with
more universal check based on jiffies clock
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Due to the process and communications issues with the 2.6.30 S3C
platform merges none of the underlying arch/arm code for S3C64xx audio
support made it into mainline, rendering the drivers useless. Disable
them in Kconfig to avoid user confusion - users patching in the required
support can always reenable this too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add DSP_A interface format support by setting the LRP bit in
DSP mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Cc: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8988 is a low power, high quality stereo CODEC designed for
portable digital audio applications.
The device integrates complete interfaces to 2 stereo headphone or line
out ports. External component requirements are drastically reduced as no
separate headphone amplifiers are required. Advanced on-chip digital
signal processing performs graphic equaliser, 3-D sound enhancement and
automatic level control for the microphone or line input.
The WM8988 can operate as a master or a slave, with various master clock
frequencies including 12 or 24MHz for USB devices, or standard 256fs
rates like 12.288MHz and 24.576MHz. Different audio sample rates such as
96kHz, 48kHz, 44.1kHz are generated directly from the master clock
without the need for an external PLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.
A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (36 commits)
ALSA: hda - Add VREF powerdown sequence for another board
ALSA: oss - volume control for CSWITCH and CROUTE
ALSA: hda - add missing comma in ad1884_slave_vols
sound: usb-audio: allow period sizes less than 1 ms
sound: usb-audio: save data packet interval in audioformat structure
sound: usb-audio: remove check_hw_params_convention()
sound: usb-audio: show sample format width in proc file
ASoC: fsl_dma: Pass the proper device for dma mapping routines
ASoC: Fix null dereference in ak4535_remove()
ALSA: hda - enable SPDIF output for Intel DX58SO board
ALSA: snd-atmel-abdac: increase periods_min to 6 instead of 4
ALSA: snd-atmel-abdac: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: cleanup registers when removing driver
ALSA: snd-atmel-ac97c: do a proper reset of the external codec
ALSA: snd-atmel-ac97c: enable interrupts to catch events for error reporting
ALSA: snd-atmel-ac97c: set correct size for buffer hardware parameter
ALSA: snd-atmel-ac97c: do not overwrite OCA and ICA when assigning channels
ALSA: snd-atmel-ac97c: remove dead break statements after return in switch case
ALSA: snd-atmel-ac97c: cleanup register definitions
...
Replace all DMA_24BIT_MASK macro with DMA_BIT_MASK(24)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_28BIT_MASK macro with DMA_BIT_MASK(28)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_30BIT_MASK macro with DMA_BIT_MASK(30)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_31BIT_MASK macro with DMA_BIT_MASK(31)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Fix for compillation error introduced by the constrain patch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add powerdown sequence for VREF using a shared jack when the headphone
is present and the microphone isn't on.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added an else part to check
SNDRV_MIXER_OSS_PRESENT_CVOLUME for MIC (slot 7)
in commit 36c7b833e5
Similarly, checks and volume control is required for
SNDRV_MIXER_OSS_PRESENT_CSWITCH and SNDRV_MIXER_OSS_PRESENT_CROUTE
as well.
Signed-off-by: Deepika Makhija <deepika.makhija@einfochips.com>
Signed-off-by: Viral Mehta <viral.mehta@einfochips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To enable periods shorter than 1 ms, we have to make sure that short
periods are only available for alternate settings that have a small
enough data packet interval. Furthermore, the code that aligns URBs to
USB frames is now superfluous.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The data packet interval needs to be available in the audioformat
structure, together with the other audio format parameters, so that it
can be used to influence ALSA hardware parameters.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This removes the check_hw_params_convention() function because
1) it is not necessary, as the hw_rule_* functions also work correctly
(i.e., as no-ops) when the device supports all combinations of the
audio format parameters; and
2) it would become too complex when adding a fourth altsetting-dependent
hardware parameter, as this would require another three loops to
check dependecies with rate/channels/format.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
