When quirks are applied, the numbers of output pins in autocfg aren't
set up properly but only pin arrays are changed. Let's fix it up so that
the rest of the parser can use autocfg.line_outs & co safely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Purpose of this virtual Detection pin is to keep codec bias on whenever the
GPIO or jack detection features are needed.
Jack detection needs a mic bias so machine drivers can construct a following
route for instance for keeping the path and codec bias on:
"Input Jack" -> "Mic Bias xV" -> "Detection" -> detection block inside codec.
For the GPIO the machine driver can force the pin on with
snd_soc_dapm_force_enable_pin.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch merges all three patch_*hdmi variants to the single HDMI
parser. There is only one snd-hda-codec-hdmi module now.
In this patch, the behavior of each parser isn't changed much.
The old ATI parser still doesn't use the dynamic parser yet.
In later patches, they'll be cleaned up.
Also, this patch gets rid of the individual snd-hda-eld module and
builds into snd-hda-codec-hdmi, since this is referred only from the
HDMI parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes multiple bugs and a typo, occurred during the multi-
component transition.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The sh/siu ASoC driver doesn't compile because of a function defined static in
the source and extern in a header. Remove the unneeded declaration in the
header.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The clkdev API doesn't use .name and .id members of struct clk for clock
lookup. Instead clocks should be added to a lookup list. Without this patch
audio om the Migo-R board fails silently.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Dzianis Kahanovich <mahatma@eu.by>
[Modified to move the location of the table]
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The external mic jack for auto-mic switch must be really an external
jack and with a presense-detection capability. This patch makes the
check more paranoia.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the helper function to give the input-pin attribute for jack
connectivity and location. This simplifies checks of input-pin jacks
a bit in some places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM proc files may open a race against substream close, which can
end up with an Oops. Use the open_mutex to protect for it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pm_qos_request isn't freed properly when OSS PCM emulation is used
because it skips snd_pcm_hw_free() call but directly releases the
stream. This resulted in Oops later.
Tested-by: Simon Kirby <sim@hostway.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"uinfo->value.enumerated.item" is an unsigned int. If it's negative
when we do the comparison:
if ((int)uinfo->value.enumerated.item >= cval->max)
then we would read past the end of the array on the next line.
I also changed the strcpy() to strlcpy() out of paranoia.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Through the transition of autocfg to individual inputs array, I forgot
to rewrite the argument passed to alc_set_input_pin(). This resulted in
wrongly setup input pins. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
List registered platforms in debugfs to improve debugability of machine
drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow the user to inspect the list of registered DAIs at runtime to
improve diagnostics for machine driver setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Help with diagnostics for machine driver setup by listing all the
registered CODECs in debugfs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
BugLink: http://launchpad.net/bugs/640254
In some cases a magic processing coefficient is needed to enable
the internal speaker on Dell M101z. According to Realtek, this
processing coefficient is only present on ALC269vb.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When user want to change the card id to the same string
on the card via /sys/class/sound/cardX/id, do not
report error. Instead return with success without
doing anything.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Used only when CONFIG_SND_DEBUG=y
sound/usb/mixer.c: In function 'get_min_max':
sound/usb/mixer.c:762: warning: unused variable 'chip'
Reported-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of Intel controllers work as generic HD-audio without quirks,
and it'll be hopefully so in future. Let's mark pci id with the
PCI_CLASS_MULTIMEDIA_HD_AUDIO for Intel so that the driver will work
with any new control chips in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the preliminary support for new Conexant audio codecs with
14f1:5097, 14f1:5098, 14f1:50a1, 14f1:50a2, 14f1:50ab, 14f1:50ac,
14f1:50b8 and 14f1:50b9.
Unlike other Conexant parsers, this is designed to be mostly automatic,
parsing from BIOS pin configurations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For avoiding the click noises at power-saving, set some COEF values
for ALC269* codecs.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8985 is a low power, high quality, feature-rich stereo
CODEC designed for portable multimedia applications that
require low power consumption and high quality audio.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Complete the phasing out of aic3x_read_reg_cache, aic3x_write_reg_cache,
aic3x_read and aic3x_write calls.
This patch uses in aic3x_read the codec->hw_read that points to a function
implemented by soc-cache. Only use for aic3x_read is if wanting to read
volatile bits from those registers that has both read-only and read/write
bits. All other cases should use snd_soc_read.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Continue phasing out aic3x_read_reg_cache, aic3x_write_reg_cache, aic3x_read
and aic3x_write calls.
This patch takes the soc-cache in use and removes aic3x_read_reg_cache and
aic3x_write.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Start phasing out aic3x_read_reg_cache, aic3x_write_reg_cache, aic3x_read and
aic3x_write calls in order to switch to soc-cache helpers.
This patch replaces aic3x_read_reg_cache and aic3x_write with snd_soc_read
and snd_soc_write. This is basically null-op since .read and .write in
soc_codec_dev_aic3x points to aic3x_read_reg_cache and aic3x_write.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Like other coworkers, I'm about leave Mandriva/Edge-It so I'm changing
my mail address to use my personal one.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This assignment is done by the snd_soc_register_codec so there is no need
to redo it in probe function of a codec driver.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The usage of the BKL in the OSS sound drivers is
trivial, and each of them only locks against itself,
so it can be turned into per-driver mutexes.
This is the script that was used for the conversion:
file=$1
name=$2
if grep -q lock_kernel ${file} ; then
if grep -q 'include.*linux.mutex.h' ${file} ; then
sed -i '/include.*<linux\/smp_lock.h>/d' ${file}
else
sed -i 's/include.*<linux\/smp_lock.h>.*$/include <linux\/mutex.h>/g' ${file}
fi
sed -i ${file} \
-e "/^#include.*linux.mutex.h/,$ {
1,/^\(static\|int\|long\)/ {
/^\(static\|int\|long\)/istatic DEFINE_MUTEX(${name}_mutex);
} }" \
-e "s/\(un\)*lock_kernel\>[ ]*()/mutex_\1lock(\&${name}_mutex)/g" \
-e '/[ ]*cycle_kernel_lock();/d'
else
sed -i -e '/include.*\<smp_lock.h\>/d' ${file} \
-e '/cycle_kernel_lock()/d'
fi
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modifying an object twice without an intervening sequence point is
undefined.
Signed-off-by: Andreas Schwab <schwab@linux-m68k.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Each of the two PCM controllers need to be registered during probe
with appropriate 'name' of the dai driver.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the SMDK64xx boards have two audio subsystems using the board
name as the card name by itself isn't so user friendly as it might
be.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add a quirk for laptop Toshiba Satellite C650D to have proper external HP and
external Mic support.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch modify FIFO_DIPSTICK value of PCM TX FIFO to be a optimal one.
Privious value (0x20) did not support 'Almost_full' of PCM FIFO for the DMA
request.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When PCM capture, sound recorded abnormally because of RX FIFO
threshold settings are missing. So, This patch modify PCM RX FIFO
setting codes same as TX.
And for DMA, if PCM RXFIFO_DIPSTICK is not '0', it doesn't effect
to DMA request, because DMA refer RX_FIFO_EMPTY flag as the DMA
request.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a simple off-by-one bug, the size of the register cache is
incorrectly set to the maximum register index. Fix it by adding one.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With this change it's not a error to call wl1273_set_audio_route
when the codec is active if the new routing value is the same
as the current active setting.
Signed-off-by: Matti J. Aaltonen <matti.j.aaltonen@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
There is only need to enable/disable once the PLL when the bias is going
between on, prepare, standby and off states.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The reg_cache_size is the number of elements in the register cache,
not the size of the cache itself. This is not a problem if the size
of each element of the cache is 1 byte but it matters in any other
case.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch solve below report from Guennadi.
But I didn't remove #include <sound/sh_fsi.h>.
Because it have FSI_PORT_B define which is used on this file.
> +#include <linux/platform_device.h>
> +#include <sound/sh_fsi.h>
> +#include <video/sh_mobile_hdmi.h>
Now that everything is done with strings - do you still need these
headers?
Reported-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM controller platform devices are registered by the
name 'samsung-pcm', so use the same in the CPU driver.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Drop the invalid -dai suffix appended to the Samsung AC97 CPU DAI.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current SND_FSI_xxx menu attributes were bool,
but it should be tristate.
This patch solve below report from Guennadi
"bool" means, if someone is linking the whole ASoC into the kernel, they
will not be able to build this as a module. Not a big deal, but you're
stealing some freedom from the user.
Reported-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the new Traktor Kontrol S4 by Native
Instruments. It features a new audio data streaming model, MIDI
in and out ports, a huge number of 174 dimmable LEDs, 96 buttons
and 46 absolute encoder axis, including some rotary encoders.
All features are supported by the driver now.
Did some code refactoring along the way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the mic pins are assigned to the same location, we can omit the
redundant location prefix like "Front" or "Rear".
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch improves the input-source label strings to be generated from
the pin information instead of fixed strings per AUTO_PIN_* type.
This gives more suitable labels, especially for mic and line-in pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We can assign multiple pins to a single role now, let's reduce the
redundant FRONT_MIC and FRONT_LINE. Also, autocfg->input_pins[] is
no longer used, so this is removed as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep char array in the input_mux item itself instead of pointing to
an external string. This is a preliminary work for improving the
input-mux name based on the pin role.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update the Xonar config texts with the latest information about the
Xonar DS, HDAV1.3 Slim, and Xense.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the select directive does not handle indirect dependencies, select
those explicitly in the driver sections.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the possibility to route a mix of the two channels of stereo data to
the center and LFE outputs. Due to a WM8766 restriction, all surround
and back channels also get the mixed L/R signal in this case.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Automatically mute the speaker outputs as long as a headphone is plugged.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the polarity of the headphone detection pin is known, replace
the debugging message with a proper jack plug input device.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the correct number, register bits, and names for the input switches.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check for the volume update latch bit was accidentally in the wrong
function, where it would prevent the MSB from being written, instead of
correctly ignoring it for cached values.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By adding the subwoofer as a speaker pin, it is treated correctly when auto-muting.
BugLink: https://launchpad.net/bugs/611803
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If we pass in a device which is higher than SNDRV_RAWMIDI_DEVICES then
the "next device" should be -1. This function just returns device + 1.
But the main thing is that "device + 1" can lead to a (harmless) integer
overflow and that annoys static analysis tools.
[fix the case for device == SNDRV_RAWMIDI_DEVICE by tiwai]
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a fixup table for ALC262 codec containing the entry for FSC
Celsius H270. Now both headphone jacks are detected properly as
headphones.
Reference: Novell bnc637263
https://bugzilla.novell.com/show_bug.cgi?id=637263
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the alc262 auto-parser to allow multiple pins
assigned for a single purpose (line-out, headphone or speaker).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently headphone auto-mute using alc_automute_pin() assumes only
the single pin used for the headphone output. Since there are devices
with multiple headphone jacks, we need to check all these pins there,
too.
Also this patch merges the common code between alc_automute_pin() and
alc_automute_amp() helper functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_hda_parse_def_config(), some unused values may remain in hp_pins[]
array during the headphone-reassignment workaround. This patch clears
the unused array members.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_parse_pin_def_config() has some workaround for re-assigning
some pins declared as headphones to line-outs. This didn't work properly
for some cases because it used memmove() stupidly wrongly.
Reference: Novell bnc#637263
https://bugzilla.novell.com/show_bug.cgi?id=637263
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes sparse warning due non declaration of static function
sound/soc/omap/omap-mcbsp.c:783:5: warning: symbol 'omap_mcbsp_st_info_volsw' was not declared. Should it be static?
Signed-off-by: G, Manjunath Kondaiah <manjugk@ti.com>
Cc: alsa-devel@alsa-project.org
Cc: Jarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Tony Lindgren <tony@atomide.com>
Cc: Nishanth Menon <nm@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Usage of 256 as clkdiv gives better rounding error (<1%)
for 16khz and 48khz
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The error handling in snd_seq_oss_open() has several bad codes that
do dereferecing released pointers and double-free of kmalloc'ed data.
The object dp is release in free_devinfo() that is called via
private_free callback. The rest shouldn't touch this object any more.
The patch changes delete_port() to call kfree() in any case, and gets
rid of unnecessary calls of destructors in snd_seq_oss_open().
Fixes CVE-2010-3080.
Reported-and-tested-by: Tavis Ormandy <taviso@cmpxchg8b.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver doesn't probe the device properly because of left-over cfg[]
that isn't used at all for msnd-classic device. This is only for msnd-
pinnacle.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changing the way the input controls are named using port connection
type and jack location info.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding support for digital MIC in 92HD83/90/91XXX codecs family.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
EeePC 1001HAG has a similar problem like other ASUS machine, which doesn't
set the codec SSID properly for indicating the beep capability.
To enable PC-beep again, put this to the whitelist.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the wrong "return" in the loop, a capture substream won't be
released at disconnection properly if the device is capture only and has
no playback substream. This caused Oops occasionally at the device
reconnection.
Reported-by: Kim Minhyoung <minhyoung.kim@lge.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Line and Mic inputs cannot be used at the same time, so the driver
has to automatically disable one of them if both are set. However, it
forgot to notify userspace about this change, so the mixer state would
be inconsistent. To fix this, check if the other control gets muted,
and send a notification event in this case.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Nathan Schagen
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the WM8776 chip, this driver uses a different sample format and
more features than the Windows driver. When rebooting from Linux into
Windows, the latter driver does not reset the chip but assumes all its
registers have their default settings, so we get garbled sound or, if
the output happened to be muted before rebooting, no sound.
To make that driver happy, hook our driver's cleanup function into the
shutdown notifier and ensure that the chip gets reset.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Nathan Schagen
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not needed with multi-component.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Clean up the playback pointer callback function a bit, and make the
pointer check more strictly to avoid bogus pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is adangling code in wm8753_probe which is never executed.
Remove it.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Acked-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Null pointer dereference will occur from *setup = pdata->setup if pdata
is not set. Fix this by moving assignments from pdata inside non-null case.
Thanks to Jiri Slaby <jirislaby@gmail.com> for noticing.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Jiri Slaby <jirislaby@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The Audio Class v2 support code in 2.6.35 added checks for the
bInterfaceProtocol field. However, there are devices (usually those
detected by vendor-specific quirks) that do not have one of the
predefined values in this field, which made the driver reject them.
To fix this regression, restore the old behaviour, i.e., assume that
a device with an unknown bInterfaceProtocol field (other than
UAC_VERSION_2) has more or less UAC-v1-compatible descriptors.
[compile warning fixes by tiwai]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk to make the BOSS ME-25 work.
Many thanks to Kees van Veen.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for the Roland/Cakewalk A-300PRO/A-500PRO/A-800PRO keyboards.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a quirk for the other logical device of the PCR-1 so that not only
the MIDI interface but also the audio interface works.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch revive ak4642_snd_controls which was removed on
f0fba2ad1b
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a call to of_node_put in the error handling code following a call to
of_parse_phandle.
This patch also moves the existing call to of_node_put tothe end of the
error handling code, to make it possible to jump to of_node_put without
doing the other cleanup operations. These appear to be disjoint
operations, so the ordering doesn't matter.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@r exists@
local idexpression x;
expression E,E1,E2;
statement S;
@@
*x =
(of_find_node_by_path
|of_find_node_by_name
|of_find_node_by_phandle
|of_get_parent
|of_get_next_parent
|of_get_next_child
|of_find_compatible_node
|of_match_node
|of_find_node_by_type
|of_find_node_with_property
|of_find_matching_node
|of_parse_phandle
)(...);
...
if (x == NULL) S
<... when != x = E
*if (...) {
... when != of_node_put(x)
when != if (...) { ... of_node_put(x); ... }
(
return <+...x...+>;
|
* return ...;
)
}
...>
(
E2 = x;
|
of_node_put(x);
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.uo.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow snd-soc-kirkwood autoloading by adding an alias.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For devices with more than one control interface, let's assume the first
one contains the audio controls. Unfortunately, there is no field in any
of the descriptors to tell us whether a control interface is for audio
or MIDI controls, so a better check is not easy to implement.
On a composite device with audio and MIDI functions, for example, the
code currently overwrites chip->ctrl_intf, causing operations on the
control interface to fail if they are issued after the device probe.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The M-Audio Fast Track Ultra series devices did not play sound correctly
at 44.1/88.2 kHz. Changing the output endpoint attribute to adaptive
fixes this.
Signed-off-by: Felix Homann <fexpop@web.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This new model adds the following functionality to HP G60:
- Automute of internal speakers
- Autoswitch of internal/external mics
- Remove SPDIF not physically present
BugLink: http://launchpad.net/bugs/587388
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow selection of the channel used for input to the AIFnDAC signals.
This isn't integrated into DAPM since we treat the data as a single
mono channel until just beyond this selection so it ends up having
no visible effect on the routing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
multicomponent support added/changed some device name but added some typos,
breaking existing OpenRD Client support.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modify dai link
- platform_name: sh_fsi/sh_fsi2 are used for FSI driver
- codec_name: ak4642/ak4643 are used for ak4642 driver
This is quick hack. I should modify it more wisely in future
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Disable some codec modules in standby mode, completely disable
codec in off mode to save some power.
Fix suspend/resume: mark mixer regs as dirty on resume to
restore mixer values, otherwise driver produces no sound
(master is muted by default).
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes up the au1x audio platform after the multi-component
merge:
- compile fixes and updates to get DB1200 platform audio working again,
- removal of global variables in AC97/I2S/DMA(PCM) modules.
The AC97 part is limited to one instance only for now due to issues
with getting at driver data in the soc_ac97_ops.
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added snd_hda_get_input_pin_label() helper function to return the
string that can be used for control or capture-source ids.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the new fields to contain all input-pins to struct auto_pin_cfg.
Unlike the existing input_pins[], this array contains all input pins
even if the multiple pins are assigned for a single role (i.e. two
front mics). The former input_pins[] still remains for a while, but
will be removed in near future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
patch_via.c has redundant codes for parsing the input-pins. Although
they are pretty similar, but all implemented in different functions
just because of hard-coded ids and slight incompatibilities.
This patch refactors the codes to use the common helper function,
resulting in the reduction of many lines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of defining each content as a separate struct, put all into the
definition of struct alc_fixup arrays so that reader doesn't go back to
see the definition again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There were some new formats added in commit 15c0cee6c8 "ALSA: pcm:
Define G723 3-bit and 5-bit formats". That commit increased
SNDRV_PCM_FORMAT_LAST as well. My concern is that there are a couple
places which do:
for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) {
if (dummy->pcm_hw.formats & (1ULL << i))
snd_iprintf(buffer, " %s", snd_pcm_format_name(i));
}
I haven't tested these but it looks like if "i" were equal to
SNDRV_PCM_FORMAT_G723_24 or higher then we might read past the end of
the array.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently output controls are not uniform. Some routes are adjusted by
mono controls that don't match to associated mixer switch, many routes are
not covered at all and stereo controls have following variants:
- L-to-L & R-to-R
- R-to-L & R-to-R
- L-to-L & R-to-L
This patch attempts to fix these issues. First, for the convenience, only
direct L-to-L, R-to-R and [L | R]-to-Mono routes are controlled by the
stereo controls. This logic is also used with the output pin mute controls
so all of them except mono output are controlled by stereo switches.
Then rest of the swapped L-to-R and R-to-L routes are controlled by the
mono controls that map to mixer switches with a same name. Mixers can then
associate these switches and volumes together.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
It turned out that the output mixers and their routes were misdefined: They
are not mixing output pins to internal signals but opposite. This has worked
for direct left-to-left and right-to-right routes since for those there are
complete routes. For swapped left-to-right and right-to-left routes this is
not working since there are no routes defined between them.
Another consequence is that those misdefined mixers are incorrectly routed
to several output pins leading unnecessary pin powerings even if there is no
route active to them.
Fix these by reimplementing the output mixers and routes as they are in
hardware. For completeness add also a few missing links between internal
signals and outputs.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Each output pin has 7 consecutive control registers in tlv320aic3x register
map. First 6 of them control the signal mixing and one is for output level
and power control.
Sort these registers as they are sorted clearly in hardware, it makes also
definitions more readable and easier to pinpoint missing register
definitions.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Bit 3 in output pin_CTRL register mutes the whole output pin not just the
route from DAC so remove misleading DAC from control name. Currently only
"Line[L | R] Playback Switch" were correct.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The spinlock lock in sound_timer.c is used without initialization.
Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If hw error is ignored, status is updated with invalid info.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I think this is a typo, debugfs_pop_time should not be executable.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimloogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The attached patch enables playback on a Sony VAIO machine.
BugLink: http://launchpad.net/bugs/618271
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "priv" allocated in pxa_ssp_probe() should be kfreed in pxa_ssp_remove().
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In synchronous mode the SSI_SRCCR values are ignored. Instead
SSI_STCCR must be used for both receiving and transmitting.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Makefile and Kconfig updates for WL1273 appear to have been mising
from the patch posted, add them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This makes it that little bit easier to spot the diagnostics in the
logs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Speaker amplifier is controlled by TWL4030 GPIO which may sleep. Therefore
use gpio_set_value_cansleep to get rid of runtime warning that is introduced
after the commit 9c4ba94 and to get a stack trace if ever executing this
code in atomic context.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
aic3x_init does a soft reset first and thus TLV320AIC3x GPIO setup must be
done after doing the basic init. Before multi-component the init was done
at i2c probe time and GPIO setup at soc probe time.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds quirk for the Lenovo S10-3t so the headphone &
microphone jacks will now work.
Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device is similar to the M-Audio Delta 1010LT in that it uses the
AK4524VF ADC/DAC, but it does not use the CS8427 for SPDIF.
The SPDIF appears to be set up correctly, but I am not able to test it
as I do not have any devices that use it.
This patch makes the ADC/DAC's and the hardware mixer visible to apps
such as alsamixer and envy24control.
Signed-off-by: Garnet MacPhee <dhubsith@comcast.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'struct of_device' no longer exists, and its functionality has been merged
into platform_device. Update the MPC8610 HPCD audio drivers (fsl_ssi, fsl_dma,
and mpc8610_hpcd) accordingly.
Also add a #include for slab.h, which is now needed for kmalloc and kfree.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Otherwise we generate worrying (but benign) warnings for amps.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This is an ALSA codec for the Texas Instruments WL1273 FM Radio.
Signed-off-by: Matti J. Aaltonen <matti.j.aaltonen@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The Freescale P1022 is a dual-core e500-based SOC with multimedia capabilities,
specifically the same SSI audio controller on the MPC8610. The P1022 DS
reference board includes a P1022 and a Wolfson Microelectronics WM8776
codec.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix reference to moved header file, which was unused anyway.
This change fixes below build error:
CC sound/soc/pxa/pxa2xx-ac97.o
sound/soc/pxa/pxa2xx-ac97.c:27:24: error: pxa2xx-pcm.h: No such file or directory
make[3]: *** [sound/soc/pxa/pxa2xx-ac97.o] Error 1
make[2]: *** [sound/soc/pxa] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Haojian Zhuang <haojian.zhuang@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the tlv320aic3007 codec to the tlv320aic3x
driver.
The tlv320aic3007 is similar to the aic31, but has an additional class-D
speaker amp. The speaker amp control register overlaps with the mono
output register of other codecs in this family, so we add logic to
identify the actual codec being registered to set things up accordingly.
Signed-off-by: Randolph Chung <tausq@parisc-linux.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some codecs have separate DAIs for playback and capture, so the DMA driver
should allocate a DMA buffer only for the streams that are valid when the
driver is opened.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In e740_init(), we call gpio_request() for
GPIO_E740_MIC_ON, GPIO_E740_AMP_ON and GPIO_E740_WM9705_nAVDD2.
We should free the these gpio accordingly in e740_exit().
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The new sticky PCM parameter introduced the delayed clean-ups of
stream- and channel-id tags. In the current implementation, this check
(adding dirty flag) and actual clean-ups are done only for the codec
chip. However, with HD-audio architecture, multiple codecs can be
on a single bus, and the controller assign stream- and channel-ids in
the bus-wide.
In this patch, the stream-id and channel-id are checked over all codecs
connected to the corresponding bus. Together with it, the mutex is
moved to struct hda_bus, as this becomes also bus-wide.
Reported-and-tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Intel and Nvidia HDMI codec drivers have own implementations of
sticky PCM parameters. Now HD-audio core part already has it,
thus both setups conflict. The fix is simply remove the part in
patch_intelhdmi.c and patch_nvhdmi.c and simply call
snd_hda_codec_setup_stream() as usual.
Reported-and-tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the process of unification of codec DAI names while implementing
multi-component, the CX20442 codec DAI has been renamed to "cx20442-hifi".
This new name seems not adequate for a 8kHz voice codec.
Use a better name, "cx20442-voice", as suggested by Liam Girdwood.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The tlv320aic3x codec driver only supports symmetric rates for capture/
playback. Set the flag in the DAI accordingly.
Signed-off-by: Randolph Chung <tausq@parisc-linux.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current code in pcm_lib.c do all checks using only the position
in the ring buffer. Unfortunately, where the interrupts gets delayed or
merged into one, we need another timing source to check when the
buffer size boundary overlaps to avoid the wrong updating of the
ring buffer pointers.
This code uses jiffies to check the right time window without any
performance impact.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
BugLink: https://bugs.launchpad.net/bugs/619439
This ThinkPad model needs External Amplifier muted for audible playback,
so set the inv_eapd quirk for it.
Reported-and-tested-by: Dennis Bell <dennis.bell@parkerg.co.uk>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just added new codec ids. These are almost compatible with existing ones.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add code that programs the DMA and SSI controllers differently based on the
FIFO depth of the SSI.
The SSI devices on the MPC8610 and the P1022 are identical in every way except
one: the transmit and receive FIFO depth. On the MPC8610, the depth is eight.
On the P1022, it's fifteen. The device tree nodes for the SSI include a
"fsl,fifo-depth" property that specifies the FIFO depth.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
88PM860x codec is used in Marvell saarb development board. 88PM860x codec
is used as master mode for SSP communication. Only I2S format is supported.
Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add 88PM860x codec driver. 88PM860x codec supports two interfaces. And it
also supports headset/mic/hook/short detection.
Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the SSC is already being registered as a device under arch and
the DMA and SSC hardware are pretty much the same provide a simplified
device registration function for the Atmel SSC which will add the
ASoC-specific devices within the ASoC code, parenting the SSC device
off the actual SSC device. Also use it in the sam9g20-ek driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
A couple of typos in the multi-component conversion.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Instead of unconditionally enabling the crystal oscillator on the WM8731
only enable it when explicitly selected via set_sysclk(), allowing machine
drivers to specify that they drive a clock into MCLK alone. This avoids
any conflicts between the oscillator and the external MCLK source and saves
power for systems which do not need the oscillator.
This should also deliver a small power saving on systems using the crystal
since the oscillator will only be enabled when the ADC or DAC is active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The current code in pcm_lib.c do all checks using only the position
in the ring buffer. Unfortunately, where the interrupts gets delayed or
merged into one, we need another timing source to check when the
buffer size boundary overlaps to avoid the wrong updating of the
ring buffer pointers.
This code uses jiffies to check the right time window without any
performance impact.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With some hardware combinations, the PCM interrupts are acknowledged
before the period boundary from the emu10k1 chip. The midlevel PCM code
gets confused and the playback stream is interrupted.
It seems that the interrupt processing shift by 2 samples is enough
to fix this issue. This default value does not harm other,
non-affected hardware.
More information: Kernel bugzilla bug#16300
[A copmile warning fixed by tiwai]
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Output size_t type as a "%Zu" to avoid warnings.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Dead pxa2xx-pcm.h includes and a missing ,
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch contains two small fixes for the sound board driver for the qi_lb60
introduced by the multi-component patches:
* Remove unnecessary includes: Those includes where only used to get the
definitions for the DAI devices and are thus not needed anymore.
* Fix a typo.
Signed-off-By: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Fix capture mixer elements for ALC680 base model
- Support auto change ADC for recording from MIC
- Cancel capture source assigned in auto mode.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The RX and TX directions were inverted.
Reported-by: Seungwhan Youn <claude.youn@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Fairly simple conflicts, the most serious ones are the i.MX ones which I
suspect now need another rename.
Conflicts:
arch/arm/mach-mx2/clock_imx27.c
arch/arm/mach-mx2/devices.c
arch/arm/mach-omap2/board-rx51-peripherals.c
arch/arm/mach-omap2/board-zoom2.c
sound/soc/fsl/mpc5200_dma.c
sound/soc/fsl/mpc5200_dma.h
sound/soc/fsl/mpc8610_hpcd.c
sound/soc/pxa/spitz.c
This is not supported by current hardware revisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
The detection and loading of firmeware on riptide driver has been broken
due to rewrite of some codes, checking the presense wrongly.
This patch fixes the logic again.
Reference: kernel bug 16596
https://bugzilla.kernel.org/show_bug.cgi?id=16596
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: sound/usb/format: silence uninitialized variable warnings
MAINTAINERS: Add Ian Lartey as comaintaner for Wolfson devices
MAINTAINERS: Make Wolfson entry also cover CODEC drivers
ASoC: Only tweak WM8994 chip configuration on devices up to rev D
ASoC: Optimise DSP performance for WM8994
ALSA: hda - Fix dynamic ADC change working again
ALSA: hda - Restrict PCM parameters per ELD information over HDMI
sound: oss: sh_dac_audio.c removed duplicated #include
The DAC OSR should be selected based on the sample clock ratio.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Drive a minimal supported number of clocks required for the current
bit format in master mode. In slave mode this setting has no effect.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Implement set_sysclk() and then rather than assuming 256fs use the
supplied value to calculate and configure the clock ratio for the
currently used sample rate. As a side effect we also end up
implementing clock selection for the ADC path.
In order to avoid confusion remove the existing set_clkdiv() based
configuration of the clock source for the DAC and update the SMDK64xx
driver (which is the only in-tree user of the CODEC).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
In the case of the BCLK rate the defines are at best misleading anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
All the cool kids are using snd_soc_update_bits() these days.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
There's much more needed but this'll get us started.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
SCFR bit is required to be always set if pxa ssp is in slave mode. This bit
indicates clock input to SSPSCLK is only active during data transfers.
Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since pxa2xx-pcm.h is removed from sound/soc/pxa, we need to update the
path in related files.
Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Tested-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Removed #include of pxa2xx-pcm.h
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the core now includes deduplication in the name of CODEC
devices there's no need to add extra for the debugfs directory name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the debugfs directory is current per CODEC we should only init
it when the CODEC is initialised, otherwise we end up with errors
being generated when an attempt is made to add duplicate debugfs
entries.
Since most of this stuff is actually for the card we should refactor
but this can come later.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Gcc complains that ret might be used uninitialized:
sound/usb/format.c: In function ‘snd_usb_parse_audio_format’:
sound/usb/format.c:354: warning: ‘ret’ may be used uninitialized in this function
sound/usb/format.c:354: note: ‘ret’ was declared here
sound/usb/format.c:414: warning: ‘ret’ may be used uninitialized in this function
sound/usb/format.c:414: note: ‘ret’ was declared here
I suppose it could be uninitialized if there is ever a UAC_VERSION_3
released. Anyway this patch is worthwhile if only to silence the gcc
warning.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is V2 of the patch, after feedback from Clemens and Daniel.
This patch adds SuperSpeed support to the USB drivers under sound/. It adds
tests for USB_SPEED_SUPER to the appropriate places that check for the USB
speed.
This patch has been tested with our SS USB3 device emulating a set of Yamaha
speakers and a Logitech microphone, but with the descriptors modified to add
USB3 support. It has also been tested with the real speakers and microphone,
to make sure that USB2 devices still work.
Signed-off-by: Paul Zimmerman <paulz@synopsys.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Any subsequent revisions will have these configuration changes applied
by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Change the chip defaults to optimise performance of some of the DSP
functionality.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Its hardware is handled more fully by the new azt1605/azt2316 drivers.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a new driver for Aztech Sound Galaxy ISA soundcards based on the
AZT1605 and AZT2316 chipsets. It's constructed as two seperate drivers
for either chipset generated from the same source file, with (very)
minimal ifdeffery.
The drivers check the SB DSP version to decide if they are being loaded
for the right chip. AZT1605 returns 2.1 by default and AZT2316 3.1.
This isn't full-proof as the DSP version can actually be set through
software but it's close enough -- as far as I've been able to see, the
DSP version can not be stored in the EEPROM and the cards will therefore
startup with the defaults.
This distinction could (with the same success rate) also be used to
decide which chip we're looking at at runtime meaning a single, merged
driver is also an option but I feel it's actually nicer this way. A
merged driver would have to postpone translating the passed in resource
values to the card configuration until it knew which one it was looking
at and would need to postpone erring out on mpu_irq=10 for azt1605 and
mpu_irq=3 for azt2316.
The drivers have been tested on various cards. For snd-azt1605:
FCC-ID I38-MMSN811: Aztech Sound Galaxy Nova 16 Extra
FCC-ID I38-MMSN822: Aztech Sound Galaxy Pro 16 II
and for snd-azt2316:
FCC-ID I38-MMSN824: Aztech Sound Galaxy Pro 16 AB
FCC-ID I38-MMSN826: Trust Sound Expert DeLuxe Wave 32 (05201)
FCC-ID I38-MMSN830: Trust Sound Expert DeLuxe 16+ (05202)
FCC-ID I38-MMSN837: Packard Bell ISA Soundcard 030069
FCC-ID I38-MMSN846: Trust Sound Expert DeLuxe 16-3D (06300)
FCC-ID I38-MMSN847: Trust Sound Expert DeLuxe Wave 32-3D (06301)
FCC-ID I38-MMSN852: Aztech Sound Galaxy Waverider Pro 32-3D
826 and 846 were also marketed directly by Aztech and then known as:
FCC-ID I38-MMSN826: Aztech Sound Galaxy Waverider 32+
FCC-ID I38-MMSN846: Aztech Sound Galaxy Nova 16 Extra II-3D
Together, these cover the AZT1605 and AT2316A, AZT2316R and AZT2316-S
chipsets. All cards work fully -- full-duplex PCM, MIDI and FM. Full
duplex is a little flaky on some.
I38-MSN811 tends to not work in full-duplex but sometimes does with the
highest success rate being achieved when you first start the capture and
then a playback instead of the other way around (it's a CS4231-KL
codec).
The cards with an AD1845XP codec (my I38-MMSN826 and one of my
I38-MMSN830s) are also somewhat duplex-challenged. Sometimes full-duplex
works, sometimes not and this varies from try to try. This seems likely
to be a timing problem somewhere inside wss-lib.
I38-MMSN826 has an additional "ICS2115 WaveFront" wavetable synth
onboard that isn't supported yet. The wavetable synths on I38-MMSN847
and I38-MMSN852 are wired directly to the standard MPU-401 UART and the
AUX1 input on the codec and work without problem.
CD-ROM audio on the cards is routed to the codec "Line" input, Line-In
to its Aux input, and FM/Wavetable to its AUX1 input. I did not rename
the controls due to the capture source enumeration: I see that
capture-source overrides are hardcoded in wss-lib and this is just too
ugly to live.
Versus the old snd-sgalaxy driver these drivers add support for the
models without a configuration EEPROM (which are common), full-duplex,
MPU-401 UART and OPL3. In the future they might grow support for that
ICS2115 WaveFront synth on 826 and an hwdep interface to write to the
EEPROM on the models that have one.
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit eb541337b7
ALSA: hda - Make converter setups sticky
changes the semantics of snd_hda_codec_cleanup_stream() not to clean up
the stream at that moment but delay the action. This broke the codes
expecting that the clean-up is done immediately, such as dynamic ADC
changes in some codec drivers.
This patch fixes the issue by introducing a lower helper,
__snd_hda_codec_cleanup_stream(), to allow the immediate clean up.
The original snd_hda_codec_cleanup_stream() is kept as is now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a device is plugged over HDMI, it passes some information in ELD
including the supported PCM parameters like formats, rates, channels.
This patch adds the check to PCM open callback of HDMI streams so that
only valid parameters the device supports are used.
When no device is plugged, the parameters the codec supports are used;
it's mostly all parameters the hardware can work. This is for apps
that are started before device plugging and do probing (e.g. a sound
daemon), so that at least, probing would work even before the device
plugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: add AD1980 obsolete information
ASoC: register cache should be 1 byte aligned for 1 byte long register
ALSA: hda - Adding support for new IDT 92HD87XX codecs
ASoC: Fix inverted mute controls for WM8580
ALSA: HDA: Use model=auto for LG R510
ALSA: hda - Update model entries in HD-Audio-Models.txt
ALSA: hda: document VIA models
ALSA: hda - patch_nvhdmi.c: Add missing codec IDs, unify names
ALSA: hda - add support for Conexant CX20584
ALSA: hda - New snd-hda-intel model/pin config for hp dv7-4000
ALSA: hda - Fix missing stream for second ADC on Realtek ALC260 HDA codec
ALSA: hda - Make converter setups sticky
ALSA: hda - Add support for Acer ZGA ALC271 (1025:047c)
sound/oss: Adjust confusing if indentation
sound: oss: au1550_ac97.c removed duplicated #include
ASoC: Fix for changed Eureka Kconfig symbol names
Nothing should be referencing this any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add back the register restore call, when the codec driver is
removed.
This does not affect normal operation, but it is usefull when
debugging audio through the twl4030 class codecs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the provision of a struct device for the CODEC is now mandatory
we can use container_of() to locate the struct i2c_client and struct
spi_device for relevant devices, removing the need to manually set it
in each driver.
A further patch will automate selection of the control type based on
the bus_type of the struct device, further reducing the amount of
driver code required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Now soc-cache.c can figure out the I2C and SPI control data from the
device for the CODEC we don't need to manually assign it in drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for adding "status = disabled" to an SSI node to incidate that it
is not wired on the board. This replaces the not-so-intuitive previous method
of omitting a codec-handle property.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Kumar Gala <galak@kernel.crashing.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The error handling code in the OF probe function of the SSI driver is not
freeing all resources correctly.
Since the machine driver no longer calls the DMA driver to provide information
about the SSI, we don't need to keep a list of DMA objects any more. In
addition, the fsl_soc_dma_remove() function is incorrectly removing *all*
DMA objects when it should only remove one.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Update the DMA driver used by the Freescale MPC8610 HPCD audio driver to
support 36-bit physical addresses, for both DMA buffers and the SSI registers.
The DMA driver calls snd_dma_alloc_pages() to allocate the DMA buffers for
playback and capture. This function is just a front-end for
dma_alloc_coherent(). Currently, dma_alloc_coherent() only allocates buffers
in low memory (it ignores GFP_HIGHMEM), so we never actually get a DMA buffer
with a real 36-bit physical address.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The immap_86xx.h header file only defines one data structure: the "global
utilities" register set found on Freescale PowerPC SOCs. Rename this file
to fsl_guts.h to reflect its true purpose, and extend it to cover the "GUTS"
register set on 85xx chips.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Kumar Gala <galak@kernel.crashing.org>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch add sound support for the Goni board based on S5PV210.
The Goni board is based on Samsung SoC(S5PV210) and include
WM8994 codec over I2S to support sound.
The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
: When TV-OUT cable is inserted on Goni board,
the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
: When TV-OUT cable is inserted on Goni board,
the TV-OUT cable is connected to television.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch add sound support for the Aquila board based on S5PC110.
The Aquila board is based on Samsung SoC(S5PC110) and include
WM8994 codec over I2S to support sound. This uses the I2Sv4 driver
compatible with I2Sv5 to run sound.
The kind of jack is below states :
* SND_JACK_HEADPHONE
* SND_JACK_HEADSET
* SND_JACK_MECHANICAL
: When TV-OUT cable is inserted on Aquila board,
the TV-OUT cable isn't connected to television.
* SND_JACK_AVOUT
: When TV-OUT cable is inserted on Aquila board,
the TV-OUT cable is connected to television.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
ASoC: multi-component: SAMSUNG: Fix wrong field name on Aquila board
This patch modify the wrong field name on Aquila board.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch extends the ASoC API to allow sound cards to have more than one
CODEC and more than one platform DMA controller. This is achieved by dividing
some current ASoC structures that contain both driver data and device data into
structures that only either contain device data or driver data. i.e.
struct snd_soc_codec ---> struct snd_soc_codec (device data)
+-> struct snd_soc_codec_driver (driver data)
struct snd_soc_platform ---> struct snd_soc_platform (device data)
+-> struct snd_soc_platform_driver (driver data)
struct snd_soc_dai ---> struct snd_soc_dai (device data)
+-> struct snd_soc_dai_driver (driver data)
struct snd_soc_device ---> deleted
This now allows ASoC to be more tightly aligned with the Linux driver model and
also means that every ASoC codec, platform and (platform) DAI is a kernel
device. ASoC component private data is now stored as device private data.
The ASoC sound card struct snd_soc_card has also been updated to store lists
of it's components rather than a pointer to a codec and platform. The PCM
runtime struct soc_pcm_runtime now has pointers to all its components.
This patch adds DAPM support for ASoC multi-component and removes struct
snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec
or runtime PCM level basis rather than using snd_soc_socdev.
Other notable multi-component changes:-
* Stream operations now de-reference less structures.
* close_delayed work() now runs on a DAI basis rather than looping all DAIs
in a card.
* PM suspend()/resume() operations can now handle N CODECs and Platforms
per sound card.
* Added soc_bind_dai_link() to bind the component devices to the sound card.
* Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove
DAI link components.
* sysfs entries can now be registered per component per card.
* snd_soc_new_pcms() functionailty rolled into dai_link_probe().
* snd_soc_register_codec() now does all the codec list and mutex init.
This patch changes the probe() and remove() of the CODEC drivers as follows:-
o Make CODEC driver a platform driver
o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core.
o Removed all static codec pointers (drivers now support > 1 codec dev)
o snd_soc_register_pcms() now done by core.
o snd_soc_register_dai() folded into snd_soc_register_codec().
CS4270 portions:
Acked-by: Timur Tabi <timur@freescale.com>
Some TLV320aic23 and Cirrus platform fixes.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
TI CODEC and OMAP fixes
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Samsung platform and misc fixes :-
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Reviewed-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
MPC8610 and PPC fixes.
Signed-off-by: Timur Tabi <timur@freescale.com>
i.MX fixes and some core fixes.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
J4740 platform fixes:-
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
CC: Tony Lindgren <tony@atomide.com>
CC: Nicolas Ferre <nicolas.ferre@atmel.com>
CC: Kevin Hilman <khilman@deeprootsystems.com>
CC: Sascha Hauer <s.hauer@pengutronix.de>
CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
CC: Kuninori Morimoto <morimoto.kuninori@renesas.com>
CC: Daniel Gloeckner <dg@emlix.com>
CC: Manuel Lauss <mano@roarinelk.homelinux.net>
CC: Mike Frysinger <vapier.adi@gmail.com>
CC: Arnaud Patard <apatard@mandriva.com>
CC: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This codec has been obsoleted by ADI, so add appropriate warnings to the
source tree to dissuade people from using in new designs based on driver
support.
Signed-off-by: Sonic Zhang <sonic.zhang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Added the entries for 92HD87B1/3 and 92HD87B2/4 codecs.
These are compatible with existing 83xxx codecs.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Should use Capture rather than ADC so the UI tools can identify their
function more readily.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Sadly these aren't soft controllable and can't be read back either :(
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Otherwise debugfs gets upset when we try to create filenames with /
in them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Two users report model=auto is needed to make the internal mic work properly.
BugLink: https://bugs.launchpad.net/bugs/495134
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Add missing codec IDs.
* Modify some existing codec names for discrete GPUs to match newly
added IDs. Note: existing names were a mixture of marketing and
engineering GPU names. Equally, there's no reason that codec IDs
have to be specific to a particular GPU or board, so identify
codecs in a less marketing-oriented fashion.
* Reformat codec ID table so it's easier to read, for me at least.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd-aloop module allows redirecting of the PCM playback in the
kernel back to the user space using the standard ALSA PCM capture API.
The module also allows time synchronization with another timing source
and notifications of playback stream parameter changes.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The Conexant CX20584 with 141f:5068 seems compatible with other
cxt5066 code. Just add the missing id.
Tested-by: Cristopher Camacho Leandro <ccamacho@linuxmail.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This provides a new model and pin config for the snd-hda-intel
92HD83XXX codec for hp laptop model dv7-4000, enabling the subwoofer.
Signed-off-by: Steven Eastland <seastland at gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I discovered tonight that ALSA no longer sets up a stream for the second ADC
provided by the Realtek ALC260 HDA codec. At some point alc_build_pcms()
started using stream_analog_alt_capture when constructing the second ADC
stream, but patch_alc260() was never updated accordingly. I have no idea
when this regression occurred. The trivial patch to patch_alc260() given
below fixes the problem as far as I can tell. The patch is against 2.6.35.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (214 commits)
ALSA: hda - Add pin-fix for HP dc5750
ALSA: als4000: Fix potentially invalid DMA mode setup
ALSA: als4000: enable burst mode
ALSA: hda - Fix initial capsrc selection in patch_alc269()
ASoC: TWL4030: Capture route runtime DAPM ordering fix
ALSA: hda - Add PC-beep whitelist for an Intel board
ALSA: hda - More relax for pending period handling
ALSA: hda - Define AC_FMT_* constants
ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
ALSA: hda - Add support for HDMI HBR passthrough
ALSA: hda - Set Stream Type in Stream Format according to AES0
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
ASoC: wm9081: fix resource reclaim in wm9081_register error path
ASoC: wm8978: fix a memory leak if a wm8978_register fail
ASoC: wm8974: fix a memory leak if another WM8974 is registered
ASoC: wm8961: fix resource reclaim in wm8961_register error path
ASoC: wm8955: fix resource reclaim in wm8955_register error path
ASoC: wm8940: fix a memory leak if wm8940_register return error
ASoC: wm8904: fix resource reclaim in wm8904_register error path
...
Call the gpio reset platform function instead of using the flawed
ac97 functionality of the MPC5200(b)
From MPC5200B User's Manual:
"Some AC97 devices goes to a test mode, if the Sync line is high
during the Res line is low (reset phase). To avoid this behavior the
Sync line must be also forced to zero during the reset phase. To do
that, the pin muxing should switch to GPIO mode and the GPIO control
register should be used to control the output lines."
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
* git://git.kernel.org/pub/scm/linux/kernel/git/brodo/pcmcia-2.6:
pcmcia: avoid buffer overflow in pcmcia_setup_isa_irq
pcmcia: do not request windows if you don't need to
pcmcia: insert PCMCIA device resources into resource tree
pcmcia: export resource information to sysfs
pcmcia: use struct resource for PCMCIA devices, part 2
pcmcia: remove memreq_t
pcmcia: move local definitions out of include/pcmcia/cs.h
pcmcia: do not use io_req_t when calling pcmcia_request_io()
pcmcia: do not use io_req_t after call to pcmcia_request_io()
pcmcia: use struct resource for PCMCIA devices
pcmcia: clean up cs.h
pcmcia: use pcmica_{read,write}_config_byte
pcmcia: remove cs_types.h
pcmcia: remove unused flag, simplify headers
pcmcia: remove obsolete CS_EVENT_ definitions
pcmcia: split up central event handler
pcmcia: simplify event callback
pcmcia: remove obsolete ioctl
Conflicts in:
- drivers/staging/comedi/drivers/*
- drivers/staging/wlags49_h2/wl_cs.c
due to dev_info_t and whitespace changes
of_device is just an alias for platform_device, so remove it entirely. Also
replace to_of_device() with to_platform_device() and update comment blocks.
This patch was initially generated from the following semantic patch, and then
edited by hand to pick up the bits that coccinelle didn't catch.
@@
@@
-struct of_device
+struct platform_device
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Reviewed-by: David S. Miller <davem@davemloft.net>
So far, we reset the converter setups like the stream-tag, the
channel-id and format-id in prepare callbacks, and clear them in
cleanup callbacks. This often causes a silence of the digital
receiver for a couple of seconds.
This patch tries to delay the converter setup changes as much as
possible. The converter setups are cached and aren't reset as long
as the same values are used. At suspend/resume, they are cleared
to be recovered properly, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Indent the branch of an if.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@r disable braces4@
position p1,p2;
statement S1,S2;
@@
(
if (...) { ... }
|
if (...) S1@p1 S2@p2
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
if (p1[0].column == p2[0].column):
cocci.print_main("branch",p1)
cocci.print_secs("after",p2)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'next-devicetree' of git://git.secretlab.ca/git/linux-2.6: (63 commits)
of/platform: Register of_platform_drivers with an "of:" prefix
of/address: Clean up function declarations
of/spi: call of_register_spi_devices() from spi core code
of: Provide default of_node_to_nid() implementation.
of/device: Make of_device_make_bus_id() usable by other code.
of/irq: Fix endian issues in parsing interrupt specifiers
of: Fix phandle endian issues
of/flattree: fix of_flat_dt_is_compatible() to match the full compatible string
of: remove of_default_bus_ids
of: make of_find_device_by_node generic
microblaze: remove references to of_device and to_of_device
sparc: remove references to of_device and to_of_device
powerpc: remove references to of_device and to_of_device
of/device: Replace of_device with platform_device in includes and core code
of/device: Protect against binding of_platform_drivers to non-OF devices
of: remove asm/of_device.h
of: remove asm/of_platform.h
of/platform: remove all of_bus_type and of_platform_bus_type references
of: Merge of_platform_bus_type with platform_bus_type
drivercore/of: Add OF style matching to platform bus
...
Fix up trivial conflicts in arch/microblaze/kernel/Makefile due to just
some obj-y removals by the devicetree branch, while the microblaze
updates added a new file.
The WM8962 is a low power, high performance stereo CODEC designed for
portable digital audio applications.
This initial driver release supports the key audio paths of the WM8962.
Extended functionality, such as microphone detection, digital microphones
and the advanced DSP signal enhancements provided by the device are not
yet supported.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The NID 0x11 on HP dc5750 with ALC260 should be a speaker although BIOS
gives it as a line-out. This patch adds a quirk to fix the pin config
so that the real line-out is used properly.
Reference: bnc#624118
https://bugzilla.novell.com/show_bug.cgi?id=624118
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My previous patch assumed that the DMA mode (represented by 3 lowest bits of
ALS4K_GCR99_DMA_EMULATION_CTRL register) is set to the default value 0. If
that's not the case, it might result in invalid mode to be set.
This patch fixes this potential problem.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/rafael/suspend-2.6:
PM / Runtime: Add runtime PM statistics (v3)
PM / Runtime: Make runtime_status attribute not debug-only (v. 2)
PM: Do not use dynamically allocated objects in pm_wakeup_event()
PM / Suspend: Fix ordering of calls in suspend error paths
PM / Hibernate: Fix snapshot error code path
PM / Hibernate: Fix hibernation_platform_enter()
pm_qos: Get rid of the allocation in pm_qos_add_request()
pm_qos: Reimplement using plists
plist: Add plist_last
PM: Make it possible to avoid races between wakeup and system sleep
PNPACPI: Add support for remote wakeup
PM: describe kernel policy regarding wakeup defaults (v. 2)
PM / Hibernate: Fix typos in comments in kernel/power/swap.c
Without this patch, an undefined/random sg->dma_length is used and
the sound will be played/captured wrongly.
Signed-off-by: Markus Pietrek <markus.pietrek@emtrion.de>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Enable burst mode to prevent dropouts during high PCI bus usage.
The card is useless in X without this because of dropouts when anything moves
on the screen (at least with PCI VGA card). Enabling this is also recommended
by the datasheet (page 48).
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In patch_alc269(), we initialize the primary capsrc so that the device
works from the beginning. It issues CONNECT_SEL verb no matter which
widget is although some widget (e.g. 0x23) has no connection selection
but a mixer, which requires unmuting instead.
This patch fixes the initialization of capsrc by re-using the code as
a helper function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the ordering problem in DAPM domain, when the user
changes between digital and analog sources during active
capture (or loopback) scenario.
Before this patch, when the user changed from analog source
to digital there were a short time, when the codec enabled
analog mic bias (2.2 volts) instead of the correct digital
mic bias (1.8 volts) to the digital microphones.
This behaviour caused by the former implementation of
selecting the correct type of bias. This was done at the
POST_REG event of the DAPM_MUX_E("TXx Capture Route")
widget.
By moving the bias type selection as DAPM_SUPPLY and
connecting it to the corresponding digimic widget the
problematic situation can be avoided.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
An Intel board needs a white-list entry to enable PC-beep.
Otherwise the driver misdetects (due to bogus BIOS info) and ignores
the PC-beep on 2.6.35.
Reported-and-tested-by: Leandro Lucarella <luca@llucax.com.ar>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the pending periods are often bogus and take long time until
actually processed, it often results in a high CPU usage of the hd-audio
workq. Overall it's better to have low CPU consumption by avoiding a
too tight loop rather than the wake-up timing accuracy.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix HDA beep frequency on IDT 92HD73xx and 92HD71Bxx codecs.
These codecs use the standard beep frequency calculation although the
datasheet says it's linear frequency.
Other IDT/STAC codecs might have the same problem. They should be
fixed individually later.
Signed-off-by: Daniel J Blueman <daniel.blueman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Passing IEC 61937 encapsulated compressed audio at bitrates over 6.144
Mbps (i.e. more than a single 2-channel 16-bit 192kHz IEC 60958 link)
over HDMI requires the use of HBR Audio Stream Packets instead of Audio
Sample Packets.
Enable HBR mode when the stream has 8 channels and the Non-PCM bit is
set.
If the audio converter is not connected to any HBR-capable pins, return
-EINVAL in prepare().
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set bit 15 (Stream Type) of HDA Stream Format to 1 (Non-PCM) when IEC958
channel status bit 1 (AES0 & 0x02) is set to 1 (non-audio).
This is a prequisite for HDMI HBR passthrough.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just as with the X301. The X300 does not have a way to do SPDIF either.
It does not have a dock connector, nor does it have the SPDIF through
the headphone jack.
This patch fixes it so X300 does not show SPDIF, since it cannot do it.
To add all Lenovo Thinkpads had different codec subsytem IDs:
X300:
http://launchpadlibrarian.net/34862838/Card0.Codecs.codec.0.txt
Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo X301 does not have the ability to connect to a docking
station to use the SPDIF port. It also does not have the ability to do
SPDIF though the headphone jack or Display Port jacks.
This patch fixes it so this is not exposed for the X301 and users do
think it has the ability to do SPDIF.
I tested both headphone & display port jacks and it is not there. I have
tested this patch and it works great.
Also to add the other Thinkpads have different subsystem codec IDs.
Here are examples:
X301:
http://launchpadlibrarian.net/31561902/Card0.Codecs.codec.0.txt
X200:
http://launchpadlibrarian.net/49055036/Card0.Codecs.codec.0.txt
W500:
http://launchpadlibrarian.net/36276057/Card0.Codecs.codec.0.txt
Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the error path in wm9081_register to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a memory leak found if wm8978_register() fail.
This patch moves the buffer allocate and release
at the same level to prevent the memory leak.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
wm8974 is allocated in wm8974_i2c_probe() but is not freed if wm8974_register()
return -EINVAL (if another WM8974 is registered).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the error path in wm8961_register to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the error path in wm8955_register to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds checking for wm8940_register return value,
and does kfree(wm8940) if wm8940_register() fail.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch includes below fixes:
1. wm8904 need to be kfreed in wm8904_register() error path before return.
2. fix the error path for snd_soc_register_codec() fail and
snd_soc_register_dai() fail to properly free resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
wm8711 is allocated in either wm8711_spi_probe() or wm8711_i2c_probe() but is
not freed if wm8711_register() return -EINVAL(if another ad1836 is registered).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch includes below fixes:
1. If another WM8523 is registered, need to kfree wm8523 before return -EINVAL.
2. If snd_soc_register_codec failed, goto error path to properly free resources.
3. Instead of using mixed in-line and goto style cleanup, use goto style error
handling if snd_soc_register_dai failed.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
da7210 should be kfreed if da7210_init() return error.
This patch also fixes the error handing in the case of snd_soc_register_dai()
fail by adding snd_soc_unregister_codec() in error path.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4642 should be kfreed if ak4642_init() return error.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Reviewed-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ad1836 is allocated in ad1836_spi_probe() but is not freed if ad1836_register()
return -EINVAL (if another ad1836 is registered).
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8741 is a very high performance stereo DAC designed for audio
applications such as professional recording systems, A/V receivers and
high specification CD, DVD and home theatre systems. The device supports
PCM data input word lengths from 16 to 32-bits and sampling rates up to
192kHz. The WM8741 also supports DSD bit-stream data format, in both
direct DSD and PCM-converted DSD modes.
TODO: Expand wm8741_set_dai_sysclk and rate_constraint members to
allow for all supported sample rate / Master Clock frequency combinations.
Fully enable control of supplies.
Signed-off-by: Ian Lartey <ian@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The use of sDMA packet mode in THRESHOLD mode removes the restriction on the
period size.
With the extended THRESHOLD mode user space can ask for any
period size it wishes, and the driver will configure the
sDMA and McBSP FIFO accordingly.
Replace the hw_rule for the period size with static constraint,
which will make sure that the period size will be always
even (to avoid prime period size, which could be possible in
mono stream)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Utilize the sDMA controller's packet syncronization mode, when
the McBSP FIFO is in use (by extending the THRESHOLD mode).
When the sDMA is configured for packet mode, the sDMA frame size
does not need to match with the McBSP threshold configuration.
Uppon DMA request the sDMA will transfer packet size number of
words, and still trigger interrupt on frame boundary.
The patch extends the original THRESHOLD mode by doing the
following:
if (period_words <= max_threshold)
Current THRESHOLD mode configuration
Otherwise (period_words > max_threshold)
McBSP threshold = sDMA packet size
sDMA frame size = period size
With the extended THRESHOLD mode we can remove the constraint
for the maximum period size, since if the period size is
bigger than the maximum allowed threshold, than the driver
will switch to packet mode, and picks the best (biggest)
threshold value, which can divide evenly the period size.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
To make the code a bit more readable, change the indexed
references to the omap_mcbsp_dai_dma_params elements with
pointer.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
In preparation for the extended threshold mode (sDMA packet mode
support), the code need to be restructured.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Match usb ids in usb/quirks-table.h for some Hauppage HVR-950Q models
and for the HVR850 model to those ids at the end of au0828-cards.c
Thanks to nhJm449 for pointing out the problem.
Signed-off-by: John S Gruber <JohnSGruber@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current ALC259/268/269 parser ignores some pins as unhandled,
but user won't notice what goes wrong. So, added a warning message
for the ignored pins as a hint.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call snd_hda_shutup_pins() for power-saving and reboot-notifier in
patch_conexant.c as well as other codecs. This will reduce the pop
noise in power-save mode.
Reference: bnc#624896
https://bugzilla.novell.com/show_bug.cgi?id=624896
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If BIOS sets up the input pin as VREF 50, use the value as is instead of
overriding forcibly to VREF 80. This fixes the quality of inputs on
some devices like Packard-Bell M5210.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since ALC259/269 use the same parser of ALC268, the pin 0x1b was ignored
as an invalid widget. Just add this NID to handle properly.
This will add the missing mixer controls for some devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make a helper function to parse the digital I/Os of all Realtek codecs
to simplify the code and to ensure the setups.
Also, initialize digital I/O pins properly in init callbacks. Some BIOS
seem to leave pins uninitialized.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ALC662-compatible codecs like ALC892 may have more than 4
connections for the input source. Use HDA_MAX_CONNECTIONS instead of
the fixed magic number 4.
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 a PC-beep workaround for ASUS P5-V
ALSA: hda - Assume PC-beep as default for Realtek
ALSA: hda - Don't register beep input device when no beep is available
ALSA: hda - Fix pin-detection of Nvidia HDMI
Current FSI driver id is not only 0
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The non-standard name "iMic" makes PulseAudio ignore the microphone.
BugLink: https://launchpad.net/bugs/605101
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS P5-V provides a SSID that unexpectedly matches with the value
compilant with Realtek's specification. Thus the driver interprets
it badly, resulting in non-working PC beep.
This patch adds a white-list for such a case; a white-list of known
devices with working PC beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
get_user() may fail, if so return -EFAULT.
[Fixed one missing place by tiwai]
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Enable PC-beep as default for hardwares that aren't compliant with the
SSID value Realtek requires. In such a case, better to enable the beep
to avoid a regression.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We check now the availability of PC beep and skip the build of beep
mixers, but the driver still registers the input device. This should
be checked as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The behavior of Nvidia HDMI codec regarding the pin-detection unsol events
is based on the old HD-audio spec, i.e. PD bit indicates only the update
and doesn't show the current state. Since the current code assumes the
new behavior, the pin-detection doesn't work relialby with these h/w.
This patch adds a flag for indicating the old spec, and fixes the issue
by checking the pin-detection explicitly for such hardware.
Tested-by: Wei Ni <wni@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The correct size should be sizeof(gRESP_HPI_SUBSYS_FIND_ADAPTERS),
sizeof(&gRESP_HPI_SUBSYS_FIND_ADAPTERS) is incorrect.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no need to handle POST_PMU, POST_PMD event with
the Capture Route widget.
It is enough to handle POST_REG event, since that will come
when the user changes the routing, and we will switch the needed
bits in the registers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
WARNING: sound/soc/au1x/snd-soc-au1xpsc-i2s.o(.data+0xa8): Section mismatch in reference from the variable au1xpsc_i2s_driver to the function .init.text:au1xpsc_i2s_drvprobe()
The variable au1xpsc_i2s_driver references
the function __init au1xpsc_i2s_drvprobe()
If the reference is valid then annotate the
variable with __init* or __refdata (see linux/init.h) or name the variable:
*_template, *_timer, *_sht, *_ops, *_probe, *_probe_one, *_console,
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
The commit afbd9b8448
ALSA: hda - Limit the amp value to write
introduced a regression for codec setups with amp offsets like IDT/STAC
codecs. The limit value should be a raw value without offset calculation.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both of_bus_type and of_platform_bus_type are just #define aliases
for the platform bus. This patch removes all references to them and
switches to the of_register_platform_driver()/of_unregister_platform_driver()
API for registering.
Subsequent patches will convert each user of of_register_platform_driver()
into plain platform_drivers without the of_platform_driver shim. At which
point the of_register_platform_driver()/of_unregister_platform_driver()
functions can be removed.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: David S. Miller <davem@davemloft.net>
HeadPhone Playback Volume control register of DA7210 has
reserved area. This patch considered it as mute.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: Select wm_hubs automatically for WM8994
ASoC: Remove duplicate AUX definition from WM8776
ASoC:: remove a redundant snd_soc_unregister_codec call in wm8988_register
ASoC: wm8727: add a missing return in wm8727_platform_probe
ASoC: fsi: fixup wrong value setting order of TDM
ASoC: fsi: fixup clock inversion operation
CONFIG_SOFTOSS* doesn't exist in Kconfig or somewhere
else, therefore removing all references for it from the source code.
Signed-off-by: Christian Dietrich <qy03fugy@stud.informatik.uni-erlangen.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When digital microphones are connected to twl, delay is
needed after enabling the digimic interface of the codec.
Add new parameter for the setup data, which can be used
to pass the apropriate delay in ms after the digimic
interface has been enabled.
Without certain delay (in certain HW configuration) the
beggining of the recorded sample contains a glitch, which
is generated by the digital microphones.
Delaying the micbias1, 2 (which is the bias for the digimic0
or 1) does not help, since the glitch is coming after
switching the digimic interface.
Reversing the micbias and digimic enable order does not
work either (in that case the wait need to be added after
the micbias enabled).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
AIF1ADC TDM mode has no effect other than causing the ADCDAT line to
be tristated rather than driven low on clock cycles where there is no
data to be transmitted. If the clock cycle is idle then there should
be no devices using the data so tristating should have no adverse
effects.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently the EDMA queue to be used by for servicing ASP through
internal RAM is fixed to EDMAQ_0 and that to service internal RAM
from external RAM is fixed to EDMAQ_1.
This may not be the desirable configuration on all platforms. For
example, on DM365, queue 0 has large fifo size and is more suitable
for video transfers. Having audio and video transfers on the same
queue may lead to starvation on audio side.
platform data as defined currently passes a queue number to the driver
but that remains unused inside the driver.
Fix this by defining one queue each for ASP and RAM transfers in the
platform data and using it inside the driver.
Since EDMAQ_0 maps to 0, thats the queue that will be used if
the asp queue number is not initialized. None of the platforms
currently utilize ping-pong transfers through internal RAM so that
functionality remains unchanged too.
This patch has been tested on DM644x and OMAP-L138 EVMs.
Signed-off-by: Sekhar Nori <nsekhar@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modify I2Sv2 driver to support Samsung SoC(S5PV210).
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Otherwise all machine drivers need to do so.
Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a follow on patch adds support for AMD based Lenovo G series
machines, such as the Lenovo G555.
Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In situation when appl_ptr is far greater then hw_ptr, the hw_avail value
can be greater than buffer_size. Check for this.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The function IDs are different for audio and modem. Do not mix them.
Also, show the unsolicited bit in the function_id register.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
All current users of pm_qos_add_request() have the ability to supply
the memory required by the pm_qos routines, so make them do this and
eliminate the kmalloc() with pm_qos_add_request(). This has the
double benefit of making the call never fail and allowing it to be
called from atomic context.
Signed-off-by: James Bottomley <James.Bottomley@suse.de>
Signed-off-by: mark gross <markgross@thegnar.org>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code checks 'davinci_vc' after kzalloc() and do not checks
'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that
it is a typo (autocompletion?).
Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Specified ID is necessary, when some codecs are used with FSI.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai
Liam Girdwood
Jarkko Nikula
Kuninori Morimoto
Guennadi Liakhovetski
Mark Brown
Dzianis Kahanovich
David Henningsson
Charles Chin
Dan Carpenter
Peter Ujfalusi
Kailang Yang
Dimitris Papastamos
Arnaud Patard (Rtp)
Jaroslav Kysela
Stephen Warren
Arnd Bergmann
Andreas Schwab
Ben Hutchings
Joe Perches
Jassi Brar
Anisse Astier
Seth Heasley
Seungwhan Youn
Matti J. Aaltonen
Axel Lin
Daniel Mack
Clemens Ladisch
G, Manjunath Kondaiah
Vasily Khoruzhick
Vitaliy Kulikov
Brian Austin
Jiri Slaby
Julia Lawall
Ian Lartey
Manuel Lauss
Akinobu Mita
Eliot Blennerhassett
Uwe Kleine-König
Sascha Hauer
Jerone Young
Garnet MacPhee
Timur Tabi
Paul Mundt
Randolph Chung
Janusz Krzysztofik
Randolph Chung
Daniel T Chen
Haojian Zhuang
Lars-Peter Clausen
Linus Torvalds
Paul Zimmerman
René Herman
Chanwoo Choi
Andrea Gelmini
Sonic Zhang
Cliff Cai
Steven Eastland
Jonathan Woithe
Benjamin Herrenschmidt
Eric Millbrandt
Grant Likely
Eric Bénard
Ondrej Zary
Jiri Kosina
Markus Pietrek
Daniel J Blueman
Anssi Hannula
Dominik Brodowski
John S Gruber
Kulikov Vasiliy
Ralf Baechle
David Henningsson
Christian Dietrich
Sekhar Nori
James Bottomley