- Support ASUS F81Se F5Q P80 U20A U80 U50 UX50 for ALC269
- Support ASUS F70SL UX20 X58LE F50Z N80Vc N81Te N505Tp Vx3V N5051A
for ALC663
- Support DELL ZM1 for ALC272
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't unmute unneeded amps for input mixers of ALC662 & co.
It caused possible recording noises.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The legacy i2c binding model is going away soon, so convert the ppc
keywest sound driver to the new model or it will break.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The legacy i2c binding model is going away soon, so convert the AOA
codec drivers to the new model or they'll break.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Tested-by: Johannes Berg <johannes@sipsolutions.net>
Tested-by: Andreas Schwab <schwab@linux-m68k.org>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather than having switch statements at point of use make the DAPM
power check a member of the widget structure and set it when we
instantiate the widget.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This also switches us to using a switch statement for the widget type
in dapm_power_widget().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This will form a basis for further power check refactoring: the overall
goal of these changes is to allow us to check power separately to
applying it, allowing improvements in the power sequencing algorithms.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add Voice DAI to support the PCM voice interface of the twl4030 codec.
The PCM voice interface can be used with 8-kHz(voice narrowband) or
16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mono
TX or stereo TX.
The PCM voice interface has two modes
- PCM mode1 : This uses the normal FS polarity and the rising edge of
the clock signal.
- PCM mode2 : This uses the FS polarity inverted and the falling edge
of the clock signal.
If the system master clock is not 26MHz or the twl4030 codec mode is not
option2, the voice PCM interface is not available.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use it to clean up snd_us122l_card_used[].
Without patch unplugging of an US122L soundcard didn't reset the
corresponding element of snd_us122l_card_used[] to 0.
The (SNDRV_CARDS + 1)th plugging in did not result in creating the soundcard
device anymore.
Index values supplied with the modprobe command line were not used correctly
anymore after the first unplugging of an US122L.
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing definition of max channels for CA0110, which resulted
in an error at opening PCM devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Firstly, this patch makes the palm27x asoc driver a little more sane. Also,
since all affected devices use GPIO95 as AC97_nRESET, this patch sets that
properly. Affected are PalmT5, TX and LifeDrive.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
* fix/hda:
ALSA: hda - Set function_id only on FG nodes
ALSA: hda - Add upper-limit of mixer amp for AD1884A-laptop model, too
ALSA: hda - Fix headphone-detection on some machines with STAC/IDT codecs
ALSA: hda_intel.c - Consolidate bitfields
I notice that the fixes were merged, minus one:
sound/soc/codecs/wm9705.c: At top level:
sound/soc/codecs/wm9705.c:445: warning: initialization from incompatible pointer type
so you might find this trivial patch useful.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
(Re)set function_id only from the value on FG nodes.
The current code overrides the value with the last widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The original implementation of the constraints were good against sane
applications.
If the opening sequence is:
stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> the
constraints are set correctly for stream2.
But if the sequence is:
stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream2
would receive constraint rate = 0, sample_bits = 0, since the stream1 has not
yet called hw_params...
The command to trigger this event:
gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=false
This patch does some 'black magic' in order to always set the correct
constraints and sets it only when it is needed for the other stream.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My email address is going to expire soon so update it. Adding also
Peter Ujfalusi <peter.ujfalusi@nokia.com> as a second contact to OMAP core
drivers since I won't have anymore access to non-public OMAP documentation
in the future and Peter is working with these drivers as well.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Those macros are just screwed as soon as CONFIG_PXA25x is enabled.
This patch
- changes ssp_set_scr to take an ssp_dev pointer instead of ssp_device
- adds a corresponding ssp_get_scr function.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Within 2.6.30's mergewindow, struct urb's transfer_buffer_length has become
unsigned. This changed an "int > int" comparision to an "unsigned > int" one
in snd_usb_122l.
Fix this by using a local int variable instead of urb->transfer_buffer_length
in comparisions.
Shorten playback_prep_freqn() a bit and tweak error-paths in
usb_stream_prepare_playback().
Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DSP_A mode is similar to the DSP_B, but the MSB is delayed with
one bclk (appears after the FS pulse and not under it).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use single-phase mode for the DSP mode and keep the dual phase
mode for the I2S mode.
The mono (1 channel) mode already used single phase mode,
now it is more cleaner. There is no need to configure the
second phase, when the single phase is used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using inverted FS polarity in OSK5912 must be an error since TLV320AIC23
do not have support for inverted polarities. This is mostly due the hassle
with the DSP formats in OMAP McBSP DAI and inversion on OMAP side probably
just made this configuration working at some point.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DSP format wasn't still correct in OMAP McBSP DAI even after the commit
bd25867a6c.
Thanks to Peter Ujfalusi <peter.ujfalusi@nokia.com> for noticing and being
part of the fix. Now the FS length definition is more clear by defining
it with FWID(0).
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix accidental change of <mach/regs-gpio.h> to
<plat/regs-gpio.h> in s3c2412-i2s.c
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the build error in s3c-i2s-v2.c caused by
a change to the snd_soc_dai ops field.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The definition of s3c_i2sv2_iis_calc_rate was never
renamed from s3c2412_iis_calc_rate, so rename this
to allow the build to work.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build errors in sound/soc/s3c24xx/jive_wm8750.c
from changes to ASoC.
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pxa_ssp_set_dai_fmt() currently has an early exit if the desired format
equals the current configuration. This is correct behaviour unless this
function is called with a zero value parameter for the first time.
Zero is a valid value for this function, but the early exit is bogus in
this case.
Hence, set priv->dai_fmt to -1 in the beginning so we can configure the
port.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: pHilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When the headphone can have no unique DAC, the current code doesn't
check the HP-detection although it should. Put the hp-detection check
before the DAC check to fix this bug.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some limited volume controls (mostly simple attenuations) have only two
settings so the ASoC info functions misreport them as booleans. Since
we currently have no better information check for " Volume" in the
control name and always report any controls matching as being integer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Check the length to copy via strlen() beforehand to avoid the stack
corruption, or use strlcpy() to be safe in HD-audio codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode.
In the HD-audio mode, no multiple streams are supported by just it
behaves like a normal HD-audio device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Subject says it all. Briefly, use hp_only for another Dell Inspiron 8600.
Reference: Ubuntu #41015 (https://launchpad.net/bugs/41015)
Signed-off-by: Daniel T Chen <seven.steps@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While cleaning up quirks, I noticed that there is a duplicated quirk for
the SSID 0x103c0934. Looking back through the bug reports, I've concluded
that there is only one necessary quirk (hp_mute_led), so this patch
removes the conflicting one.
Reference: Ubuntu #44066 (https://launchpad.net/bugs/44066)
Signed-off-by: Daniel T Chen <seven.steps@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit fa00e046b4
added a new bitfield not adjacent to other
bitfields in the same struct. Moved the new one.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the key value generation for get/set amp verbs. The upper bits of
the parameter have to be combined with the verb value to be unique for
each direction/index of amp access.
This fixes the resume problem on some hardwares like Macbook after
the channel mode is changed.
Tested-by: Johannes Berg <johannes@sipsolutions.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* topic/memdup_user:
ALSA: sound/pci: use memdup_user()
ALSA: sound/usb: use memdup_user()
ALSA: sound/isa: use memdup_user()
ALSA: sound/core: use memdup_user()
