Currently, eld_valid is never set to false, except at kernel module
load time. This patch makes sure that eld is no longer valid when
the cable is (hot-)unplugged.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the recent update, Fujitsu S7020 laptop with ALC260 codec lost the
speaker output, no matter how the amps and the pins are set. After a
long debugging session, we found out that the default codec init code
is harmful for this machine, and we have to reset it to
ALC_INIT_NONE.
Reported-and-tested-by: Jonathan Woithe <jwoithe@just42.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These two machines have no mute LED string in BIOS.
BugLink: https://bugs.launchpad.net/bugs/1128934
Tested-by: Tammy Yang <tammy.yang@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
bootresponse in snd_usb_mbox2_boot_quirk is only 12 (decimal) u8's
long, but i9s passed to snd_usb_ctl_msg as it would be 0x12 (hexa)
long. Fix that by having proper size of the array, i.e. 0x12.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more updates from the past week - a new driver from Dialog and
some small fixes and tweaks.
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Merge tag 'asoc-3.9-updates' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Final updates for v3.9
A few more updates from the past week - a new driver from Dialog and
some small fixes and tweaks.
This patch adds support for the Dialog DA7213 audio codec.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
this add new API for sound compress to support gapless playback.
As noted in Documentation change, we add API to send metadata of encoder and
padding delay to DSP. Also add API for indicating EOF and switching to
subsequent track
Also bump the compress API version
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This chip needs the speaker pin to go to D3 to avoid clicks,
so default_power_filter does not work here.
This was found on Thinkpad R61i/T61i but I guess it applies to
the entire chip. If not, quirks should be set for at least
PCI SSID 17aa:20ac.
Thanks to c4pp4 for testing.
BugLink: https://bugs.launchpad.net/bugs/886975
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a regression report wrt the IRQ issue related with the
power-save on a Dell machine, and disabling runtime PM works around.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=53441
Cc: <stable@vger.kernel.org> [v3.7+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current badness value used for the missing multi-io seems too
weak, and the multi-io tends to be skipped for desktop configurations
when no enough DACs are available. It's because the total badness of
the multi-io becomes often larger than the badness with assigning an
individual DAC to a headphone jack. This is good for one side, but it
seems that the surround outputs are more demanded by that.
This patch increases the badness value for the missing multi-io
slightly so that the multi-io would be preferred than the individual
headphone DAC if they conflict. Through the tests with hda-emu,
mostly only desktop configurations with ALC662/663 and CMI codecs are
affected by this change, and all look reasonable.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For optimal performance the inputs should be kept muted until after power
up. Since there are few use cases for muting inputs during capture move
the mutes to automatic control.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit [dcda58061: ALSA: hda - Add workaround for conflicting
IEC958 controls] introduced a workaround for cards that have both
SPDIF and HDMI devices for giving device=1 to SPDIF control elements.
It turned out, however, that this workaround doesn't work well -
- The workaround checks only conflicts in a single codec, but SPDIF
and HDMI are provided by multiple codecs in many cases, and
- ALSA mixer abstraction doesn't care about the device number in ctl
elements, thus you'll get errors from amixer such as
% amixer scontrols -c 0
ALSA lib simple_none.c:1551:(simple_add1) helem (MIXER,'IEC958
Playback Switch',0,1,0) appears twice or more
amixer: Mixer hw:0 load error: Invalid argument
This patch fixes the previous broken workaround. Instead of changing
the device number of SPDIF ctl elements, shift the element indices of
such controls up to 16. Also, the conflict check is performed over
all codecs found on the bus.
HDMI devices will be put to dev=0,index=0 as before. Only the
conflicting SPDIF device is moved to a different place. The new place
of SPDIF device is supposed by the updated alsa-lib HDA-Intel.conf,
respectively.
Reported-by: Stephan Raue <stephan@openelec.tv>
Reported-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org> [v3.8]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The chip address is 32bit long but INVALID_CHIP_ADDRESS is defined as
an unsigned long. This makes dsp_chip_to_dsp_addx() misbehaving on
64bit architectures. Fix the INVALID_CHIP_ADDRESS definition to be
32bit.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The problem addressed by this fixup is not specific to Vaio Z, affecting
some Vaio all-in-one desktop PCs too. Update the code comments accordingly.
Signed-off-by: Fernando Luis Vazquez Cao <fernando@oss.ntt.co.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Vaio all-in-one desktop PCs (for example VGC-LN51JGB) are affected by
the same issue that caused Vaio Z laptops to become silent: the speaker pin
must be connected to the first DAC even though the codec itself advertises
flexible routing through any of the DACs.
Use the no-primary-hp fixup for choosing the speaker pin as the primary so
that the right DAC is assigned on this device.
Cc: stable@vger.kernel.org
Signed-off-by: Fernando Luis Vazquez Cao <fernando@oss.ntt.co.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unused memory ressource get from McPDM driver.
Signed-off-by: Sebastien Guiriec <s-guiriec@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the following sparse warning:
sound/soc/fsl/imx-audmux.c:182:3: warning: symbol 'audmux_type' was not declared. Should it be static?
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_ali_pointer function is called with local
interrupts disabled. However it seems very strange to
reenable them in such way.
Found by Linux Driver Verification project (linuxtesting.org).
Signed-off-by: Denis Efremov <yefremov.denis@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the other code in this driver and similar
code in rme96 it seems, that spin_lock_irq in
snd_rme32_capture_close function should be paired
with spin_unlock_irq.
Found by Linux Driver Verification project (linuxtesting.org).
Signed-off-by: Denis Efremov <yefremov.denis@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_compr_update_tstamp() can only fill in the snd_compr_tstamp
if the codec implements the pointer() function. If that happened
the code was previously returning uninitialized garbage in the
tstamp because it wasn't initialized anywhere.
This change zero-fills the tstamp in the two places it is used
before calling snd_compr_update_tstamp(), and also has
snd_compr_update_tstamp() return an error indication if it
can't provide a tstamp. For the case of snd_compr_calc_avail()
it ignores this error because we still need to return info on
the available buffer space even if we can't provide tstamp
info - when the tstamp is not valid all fields are now
guaranteed to be zero.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A fairly quiet release for ASoC:
- Support for a wider range of hardware in the compressed stream code.
- The ability to mute capture streams as well as playback streams while
inactive.
- DT support for AK4642, FSI, Samsung I2S and WM8962.
- AC'97 support for Tegra.
- New driver for max98090, replacing the stub which was there.
Due to dependencies we've also got support for asynchronous I/O in regmap
and DTification of DMA support for Samsung platforms (used only by the
I2S driver and SPI) merged here as well.
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Merge tag 'asoc-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.9
A fairly quiet release for ASoC:
- Support for a wider range of hardware in the compressed stream code.
- The ability to mute capture streams as well as playback streams while
inactive.
- DT support for AK4642, FSI, Samsung I2S and WM8962.
- AC'97 support for Tegra.
- New driver for max98090, replacing the stub which was there.
Due to dependencies we've also got support for asynchronous I/O in regmap
and DTification of DMA support for Samsung platforms (used only by the
I2S driver and SPI) merged here as well.
Adds quirks and mixer support for the M-Audio Fast Track C600 USB
audio interface. This device is very similar to the C400 - the C600
simply has some more inputs and outputs, so the existing C400 support
is extended to support this device as well.
Signed-off-by: Matt Gruskin <matthew.gruskin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
