Extend the existing auto-parser for CX2064x for cxt5051 codec.
Now the auto-parser supports ADC-switching for this codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of keeping always EAPD on, turn on/off appropriately at jack
plugging in Conexant auto-parser mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Autodetect TEA575x tuner connection type during init. This allows tuner to
work out-of-the box.
tea575x_tuner module parameter remains functional to force tuner type.
Tested with SF256-PCP and SF64-PCR.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use common functions to access TEA575x tuner - remove original read/write
functions and provide new pin manipulation functions instead.
Also convert the original triple implementation to a simple GPIO pin map.
Tested with SF256-PCP and SF64-PCR (added the GPIO pin for MO/ST signal
for them).
SF256-PCS untested (pin for MO/ST signal is a guess).
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use common functions to access TEA575x tuner - remove original read/write
functions and provide new pin manipulation functions instead.
Tested with SF64-PCE2 card.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AMD chipsets often behave pretty badly regarding the DMA position
reporting. It results in the bad quality audio recording.
Using position_fix=3 works well in general for them, so let's enable
it as default for AMD.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Renamed to Digial SRC Capture Switch for more correct representation.
Also fixed analog volume control on Lola161611 and lola881.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For assuring the synchronized state with the pause operation,
loop over the all linked streams and waits until all get ready
in a loop.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the refcounting for the exclusive SRC control.
Also, fixed the possible stall after PCM pause operations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added granularity and sample_rate_min module options.
The former controls the h/w access granularity. As default, it's set
to the max value 32.
The latter controls the minimum sample rate in Hz, as default 16000.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a single BDL for both buffers instead of allocating for each.
Also a few tune-up to avoid the stream stalls in the PCM code and
the prelimianry work for SG-buffer support are added, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec proc file becomes a read only that shows the codec widgets
in a text form. A new proc file, codec_rw, is introduced instead for
accessing the Lola verb directly by reading and writing to it.
Also, regs proc file shows the contents of DSD, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new driver for supporting Digigram Lola PCI-e boards.
Lola has a similar h/w design like HD-audio but with extended verbs.
Thus the driver is written similarly like HD-audio driver in the bus
part. The codec part is rather written in a fixed way specific to the
Lola board because of the verb incompatibility.
The driver provides basic PCM, supporting multi-streams and mixing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix NULL-dereference when try to use alt_playback since those codecs
which support multistreaming playback usually have more than 1 adc but
the driver should create alt_capture when spec->stream_analog_alt_capture
is also defined.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check of chained fixup list entry was done against the wrong element.
A stupid mistake during refactoring.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit c6b358748e.
It turned out that there are different pin configurations for this
PCI SSID, including multi-channel modes. And more proper fix for
allowing line-out mutes will come up in 2.6.40 tree, so we won't need
this fixup any more there.
Reported-by: Andrew Clayton <andrew@digital-domain.net>
Reported-by: Emmanuel Benisty <benisty.e@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI SSID is 1025:031c and the codec SSID is 1025:031d,
so the driver mistakes this for a SKU value, but looking at
the numbers, this is obviously wrong.
Cc: stable@kernel.org (2.6.38+)
BugLink: http://bugs.launchpad.net/bugs/761861
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PC Beep was not being reported as enabled on my EeePC 901:
SKU: enable_pcbeep=0x0
Signed-off-by: Daniel Cordero <danielcordero@lavabit.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Auto-Mute Mode control is useful even when only two outputs
(e.g. HP and speaker) are available. Then user can enable/disable
the auto-mute behavior on the fly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not only supporting the line-out automute as additional feature
to the existing headphone automute, now the headphone jack can
mute the line-out alone even without the speaker outs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By popular demands, I add the functionality to mute / unmute the
line-out jacks per the headphone plug / unplug. For achieving this
and keeping the compatibility with the old behavior, the new mixer
enum "Auto-Mute Mode" is added. With this, user can control the
auto-mute behavior either disabled, speaker-only or lineout+speaker.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another consolidation of auto-mute functions for the devices
controlling the output muts together with the master mixer switch,
typically found for ALC262 machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the common helper function and flags to support the auto-mute
per line-out jack detection, and also the mute of line-out jacks.
A few model-specific implementations are replaced with the common
helpers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models do mute on/off the connected mixer widget for the automatic
muting, instead of controlling the pin widget itself. This patch adds
the implementation of such type of auto-mute in the common helper
function, and reduces the redundant codes for each model preset.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two entry points for the headphone automute functions for
Realtek, alc_automute_amp() and alc_automute_pin(). These call the
same function in the end, so we can basically consolidate these
with a flag in spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In notify_aa_path_ctls(), adds 'rear mic' item and confirms the A-A
path control existing before notifying card that the A-A path volume
is muted if smart5.1 is enabled.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allow application such as gstreamer and wine which use
snd_pcm_hw_params_set_buffer_time_near() won't fail any more
since sound chips require special containt power 2 period bytes
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the support of "Channel Mode" enum control to Realtek
auto-parser. When line-in or mic-in jacks are capable to output and
free DACs are available, the driver allows to switch to multi-channel
mode via "Channel Mode" enum switch, as already implemented in some
preset cases.
Not implemented in all Realtek codecs. Currently, ALC880, 882, 861,
662 and the compatible codecs are supported.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow alc662_dac_to_mix() and alc662_look_for_dac() to parse
down the selector widget that is found in ALC880-type codecs,
and rename them to alc_auto_*() accordingly.
This is for the next coming multi-io extensions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For some motherboards with 5 or 6 audio jacks which had six or eight multiple
channels output, smart5.1 item is no useful and should be removed.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The workaround for AMD chipset via sync_write flag seems needed for
machines with Realtek codecs. So, it's better to activate it
generically in hda_intel.c from the beginning.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
EAPD power-down should be called also for normal shutup cases.
Let's move to there. This also fixes the compile warnings when
CONFIG_PM isn't set automatically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AMD chipset seems unstable in the normal operation mode, and it
seems requring more sensible access for each verb. Enabling sync_write
mode and allowing bus-reset is a sort of workaround for these chipset
stability issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When SND_HDA_NEEDS_RESUME is not defined, the compiler identifies that
the following symbols are static but not used:
restore_shutup_pins
hda_cleanup_all_streams
Fix warnings by adding SND_HDA_NEEDS_RESUME guards.
Signed-off-by: Mike Waychison <mikew@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This was reverted mistakenly in the recent update patch.
Fixed again.
Reported-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove "Front Playback Volume" and "Front Playback Switch" from emu10k1 only
for STAC9758/59
Since commit 7eae36fbd5
"Fix the confliction of 'Front' control",
the "Front Playback Volume" control created by commit
edf8e4565c
"emu10k1: Front channels via fxbus 8 and 9"
was removed
"Front Playback Volume" and "Surround Playback Volume" have same dB range
since I2S DAC of SB Live! and SB Live! Platinum does not has any hardware
volume control.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer laptops with ALC271x needs a magic initialization for digital-mic
to make it working with mono streams (and PulseAudio).
Added a fix-up applied to Acer with ALC271x generically.
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: hda - Don't query connections for widgets have no connections
ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums
ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
ALSA: HDA: Fix dock mic for Lenovo X220-tablet
ASoC: format_register_str: Don't clip register values
ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare
ASoC: zylonite: set .codec_dai_name in initializer
The connection lists are static and we can reuse the previous results
instead of querying via verb at each time. This will reduce the I/O
in the runtime especially for some codec auto-parsers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we now set up the connections and mutes dynamically in the
auto-parser, all static initializations via alc662_init_verbs & co are
no longer needed. Let's drop them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of static init array, better to determine the connection and
the mute status of the pin/mixer/DAC route dynamically. This fixes the
uninitialized mixer 0x0f on ALC892.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In cases where there is only one internal mic connected to ADC 0x11,
alc275_setup_dual_adc won't handle the case, so we need to add the
ADC node to the array of candidates.
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/752792
Reported-by: Vincenzo Pii
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MCP7x hardware computes the audio infoframe channel count
automatically, but requires the audio driver to set the audio
infoframe checksum manually via the Nv_VERB_SET_Info_Frame_Checksum
control verb.
When audio starts playing, nvhdmi_8ch_7x_pcm_prepare sets the checksum
to (0x71 - chan - chanmask). For example, for 2ch audio, chan == 1
and chanmask == 0 so the checksum is set to 0x70. When audio playback
finishes and the device is closed, nvhdmi_8ch_7x_pcm_close resets the
channel formats, causing the channel count to revert to 8ch. Since
the checksum is not reset, the hardware starts generating audio
infoframes with invalid checksums. This causes some displays to blank
the video.
Fix this by updating the checksum and channel mask when the device is
closed and also when it is first initialized. In addition, make sure
that the channel mask is appropriate for an 8ch infoframe by setting
it to 0x13 (FL FR LFE FC RL RR RLC RRC).
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add shutup callback to be called codec-specifically for avoiding pop
noises at suspend or shutdown. As a generic callback, just turn EAPD
off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, alc662_init_verbs[] is used for all ALC662-compatible chips,
but the EAPD controls for 0x15 in there is invalid for ALC892.
Also, since EAPDs should be set up in alc_auto_init_amp(), these static
elements aren't needed for auto-parser, too.
In this patch, the EAPD init verbs are split from alc662_init_verbs,
and applied only to static quirks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current alc662 parser doesn't set the DAC for the mixer 0x0f
properly for ALC892, which has 4 DACs while ALC662 has 3.
Fixed by implementing alc662_mix_to_dac() more genericly with the
dynamic widget list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ALC272-quirks use alc662_dac_nids instead of alc272_dac_nids.
This patch fixes these entries. No functional change since the first
two elements are identical in both arrays.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SB Live! Platinum CT4760P is just a 4 channels sound card with STAC9721 and
Philips UDA1334 DAC.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc662 series only have 3 DAC, so it can only support 5stack-dig
instead of 6stack-dig.
[updated HD-Audio-Models.txt as well by tiwai]
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some unneeded defintions
Use %pR to print resources
Make some data const
Consistent braces for else
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define and use pcm_debug_name if CONFIG_SND_DEBUG
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow older non DMA capable cards to use MMAP by
emulating the DMA using read and write functions,
and getting rid of copy & silence callbacks that
were used only by older cards.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the card drained status reporting for playback,
but allow it to persist for a few timer cycles before
signalling XRUN, to allow card to recover by itself.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clock source is neither capture nor playback,
so change 'Capture Clock' to 'Clock'.
Add spaces to control name string for consistency,
always 'PCM 0' , never 'PCM0'
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without the "thinkpad" quirk, the dock mic in
Lenovo X220 tablet edition won't work.
BugLink: http://bugs.launchpad.net/bugs/751033
Cc: stable@kernel.org
Tested-by: James Ferguson <james.ferguson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This quirk is needed for the docking station mic of
Lenovo Thinkpad X220 to function correctly.
BugLink: http://bugs.launchpad.net/bugs/746259
Cc: stable@kernel.org
Tested-by: James Ferguson <james.ferguson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To make the EV1938 chip work, add a magic bit and an extra delay.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Tino Schmidt <mailtinoshomepage@gmx.net>
Cc: all 2.6.x <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: Fix yet another race in disconnection
ALSA: asihpi - Update verbose debug print macros
ALSA: asihpi - Improve non-busmaster adapter operation
ALSA: asihpi - Support single-rate no-SRC cards
ALSA: HDA: New AD1984A model for Dell Precision R5500
ALSA: vmalloc buffers should use normal mmap
ALSA: hda - Fix SPDIF out regression on ALC889
ALSA: usb-audio - Support for Boss JS-8 Jam Station
ALSA: usb-audio: add Cakewalk UM-1G support
sound/oss/opl3: validate voice and channel indexes
sound/oss: remove offset from load_patch callbacks
Replace local VPRINTK1 with snd_printdd.
Create local snd_printddd instead of VPRINTK2 for most verbose debug.
In most cases let snd_printk supply default level for messages.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make playback silence callback a no-op, card automatically outputs
silence when written data runs out.
Increasing update interval and thus minimum period avoids xrun on startup
or because of timer jitter.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cards without settable local samplerate and without SRC
still must have a valid samplerate.
This fixed rate is determined by reading the current rate for the card.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For codec AD1984A, add a new model to support Dell Precision R5500
or the microphone jack won't work correctly.
BugLink: http://bugs.launchpad.net/bugs/741516
Tested-by: Kent Baxley <kent.baxley@canonical.com>
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update unsolicited event process function via_unsol_event() to
make it can process more unsolicited events.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add some hardware related verbs in VT2002P initial verbs.
These verbs are used to fix Class-D speaker no sound issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a verb to enable control amplifier of stereo mixer in VT1718S
initial verbs. Set stereo mixer default amplifier value as un-mute.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a verb of power down jack detect in VT1708 initial verbs.
This verb is used to avoid noise caused by hardware issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modify side_mute_channel() and update_side_mute_status() functions
to fix invalid side channel mute issue of VT2002P, VT1812 and VT1802
codecs.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 5a8cfb4e8a
ALSA: hda - Use ALC_INIT_DEFAULT for really default initialization
changed to use the default initialization method for ALC889, but
this caused a regression on SPDIF output on some machines.
This seems due to the COEF setup included in the default init procedure.
For making SPDIF working again, the COEF-setup has to be avoided for
the id 0889.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=24342
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: HDA: Realtek: Avoid unnecessary volume control index on Surround/Side
ASoC: Support !REGULATOR build for sgtl5000
ALSA: hda - VIA: Fix VT1708 can't build up Headphone control issue
ALSA: hda - VIA: Correct stream names for VT1818S
ALSA: hda - VIA: Fix codec type for VT1708BCE at the right timing
ALSA: hda - VIA: Fix invalid A-A path volume adjust issue
ALSA: hda - VIA: Add missing support for VT1718S in A-A path
ALSA: hda - VIA: Fix independent headphone no sound issue
ALSA: hda - VIA: Fix stereo mixer recording no sound issue
ALSA: hda - Set EAPD for Realtek ALC665
ALSA: usb - Remove trailing spaces from USB card name strings
sound: read i_size with i_size_read()
ASoC: Remove bogus check for register validity in debugfs write
ASoC: mini2440: Fix uda134x codec problem.
Add support for VT1802 codec, which is similiar with VT2002P
except VT1802 has no Class-D and has some different pin widget
id.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for VT1705 codec, which is similiar with VT1708S
except it has 6 channels output.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use set_widgets_power_state() function to seperately control different
codecs' power management actions and to replace the original large
function. Also fix some wrong widgets power up sequence which caused
no sound issue under Smart5.1 mode and Independent HP mode.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar to commit 7e59e097c0, this patch
avoids unnecessary volume control indices for more
Realtek auto-parsers, e g the ALC66x family, on the "Surround" and "Side"
controls.
These indices cause these volume controls to be ignored by PulseAudio and
vmaster and should be removed whenever possible.
Cc: stable@kernel.org
Reported-by: Jan Losinski <losinski@wh2.tu-dresden.de>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since VT1708 didn't support the control of getting connection number,
building of headphone control will fail in via_hp_build() function.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct stream names of analog playback and capture streams
for VT1818S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add get_codec_type() in via_new_spec() function to make sure getting
correct codec type before building mixer controls.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modify mute_aa_path() function to support VT1718S codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modify via_independent_hp_put() function to support VT1718S and VT1812
codecs, and fix independent headphone no sound issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add TEA5757 radio tuner support to es1968 driver. This is found at least on
MediaForte SF64-PCE2 sound cards.
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/jikos/trivial: (47 commits)
doc: CONFIG_UNEVICTABLE_LRU doesn't exist anymore
Update cpuset info & webiste for cgroups
dcdbas: force SMI to happen when expected
arch/arm/Kconfig: remove one to many l's in the word.
asm-generic/user.h: Fix spelling in comment
drm: fix printk typo 'sracth'
Remove one to many n's in a word
Documentation/filesystems/romfs.txt: fixing link to genromfs
drivers:scsi Change printk typo initate -> initiate
serial, pch uart: Remove duplicate inclusion of linux/pci.h header
fs/eventpoll.c: fix spelling
mm: Fix out-of-date comments which refers non-existent functions
drm: Fix printk typo 'failled'
coh901318.c: Change initate to initiate.
mbox-db5500.c Change initate to initiate.
edac: correct i82975x error-info reported
edac: correct i82975x mci initialisation
edac: correct commented info
fs: update comments to point correct document
target: remove duplicate include of target/target_core_device.h from drivers/target/target_core_hba.c
...
Trivial conflict in fs/eventpoll.c (spelling vs addition)
The user-supplied index into the adapters array needs to be checked, or
an out-of-bounds kernel pointer could be accessed and used, leading to
potentially exploitable memory corruption.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch replaces use of the harcoded arrays of pins, muxes, digital
mics and adcs with the auto-generated ones using codec parsing and
auto-discovers all actually connected digital mic pins on 92HD8X-like
codecs
This patch also adds the support for d-mic on pin 0x20.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the mux for digital mic is different from the mux for other mics,
the current auto-parser doesn't handle them in a right way but provides
only one mic. This patch fixes the issue.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the default input-src selection code for alc268/269 to the init
part instead of the parser. The input-src selection might be overwritten
by init verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently some special handling for the unusual case like dual-ADCs
or a single-input-src is done in the tree-parse time in
set_capture_mixer(). But this setup could be overwritten by static
init verbs.
This patch moves the initialization into the init phase so that
such input-src setup won't be lost.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SDPIF status retrieval always returned the default settings instead of
the actual ones.
Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
microphone boost was set at +12dB, not +20dB (like in Windows driver
and in adc_conf structure declaration), some comments added.
Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The time-out in snd_atiixp_aclink_reset() is wrongly checked, and
it resulted in exiting from the loop at the first iteration.
Reported-by: Amir Shamsuddin <AmirS2+alsa@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Appending an 'm' will distinguish it from a similar struct in intel8x0.c
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
