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>
Adding an 'm' will distinguish them from identical names in intel8x0.c.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At every resume a laptop I use prints this message (at KERN_ERR level):
ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2]
The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what
seems to be happening is that the register involved indicated a warm
reset for some time (as the ICH_AC97WARM bit was set) but by the time
the warning is printed, and that same register is checked again, that
bit is already cleared and only the ICH_AC97COLD bit is still set.
It turns out a warm reset needs some time to settle, but it is currently
checked right away. The test therefore fails the first time it is done
and schedule_timeout_uninterruptible() will be called. Once we return
from that jiffies is already (far) past end_time on this laptop, so we
exit the loop, print a warning, and exit the function while the warm
reset actually succeeded.
A way to fix this is to call usleep_range() after writing to the
register involved. A handful of tests suggest 500 usecs is a safe value.
(This might punish the "finish cold reset" case, but on this laptop such
a cold reset apparently never happens, so I can't say for sure.)
While we're at it drop the extra single tick from end_time, as it looks
rather silly.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
azx_init_pci() always writes PCI config register ICH6_PCIREG_TCSEL
although this looks to be only defined on Intel systems and has a
different meaning on AMD systems. On AMD systems the PCI interrupt pin
control register is modified instead.
Since the meaning of offset 0x44 in device specific configuration space is
unknown for devices by other vendors, we only exclude AMD systems to
retain the current behaviour.
Signed-off-by: Adam Lackorzynski <adam@os.inf.tu-dresden.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not initialize again the what has already been initialized as
multi outs, as this breaks surround speakers.
Tested-by: Bartłomiej Żogała <nusch88@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this change, a volume control named "Surround" or "Side" would
get an unnecessary index, causing it to be ignored by the vmaster and
PulseAudio.
Tested-by: Bartłomiej Żogała <nusch88@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When more than one pair of internal speakers is present, allow names
according to their channels.
Tested-by: Bartłomiej Żogała <nusch88@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pin config values would change the association instead of the
sequence, this commit fixes that up.
Tested-by: Bartłomiej Żogała <nusch88@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current AES32 firmware revision ID is 234, however, a user confirmed
that everything works fine with the previous revision, too.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The errata init verbs for CS42xx codecs contain the verbs to set
the power-state of SPDIF nodes to D3, which seem to break the SPDIF
output on some MacBooks. Since this is executed during the power-up
initialization, we shouldn't turn them down there.
Reported-by: Arun Raghavan <arun.raghavan@collabora.co.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since multiple codec drivers already use the input-jack stuff, let's
make common helper functions to reduce the duplicated codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing HDMI codec IDs for new Nvidia stuff.
Note that ID 0x17 isn't assigned to anything so far, as suggested by
Stephen.
[Modified to get rid of 0x17 by tiwai]
Signed-off-by: Richard Samson <samson.richard@gmail.com>
Acked-by: Acked-By: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes an error in the jack detection reporting,
causing the jack detection sometimes not to be reported
correctly to the input subsystem. It should apply to several
Realtek codecs.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Depending on the model and the presence of a TCO module, the number of
midi ports varies. Some have 1 port (MADIface), some have 2 (default),
with TCO, there are 3.
Don't hardcode the number of midi ports to initialize.
This patch also fixes a boot lockup on MADIface.
[Coding-style fixes by tiwai]
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In contrast to the PCIe version (RME AES), the PCI version (RME AES32)
has a different firmware revision.
This patch adds the missing PCI revision.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch below removes an extra "l" in the word.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch below removes an extra "l" in the word.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also fix number of 92HD87 pins to exclude invalid pins.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lately I sent patch that switched lfe with side in mixer for
acer-aspire-4930g. Then I connected 5.1 speaker system and noticed that
lfe slider wasn't working and that old lfe slider worked. What I'm doing
now is:
- reverting old patch
- adding internal lfe slider
- removing side as it is superfluous (ALC888S-VC is 7.1 but in fact
laptop can only do 5.1 and it is so in drivers for MS Windows)
Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These two Dell machines have been reported working well with
the ideapad model.
BugLink: http://bugs.launchpad.net/bugs/723676
Cc: stable@kernel.org
Tested-by: David Chen <david.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant 506e/20590 has the same graph as the rest of the 5066 family.
BugLink: http://bugs.launchpad.net/bugs/723672
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Original patch by Dan Rosenberg <drosenberg@vsecurity.com> under commit
e68d3b316a. I'm copying his text here:
The SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO ioctl in hdspm.c allow unprivileged
users to read uninitialized kernel stack memory, because several fields
of the hdspm_config struct declared on the stack are not altered
or zeroed before being copied back to the user. This patch takes care
of it.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fredrik Lingvall <fredrik.lingvall@gmail.com> has discovered wrong
frequency and sync detection on AES32. According to him, the provided
patch fixes these issues.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In contrast to the RME MADI card, coax/optical selection on the MADIface
is done via a physical switch located at the breakout box. Obviously,
the driver cannot switch ports in software.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Caused by two typos, no output channel mappings were assigned for
MADI/MADIface at double/quad speed.
The channel mapping is indeed identical to the single speed mapping, the
cards will simply use the first N channels.
Signed-off-by: Florian Faber <faber@faberman.de>
Signed-off-by: Fredrik Lingvall <fredrik.lingvall@gmail.com>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without calling an appropriate rule, AES/AES32 cards would announce a
theoretical channel count of 64 (HDSPM_MAX_CHANNELS), leading to the
already known bug:
[37422.640481] ------------[ cut here ]------------
[37422.640487] WARNING: at sound/pci/rme9652/hdspm.c:5449
snd_hdspm_ioctl+0x18f/0x202 [snd_hdspm]()
[37422.640489] Hardware name: PRIMERGY RX100 S6
[37422.640490] BUG? (info->channel >= hdspm->max_channels_in)
[37422.640492] Modules linked in: snd_hdspm snd_seq_midi ipmi_watchdog
ipmi_poweroff ipmi_si ipmi_devintf ipmi_msghandler i2c_i801 e1000e
snd_rawmidi power_meter [last unloaded: snd_hdspm]
[37422.640501] Pid: 22231, comm: jackd Tainted: G D W
2.6.36-gentoo-r5 #5
[37422.640502] Call Trace:
[37422.640508] [<ffffffff8103db3a>] warn_slowpath_common+0x80/0x98
[37422.640511] [<ffffffff8103dbe6>] warn_slowpath_fmt+0x41/0x43
[37422.640514] [<ffffffff81034306>] ? get_parent_ip+0x11/0x42
[37422.640518] [<ffffffffa0055763>] snd_hdspm_ioctl+0x18f/0x202
[snd_hdspm]
[37422.640522] [<ffffffff813fd626>] snd_pcm_channel_info+0x73/0x7c
[37422.640525] [<ffffffff814001e9>] snd_pcm_common_ioctl1+0x326/0xb01
[37422.640527] [<ffffffff81034306>] ? get_parent_ip+0x11/0x42
[37422.640531] [<ffffffff8105be6c>] ? __srcu_read_unlock+0x3b/0x59
[37422.640533] [<ffffffff81400bce>] snd_pcm_capture_ioctl1+0x20a/0x227
[37422.640537] [<ffffffff811e599c>] ? file_has_perm+0x90/0x9e
[37422.640540] [<ffffffff81400c15>] snd_pcm_capture_ioctl+0x2a/0x2e
[37422.640543] [<ffffffff810f2c69>] do_vfs_ioctl+0x404/0x453
[37422.640546] [<ffffffff810f2d09>] sys_ioctl+0x51/0x74
[37422.640549] [<ffffffff81002aab>] system_call_fastpath+0x16/0x1b
[37422.640552] ---[ end trace 0cd919cd68118082 ]---
We already have all the right values in place, we simply have to inform
the upper layers about this restriction.
Note that snd_hdspm_hw_rule_rate_out_channels and
snd_hdspm_hw_rule_rate_in_channels must not be called on AES32, because
the channel count is always 16, no matter of the samplerate in use.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only RayDAT and AIO provide sane buffer pointers that can be used with
HDSPM_BufferPositionMask, on all other cards, this would result in a
wrong HW pointer leading to xruns and these messages:
[260808.916788] BUG: pcmC0D0p:0, pos = 2976, buffer size = 1024, period size = 512
[260808.961124] BUG: pcmC0D0c:0, pos = 4944, buffer size = 1024, period size = 512
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This typo caused some microphone inputs not to be correctly
initialized on VIA codecs.
Reported-By: Mark Goldstein <goldstein.mark@gmail.com>
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add " Playback Volume" to 10 bands Equalizer Controls of au88x0 so that
alsa-lib won't regard them as "Capture Volume".
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without this patch, one line-out and one speaker and
Conexant's auto parser would announce (non-working) surround
capabilities.
BugLink: http://bugs.launchpad.net/bugs/721126
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant 20641 has several inputs to its ADC node, with one selector
and individual amps for all inputs. This patch adds support in the
Conexant auto parser to handle that case.
It also means that the pin node's volume is being renamed to "Boost"
to avoid name clash with the new volume controls on the ADC node.
BugLink: http://bugs.launchpad.net/bugs/719524
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make newly created AC97 emulation of azt3328 known to the AC97 layer
side.
- relocate common functions to the top (due to definition after use)
- rename control names
- adjust 3D settings to the card's custom layout of this register
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make use of the very flexible ALSA ac97 layer (hooks for custom I/O!)
on this weird AC97 copycat hardware,
via semi-extended I/O translation/emulation.
Some 5kB binary/loaded size saved (well... additional huge AC97 module
penalty not factored in, of course ;-P).
Given that the driver previously had 20kB that's not bad,
but the much more important thing is to have AC97 layer stress-tested
with a thoroughly weird AC97 copycat (or, simply put, if it were not for
this AC97 test aspect, this effort would merely have been a nut job ;).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use AC97 macros (sometimes already existing, or newly added)
instead of error-prone repetition of open-coded values.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bug reporter claims that position_fix=1 is needed for his
microphone to work. The controller PCI vendor-id is [1002:4383] (rev 40).
Reported-by: Kjell L.
BugLink: http://bugs.launchpad.net/bugs/718402
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This netbook has a only one jack output and an internal mic.
By default, mic and jack sense aren't working. Using lenovo-101e
parameters makes both work.
The device seems based on a Sharetronic Q70, so this should fix audio for
this model too.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Difference in major.minor between driver and firmware is an error now.
Release version mismatch give a warning.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Interrupt flag used for message handshake will be required for
stream interrupts, so conditionally compiled code without
HPI6205_NO_HSR_POLL defined can never be used; removing it.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create and use HPI_ERROR_DSP_COMMUNICATION _DSP_BOOTLOAD, rather than
backend-specific error codes (now returned as data with the error).
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
asihpi.c don't link playback and capture streams, there is too much
offset between them.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some error codes had duplicate meanings. Just use one.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reported samples_played from card may be inaccurate, so don't use it.
Update control names to be closer to alsa standard practice.
Also fixed some accidentally lowercased strings.
[Removed adriver.h inclusion for external module builds by tiwai]
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove many unused functions.
Update some message and cache structs.
Use pci info directly from pci_dev.
Allow control cache elements with variable size, and handle
large message/response from dsp.
hpi6000 and hpi6205: fix error path when adapter bootload fails.
hpimsgx.c get rid of code duplicated in hpicmn.c
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Log HPI messages and responses in consistent numeric format,
which can be post-processed to get strings.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the reporter, node 0x15 needs to be muted for subwoofer
to stop sounding. This pin is marked as unused by BIOS, so fix that.
BugLink: http://bugs.launchpad.net/bugs/715877
Cc: stable@kernel.org (2.6.37+)
Reported-by: Hans Peter
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an empty string is passed to patch option, the driver should
ignore it. Otherwise it gets an error by trying to load it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 53d7d69d8f
ALSA: hdmi - support infoframe for DisplayPort
dropped the initialization of CA field accidentally.
This resulted in only two-channel LPCM mode on Nvidia machines.
Reference: kernel bug 28592
https://bugzilla.kernel.org/show_bug.cgi?id=28592
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Built-in sub-woofer can now be controlled by lfe slider instead of
side slider on Acer Aspire 5930g
Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Conexant codec driver adds the jack arrays in init callback which
may be called also in each PM resume. This results in the addition of
new jack element at each time.
The fix is to check whether the requested jack is already present in
the array.
Reference: Novell bug 668929
https://bugzilla.novell.com/show_bug.cgi?id=668929
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This card uses separate I2S outputs for the front speakers and
headphones, and reverses the order of the three speaker outputs.
To work around this, add a model-specific callback to adjust the
controller's playback routing.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/708521
This Edge 13 model has an internal mic at 0x1a and should
therefore use the asus quirk.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As requested by Takashi and Jaroslav, these arrays should not be in the
header file.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Inform users about the newly added support for RayDAT and AIO.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Incorporate changes by Florian Faber into hdspm.c. Code taken from
http://wiki.linuxproaudio.org/index.php/Driver:hdspe
Heavily reworked to mostly comply with the coding standard (whitespace
fixes, line width, C++ style comments)
The code was tested and confirmed to be working on RME RayDAT.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/707902
More Thinkpad machines with invalid SKU found, that disables
automute between speakers and headphones on these machines.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This typo caused the dmesg output of the supported bits of HDMI
to be cut off early.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cleanly revert to non-macro implementation of
snd_azf3328_codec_setfmt(), to fix last-minute functionality breakage
induced by following checkpatch.pl recommendations without giving them
their due full share of thought ("revolting computer, ensuing PEBKAC").
I would like to thank Jiri Slaby for his very timely (in -rc1 even)
and unexpected (uncommon hardware) "recognition of the dangerous situation"
due to his very commendable static parser use. :)
Reported-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed the Asus A52J quirk to use the asus model instead of the
hp_laptop model, which fixes the external mic input. Added an Asus
U50F quirk to use the asus model. For the cxt5066 codecs, added
checking of the digital output pins to determine which digital output
nodes to use instead of always using node 0x21, since some systems
have node 0x12 connected to a SPDIF out jack.
[A slight modification for better readability by tiwai]
Signed-off-by: Andy Robinson <ajr55555@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/701271
This new model, named "asus", is identical to the "hp_laptop" model,
except for the location of the internal mic, which is at pin 0x1a.
It is used for Asus K52JU and Lenovo G560.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Four very similar procedures - one for each model - now
refactored into one. This isn't all duplicated code, but a step
in the right direction.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'cs4398_regs' in 'struct xonar_cs43xx' is an array of 'u8' with a size of
8. So, this code in sound/pci/oxygen/xonar_cs43xx.c::dump_d1_registers()
for (i = 2; i <= 8; ++i)
snd_iprintf(buffer, " %02x", data->cs4398_regs[i]);
will overrun the array when 'i == 8'.
I guess that what's needed to fix it is the trivial patch below, but I
must admit that I have no idea about this code, so I may very well be
wrong. Additionally, I have no way to actually test this, so all I know is
that the below compiles. Someone who actually knows this code should take
a look before anything is comitted - consider the below (not much more
than) a bug report.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/705323
Thinkpad Edge 14 has one more SSID that suffers from disabled auto-mute.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_realtek.c: In function ‘alc_apply_fixup’:
sound/pci/hda/patch_realtek.c:1724:14: warning: unused variable ‘modelname’
snd_printdd() is evaluated only when CONFIG_SND_DEBUG_VERBOSE=y.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a bunch of
warning: ‘inline’ is not at beginning of declaration
messages when building a 'make allyesconfig' kernel with -Wextra.
These warnings are trivial to kill, yet rather annoying when building with
-Wextra.
The more we can cut down on pointless crap like this the better (IMHO).
A previous patch to do this for a 'allnoconfig' build has already been
merged. This just takes the cleanup a little further.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
This reverts commit 03b7a1ab55.
This commit was mistakenly re-introduced. While the change is harmless
(as ALC887 uses patch_alc888() now), we should get rid of any wrong code.
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 - Fix initialization for HP 2011 notebooks
ALSA: hda - Add support for VMware controller
ALSA: hda - consitify string arrays
ALSA: hda - Add add multi-streaming playback for AD1988
ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code
ASoC: WM8990: msleep() takes milliseconds not jiffies
ALSA : au88x0 - Limit number of channels to fix Oops via OSS emu
ALSA: constify functions in ac97
ASoC: WL1273 FM radio: Fix breakage with MFD API changes
ALSA: hda - More coverage for odd-number channels elimination for HDMI
ALSA: hda - Store PCM parameters properly in HDMI open callback
ALSA: hda - Rearrange fixup struct in patch_realtek.c
ALSA: oxygen: Xonar DG: fix CS4245 register writes
ALSA: hda - Suppress the odd number of channels for HDMI
ALSA: hda - Add fixup-call in init callback
ALSA: hda - Reorganize fixup structure for Realtek
ALSA: hda - Apply Sony VAIO hweq fixup only once
ALSA: hda - Apply mario fixup only once
ALSA: hda - Remove unused fixup entry for ALC262
The driver was using an initial value for the clock on the SPI bus
which was read from ICE1712 EEPROM,
ice->eeprom.data[ICE_EEP1_GPIO_STATE] & ICE1712_DELTA_AP_CCLK (0x02)
It appears some cards have it default high, some cards
have it default low. On my Delta 66 rev. E:
$ cat /proc/asound/M66/ice1712 | grep 'GPIO state'
GPIO state : 0x70 /* ICE1712_DELTA_AP_CCLK bit is zero */
On my Audiophile 2496:
$ cat /proc/asound/M2496/ice1712 | grep 'GPIO state'
GPIO state : 0xfe /* ICE1712_DELTA_AP_CCLK bit is one */
It must be raised before the first SPI write happens, or the write will
fail, leading to:
[ 23.248721] invalid CS8427 signature 0x0: let me try again...
I theorize that 4eb4550ab3
is no longer needed, it was a different way to workaround
the problem.
[fixed variable decleration by tiwai]
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes for HP 2011 notebooks: enable dock ports and disable BTL
initialization in the driver.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the new PCI ID 0x15ad and device ID 0x1977 for VMware HDAudio
Controller.
[changed to use AZX_DRIVER_GENERIC by tiwai]
Signed-off-by: Bankim Bhavsar <bbhavsar@vmware.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Attached a patch which add a new model to support multi-streaming
playback for ad1988.
playback another stereo stream through the front panel headphone on
device 2 while playback through the speakers connected to rear panel
on device 0 at the same time.
Tested with ad1988a rev2 codec on asus P5B-V motherboard.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix playback/capture channels patch to change supported playback
channels of au8830 to 1,2,4 and capture channels to 1,2.
This prevent oops when oss emulation use SNDCTL_DSP_CHANNELS to
set 3 Channels
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'linux-next' of git://git.kernel.org/pub/scm/linux/kernel/git/jbarnes/pci-2.6:
PCI/PM: Report wakeup events before resuming devices
PCI/PM: Use pm_wakeup_event() directly for reporting wakeup events
PCI: sysfs: Update ROM to include default owner write access
x86/PCI: make Broadcom CNB20LE driver EMBEDDED and EXPERIMENTAL
x86/PCI: don't use native Broadcom CNB20LE driver when ACPI is available
PCI/ACPI: Request _OSC control once for each root bridge (v3)
PCI: enable pci=bfsort by default on future Dell systems
PCI/PCIe: Clear Root PME Status bits early during system resume
PCI: pci-stub: ignore zero-length id parameters
x86/PCI: irq and pci_ids patch for Intel Patsburg
PCI: Skip id checking if no id is passed
PCI: fix __pci_device_probe kernel-doc warning
PCI: make pci_restore_state return void
PCI: Disable ASPM if BIOS asks us to
PCI: Add mask bit definition for MSI-X table
PCI: MSI: Move MSI-X entry definition to pci_regs.h
Fix up trivial conflicts in drivers/net/{skge.c,sky2.c} that had in the
meantime been converted to not use legacy PCI power management, and thus
no longer use pci_restore_state() at all (and that caused trivial
conflicts with the "make pci_restore_state return void" patch)
The commit ad09fc9d21 didn't cover the
case for Intel and Nvidia HDMIs, where hdmi_pcm_open() is called.
Put the hw_constraint there, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In hdmi_pcm_open(), the evaluated PCM hw parameters are stored in
hinfo, but these aren't properly set back to the current runtime
record since these have been set beforehand in azx_pcm_open().
This patch fixes the behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It looks like that HDMI codecs don't support the odd number of channels
although HD-audio spec doesn't have the restriction. Add the
hw_constraint to limit to only the even number of channels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some cases, the fix-up is required in the init callback to be called
both at the first initialization and at the resume. The new action type
ALC_FIXUP_ACT_INIT is used for this case.
So far, only ALC275_FIXUP_SONY_HWEQ uses this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of keeping various data types in a single record, put the
type field and keep a single value in each entry, but allows chaining
multiple fixup entries. This allows more flexible data management
(see ALC275_FIXUP_SONY_HWEQ for example).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When only one mic is available and it's an analog mic, the current
IDT/STAC parser may give an Oops.
Reference: bko#25692
https://bugzilla.kernel.org/show_bug.cgi?id=25692
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
With GPIO2-fixup, another fixup for NID 0x19 was missing because the
fixup is applied only once. Add the corresponding verb to the entry.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
SONY VAIO ALC275 default BIOS verb set the hardware EQ to disable.
Enable it when driver is loading.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo NB 0x9e54 use the external AMP in an inverted manner.
Set EAPD to low will enable the AMP.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added hardware constraint in patch_hdmi.c to disable
channels 4/6 which are not supported by some older
NVIDIA GPUs.
Signed-off-by: Nitin Daga <ndaga@nvidia.com>
Acked-By: Stephen Warren <swarren@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dynamic PCM restriction based on ELD information may lead to the
problem in some cases, e.g. when the receiver is turned off. Then it
may send a TV HDMI default such as channels = 2. Since it's still
plugged, the driver doesn't know whether it's the right configuration
for future use. Now, when an app opens the device at this moment,
then turn on the receiver, the app still sends channels=2.
The right solution is to implement some kind of notification and
automatic re-open mechanism. But, this is a goal far ahead.
This patch provides a workaround for such a case by providing a new
module option static_hdmi_pcm for snd-hda-codec-hdmi module. When
this is set to true, the driver doesn't change PCM parameters per
ELD information. For users who need the static configuration like
the scenario above, set this to true.
The parameter can be changed dynamically via sysfs, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Add a mixer control to switch between the optical and coaxial S/PDIF
inputs on the HT-Omega Claro and Claro halo cards.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable the X-Meridian's CD input and the X-Meridian 2G's potential
MIDI ports.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of the generic Oxygen, use the actual card name, if known.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apparently, the revision is 2 on all sold sound cards, so this
information is not actually useful.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to select between the on-board and extension board
S/PDIF inputs for the X-Meridian (2G).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a mixer control to prevent capturing S/PDIF samples that are not
marked as valid (non-audio or corrupted samples).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the info callback by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify info callbacks by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the info callback by using the snd_ctl_enum_info() helper function.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the helper function snd_ctl_enum_info() to fill out the
elem_info fields for an enumerated control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add experimental support for the Asus Xonar HDAV1.3 Slim sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add experimental support for the Asus Xonar DG sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the AuzenTech X-Meridian 7.1 2G sound card.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the CSxxxx and AKxxxx DAC/ADC chips, the MCLK factor in double rate
modes (64-96 kHz) can be reduced to 128x without reducing sound quality.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the get_i2s_mclk callback with tables of MCLK values. This
simplifies the MCLK-handling code in both the framework and the model-
specific drivers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not apply the headphone gain offset to any but the front DAC. These
DACs would not be used in headphone mode, so this saves a few register
writes.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the DAC Oversampling mixer control because this setting does not
make much sense.
For cards with the H6 daughterboard, 128x oversampling was disabled
anyway because these high MCLK frequency would not be compatible with
the connector cable.
For cards without the H6 daughterboard, 128x gives a slightly higher
output quality; there is no reason to reduce it to 64x except for saving
power, but then these cards have not been designed to be power efficient
anyway (the D2's blinkenlights cannot be disabled).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because of the unshielded connector cable, it is important to use as low
a master clock frequency as possible with the H6.
For double rate modes (64-96 kHz), the MCLK rate is unconditionally
lowered from 512x to 256x because the higher rate would not improve
anything.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The clock output of the CS2000, which is used as master clock for the
DACs, was using half the actual master clock frequency for some reason.
Using the theoretically correct frequency seems also to work in practice.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Xonar Essence ST Deluxe, remove all mixer controls that would
require I2C communication with the third DAC, which does not work
because of an addressing conflict with the CS2000 chip.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the PCM format used for the PCM1796 from left-justified to I2S to
ensure that the correct format is used even for the Essence ST Deluxe's
center/LFE DAC, where I2C does not work because of an address conflict
with the CS2000 chip.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM1796 needs the master clock for I2C communication to work, so
add delays after clock changes to ensure that the clock is stable when
we try to write the DACs' registers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To make the I2C communication reliable when using the H6 daughterboard,
reduce the I2C clock frequency.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix wrong register bits for SPI clock cycle times longer than 160 ns,
and adjust the polling loop timeout for these speeds.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of DACs can now be deduced from the dac_channels_mixer field,
so the private_data field is no longer needed.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For cards like the Xonar HDAV1.3, differentiate between the number of
PCM channels that can be played and the number of channels whose volume
can be adjusted.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, Realtek auto-parser assumed that the multiple pins are only for
line-outs, and assigned the channel names like Front, Surround, etc for
the multiple outputs. But, there are devices that have multiple
headphones, and these can be better controlled with the corresponding
control-name like "Headphone" with indicies.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple headphone pins are defined without line-out pins, the
driver takes them as primary outputs. But it forgot to set line_out_type
to HP by assuming there is some rest of HP pins. This results in some
mis-handling of these pins for Realtek codec parser. It takes as if
these are pure line-out jacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
BugLink: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5184
A user reported on the alsa-devel mailing list that he needs to use
the vostro model quirk to have audible playback, so apply it for his
PCI SSID.
Reported-and-tested-by: Fernando Lemos <fernandotcl@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/689036
Many new Lenovos need the ideapad quirk. Also, since the
auto parser for this chip is far from optimal, the regression
risk is low (although not zero).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If more than one mic is present with different locations,
e g "Front Mic" and "Rear Mic", they can use the same index (0),
since their names are different.
Previous behavior was to have "Front Mic" as index 1, causing it
to be ignored by e g PulseAudio.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/697240
If the "Volume" suffix is not given, alsa-lib gets confused and
loses the dB information at the simple element level.
Boosts generally affects both playback and capture, as they are
applied early in the chain. Hence no "Playback" or "Capture" in
the suffix.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/696493
According to datasheet (and real-world testing), IDT 92HD88B can
have internal mics at NID 0x11 and 0x20, so enable them accordingly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- much improved implementation due to clean codec hierarchy
- preparation for potential per-codec spinlock change
NOTE: additionally removes a chip->pcm[codec_type] NULL ptr check
(due to it requiring access to external chip struct),
however I believe this to be ok since this condition should not occur
and most drivers don't check against that either.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- use a separate variable for the frequency part, don't always "or" it
- use a "clever"(?) macro to shorten the code
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- correct samples to be POSIX shell compatible
- add logging of jiffies value in _pointer()
- several comments
- cleanup
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sjoerd Simons reports that, without using position_fix=1, recording
experiences overruns. Work around that by applying the LPIB quirk
for his hardware.
Reported-and-tested-by: Sjoerd Simons <sjoerd@debian.org>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The assignment to the local variable 'channel' in
snd_ca0106_pcm_pointer_capture() is a little crazy. Order of assignment is
undefined. This fixes it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
pci_restore_state only ever returns 0, thus there is no benefit in
having it return any value. Also, a large majority of the callers do
not check the return code of pci_restore_state. Make the
pci_restore_state a void return and avoid the overhead.
Acked-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Signed-off-by: Jon Mason <jon.mason@exar.com>
Signed-off-by: Jesse Barnes <jbarnes@virtuousgeek.org>
The fix-up entries by the commit 2785591a97
ALSA: hda - Add fix-up for Sony VAIO with ALC275 codecs
weren't applied in the right position. They had to be before the quirk
entry matching to all Sony devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a mixer control element was already created with the given name,
try to find another index for avoiding conflicts, instead of breaking
with an error. This makes the driver more robust.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the d-mics are assigned to the same purpose of another analog mic
pins, the driver doesn't compute the index properly, resulting in an
error with "existing control". This patch fixes it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conflicts:
MAINTAINERS
arch/arm/mach-omap2/pm24xx.c
drivers/scsi/bfa/bfa_fcpim.c
Needed to update to apply fixes for which the old branch was too
outdated.
ALC275 doesn't require the ALC269 (and its variants) specific init
sequences. Add the check of codec id.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set GPIO2 for some Sony VAIO with ALC275 to fix speaker output.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change non-standard mic control names to standard control names
to clean up the namespace.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Usually external microphones are just labelled "Mic", so rename
"Ext Mic" and "External Mic" to "Mic" to clear up the namespace.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Int Mic" and "Internal Mic" both mean the same thing, so rename
the former to the latter in order to clean up the namespace a little.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/580006
SKU turns off auto-mute for these machines, so ignore the SKU.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the recent change for multiple mics assignment, we need to handle
the index of each Mic Boost control respectively. Otherwise the driver
gets the control element conflicts, and gives the unsable state.
Reference: kernel bug 25002
https://bugzilla.kernel.org/show_bug.cgi?id=25002
Reported-and-tested-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Multiple quirk functions were using the exact same code to verify if the Mic
jack was plugged and mute the Mic accordingly
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set default association/sequence right on pin 0x17 in order for
the automatic parser to recognize the subwoofer correctly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/690530
The SKU value of this machine dictates that auto-mute should be
disabled. Since the SKU value is similar to the PCI SSID, the most
likely conclusion is that the SKU value should be ignored.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mute speakers when a line-out jack is plugged as well as headphone jacks
with the new Conexant codec parser in the auto mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
flush_scheduled_work() is deprecated and scheduled to be removed.
* cancel[_delayed]_work() + flush_scheduled_work() ->
cancel[_delayed]_work_sync().
* wm8350, wm8753 and soc-core use custom code to cancel a delayed
work, execute it immediately if it was pending and wait for its
completion. This is equivalent to flush_delayed_work_sync(). Use
it instead.
Signed-off-by: Tejun Heo <tj@kernel.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rev. E of the M-Audio Delta 66 is partially supported (commit
ef2cd2ccad), but the layout of the GPIO
pins was still unclear. This patch adds the GPIO definitions so that
communication to the CS8247 & 2x AK4524 works correctly.
ALSA bug#3327 has more details; users cap & jhunt report there that the
GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 =
CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1). There has been a lot of
conflicting information in the bug, but given these definitions, my
Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain
settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz.
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/497546
Confirmed that the ideapad model works better than the current
quirk for Dell Vostro 320.
Cc: stable@kernel.org (2.6.35+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
create fixup function for the mario model and override amp capabilities
for NID 0x2
Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Facilitate fixup for realtek codecs via modelname lookup of fixup
data. Fallback to quirk based lookup in absence of model definition.
Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some newer chips have more than one HDMI output, but usually not
all of them are exposed as physical jacks. Removing the unused
PCM devices (as indicated by BIOS in the pin config default) will
reduce user confusion as they currently have to choose between
several HDMI devices, some of them not working anyway.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc()
is called for every SAD (Short Audio Descriptor) in the ELD data. For
LPCM coding type SAD defines the supported sample sizes. For several
other coding types (such as AC-3), a maximum bitrate is defined.
The maximum bitrate and sample size fields are not always cleared.
Therefore, if a device is unplugged and a different one is plugged in,
and the coding types of some SAD positions differ between the devices,
the old max_bitrate or sample_bits values will persist if the new SADs
do not define those values.
The leftover max_bitrate and sample_bits do not cause any issues other
than wrongly showing up in eld#X.Y procfs file and kernel log.
Fix that by always clearing sample_bits and max_bitrate when reading
SADs.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit bbbe33900d added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, according to CEA-861-D no SAD is needed for basic audio
(32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a
basic audio flag in the CEA EDID Extension.
The flag is not present in ELD. However, as all audio capable sinks are
required to support basic audio, we can assume it to be always
available.
Fix allowed audio formats with sinks that have SADs (Short Audio
Descriptors) which do not completely overlap with the basic audio
formats (there are no reports of affected devices so far) by always
assuming that basic audio is supported.
Reported-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit bbbe33900d added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, it wrongly assumes that the bits 0-2 of the first byte of
CEA Short Audio Descriptors mean a supported number of channels. In
reality, they mean the maximum number of channels (as per CEA-861-D
7.5.2). This means that the channel count can only be used to restrict
max_channels, not min_channels.
Restricting min_channels causes us to deny opening the device in stereo
mode if the sink only has SADs that declare larger numbers of channels
(like Primare SP32 AV Processor does).
Fix that by not restricting min_channels based on ELD information.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: Jean-Yves Avenard <jyavenard@gmail.com>
Tested-by: Jean-Yves Avenard <jyavenard@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reformat and update the comments that describe the hardware connections
on the various models.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of the hardcoded "CMI8788", show the actual chip name.
Note: This is neither what the chip is (it's always the same),
nor what the chip is actually called.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To help with debugging, add the registers of the model-specific
codecs to the controller and AC97 register dump in the proc file.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "Front Panel" switch on the Xonar D1/DX actually switches only the
output direction, so mark it appropriately.
The front panel microphone is controlled by the FMIC2MIC bit of the
CM9780. It was unconditionally enabled on the D1/DX and never set on
the ST(X); add a control for it. Selecting the front panel microphone
as source does not actually disable the microphone jack, but this is
bug-compatible with the Windows driver, and users rely on it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The GPIO bit that enables analog output on the Xonar HDAV1.3 also
disables the HDMI audio output, so we better add a switch for it.
Hopefully, this is sufficient to make the HDMI output work.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize the configuration of some unknown GPIO output bits (that
might not be used at all) to be the same as in the Windows driver, just
to be sure.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/685161
The reporter of the bug states that he must use position_fix=1 to enable
capture for the internal microphone, so set it for his machine's PCI
SSID. Verified using 2.6.35 and the 2010-12-04 alsa-driver build.
Reported-and-tested-by: Ralph Wabel <rwabel@gmx.net>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Switch to the generic hdmi parser for codec id 1002:aa01 (ATI R6xx
HDMI), as the codec appears to work fine with it.
Note that the codec is still limited to stereo output only, despite it
reportedly being multichannel capable. Some as of yet unknown quirks
will be needed to get that working.
Testing was done on 2.6.36 by John Ettedgui.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Tested-by: John Ettedgui <john.ettedgui@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a quirk to cxt5066_cfg_tbl to enable jack sense for ThinkPad Edge 13.
Reference: http://launchpad.net/bugs/685015
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of hard-coded magic numbers, properly define and use macros
for improve the readability. Also, dell_automute is handled samely
as thinkpad, since it also sets port_d_mode, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the docking station for the Lenovo T410 and T410s, the line-out
doesn't work. The trouble seems to be that it generates a plug event,
but then doesn't report that the jack is connected. So automute mutes
the jack when you plug something into it. The following patch (next
message) fixes it.
Signed-off-by: John Baboval <john.baboval at virtualcomputer.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/595482
The original reporter states that audible playback from the internal
speaker is inaudible despite the hardware being properly detected. To
work around this symptom, he uses the model=lg quirk to properly enable
both playback, capture, and jack sense. Another user corroborates this
workaround on separate hardware. Add this PCI SSID to the quirk table
to enable it for further LG P1 Expresses.
Reported-and-tested-by: Philip Peitsch <philip.peitsch@gmail.com>
Tested-by: nikhov
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai