The free callback is called at the state where no extra verbs are
executed, thus calling *_shutup() is useless, as it's checking the
shutdown flag. Remove such superfluous calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer Aspire AO756 has an analog built-in mic on nid 0x1b and an
external mic on nid 0x19. The BIOS doesn't set this up.
The mic detect on this Acer Aspire netbook and Acer C7 ChromeBook is
only valid when the headphone is plugged. The detect circuit relies on
the tip detect switch being closed on the jack. Tell hda_jack to ignore
the mic sense unless the headphones are plugged.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When 2.1 speakers are detected, use the corresponding channel map
instead of the standard map with front+rear surrounds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc269_toggle_power_output() was only use in ALC269VB. I rename it to
alc269vb_toggle_power_output().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I have a Lenovo ThinkPad T430 and an UltraBase Series 3 docking
station.
Without this patch, if I plug my headphones into the jack on the
computer, everything works fine. The computer speakers mute and the
audio is played in the headphones. However, if I plug into the docking
station headphone jack the computer speakers are muted but there is no
audio in the headphones.
Addresses https://bugs.launchpad.net/bugs/1060372
Signed-off-by: Joseph Salisbury <joseph.salisbury@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even when CONFIG_SND_DEBUG is not enabled, we don't want to
return an arbitrary memory location when the channel count is
larger than we expected.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When both an SPDIF and an HDMI device are created on the same card
instance, multiple IEC958 controls are created with indices=0, 1, ...
But the alsa-lib configuration can't know which index corresponds
actually to which PCM device, and both the SPDIF and the HDMI
configurations point to the first IEC958 control wrongly.
This patch introduces a (hackish and ugly) workaround: the IEC958
controls for the SPDIF device are re-labeled with device=1 when HDMI
coexists. The device=1 corresponds to the actual PCM device for
SPDIF, so it's anyway a better representation. In future, HDMI
controls should be moved with the corresponding PCM device number,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define()
always after fixup"), alc_auto_parse_customize_define was moved after
detection of ALC271X.
The problem is that detection of ALC271X relies on spec->cdefine.platform_type,
and it's set on alc_auto_parse_customize_define.
Move the alc_auto_parse_customize_define and its required fixup setup
before the block doing the ALC271X and other codec setup.
BugLink: https://bugs.launchpad.net/bugs/1006690
Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For less duplication of code between codecs, and to make it easier
in the future to improve code for all codecs simultaneously.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although HD-audio allows pair-wise channel configurations, only the
fixed channel positions are used in this version. In future, this can
be changed and allow user to modify the channel positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_HDA_POWER_SAVE is no longer an experimental feature and its
behavior can be well controlled via the default value and module
parameter. Let's just replace it with the standard CONFIG_PM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling the jack sync in the init callback of each codec,
call it generically at initialization and resume. By calling it at
the last of resume sequence, a possible race between the jack sync and
the unsol event enablement in the current code will be closed, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As with the ThinkPad Models X230 Tablet and T530 the X230 needs a qurik to
correctly set up the pins for the dock port.
Signed-off-by: Felix Kaechele <felix@fetzig.org>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a model/fixup string "lenovo-dock", for Thinkpad T430s, to allow
sound in docking station.
Tested on Lenovo T430s with ThinkPad Mini Dock Plus Series 3
Cc: stable@kernel.org
Signed-off-by: Philipp A. Mohrenweiser <phiamo@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo Thinkpad T530 with ALC269VC codec has a dock port but BIOS
doesn't set up the pins properly. Enable the pins as well as on
Thinkpad X230 Tablet.
Reported-and-tested-by: Mario <anyc@hadiko.de>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On recent kernels, Realtek codec parser tries to optimize the routing
aggressively and take the headphone output as primary at first. This
caused a regression on VAIO Z with ALC889, the silent output from the
speaker.
The problem seems that the speaker pin must be connected to the first
DAC (0x02) on this machine by some reason although the codec itself
advertises the flexible routing with any DACs.
This patch adds a fix-up for choosing the speaker pin as the primary
so that the right DAC is assigned on this device.
Reported-and-tested-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This merges the changes for converting to new PM ops for platform
and some other drivers.
Also move some header files to local places from the public
include/sound.
With the model parsers out of the way, we have no custom unsol
events to worry about, we can therefore simplify the code.
In addition, this fixes a bug on the ASUS TC710, which has only
a headphone jack and a mic jack, but no internal mic or speakers.
Therefore the unsol_event pointer was not set, and as a result,
the jack kcontrols were not correctly updated.
BugLink: https://bugs.launchpad.net/bugs/1021192
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.
Since we change vx222, vx_core and vxpocket have to be converted,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Windows use hidden register to control EAPD.
Linux use verb to control EAPD.
If windows reboot to Linux, it must change the EAPD control to verb
control.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This 3-pin jack was labeled "Headphone Jack", but it could also be
used as a mic jack just by switching "Input Source". Therefore we need
to call the jack something else, to make sure PulseAudio can use the
speaker together with the external mic. (PulseAudio might mute the
speaker if it detects a headphone being plugged in.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For convenience, add "inv-dmic" model string for enabling the inverted
internal mic workaround to possible Realtek codecs, so far,
ALC882-variants, ALC262, ALC268, ALC269-variants, and ALC662-variants.
Also, the model strings for hardware inv-dmic workarounds,
"alc269-dmic" and "alc271-dmic", are added for ALC269(VA) and ALC271
codecs as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some laptops are equipped with ForteMedia digital mics that give the
differential input. With such devices, summing stereo streams into a
mono (like PulseAudio does) results in almost silence.
This patch provides a workaround for this bug by adding a new mixer
switch to turn on/off the right channel of digital mic, just like a
similar fix for Conexant codecs.
When the new switch "Inverted Internal Mic Capture Switch" is off and
the current input source is the digital mic, the right channel of the
recording stream is muted. When another input source is selected, the
right channel is restored.
Tested-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec ALC272X is a special codec for some Dell machines, and its
detection got broken in the recent kernel because SSID check (required
by ALC272X check) was moved to the later point. Now we need to move
this codec check to the right place, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull sound update from Takashi Iwai:
"This is the second updates for 3.5-rc1. It's mainly for OMAP4 HDMI
updates and the device tree updates for OMAP, in addition to a couple
of PCM accuray improvement and Realtek ALC269VD codec support."
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits)
ALSA: hda/realtek - Add new codec support for ALC269VD
ALSA: core: group read of pointer, tstamp and jiffies
ASoC: OMAP: HDMI: Rename sound card source file
ASoC: OMAP: HDMI: Make sound card naming more generic
ASoC: OMAP: HDMI: Make build config options more generic
ASoC: OMAP: HDMI: Expand capabilities of the HDMI DAI
ASoC: OMAP: HDMI: Improve how the display state is verified
ASoC: OMAP: HDMI: Expand configuration of hw_params
ASoC: OMAP: HDMI: Use the DSS audio interface
ASoC: OMAP: HDMI: Create a structure for private data of the CPU DAI
ASoC: OMAP: HDMI: Change error values in HDMI CPU DAI
ASoC: OMAP: HDMI: Update the platform device names
ASoC: omap-abe-twl6040: Introduce driver data for runtime parameters
ASoC: omap-abe-twl6040: Move Digital Mic widget into dapm table
ASoC: omap-abe-twl6040: Keep only one snd_soc_dai_link structure
ASoC: omap-dmic: Add device tree bindings
ASoC: omap-mcpdm: Add device tree bindings
ASoC: omap-mcbsp: buffer size constraint only applies to playback stream
ASoC: omap-mcbsp: Use the common interrupt line if supported by the SoC
ASoC: omap-mcbsp: Remove unused FRAME dma_op_mode
...
Pull sound updates from Takashi Iwai:
"This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten for
the better support of "implicit feedback". If anything about USB got
broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up
immediately at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital
links between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal
routing through components with tight sequencing and formatting
constraints within their internal paths or where there are multiple
components connected with CPU managed DMA controllers inside the
SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like
digital basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
Texas Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be sent
slightly later as partly depending on the changes of DRM."
Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree). Also some
fairly trivial context conflicts in the imx and mcx soc drivers.
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: snd-usb: fix stream info output in /proc
ALSA: pcm - Add proper state checks to snd_pcm_drain()
ALSA: sh: Fix up namespace collision in sh_dac_audio.
ALSA: hda/realtek - Fix unused variable compile warning
ASoC: sh: fsi: enable chip specific data transfer mode
ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
ASoC: sh: fsi: use same format for IN/OUT
ASoC: sh: fsi: add fsi_version() and removed meaningless version check
ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
ASoC: tegra: Add machine driver for WM8753 codec
ALSA: hda - Fix possible races of accesses to connection list array
ASoC: OMAP: HDMI: Introduce codec
ARM: mx31_3ds: Add sound support
ASoC: imx-mc13783 cleanup
mx31moboard: Add sound support
ASoC: mc13783 codec cleanups
ASoC: add imx-mc13783 sound support
ASoC: Add mc13783 codec
mfd: mc13xxx: add codec platform data
ASoC: don't flip master of DT-instantiated DAI links
...
