Implement new fixup entries for Quanta FL1 and Fujitsu Lifebook
specific COEF and pin configurations. Removed the model entries
from alc269_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the duplicated creation of capture-mixer elements for some static
ALC268 configurations. The capture mixers must be put to cap_mixer field
instead of mixers array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Refactoring the code using snd_pcm_hw_constraint_pow2() helper function.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AES32 supports the non-standard 128kHZ, and this is enabled only when
SNDRV_PCM_RATE_KNOT is set in hw.rates field.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Xonar Essence ST/STX, the connector J14 has been confirmed to be
a digital input, so enable it in the driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC260 HP models work with the BIOS auto-parser. Let's cut them off.
Also move alc260_hp_master_*() to alc262_quirks.c as these are still
referred from there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/826081
The original reporter needs 'Headphone Jack Sense' enabled to have
audible audio, so add his PCI SSID to the whitelist.
Reported-and-tested-by: Muhammad Khurram Khan
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Newer RME cards like RayDAT and AIO support 32 samples per period. This
value is encoded as {1,1,1} in the HDSP_LatencyMask bits in the control
register.
Since {1,1,1} is also the representation for 8192 samples/period on
older RME cards, we have to special case 32 samples and 32768 bytes
according to the actual card.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, hdspm_decode_latency is called several times, violating the
DRY principle. Given that we need to distinguish between old and new
cards when decoding the latency bits in the control register, introduce
hdspm_get_latency() to provide the required functionality.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On newer RME cards like RayDAT and AIO, the 8192 samples per period size
are no longer supported. Instead, setting all three bits of
HDSP_LatencyMask to one ({1,1,1}) now corresponds to 32 samples per
period.
To make this more obvious to future developers, let's reorder the array
according to their bit representation, starting at 64 ({0,0,0}) up to
4096 ({1,1,0}) and finally 32 ({1,1,1}).
Note that this patch doesn't change semantics.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On newer RME cards like RayDAT and AIO, the lower bound is 32 samples
per period in contrast to 64 samples as seen on older cards.
We hence lower period_bytes_min to 32 * 4. Four bytes per sample.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Older RME cards like MADI and AES support period sizes of 8192 samples.
The original hdspm driver already featured this value, apparently, it
was lost during the rewrite.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_azf3328_dbgcallenter is called at the very beginning of the function,
so it could be useful to call snd_azf3328_dbgcallleave at all exit points.
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In commit 45eebda7, it add new function stac_vrefout_set, but it
is only used in code between CONFIG_SND_HDA_POWER_SAVE macro, so
add the macro to avoid such warning:
sound/pci/hda/patch_sigmatel.c:676:12: warning: 'stac_vrefout_set' defined but not used
Signed-off-by: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_TEA575X is enabled by RADIO_SF16FMR2, but the latter one is
no PCI device. Since tea575x-tuner itself is independent from the board
bus type, the config should be moved out of SND_PCI dependency.
Reported-by: Randy Dunlap <rdunlap@xenotime.net>
Acked-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use kzalloc rather than kmalloc followed by memset with 0
This considers some simple cases that are common and easy to validate
Note in particular that there are no ...s in the rule, so all of the
matched code has to be contiguous
The semantic patch that makes this output is available
in scripts/coccinelle/api/alloc/kzalloc-simple.cocci.
More information about semantic patching is available at
http://coccinelle.lip6.fr/
Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_via.c:2087: warning: 'dac' may be used uninitialized in this function
Signed-off-by: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put the exception checks for io_type switch() for possible mistakes in
future. Also this shuts up annoying compile warnings.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new parameter to disable rounding of buffer/period sizes to
multiples of 128 bytes. This is more efficient in terms of memory
access but isn't required by the HDA spec and prevents users from
specifying exact period/buffer sizes. For example for 44.1kHz, a
period size set to 20ms will be rounded to 19.59ms.
Tested and enabled on Intel HDA controllers. Option is disabled by
default for other controllers.
Tested-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It works fine with auto-parser and now the digital mic workaround was
implemented in auto-parser fixup, let's drop the static model quirks for
these models.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The digital-mic unit on ASUS Eee PC gives PDM signals instead of the
normal stereo PCM, thus you can't record a mono stream from the stereo
stream as is; the summed stereo signal results in almost zero level, and
you'll hear only soft noise.
As a workaround, use ALC269-specific COEF to manipulate the dmic route
for mono, like used for ALC271x. This is implemented as a fix-up, thus
it works only with model=auto or without REALTEK_QUIRKS Kconfig.
Reported-and-tested-by: Pavel Roskin <proski@gnu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoids assigning possibly invalid address to pa, even if it
is never dereferenced.
Correct error response to reflect request object/function ids.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We leak the memory allocated to 'firmware' when we fail to
release_firmware() after a kmalloc() failure in hpi_dsp_code_open().
This patch should take care of the leak.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (430 commits)
[media] ir-mce_kbd-decoder: include module.h for its facilities
[media] ov5642: include module.h for its facilities
[media] em28xx: Fix DVB-C maxsize for em2884
[media] tda18271c2dd: Fix saw filter configuration for DVB-C @6MHz
[media] v4l: mt9v032: Fix Bayer pattern
[media] V4L: mt9m111: rewrite set_pixfmt
[media] V4L: mt9m111: fix missing return value check mt9m111_reg_clear
[media] V4L: initial driver for ov5642 CMOS sensor
[media] V4L: sh_mobile_ceu_camera: fix Oops when USERPTR mapping fails
[media] V4L: soc-camera: remove soc-camera bus and devices on it
[media] V4L: soc-camera: un-export the soc-camera bus
[media] V4L: sh_mobile_csi2: switch away from using the soc-camera bus notifier
[media] V4L: add media bus configuration subdev operations
[media] V4L: soc-camera: group struct field initialisations together
[media] V4L: soc-camera: remove now unused soc-camera specific PM hooks
[media] V4L: pxa-camera: switch to using standard PM hooks
[media] NetUP Dual DVB-T/C CI RF: force card hardware revision by module param
[media] Don't OOPS if videobuf_dvb_get_frontend return NULL
[media] NetUP Dual DVB-T/C CI RF: load firmware according card revision
[media] omap3isp: Support configurable HS/VS polarities
...
Fix up conflicts:
- arch/arm/mach-omap2/board-rx51-peripherals.c:
cleanup regulator supply definitions in mach-omap2
vs
OMAP3: RX-51: define vdds_csib regulator supply
- drivers/staging/tm6000/tm6000-alsa.c (trivial)
Apparently, there are multiple old firmware revisions in the wild for
the PCI RME MADI cards. Just add them to the list of supported devices
and treat them like their modern counterparts.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In slave mode, the card can only detect the base frequency (32..48kHz)
on the MADI link (exception: 96k frames), so the real external sample
rate is this base frequency multiplied by 1, 2 or 4 depending on the
speed mode.
This patch enables 64..192kHz sample rates in clock slave mode, which
failed before due to an alleged sample rate mismatch between the MADI
link (e.g., 48kHz) and the application in DS/QS mode (e.g., 96kHz,
192kHz).
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When running in slave mode (no clock master), there is no way to
determine the real wirespeed on the MADI link (single/double/quad
speed). Like physical gear, simply provide the user with a tristate
switch to select the appropriate format.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
sound: oss: rename local change_bits to avoid powerpc bitsops.h definition
ALSA: hda - Fix duplicated DAC assignments for Realtek
ALSA: asihpi - off by one in asihpi_hpi_ioctl()
ALSA: hda - Fix Oops with Realtek quirks with NULL adc_nids
ALSA: asihpi - bug fix pa use before init.
ALSA: hda - Add support for vref-out based mute LED control on IDT codecs
Convert radio-sf16fmr2 to use generic TEA575x implementation. Most of the
driver code goes away as SF16-FMR2 is basically just a TEA5757 tuner
connected to ISA bus.
The card can optionally be equipped with PT2254A volume control (equivalent
of TC9154AP) - the volume setting is completely reworked (with balance control
added) and tested.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Copying hp_pins and speaker_pins from line_out_pins may confuse the
parser, and it can lead to duplicated initializations for the same pin
with a wrong DAC assignment. The problem appears in 3.0 kernel code.
Cc: <stable@kernel.org> (for 3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"adapter" is used as an array index in the adapters[] array so
the off by one would make us read past the end.
1c073b6797 "ALSA: asihpi - Remove spurious adapter index check"
reverted Dan Rosenberg's check that would have prevented the
overflow here.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Somce quirk models don't set adc_nids but let the parser filling it.
But the recent code has unnecessary NULL-checks of spec->input_mux,
and it resulted in NULL dereferences.
This patch fixes that regression.
Reported-and-tested-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes bug introduced by 1c073b67.
Also declare pa local to block in which it is used.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch also registers all necessary callbacks to support mute LED
only when such control is enabled. And it keeps codec AFG in D0 or D1
state all the time when aggressive power managemnt is enabled for vref-out
control (and mute LED) work correctly.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows us to move duplicated code in <asm/atomic.h>
(atomic_inc_not_zero() for now) to <linux/atomic.h>
Signed-off-by: Arun Sharma <asharma@fb.com>
Reviewed-by: Eric Dumazet <eric.dumazet@gmail.com>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: David Miller <davem@davemloft.net>
Cc: Eric Dumazet <eric.dumazet@gmail.com>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This update includes the changes necessary for supporting the
CS421x family of codecs. Previously this file only supported
the CS420x family of codecs.
This file also contains init verbs to correct several issues in
the CS421x hardware.
Behavior between the CS421x and CS420x codec families is similar,
so several functions have been reused with "if" statements to
determine which codec family (CS421x or CS420x) is present.
Also, this file will be updated sometime in the near future in
order to add support for a system using CS421x that requires
mono mix on the speaker output only.
[Fix const usages and adaption for new APIs by tiwai]
Signed-off-by: Tim Howe <tim.howe@cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The procedure for codec D-state change may have exceptional cases
depending on the codec chip, such as a longer delay or suppressing D3.
This patch adds a new codec ops, set_power_state() to override the system
default function. For ease of porting, snd_hda_codec_set_power_to_all()
helper function is extracted from the default set_power_state() function.
As an example, the Conexant codec-specific delay is removed from the
default routine but moved to patch_conexant.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new ops, post_suspend(), which is called after suspend() ops is
performed. This is called only in the case of the real PM suspend, and
the codec driver can use this for further changing of D-state or
clearing the LED, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It makes little sense to enable power-saving without PM.
This removes SND_HDA_NEEDS_RESUME define so that we can use CONFIG_PM
in all places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds checking of mute state on all outputs besides just
speakers to calculate the master mute state for mute led support.
It also renames and splits the function that does it for better code
clarity.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Codec state is not restored immediately on resume but on the first
access when power-save is enabled. That leads to an invalid mute led
state after resume until either sound is played or some control is
changed. This patch adds a possibility for a vendor specific patch to
restore codec state immediately after resume if required. And it adds
code to restore IDT codecs state immediately on resume on HP systems
with mute led support.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (43 commits)
fs: Merge split strings
treewide: fix potentially dangerous trailing ';' in #defined values/expressions
uwb: Fix misspelling of neighbourhood in comment
net, netfilter: Remove redundant goto in ebt_ulog_packet
trivial: don't touch files that are removed in the staging tree
lib/vsprintf: replace link to Draft by final RFC number
doc: Kconfig: `to be' -> `be'
doc: Kconfig: Typo: square -> squared
doc: Konfig: Documentation/power/{pm => apm-acpi}.txt
drivers/net: static should be at beginning of declaration
drivers/media: static should be at beginning of declaration
drivers/i2c: static should be at beginning of declaration
XTENSA: static should be at beginning of declaration
SH: static should be at beginning of declaration
MIPS: static should be at beginning of declaration
ARM: static should be at beginning of declaration
rcu: treewide: Do not use rcu_read_lock_held when calling rcu_dereference_check
Update my e-mail address
PCIe ASPM: forcedly -> forcibly
gma500: push through device driver tree
...
Fix up trivial conflicts:
- arch/arm/mach-ep93xx/dma-m2p.c (deleted)
- drivers/gpio/gpio-ep93xx.c (renamed and context nearby)
- drivers/net/r8169.c (just context changes)
Formatting a PCM name is useful for module debug too.
Add snd_prefix when making function public.
[minor coding-style fixes by tiwai]
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a regression in the DAC filling code in patch_realtek.c. The already
filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0,
thus always pointed to the first DAC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HPI Version is used to check for firmware compatibility.
This version will accept 4.08.xx released firmware,
and will also accept 4.09.xx beta firmware
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add names corresponding to new HPI node types.
Shorten some names so that constructed names don't overflow the
maximum name length.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because mutex is used in adapter struct defined here.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Subsystem requests don't have or need a valid adapter index.
The adapter index is already checked further on, before it is used to index
the adapters array. (Reverts 4a122c10f)
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Work towards moving the function into alsa common header.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loader API has been revised so that OS specific data is kept
local to hpidspcd.c, and the public API is unchanged across OSes.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some cobranet control data would not fit in an original HPI message.
Now that HPI is able to transfer larger messages, this special handling
is no longer required.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow for up to 256 bytes of extra data on top of standard hpi
request and response sizes.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Having a 'request message' makes more sense than a 'message message'
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes non-working indep-HP control on VT1708* codecs.
The problems are that via_independent_hp_put() wasn't fixed to follow
the recent change of three HP paths, and hp_indep_path didn't contain
the amp nids of mixer elements.
Together with the fixes, a few code clean-ups are done.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the behavior of independent-HP enum switch. Now
instead of returning a busy error, the driver switches dynamically the
stream of the HP (and shared) DACs according to the current mode.
The logic is similar like the dual-mic ADC switch, but a bit more
complicated because of the presence of shared DAC.
Together with the change, a mutex is introduced to protect against the
possible races for the indep-HP mode setting.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the dynamic control of analog-loopback for VIA codecs.
When the loopback is enabled, the inputs from line-ins and mics are
mixed with the front DAC, and sent to the front outputs. The very same
input is routed to the headhpones and speakers in loopback mode.
However, since the loopback mix can't take other than the front DAC,
there is no longer individual volume controls for headphones and
speakers. Once when the loopback control is off, these volumes take
effect.
Since the individual volumes are more desired in general use caess, the
loopback mode is set to off as default for now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit dd203fa97b (ALSA: virtuoso: remove non-working controls on
Essence ST Deluxe) made it impossible to adjust the volume after the
driver initialized it to muted.
Ensure that those DACs that can be accessed with I2C are initialized
to the same volume that is the reset default of the DAC without I2C.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.38+ <stable@kernel.org>
BugLink: https://bugs.launchpad.net/bugs/774895
The original reporter states that his volume keys do not change the
desired Master and PCM mixer elements together, so apply the hp+mute led
quirk for his PCI SSID.
Reported-by: Jeffrey Finkelstein
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the rewrite, the check of spec->need_dac_fix and the corresponding
num_dacs change was dropped from the channel-mode control.
This patch re-adds it, and also enables need_dac_fix for ALC880 as default,
as this feature was originally introduced to fix h/w bugs of this chip.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a fix up for commit acfa634f.
commit acfa634f7e
Author: Takashi Iwai <tiwai@suse.de>
Date: Tue Jul 12 17:27:46 2011 +0200
ALSA: hda - Add Kconfig for the default buffer size
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a Kconfig entry to specify the default buffer size.
Distros using PulseAudio can choose a larger value here.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1718S and co have a secret connection from DAC to AA-mix, which
doesn't appear in the connection list obtained from the h/w.
Currently the driver fixes the connection index locally at init, but
now we can expose it statically via snd_hda_override_connections()
so that this conection can be checked better by the parser in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the codec proc outputs, read the raw connections instead of the
cached connection list, i.e. proc files contain only raw values.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a function to add/modify the connection-list cache entry.
It'll be useful to fix a buggy hardware result.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some machines seem to use EAPD control of the unused pin for controlling
the overall EAPD. Since the driver currently doesn't check the EAPD of
unused pins, the EAPD isn't enabled. For avoiding such a problem, turn
all extra EAPDs on as default.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For smart51 pins, we need to preserve the input pin-control bits at
auto-mute controls instead of overwriting zero or pin-out-only.
Otherwise the VREF won't be set properly when smart51 is disabled
again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When Independent-HP mode is changed for VIA, the driver needs to
re-issue the auto-mute check so that the line-out pins are set properly
without influence of HP pin state.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the line-jack is plugged/unplugged, the driver must check also
the headphone jack state in addition to the line-out jack. Currently
it checks only the line-out state and ignores the headphone.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of checking the model quirk, use a fixup table for workaround
of 44kHz-fixed PCM for Lenovo IdeaPad with ALC269.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's harmless but annyoing.
sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now all alc*_parse_auto_config() do almost same thing except for the
NID list to ignore and the PINs for SSID-check, we can merge all these
to a single function. A good amount of code reduction.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One more code reduction. This codec has less DACs, thus the wiring
to DAC can't be filled uniquely for all output pins, i.e. some outputs
share the same volume control.
Except for that, all seems working fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge more auto-parser code in patch_realtek.c, now for ALC861.
The topology of this codec is pretty simple, and can be parsed well
by the current starndard parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
query_amp_caps() may return non-zero if the amp cap isn't supported
by the codec. Thus one needs to check widget-caps first, then check
the corresponding amp-caps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A regression fix from commit 21268961d3
ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs
The auto-mic wasn't detected properly when no ADC-switch is needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, to create Independent HP
control.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, there're only two DACs. So smart51
control shouldn't be created.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, the original activate_output_path()
function can't initialize output and hp path correctly, since mixers connected to
output pin widgets are not considered. So modify the activate_output_path()
function to satisify this kind of codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put the all static quirk codes out of patch_realtek.c, split into the
file for each codec model. For controlling the build of quirk codes,
a new Kconfig, CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS is introduced.
By setting this off, all quirk codes won't be built, thus you can save
lots of memory.
The codes in patch_realtek.c are also shuffled and more comments are
given, but the contents aren't changed. This is just a refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the auto-parser and the auto-mic handling codes to
allow more flexible dynamic ADC-switching with Realtek codecs.
In the new code, the following strategy is taken:
- When a cap-src can't handle all input-sources, either skip it, or
switch to the ADC-switching mode. In ADC-switching mode, like the
former dual-ADC mode for ALC275, it changes ADC on the fly according
to the current input source.
- When auto-mic is possible, always assign imux. If the mic pins are
set statically via a quirk, rebuild imux according to the pins.
In the auto-mic mode, the driver always changes the imux (although
the imux isn't exposed as a mixer element).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of assigning each default hda_pcm_stream pointers, do NULL-checks
and assign default values in alc_build_pcms().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only different implmentation was alc880_auto_init_input_src(),
and now it covers this variant, and we can use the single function
for all codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now with the new code for looking for ADCs and MUXs, we can replace
the whole ADC assignment with the parsed results.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All alc*_auto_init_analog_input() calls are identical, so let's use
the same function more clearly without aliases.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Parse ADCs and cap-srcs in alc_auto_create_input_ctls() by itself
instead of passing explicitly from the caller. By this change, all
alc*_auto_create_input_ctls() can be unified to the same calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the same common code for auto-parsing the output paths and their
initializations, based on the existing ALC662 code, which is smarter
than the old ALC880/2 code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple inputs are present on the mixer widget (typically a DAC
and a loopback), mute/unmute both inputs with the corresponding mixer
element.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In alc662_auto_fill_dac_nids(), the HP and speaker DACs aren't parsed
when the corresponding pins aren't fixed with single DACs.
Now check these DACs even for non-fixed pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the dual-adc switching mode is active in Realtek auto-parser,
we need to couple all ADCs as a single capture-volume. Currently, the
volume control changes only the first ADC, thus others may remain silent.
This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of ALC269VC codec.
Also delete the unnecessary codec_variant type enum list:
now only three variants (ALC269VA ALC269VB ALC269VC) are needed.
In addition, added some aliases:
- Add ALC269VB alias name ALC277
- Add ALC269VC alias name ALC259 ALC281X
- Add ALC269VC for Lenovo device 0x21f3 name ALC3202
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we now route the front DAC via aa-mix widget, adding the aa-mix
to surrounds will result in a mix-up of both front and surround PCM
signals. For avoiding this, the aa-mix routes have to be disabled
for surround paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the individual DAC is available for the headphone output, the driver
should create the DAC for its volume control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, the multi-channel path should be like following:
DAC 0-->Mixer 9(index 5)-->Mixer 0(index 1)-->Front Pin;
DAC 1-->Mixer 1(index 0)-->Surround Pin;
DAC 2-->C/LFE Pin;
DAC 3-->Mixer 2(index 0)-->Side Pin;
But current code built Surround and Side path through index 1 of
Mixer 1 and 2. So Adjusting Surround and Side channel amplifier is
invalid. This patch fixes the issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, Mixer 9 doesn't expose the connection to DAC 0. So when
building up a 'PCM Playback' amplifier control, it will fail since
getting DAC 0 index of Mixer 9 returned -1. So I added a dac_mixer_idx
to indicated the actual index of DAC 0 to Mixer 9. Following is the
patch and next mail is another.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unmute DAC on front speaker path when Independent HP is enabled.
When to enable Independent HP, the front speaker won't output any sound
for VT1708, VT1708B, VT1708S and VT1702.
I find the via_independent_hp_put() routine will mute DAC 0 path in Mixer 0.
For these codecs, when using Independent HP, there could have two
independent streams, one is from DAC0-->Mixer0-->Front Pin, the other is
from DAC3-->GainSW3-->Side Pin.
So I added a check for DAC-->Mixer path in activate_output_path().
If current path is DAC-->Mixer, no need to mute DAC index in Mixer.
In fact, to change connection of Headphone pin or Mux connected with HP
is enough.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the re-implementation of Independent-HP mode in the
case where the DAC is shared between HP and side-channel streams.
Now the driver tries to parse the output-path using the pre-parsed
side-channel DAC for the independent HP output, too.
When a playback PCM stream is opened with this shared mode, the
Independent-HP mixer switch can't be changed for avoiding the conflict,
thus it returns -EBUSY error.
One remaining unintuitive issue is that the DAC volume is still
controlled as "Side" volume although it's shared by both independent-HP
and side streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The char can be unsigned on some architectures. Since the code checks
the negative values, they should be declared as signed char explicitly.
sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type
sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrongly converted short values:
sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type
sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I noticed that the last character of the ELD monitor name is lost,
this fixes the issue.
This fix should be confirming to the HDA spec, and works together with
the DRM part of the ELD patch.
The HDA spec does not mention that Monitor_Name_String is an '\0'
ending string, and it allows NML to be 1, which is only valid when MNL
does not count the possible ending '\0'.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move codec init verb which is only applicatable for VT1708.
I've found the root cause that jack plugged in can't be detected.
The verb in vt1708_init_verbs is used to power down jack detect circuit.
This verb is only applicable to VT1708. vt1708 didn't implement jack
detect function in hardware, so we should shut down this function to
avoid noise. But for other codecs, hardware implement jack detect
function. If sending this verb during initialization, jack detect will
be invalid. So I move this verb from via_parse_auto_config() to
patch_vt1708().
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create snd_hda_get_conn_index() helper function for obtaining the
connection index of the widget. Replaced the similar codes used in
several codec-drivers with this common helper.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to enable unsolicited response to line-out pins.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we're not using the new auto parser as a fallback yet,
add it manually as a quirk.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant ID 506c was found on Acer Aspire 3830TG. As users report
no playback, sending to stable should be safe.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/783582
Reported-by: andROOM
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use static inline for dummy function to fix the warnings like below
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_init’:
sound/pci/hda/patch_sigmatel.c:4387:3: warning: statement with no effect
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_resume’:
sound/pci/hda/patch_sigmatel.c:4927:3: warning: statement with no effect
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More similar fixes like previous commits: handle the exceptional case
like ALC267 where no volume amp is found in ADC widget but in the
capsrc widget instead.
Also minor checks for avoiding possible erros: no connection-select
when the pin has a single selection, and add beep verbs only when the
0x1d is used for beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC259 seems to provide an invalid widget capability for the input-src
selector widget. The widget shows the input-amp while it's a selector,
and this confuses the current ALC882 initialization code that is used
for ALC259, too. For fixing this, check the amp capability and handle
the connection selection individually.
Also, ALC259 has no mute bit in DAC volume, so we need to initialize
it as ZERO instead of MUTE.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 and compatible codecs have the output volume in DACs, thus we
can't use the ALC880's code as is. Fixed by checking the amp caps and
picking up the right widget for initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Look through the known NIDs that may have EAPD capabilities and turn
on/off them appropriately instead of checking the individual vendor ids.
This will also avoid the forgotten entries of newly added codec ids
in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the regression introduced by the commit
1f0f4b8036
ALSA: hda - Reduce static init verbs for Realtek auto-parsers
The input amps of mixer widgets should be unmuted as default (as
usually they have no assigned mixer switches).
More fixes in this commit are, however, for ALC260: ALC260 codec can
have multiple output mixers connnected to a single DAC althouh the
driver didn't pick up them properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Realtek codecs have the output pins hardwired with certain DACs.
These DACs have to be assigned at first and assign the rest for
multi-DAC pins so that all DACs can be assigned properly.
Without such an optimization, speaker outputs may be assigned to the
same DAC as the headphone or others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In alc_auto_add_multi_channel_mode(), when the primary HP workaround
is enabled, it re-initializes the DAC list but calls alc662's function
in a fixed way. This isn't pretty suitable for other codecs, of course.
Now we call it with fill_dac function pointer so that the proper
function can be called at that point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using fixed init verbs, initialize DACs, ADCs and mixers
more dynamically for Realtek auto-parsers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper function snd_hda_get_conn_list().
Unlike snd_hda_get_connections(), this function doesn't copy the
connection-list but gives the raw pointer for the cached list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
each device has a unique mac address, which can be used to distinguish
multiple devices in the same machine. we therefore include the full mac
address in the device shortname and the last 6 bytes in the device id.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the sound/ directory there are two files (flagged by 'make
versioncheck'); sound/pci/asihpi/asihpi.c and
sound/soc/codecs/wm8991.c that include linux/version.h although they
don't need it. This patch removes the unneeded includes.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change alc_get_line_out_pfx() in patch_realtek.c to provide the channel
specific name and assign the index so that each caller doesn't have to
set the channel name by itself.
Also, check the multi-io case with the primary hp-out; for the multi-io
channels, assign the channel name instead of "Headphone" with indices.
This makes the mixer names more intuitive and reduces confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some BIOS set up the pin config wrongly as line-out although it's
supposed to be a speaker out. In most cases, though, we can judge
the validity by checking the connection type -- when it's FIXED,
mostly it's an invalid line-out but a speaker.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even if the machine has no line-out but only HP-out, try to detect the
multi-io. It'll allow more possibilities for 5.1 outputs on laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS lists the internal speaker as an internal line-out. Change to
internal speaker + model=auto for better auto-mute capabilities.
BugLink: http://bugs.launchpad.net/bugs/754964
Reported-by: Marc Legris <marc.legris@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some VIA codecs like VT1702 provide the input-route only to specific
ADCs such as digital-mic inputs. These routes aren't covered by the
normal primary ADC, and for now, user had to open the capture stream
assigned to that special ADC manually for using such inputs.
This patch implements a way to switch the current ADC dynamically per
the input-source selection in such a case. When this workaround is
activated, the driver provides only one capture stream and one input-
source control but with the full possible inputs. The driver switches
the ADC to be used (or being used) according to the input-source on the
fly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When smart51 mode is enabled, auto-mute these surround outputs
as well as the primary line-out. Also this patch includes minor
clean-ups.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unify the VT1709 10ch and 6ch parsers, as well as VT1708B 8ch and 4ch
parsers. They have no difference now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codecs like VT1708 needs more complicated routing using the mixer
widget rather than the simple selector widgets.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The surround/CLFE/side DACs on VT1708B and co have no amp but the
connected selector widgets have the amp instead. Fix the parser to
check these selector widgets for the possible mixer controls as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the check of the multiple loopback-mixer, which gave sometimes
a wrong index assigned to an element even for different names, e.g.
Mic and Front Mic. Now check the label properly for avoid duplication.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input jacks assigned as the smart51 outputs must be in the same
stack, either rear, front or other. Also, prefer line-in as the surround
to mic-in.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to create playback volume control if pin has amplifier capability
but not DAC.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the order of the output-path list in a way from the DAC to the
target pin. Also now the list include the target pin, too.
Together with this format change, simplify the arguments of
parse_output_path() function, and fix the initialization in
via_auto_init_output().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop "Capture" prefix from the mic-boost names.
Otherwise some control names can overflow the max name length.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create patch_ca0132.c, to add support for devices featuring the
Creative CA0132 HD-audio codec.
This driver implements :-
* 1 playback subdevice to headphone and speaker
* 2 capture subdevices:
i - Mic-in
ii- Line-in
* mixer device
Advanced DSP features are not yet included.
Developed and maintained by Creative Labs, Inc.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a master volume and mute control of playback for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When switch HP independent mode, mute/unmute connctions of mixer which is
connected to headphone for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some invalid initial verbs and correct some wrong initial verbs
for VT1718S codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER.
This keeps headphone automute and microphone input from operating
on at least one laptop from Opti Systems.
Without the override, the BIOS parser does a fine job setting the
card up and everything works.
Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com>
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reporter, who is running kernel 2.6.38, reports that
he needs to set model=auto for the headphone output to work
correctly.
BugLink: http://bugs.launchpad.net/bugs/761022
Cc: stable@kernel.org (v2.6.38+)
Reported-by: Jo
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the existing aa-loop list for simplifying the check for analog
low-current mode. Also fix the stream count test for playback streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Issue the init verbs of unsolicited events dynamically from the parsed
results for VIA codecs. Also, consolidate the unsol handlers for HP
and line-out mutes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
