In case there are speakers or headphones as well, anything that only
covers the line out should not be labelled "PCM". Let's name it
"Line Out" instead for clarity.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the scenario where there is one "Line Out", one "Speaker" and one
"Headphone", and there are only two DACs, two outputs will share a DAC.
Currently any mixer on such a DAC will get the "PCM" name, which is
misleading. Instead use "Headphone+LO" or "Speaker+LO" to better
specify what the volume actually controls.
[fixed missing slave string additions by tiwai]
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The next patch will use it, so make it visible across modules.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, hda_jack infrastructure allows only one callback per jack, and
this makes things slightly complicated when a driver wants to assign
multiple tasks to a jack, e.g. the standard auto-mute with a power
up/down sequence. This can be simplified if the hda_jack accepts
multiple callbacks.
This patch is such an extension: the callback-specific part (the
function and private_data) is split to another struct from
hda_jack_tbl, and multiple such objects can be assigned to a single
hda_jack_tbl entry.
The new struct hda_jack_callback is passed to each callback function
now, thus the patch became bigger than expected. But these changes
are mostly trivial.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The action value assigned to each hda_jack_tbl entry is mostly
superfluous. The actually used values are either the widget NID or a
value specific to the callback.
The former case can be simply replaced by a reference to widget NID
itself. The only place doing the latter is STAC/IDT codec driver for
the powermap handling. But, the code doesn't need to check the action
field at all -- the function jack_update_power() is called either with
a specific pin or with NULL. So the check of jack->action can be
removed completely there, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DACs on Sigmatel/IDT codecs do mute at the lowest volume level,
and in the earlier drivers, we passed TLV_DB_SCALE_MUTE bit for each
volume control element like Speaker and Headphone as well as Master.
Along with the translation to the generic parser, however, the TLV bit
was lost for the slave controls (e.g. Speaker) but set only to
Master. In theory this should have sufficed, but apps, particularly
PA, do care the slave volume bits, so we seem to see a regression in
the volume controls.
This patch adds a flag to hda_gen_spec to specify the DAC mute
feature, and adds the TLV bit properly for all relevant volume
controls. Also, the TLV bit for vmaster is set in hda_generic.c, so
that we can get rid of all tricks from the codec driver side.
As the similar hack is applied to Conexant 5051 stuff, we can get rid
of it as well.
BugLink: https://bugs.launchpad.net/bugs/1357928
Signed-off-by: Takashi Iwai <tiwai@suse.de>
print_nid_path has a possible buffer overflow if
struct nid_path.path values are > 256.
Avoid this and neaten the output to remove the leading ':'
Neaten debug_badness to always verify arguments.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pass the codec object so that we can replace all the rest of
snd_print*() usages with the proper device-specific print helpers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The beep input device is registered via input_register_device(), but
this is called in snd_hda_attach_beep_device() where the sound devices
aren't registered yet. This leads to the binding to non-existing
object, thus results in failure. And, even if the binding worked
(against the PCI object), it's still racy; the input device appears
before the sound objects.
For fixing this, register the input device properly at dev_register
ops of the codec object it's bound with. Also, call
snd_hda_detach_beep_device() at dev_disconnection so that it's
detached at the right timing. As a bonus, since it's called in the
codec's ops, we can get rid of the further call from the other codec
drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() and co for messages from HD-audio controller and codec
drivers. The codec drivers are mostly bound with codec objects, so
some helper macros, codec_err(), codec_info(), etc, are provided.
They merely wrap the corresponding dev_xxx().
There are a few places still calling snd_printk() and its variants
as they are called without the codec or device context.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last user of snd_hda_gen_spec_free() is patch_via.c, and we can
rewrite it safely with snd_hda_gen_free(), so that
snd_hda_gen_spec_free() can be a local function in hda_generic.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code for controlling mic mute LED in patch_sigmatel.c
blindly assumes that there is a single capture switch. But, there can
be multiple multiple ones, and each of them flips the state, ended up
in an inconsistent state.
For fixing this problem, this patch adds kcontrol to be passed to the
hook function so that the callee can check which switch is being
accessed. In stac_capture_led_hook(), the state is checked as a
bitmask, and turns on the LED when all capture switches are off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... by using snd_Hda_codec_update_cache() instead of *_write_cache().
Since all path elements should have been updated by this function,
we are safe to assume that the cache contents are consistent.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Apply the codec->power_filter to the FG nodes in general for reducing
hackish set_power_state ops override in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD1986A mic pins (0x1d and 0x1f) share the same widget for controlling
the loopback volume/mute, but the generic parser didn't check it.
This ended up with the duplicated controls for the same effect.
This patch adds the check of the duplication for avoiding it.
After this fix, there will be only one control although it affects
both paths; this remaining issue should be fixed later in a different
patch.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD1986A codec is a pretty old codec and has really many hidden
restrictions. One of such is that each DAC is dedicated to certain
pin although there are possible connections. Currently, the generic
parser tries to assign individual DACs as much as possible, and this
lead to two bad situations: connections where the sound actually
doesn't work, and connections conflicting other channels.
We may fix this by trying to find the best connections more harder,
but as of now, it's easier to give some hints for paired DAC/pin
connections and honor them if available, since such a hint is needed
only for specific codecs (right now only AD1986A, and there will be
unlikely any others in future).
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all channels have been initialized, so far, especially when aamix
NID itself doesn't have amps but its leaves have. This patch fixes
these holes. Otherwise you might get unexpected loopback inputs,
e.g. from surround channels.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD and VIA codecs had stereo mixer input enabled as default before
moving to the generic parser, and people think the lack of such a
regression. In this patch, the stereo mixer input is added back to
the input selection if no auto-mic is available, and if it's not
disabled explicitly via hint. This should satisfy most of demands,
i.e. stereo mix on desktop machines like what it worked before, and it
still keeps the new auto-mic feature on laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loopback mixing paths aren't initialized correctly at init
callback. Mostly this is harmless as codecs usually set the mute
state as default, but we still should make sure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have blindly assumed that all valid configurations should have
either analog or digital playback, but there can be capture-only
configurations. The parser shouldn't escape in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current generic parser assumes blindly that the volume and mute
amps are found in the aamix node itself. But on some codecs,
typically Analog Devices ones, the aamix amps are separately
implemented in each leaf node of the aamix node, and the current
driver can't establish the correct amp controls. This is a regression
compared with the previous static quirks.
This patch extends the search for the amps to the leaf nodes for
allowing the aamix controls again on such codecs.
In this implementation, I didn't code to loop through the whole paths,
since usually one depth should suffice, and we can't search too
deeply, as it may result in the conflicting control assignments.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65641
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the hp mic pin has no VREF bits, the driver forgot to set PIN_IN
bit. Spotted during debugging old MacBook Airs.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a headphone jack is configurable as input, the generic parser
tries to make it retaskable as Headphone Mic. The switching can be
done smoothly if Capture Source control exists (i.e. there is another
input source). Or when user explicitly enables the creation of jack
mode controls, "Headhpone Mic Jack Mode" will be created accordingly.
However, if the headphone mic is the only input source, we have to
create "Headphone Mic Jack Mode" control because there is no capture
source selection. Otherwise, the generic parser assumes that the
input is constantly enabled, thus the headphone is permanently set
as input. This situation happens on the old MacBook Airs where no
input is supported properly, for example.
This patch fixes the problem: now "Headphone Mic Jack Mode" is created
when such an input selection isn't possible.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop the hard dependency on the generic parser code and load / unload
the generic parser code dynamically if built as a module. This allows
us to avoid the generic parser if only HDMI/DP codecs are found.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We don't change the EAPD bit in set_pin_eapd() if keep_eapd_on flag is
set by the codec driver and enable is false. But, we also apply the
flipping of enable value according to inv_eapd flag in the same
function, and this confused the former check, handled as if it's
turned ON. The inverted EAPD check must be applied after keep_eapd_on
check, instead.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a bitmask to hda_gen_spec indicating NIDs to exclude from the
possible volume controls. That is, when the bit is set, the NID
corresponding to the bit won't be picked as an output volume control
any longer.
Basically this is just a band-aid for working around the issue found
with CS4208 codec, where only the headphone pin has a volume AMP with
different dB steps.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=60811
Cc: <stable@vger.kernel.org> [v3.12+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The generic parser has a support of vmaster hook, but this is
initialized only in the init callback with the check of the presence
of the corresponding kctl. However, since kctl is NULL at the very
first init callback that is called before build_controls callback, the
vmaster hook sync is skipped there. Eventually this leads to the
uninitialized state depending on the hook implementation.
This patch adds a simple workaround, just calling the sync function
explicitly at build_controls callback.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The create_bind_cap_vol_ctl does not create any control indicating
that an inverted dmic is present. Therefore, create multiple
capture volumes in this scenario, so we always have some indication
that the internal mic is inverted.
This happens on the Lenovo Ideapad U310 as well as the Lenovo Yoga 13
(both are based on the CX20590 codec), but the fix is generic and
could be needed for other codecs/machines too.
Thanks to Szymon Acedański for the pointer and a draft patch.
BugLink: https://bugs.launchpad.net/bugs/1239392
BugLink: https://bugs.launchpad.net/bugs/1227491
Reported-by: Szymon Acedański <accek@mimuw.edu.pl>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
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Merge tag 'asoc-v3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.12
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
The current generic parser code assumes that always a pin widget
controls the mute for an output blindly although it might be a
different widget in the middle. Instead of the fixed assumption,
check each parsed path and just pick up the right widget that has been
already defined as a mute control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-mute using the amp currently works only for a single amp on a
pin. Make it working also with HDA_CTL_BIND_MUTE type, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've added a fake mute control (setting the amp volume to zero) for
CX5051 at commit [3868137e: ALSA: hda - Add a fake mute feature], but
this feature was overlooked in the generic parser implementation. Now
the driver lacks of mute controls on these codecs.
The fix is just to check both AC_AMPCAP_MUTE and AC_AMPCAP_MIN_MUTE
bits in each place checking the amp capabilities.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59001
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VAIO-Z laptops need to use the specific DAC for the speaker output
by some unknown reason although the codec itself supports the flexible
connection. So we implemented a workaround by a new flag,
no_primary_hp, for assigning the speaker pin first.
This worked until 3.8 kernel, but it got broken because the driver
learned for a better multi-io pin mapping, and not it can assign two
mic pins for multi-io. Since the multi-io requires to be the primary
output, the hp and two mic pins are assigned in prior to the speaker
in the end.
Although the machine has two mic pins, one of them is used as a noise-
canceling headphone, thus it's no real retaskable mic jack. Thus, at
best, we can disable the multi-io assignment and make the parser
behavior back to the state before the multi-io.
This patch adds again a new flag, no_multi_io, to indicate that the
device has no multi-io capability, and set it in the fixup for
VAIO-Z. The no_multi_io flag itself can be used generically, added
via a helper line, too.
Reported-by: Tormen <my.nl.abos@gmail.com>
Reported-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_jack_detect() function returns a boolean value for a jack
plugged in or not, but it also returns always true when the
corresponding pin is phantom (i.e. fixed). This is OK in most cases,
but it makes the generic parser misbehaving about the auto-mute or
auto-mic switching, e.g. when one of headphone pins is a fixed.
Namely, the driver decides whether to mute the speaker or not, just
depending on the headphone plug state: if one of the headphone jacks
is seen as active, then the speaker is muted. Thus this will result
always in the muted speaker output.
So, the problem is the function returns a boolean, after all, although
we need to think of "phantom" jack. Now a new function,
snd_hda_jack_detect_state() is introduced to return these tristates.
The generic parser uses this function for checking the headphone or
mic jack states.
Meanwhile, the behavior of snd_hda_jack_detect() is kept as is, for
keeping compatibility in other driver codes.
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The char arrays with size 44 are for the name string of
snd_ctl_elem_id. Define the constant and replace the raw numbers with
it for clarifying better.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
add_control_with_pfx() in hda_generic.c assumes a shorter name string
for the control element, and this resulted in the truncation of the
long but valid string like "Headphone Surround Switch" in the middle.
This patch aligns the max size to the actual limit of snd_ctl_elem_id,
44.
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag, auto_mute_via_amp, to determine the behavior of the
headphone / line-out auto-mute. When this flag is set, the generic
driver mutes the speaker and line outputs via the amp mute of each
pin, instead of changing the pin control values.
This is introduced for devices that don't work expectedly with the pin
control values; for example, some devices are known to keep enabling
the speaker outputs no matter which pin control values are set on the
speaker pins.
The driver doesn't check actually whether the pins have the output amp
caps, but assumes that the proper mixer (mute) controls are created on
all these pins. If not the case, you can't use this flag for your
device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit was written in the way to make the backport to
3.9.y easier, and left the duplicated open codes intentionally.
Now let's clean up the duplicated codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1802 codec seems to reset EAPD of other pins in the hardware level,
and this was another reason of the silent headphone output on some
machines. As a workaround, introduce a new flag indicating to keep
the EPAD on to the generic parser, and set it in patch_via.c.
Reported-by: Alex Riesen <raa.lkml@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codec drivers (VIA codecs and some Realtek fixups) set the
automute and automic hooks after calling
snd_hda_gen_parse_auto_config(). In the current code, the hook
pointers are referred only in snd_hda_gen_parse_auto_config() and
passed to snd_hda_jack_detect_enable_callback(), thus changing the
hook values won't change the actually called callbacks properly.
This patch fixes this bug by setting the static functions as the
primary callback functions for the jack detection, and let them
calling the appropriate hooks dynamically.
Cc: <stable@vger.kernel.org> [v3.9]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an inactive path is powered down with spec->power_down_unused
flag, we should check the activity of each widget in the path whether
it's still referred from any active path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When setting up the aamix output paths, use the primary DAC instead of
the individual DAC for each output as default. Otherwise multiple
DACs will be turned on for a single aamix widget, which results in
doubly or more volumes, because the duplicated signals will be sent
through all these DACs for a single stream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we have a loopback mixer control, this should manage the state
whether the output paths include the aamix or not. But the current
code blindly initializes the output paths with aamix = true, thus the
aamix is enabled unless the loopback mixer control is changed.
Also, update_aamix_paths() called by the loopback mixer control put
callback invokes snd_hda_activate_path() with aamix = true even for
disabling the mixing. This leaves the aamix path even though the
loopback control is turned off.
This patch fixes these issues:
- Introduced aamix_default() helper to indicate whether with_aamix is
true or false as default
- Fix the argument in update_aamix_paths() for disabling loopback
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
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Merge tag 'asoc-v3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
Now that we have a flag for headphone mics, we can use that flag
in the jack creation instead of creating the jack manually.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows a specific mic to get the "Headphone Mic" name, in addition
to the existing "Headset Mic" name.
Also, it allows for a special mark: if the sequence number is set
to 0xc, that's an indication to prefer it for headset mic, and if it's
set to 0xd, that's an indication to prefer it for headphone mic.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
changed is not initialized in path_power_down_sync, but it is expected
to be false in case no change happened in the loop. So set it to
false.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The lack of independent HP mode shouldn't be too bad, but currently
its badness is set a bit too high. Let's lower it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard badness values don't seem to fit to all preferences.
Some configuration prefer the side output over the headphone, some
want the speaker over the surround, etc.
This patch moves the badness table pointers into hda_gen_spec, so that
the codec driver can override them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge back for-linus branch for the badness table adjustment for VIA codecs
* for-linus:
ALSA: hda - Fix DAC assignment for independent HP
ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
ALSA: hda - Fix typo in checking IEC958 emphasis bit
ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
ALSA: snd-usb: mixer: propagate errors up the call chain
ALSA: usb: Parse UAC2 extension unit like for UAC1
ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
The generic parser should evaluate the availability of the independent
HP when specified. Otherwise a DAC without the direct connection to
the corresponding pin may be assigned for the HP, but the driver
doesn't check it at all. The problem was actually seen on some
machines with VT1708s or equivalent codec, where DAC0 is assigned to
HP although it can be connected only via aamix.
This patch adds the badness evaluation for the independent HP to make
it working properly.
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling snd_hda_attach_beep_device() and
snd_hda_detach_beep_device() in each codec driver, move them to the
generic parser. The codec driver just needs to set spec->beep_nid for
activating the digital beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a headphone pin is set up as a shared hp/mic pin, we rather want
to keep it as a headphone primarily as default, but the driver
overrides it always as a mic pin, just because the input controls are
created after outputs. Add a check of pin NID and skip the
re-initialization of pinctl for such a shared hp/mic pin.
Reported-by: Jonathan Woithe <jwoithe@just42.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone mic jack enum control is created (via explicitly
specification by user), it doesn't make much sense to change the I/O
direction dynamically per capture source change, since the I/O
direction is rather controlled over the enum ctl.
This also reduces the implicit dependency between the capture source
and the hp mic jack enum ctls, which might confuse a program accessing
the whole control elements at once like alsactl.
In addition, this patch introduces update_hp_automute_hook() function
to call the proper hook function. It's just to remove the open codes
in multiple places in hda_generic.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no big merit to distinguish these two hints. Instead, just
have a single flag, add_jack_modes, for creating the jack mode enum
ctls for both I/O directions.
The hint string parser code is left and translated as add_jack_modes
just for keeping compatibility.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commits added the capability to change the pin control of
hp/mic shared jack, but it actually didn't work as expected when the
value is changed from the output to the input, since I forgot to reset
the pin I/O bit in that case. This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a headphone jack is configured as a shared hp/mic jack, the jack
mode enum needs to handle both input and output directions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch improves the generic parser code to allow to set up the
headphone jack as a mic input. User can enable this feature by giving
hp_mic hint string.
The former shared hp/mic feature for the single built-in mic is still
retained. This detection can be disabled now via hp_mic_detect hint
string, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current badness value used for the missing multi-io seems too
weak, and the multi-io tends to be skipped for desktop configurations
when no enough DACs are available. It's because the total badness of
the multi-io becomes often larger than the badness with assigning an
individual DAC to a headphone jack. This is good for one side, but it
seems that the surround outputs are more demanded by that.
This patch increases the badness value for the missing multi-io
slightly so that the multi-io would be preferred than the individual
headphone DAC if they conflict. Through the tests with hda-emu,
mostly only desktop configurations with ALC662/663 and CMI codecs are
affected by this change, and all look reasonable.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loopback list is referred by the VIA codec driver no matter
whether CONFIG_PM is set or not, thus we need to enable it always.
Otherwise it gets compile errors.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]
For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros. This should have the same effect but shut up warnings like
above.
But since we had already put ifdefs, changing to inline functions
would trigger compile errors. So, such ifdefs is removed in this
patch.
In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too. For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.
Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a better power filter hook for powering down unused
widgets in the generic parser.
The feature is enabled by setting hda_gen_spec.power_down_unused
flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The arguments to call is_active_nid() in activate_amp() were swapped,
and this resulted in the muted amp on some SPDIF output pins.
Also, the index to be passed to is_active_nid() must be idx_to_check.
Otherwise it checks the wrong connection in the case of implicit aamix
connection paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD1988 family and AD1882 codecs have another mixer widget (0x21)
between the analog-loopback mixer widget (0x20) and the actual
outputs. Due to this hole, the analog-loopbacks aren't sent properly
to the output pins.
As a band-aid fix, introduce another fields holding the aamix merge
path, and activate it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since both snd_hda_codec_flush_amp_cache() and
snd_hda_codec_flush_cmd_cache() are called usually at the same time,
we can simply combine them to a single function,
snd_hda_codec_flush_cache().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture volume put callback may call the node selection change,
and its actual call won't be triggered unless flushed. In general,
we always need to call both snd_hda_codec_flush_amp_cache() and
snd_hda_codec_flush_cmd_cache() at the same place...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both the HP auto-mute and the independent HP mode conflict with each
other. Make HP auto-mute disabled (only for the affected HP jack)
during the driver is in HP independent mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It'd be better to give another name to the secondary (alt) analog PCM
stream, which is dedicated for the independent HP out and extra
inputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch eventually fixes two issues:
- Handle the case where the primary output is a headphone and can have
independent HP mode;
so far we checked only the case where the headphone is the secondary
output.
- Fix the conflict of HP independent mode and aamix mode;
when switched to aamix mode, the DAC might be also switched to
another widget shared with other outputs. Then even if we disable
the DAC for the original output, it doesn't change -- because the
active route is from another (shared) DAC to HP pin through aamix.
So, in such a case, we have to prohibit the switch to aamix for HP
routes.
This fixes issues appearing on VT codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many codecs provide routes to multiple output pins through an aamix
widget, but most of them do it only from a single DAC. However, the
current generic parser checks only the aamix paths from the original
(directly bound) DACs through aamix NID, and miss the path:
primary DAC -> aamix -> target out pin
This patch adds a more check for the routes like the above.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a patch couldn't be resolved in try_assign_dacs() although the
target DAC is expected, we forgot to add a proper badness value but
continued.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since fill_and_eval_dacs() may be called repeatedly with different
configurations, setting pinctls at each time there isn't optimal.
We can set it better only once after deciding the output configuration
in parse_output_paths().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Print the information of outputs in a bit more details and concisely
in a single place instead of printing the path at each time when
detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch "ALSA: hda - fix wrong adc_idx in generic parser" fixed the
adc_idx for the capture volume and capture switch controls. But also
modified the adc_idx retrieval for the capture source controls
wrongly. As multiple capture source controls are created in a single
shot with counts > 1, the id.index doesn't contain the real value.
The real index has to be taken via snd_ctl_get_ioffidx() as in the
original code.
This patch reverts the fixes partially to recover from the
regression.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just stumbled over this one while reading the code.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I found a codec configuration which had six inputs, so the max of
five was not appropriate.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although I commented that boost volumes would be added only for
line-in and mic pins in the source code, the actual code excludes but
for mic-in. Fix it to accept the line-ins, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two hooks in hda_gen_spec, cap_sync_hook and capture_switch_hook, play
very similar roles. The only differences are that the former is
called more often (e.g. at init or switching capsrc) while the latter
can take an on/off argument.
As a more generic implementation, consolidate these two hooks, and
pass snd_ctl_elem_value pointer as the second argument. If the
secondary argument is non-NULL, it can take the on/off value, so the
caller handles it like the former capture_switch_hook. If it's NULL,
it's called in the init or capsrc switch case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a standard capture switch without multiple binding is used, the
call for capture_switch_hook isn't called properly. Replace the put
ops to add the hook call in that case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current generic parser code, we look for the (mic) boost
controls only on input pins. But many codecs assign the boost volume
to a widget connected to each input pin instead of the input amp of
the pin itself.
In this patch, the parser tries to look through more widgets connected
to the pin and find a boost amp.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an amp in the activation path is associated with mixer controls,
activate_amp() tries to skip the initialization. It's good, but only
if the mixer really initializes both mute and volume. Otherwise,
either the mute of the volume is left uninitialized.
This patch adds this missing check and properly initialize the
partially controlled amps in an activation path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places creating the labels and indices of kctls for
each input pin in the current generic parser code. This is redundant
and makes harder to maintain. Let's create the labels and indices at
once and keep them in hda_gen_spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the imux table entries can be a subset of autocfg.input table,
the indices of these aren't always same. For passing the proper index
value of autocfg.input at creating input ctl labels (via
snd_hda_autocfg_input_label()), keep the corresponding autocfg.input
idx value in the index field of each imux item, which isn't used in
the generic driver.
Also, this makes easier to check the invalid imux pin for stereo mix.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the volume or mute control of the primary output is shared with
other (headphone or speaker) outputs, we shouldn't name it as a
specific output type but rather name it with the channel name or a
generic name like "PCM".
Also, this check should be performed individually for the volume and
the mute controls because some codecs may have shared volumes but
separate mute controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the generic parser reduces the ADC list, copy the list of the
all detected ADCs and keep it.
This list can be later referred by the codec driver for finer power
controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add vmaster_tlv[] to hda_gen_spec and store the suggested TLV data
in snd_hda_gen_parse_auto_config(). This allows the codec driver to
correct the TLV data (e.g. mute capability) before actually creating
vmaster instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like the jack mode enum ctls for output jacks, add the support
for similar enum ctls for input pins to control the bias Vref.
The new controls will be added when spec->add_in_jack_modes is set
either by the codec driver or by a hint string.
Note that ground and 100% vrefs are excluded from the list for
simplicity, currently. We may add a new flag to allow them, too.
But I guess it's easier to put a value override in the pinfix in such
a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A new flag to skip the auto-mute handling in the generic parser, just
like suppress_auto_mic flag. It has to be set before calling
snd_hda_gen_parse_auto_config().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai
David Henningsson
Joe Perches
Dan Carpenter
Jiri Slaby