Now some codes and functionalities of hda_codec struct are moved to
hdac_device struct. A few basic attributes like the codec address,
vendor ID number, FG numbers, etc are moved to hdac_device, and they
are accessed like codec->core.addr. The basic verb exec functions are
moved, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David suggested that the name "power_mgmt" is too ambiguous. Rename
the flag with a bit clearer one "power_save_node".
Also, add the corresponding description to HD-Audio.txt, too.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some pins are used for controlling the LED with the VREF value.
This patch changes the power behavior of such pins to be constantly
up. A new state, pin_fixed, is introduced to nid_path to indicate
that the path contains the fixed pin. This improves also the
readability a bit for other static routes, too.
Then a helper function snd_hda_gen_fix_pin_power() is called from the
codec driver for such fixed pins, and it will create fake paths
containing only these pins with pin_fixed=1 flag.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the widget PM may turn off the pins, this might lead to the silent
output for beep when no explicit paths are given. This patch adds
fake output paths for the beep widget so that the output pins are
dynamically powered upon beep on/off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables the finer power state control of each widget
depending on the jack plug state and streaming state in addition to
the existing power_down_unused power optimization. The new feature is
enabled only when codec->power_mgmt flag is set.
Two new flags, pin_enabled and stream_enabled, are introduced in
nid_path struct for marking the two individual power states: the pin
plug/unplug and DAC/ADC stream, respectively. They can be set
statically in case they are static routes (e.g. some mixer paths),
too.
The power up and down events for each pin are triggered via the
standard hda_jack table. The call order is hard-coded, relying on the
current implementation of jack event chain (a la FILO/stack order).
One point to be dealt carefully is that DAC/ADC cannot be powered
on/off while streaming. They are pinned as long as the stream is
running. For controlling the power of DAC/ADC, a new patch_ops is
added. The generic parser provides the default callback for that.
As of this patch, only IDT/Sigmatel codec driver enables the flag.
The support on other codecs will follow.
An assumption we made in this code is that the widget state (e.g. amp,
pinctl, connections) remains after the widget power transition (not
about FG power transition). This is true for IDT codecs, at least.
But if the widget state is lost at widget power transition, we'd need
to implement additional code to sync the cached amp/verbs for the
specific NID.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch does two things:
- code refactoring with a local helper function,
- allow codec drivers to provide the specific PCM stream info pointers
only for overriding the non-NULL entries, instead of copying the
whole.
This simplifies the codec driver side (currently the only user is
alc269's 44kHz fixed rate).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [ef403edb75: ALSA: hda - Don't access stereo amps for
mono channel widgets] fixed the handling of mono widgets in general,
but it still misses an exceptional case: namely, a mono mixer widget
taking a single stereo input. In this case, it has stereo volumes
although it's a mono widget, and thus we have to take care of both
left and right input channels, as stated in HD-audio spec ("7.1.3
Widget Interconnection Rules").
This patch covers this missing piece by adding proper checks of stereo
amps in both the generic parser and the proc output codes.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current HDA generic parser initializes / modifies the amp values
always in stereo, but this seems causing the problem on ALC3229 codec
that has a few mono channel widgets: namely, these mono widgets react
to actions for both channels equally.
In the driver code, we do care the mono channel and create a control
only for the left channel (as defined in HD-audio spec) for such a
node. When the control is updated, only the left channel value is
changed. However, in the resume, the right channel value is also
restored from the initial value we took as stereo, and this overwrites
the left channel value. This ends up being the silent output as the
right channel has been never touched and remains muted.
This patch covers the places where unconditional stereo amp accesses
are done and converts to the conditional accesses.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=94581
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the hda_codec object kept the hda_pcm list in an array, and
the codec driver was expected to assign the array. However, this
makes the object life cycle management harder, because the assigned
array is freed at the codec driver detach while it might be still
accessed by the opened streams.
In this patch, we allocate each hda_pcm object dynamically and manage
it as a linked list. Each object has a kref refcount, and both the
codec driver binder and the PCM open/close touches it, so that the
object won't be freed while in use.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we create the standard HD-audio bus (/sys/bus/hdaudio), and bind
the codec driver with the codec device over there. This is the first
step of the whole transition so that the changes to each codec driver
are kept as minimal as possible.
Each codec driver needs to register hda_codec_driver struct containing
the currently existing preset via the new helper macro
module_hda_codec_driver(). The old hda_codec_preset_list is replaced
with this infrastructure. The generic parsers (for HDMI and other)
are also included in the preset with the special IDs to bind
uniquely.
In HD-audio core side, the device binding code is split to
hda_bind.c. It provides the snd_hda_bus_type implementation to match
the codec driver with the given codec vendor ID. It also manages the
module auto-loading by itself like before: when the matching isn't
found, it tries to probe the corresponding codec modules, and finally
falls back to the generic drivers. (The special ID mentioned above is
set at this stage.)
The only visible change to outside is that the hdaudio sysfs entry now
appears in /sys/bus/devices, not as a sound class device.
More works to move the suspend/resume and remove ops will be
(hopefully) done in later patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... for distinguishing whether it's explicitly enabled via a user hint
or enabled by a driver as a fallback. Now the former case corresponds
to HDA_HINT_STEREO_MIX_ENABLE while the latter to
HDA_HINT_STEREO_MIX_AUTO.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the stereo mix input is explicitly enabled via a user hint, the
driver should create always a capture source enum ctl and disable the
auto-mic switch. Otherwise the behavior gets confused. For doing it,
this patch just sets spec->suppress_auto_mic flag appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
