Taking another look at the C400 descriptors, I see now that there is
a clock selector (0x80) for this device.
Right now, the clock source points to the internal clock (0x81), which
is also valid. When the external clock source (0x82) is selected in the
mixer, and the rates mismatch (if it's free-running it is fixed to
48KHz), xruns will occur.
Set the clock ID to the clock selector unit (0x81), which then
allows the validation code to function correctly.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A patch in the 3.2 kernel caused regression with hotplugging the
M-Audio Fast track pro, or sound after suspend. I don't have the
device so I haven't done a full analysis, but it seems userspace
(both udev and pulseaudio) got confused when a card was created,
immediately destroyed, and then created again.
However, at least one person in the bug report (martin djfun)
reports that this patch resolves the issue for him. It also leaves
a message in the log:
"snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is
a bit misleading. It is better than non-working audio, but maybe
there's a more elegant solution?
BugLink: https://bugs.launchpad.net/bugs/1095315
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another step forward. As all features for VIA codecs have been
implemented in the generic driver, we can move on to migrate the VIA
codec parser, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the support for the generic auto-parser to AD codec
driver. For AD1988, the old code is replaced simply with the new
generic parser. For other codecs, new model "auto" is added and
directed to use the generic parser.
No fixup codes have been implemented yet as of now. Eventually we'd
replace each static quirk with the generic parser + fixup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just shuffle the codes and add ifdefs for testing to drop the static
quirk codes from patch_conexant.c.
By commenting out ENABLE_CXT_STATIC_QUIRKS define at the beginning of
the file, you can disable the whole static codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time, the target is Cirrus codec. Its parser is a subset of
generic parser, so we can migrate fully with it now.
The only tricky part is the handling of SPDIF automute.
Cirrus driver sets the SPDIF out plug over the headphone. As a
workaround, set spec->gen.master_mute for toggling the headphone (and
other) mute.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CA0110 codec is a fairly straightforward hardware implementation,
and we can use the generic parser almost as is.
Just set spec->multi_cap_vol flag to follow the current behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the old parser code for C-Media auto-parser with the latest
generic parser. For compatibility reason, the static bindings are
still left, but they could be cleaned up in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pincfgs, init_verbs and hints set by sysfs or patch might be
changed dynamically on the fly, thus we need to protect it.
Add a simple protection via a mutex.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As David Henningsson recently suggested, some HP laptops use an unused
mic pin for controlling a mute LED, and this information is provided
via DMI string "HP_Mute_LED_X_Y" string. This patch adds the generic
support for such cases, as we've already done in patch_sigmatel.c.
This is applied generically to all devices with ID 0x103c.
But as we don't know whether the device 103c:1586 really contains
HP_Mute_LED_X_Y DMI string, still keep the static setup for this
device using the mic2 pin 0x19.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some fixups such as setting the flags influencing on the parser
behavior should be applied before actually parsing the tree.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Try to recover from the regression: set the HP amp for the speaker and
add the hp jack mode enum as default.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the enum controls for changing the headphone amp bits of output
jacks, such as "Headphone Jack Mode". This feature isn't enabled as
default, so far, unless spec->add_out_jack_modes flag is set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a multi-io jack is switched to another direction, call the
automute and autoswitch update functions, as this jack won't be used
as the headphone or the mic jack that may turn off others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new fixup type, HDA_FIXUP_PINCTLS, for overriding the pinctl
values of the given pins. It takes the same array of struct pintbl
like HDA_FIXUP_PINS, but each entry contains the pinctl value instead
of the pin default config value.
This patch also replaces the corresponding codes in patch_realtek.c.
Without this change, the direct call of verbs may be overridden again
by the later call of pinctl restoration by the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now the whole codebase has been changed from the earlier kernels, it
makes little sense to keep these aliases. Simply replace with the
official names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a jack is retasked as a different direction (e.g. a mic jack is
used as a CLFE output), such a jack shouldn't be counted as the target
for the automatic jack switching. Skip the automute or the autoswitch
when the current pinctl direction is different from what we suppose.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new pin target accessors for managing the current pinctl
values in the generic parser. The pinctl values of all active pins
are once determined at the initialization phase, and stored via
snd_hda_codec_set_pin_target(). This will be referred again in the
codec init or resume phase to set the actual pinctl.
This value is kept while the auto-mute. When a line-out or a speaker
pin is muted by auto-mute, the driver simply disables the pin, but it
doesn't touch the cached pinctl target value. Upon unmute, this value
is used to restore the original pinctl in return.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check more strictly about the validity of pinctl values in
snd_hda_set_pin_ctl() and correct the wrong bits automatically.
Also provide the helper function to correct pinctl bits to codec
drivers.
This automatically fixes the invalid pinctl writes that are found in
a few Realtek fixups for NID 0x0f amp like ASUS A6Rp.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We already have the list of whole pin widgets and there is an unused
space in the list; let's use it for caching the current pinctl value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a DAC is reassigned from surrounds to front or ADCs are reduced
due to incomplete imux, we clear the path indices but the path
instances remain as is. Since the paths might be still referred
through the whole path list parsing (e.g. is_active_nid()), we should
clear these path instances as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since some codecs can choose the aamix as a capture source, we should
support it as well. When spec->add_stereo_mix_input flag is set, the
parser checks the availability of aamix as the input source, and adds
the paths automatically when possible.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current parser code, the input_paths[] may become inconsistent
when some of detected ADCs are dropped due to incomplete inputs, since
the driver rearranges only adc_nids[] but doesn't touch input_paths[].
This patch fixes the issue, and also it optimizes the reachability
checks by simply referring to the parsed input_paths[] instead of
calling is_reachable() again for each connection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of handling special cases in the caller side, give a proper
name string "Headphone Mic" from hda_get_autocfg_input_label() when
the headhpone jack pin is specified as an input.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture paths shouldn't contain the analog loopback mixer.
Pass a proper argument to exclude the aamix NID.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag spec->suppress_mic_auto_switch for codecs that don't
support unsol events properly like VT1708.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the default config value shows the connection AC_JACK_PORT_BOTH,
it's better to handle it as a speaker pin. This makes the behavior
consistent in snd_hda_get_pin_label() and snd_hda_parse_pin_defcfg().
There are only few old machines showing this attribute, and all of
them are actually fixed speaker pins, as far as I know.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit modifies the definition of snd_hda_parse_nid_path()
slightly, now with_aa_mix argument is changed to anchor_nid, so that
it can handle any NID generically as an anchor point to include or
exclude.
The with_aa_mix field in struct nid_path is removed again by this
change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The multi-io channels can vary not only from 1 to 6 but also may vary
from 6 to 8 or such. At the same time, there are more speaker pins
available than the primary output pins. So, we need three variables
to check: the minimum channel counts for primary outputs, the current
channel counts for primary outputs, and the minimum channel counts for
all outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of looking through paths with the dac -> pin connection at
each time, just pass the already parsed path index to
assign_out_path_ctls(). This simplifies the code a bit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The path indices must be reset at each evaluation of DAC assignment.
Otherwise the badness value will be wrongly calculated and mixers may
be inconsistently assigned.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let is_jack_detectable() return true when the jack polling is enabled
for the codec. VT1708 uses the polling explicitly so we need to allow
it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new hook which is called at each PCM playback ops.
It can be used to control the codec-specific power-saving feature in
each codec driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bind-volume workaround was introduced for simplifying the mixer
abstraction in the case where one or more pins of multiple outputs
lack of individual volume controls. This was essentially the case
like Acer Aspire 5935, which has 5.1 speakers and 5.1 (multi-io)
jacks although there are 5 DACs, so some of them must share a DAC.
However, the recent code rewrite changed the DAC assignment policy to
share with the same channel instead of binding to the front, thus
binding the volumes for all channels makes little sense now, rather
it's confusing. So in this patch, the ugly workaround is finally
dropped and simply create the volume control corresponding to the
parsed path position.
For dual headphones or 2.1 speakers with a shared volume control, it's
anyway bound to the same DAC if needed, so this change shouldn't bring
any practical difference.
And, as a good bonus, we can cut off the whole code handling the bind
volume elements.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When 5.1 or more multiple speakers with found but not enough DACs are
available, it's better to bind such pins to the DACs of the primary
outputs with the same channels rather than binding to the first DAC
(i.e. the front channel). For the cases with two speaker pins, it's
rather regarded as front + bass combination, thus it's more practical
to still bind to the front, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... like "Speaker Surround Playback Switch".
This fix had been already applied to patch_conexant.c but was
forgotten in other places, then migrated to hda_generic.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For codecs that have individual routes going through a loopback mixer
(like VIA codecs), we need to provide an explicit switch to choose
whether the output goes through mixer or directly from DAC.
This patch adds the check for such paths and creates "Loopback Mixing"
enum control when available.
It won't influence on codecs like Realtek or others where the loopback
mixer is connected independently from the primary output routes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The output paths including aamix should be parsed only for the first
output. The surround paths including aamix must be wrong, since it
would mix all streams, i.e. all channels would be mixed into a single
and multiplexed again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call the path activation for the digital input pin properly, not only
setting the pin control. Also add spec->digin_path for keeping the
path index.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of search for the path with the certain route at each time,
keep the path index for each output and loopback, and just use it when
referred.
In this implementation, the path index number begins with one, not
zero (although I've been writing in C over decades). It's just to
make the check for uninitialized values easier.
So far, the input paths aren't handled with indices yet, but still
picked up via snd_hda_get_nid_path() at each time.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When speakers are chosen as the the primary output during evaluation,
we did some tricks to assign the possible multi-io jacks with a
certain offset value to multi_out dacs. This was a workaround for the
case with multiple speakers like Acer Aspire. But this is quite ugly
at the same time and the resultant code is hard to understand. More
badly, it works wrongly for 2.1 speakers like Apple iMac91.
In this patch, instead of fiddling with the offset to multi_out dacs,
simply add a certain badness number if headphone(s) + multi-ios are
possible. This simplify the code a bit, and it's more robust.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the requested path has been already added, return the existing path
instance instead of adding a duplicated instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the paths are created in map_singles(), we don't have to
re-create new paths in try_assign_dacs(). Just evaluate the badness
and skip to the next item.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set path->active flag at the path creation time and let the paths
initialized according to the current path->active state in
set_output_and_unmute(). This allows to modify the active flag of
some output paths dynamically, e.g. switching the front output route
with or without aamix like patch_via.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
activate_amp() in the generic parser checks whether the given NID is
included in any active paths and skips it if found. This was a
workaround for avoiding disabling the widgets in the active paths when
one path is disabled, thus it shouldn't be applied to the case for
path activation. Due to this wrong check, some analog loopback paths
haven't been initialized correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Manage the connection list cache using linked lists instead of
snd_array, and revive snd_hda_get_conn_list() again, so that we don't
have to keep the expanded values locally.
This will reduce the stack usage by recursive call of
snd_hda_get_conn_index() or parse_nid_path() of the generic parser.
The list management doesn't include any mutex protection, thus the
caller needs to take care of race appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another broken hardware workaround: there are hardware indicating
the inverted jack detection bit result.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the new flag, codec->inv_eapd, indicating that the EAPD
implementation is inverted.
There are always broken hardware in the world.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar like the implementation in patch_analog.c and patch_via.c,
the generic parser can provide the independent HP PCM stream now.
It's enabled when spec->indep_hp is set by the caller while parsing.
Currently no dynamic PCM switching as in patch_via.c is implemented
yet. The control returns -EBUSY when the value is changed during PCM
operations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the path including the loopback mixer widget in the primary
output channel as an alternative in the generic codec parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a better debug print code to show the new assigned paths in
generic parser. It appears only with CONFIG_SND_DEBUG_VERBOSE=y.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's never used in the generic parser. It was there from the old
Realtek code, which has been dropped quite ago, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a PCM name string is generated from the chip name, it might
become strange like "CX20549 (Venice) Analog". In this patch, the
parser tries to drop the invalid words like "(Venice)" in the PCM name
string. Also, when the name string is given beforehand by the caller,
respect it and use it as is.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There were some old codes that look not stable enough, which was
derived from the old Realtek code. The initialization for primary
output in init_multi_out() needs to consider the case of shared DAC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For preliminary works to migrate the generic parser for Conexant
codecs: the same function is ported to hda_generic.c.
But now it looks through the jack detect table so that it can cover
better.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a flag to indicate whether the vmaster mute hook enum is exposed
or not. Conexant codecs may want not to expose the control depending
on the model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Old codecs like AD1986A tend to have long paths as they were just made
to be the way like AC97. The current max depth 5 can be too short for
such devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The access to a cache array element could be invalid outside the
mutex, so copy locally for the later references.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dirty entry has to be checked at the beginning in the loop, not in
the inner loop for channels. This caused a regression that the right
channel isn't properly written.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bound capture volume and switch controls use the cached amp
updates, but it's missing the flushing at the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The inverted dmic fix overwrites the right channel amp value, but it
would work only when the amp values have been already actually
written. Put snd_hda_codec_resume_amp() before the amp write for
flushing caches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add an overflow check of CORB in HD-audio controller and codec drivers
so that flood of sequential writes would work properly.
In the controller side, add a check of CORB read-pointer to make
returning -EAGAIN when it's full. Meanwhile in the codec side, when
-EAGAIN error is received, it retries the write after flushing the
pending verbs (calling get_response() essentially does it).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These functions are supposed to be called at finishing the cached
sequential writes, so clear the flag properly for lazy developers who
often forget details.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When verbs or amps are actually written to the hardware, we can clear
dirty flag so that the later snd_hda_codec_resume_*() calls can skip
these verbs / amps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [2e9bf24: ALSA: hda_codec: Check for invalid zero
connections] trims the whole connection list when an invalid value is
reported by the hardware. But some codecs (at least AD1986A) may give
a zero NID in the middle of the connection list, so dropping the whole
list isn't good for such cases.
In this patch, as a workaround, allow a single zero NID in the read
connection list. If it hits zero twice, it's handled as an error, so
that we can avoid "too many connections" errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In general we prefer "Capture Source" to "Input Source".
The latter was chosen in many places just because "Capture Source"
label doesn't work well with the current alsa-lib mixer abstraction
when multiple instances are present. But when we know that there is a
single input-source element, we can safely choose "Capture Source"
label.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The next migration step is to use the common code in generic driver
for Realtek driver. This is no drastic change and there should be no
real functional changes, as the generic parser code comes from Realtek
driver originally.
As Realtek driver requires the generic parser code, it needs a
reverse-selection of CONFIG_SND_HDA_GENERIC kconfig.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These handlers are supposed to be called externally from the codec
drivers once when they need to handle own jack events.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When no controls are assigned in the parser (e.g. no analog path),
spec->kctls.list is still NULL. We need to check it before passing to
snd_hda_add_new_ctls().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some cases, we want to manipulate the auto_pin_cfg table before
passing to snd_hda_gen_parse_auto_config() (e.g. Realtek SSID check
code fiddles with the headphone pin). Also passing ignore_pins just
for snd_hda_parse_pin_defcfg() isn't good.
In this patch, snd_hda_gen_parse_auto_config() is changed to receive
the auto_pin_cfg table to be parsed. The passed auto_pin_cfg table
must have been initialized (typically by calling
snd_hda_gen_parse_auto_config()) beforehand by the caller.
Also together with this change, spec->parse_flags is also removed.
Since this was referred only at the place calling
snd_hda_parse_pin_defcfg(), no longer needed to be kept in spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally the whole generic parser code in Realtek driver is moved into
hda_generic.c so that it can be used for generic codec driver.
The old dumb generic driver is replaced. Yay.
The future plan is to adapt this generic parser for other codecs,
i.e. the codec driver calls the exported functions in generic driver
but adds some codec-specific fixes and setups.
As of this commit, the complete driver code is still duplicated in
Realtek codec driver. The big code reduction will come from now on.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch extends the capability of the auto-mic feature.
Instead of limiting the automatic input-source selection only to the
mics (internal, external and dock mics), allow it for generic inputs,
e.g. switching between the rear line-in and the front mic.
The logic is to check the attribute and location of input pins, and
enable the automatic selection feature only if all such pins are in
different locations (e.g. internal, front, rear, etc) and line-in or
mic pins. That is, if multiple input pins are assigned to a single
location, the feature isn't enabled because we don't know the
priority.
(You may wonder why this restriction doesn't exist for the headphone
automute. The reason is that the output case is different from the
input: the input source is an exclusive selection while the output
can be multiplexed.)
Note that, for avoiding regressions, the line-in auto switching
feature isn't activated as default. It has to be set explicitly via
spec->line_in_auto_switch flag in a fixup code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put INPUT_PIN_ATTR_FRONT after INPUT_PIN_ATTR_REAR, and define
INPUT_PIN_ATTR_LAST to point to the last element.
This is a preliminary work for cleaning up Realtek auto-mic parser.
In the auto-mic implementation, the front panel is preferred over the
rear panel. By arranging the attr definitions like in this commit, we
can simply use sort() for figuring out the priority order.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit passed an utterly wrong value for checking the
split inv dmic pin. This patch fixes it and also tries to remove
inv_dmic_split_idx field.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the fixup code is used commonly, it's worth to move it to the
common place, struct hda_codec, instead of keeping in hda_gen_spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To make the parser more generic, a few codes to handle the inverted
stereo dmic in a way Conexant parser does is added in this patch.
The caller should set spec->inv_dmic_split flag appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we create only "Capture Volume" and "Capture Switch" controls
for binding all possible amps, but we'd prefer creating individual
capture volume and switch controls per input in some cases
(e.g. conexant parser does it).
Add a new flag, spec->multi_cap_vol, to follow that policy.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge a few functions that have been split due to historical reasons
to single functions. Splitting too much (and placing too far away)
actually worsens the readability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few more cases where we can assign "Master" mixer element
safely, e.g. when a single DAC is used in the whole output paths.
Also, when vmaster hook is present, avoid "Master" but assign "PCM"
instead. Otherwise vmaster hook won't work properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... so that the fixup just needs to set the hook function in
FIXUP_ACT_PROBE. This will make easier to port for other codecs,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new field to indicate the possible pin NID for alternative vref
setup for the shared hp/mic. Although 0x18 is valid for all Realtek
codecs, it'll be different on other vendor's codecs.
Also, drop the sanity check in update_shared_mic_hp() since the
reference pin is set explicitly in the caller side.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the function more generic for both input and output directions,
and returns the assigned path pointer. The argument order is changed
to follow the standard (from, to) way.
Now this new function is used for analog input and loopback path
parser codes, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amps will be initialized via activate_path(), thus it's
superfluous to set in alc_auto_init_analog_input().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some fields from struct alc_spec, and clean up the usage.
Namely,
- spec->input_mux becomes a single element, private_imux[] is removed
- spec->adc_nids becomes an array by itself, and private_adc_nids[]
gets removed, too
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we reached to the final big piece of parser rewrite: the input
paths. While the old parser code assumes the more-or-less direct and
similar connections from input pin to ADC, the new code handles the
complete input paths. The capture source is switched by simple calls
of activate_path() function.
The parsing of capture volume and capture switches is, however, not
fully generalized. It assumes that amps are available in the vicinity
of ADCs (in three depth). This isn't perfect but it should cover all
codecs I know of.
Also, this commit removes some NID mapping of capture-related controls
temporarily for simplicity. It'll be restored in later commits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we have a complete list of loopback paths, thus we can initialize
the paths more completely based on it, instead of assuming a direct
connection from pin to mixer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't forget to take boost volumes into account in the managed path
list. Since it's an additional volume, we need to extend the ctls[]
array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The paths used for multi-io haven't been initialized properly, so
far. It's usually no big matter because the pins are set to input as
default, but it's still cleaner to initialize the paths properly.
Now with the path active/inactive check, we can do it easily.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pin widget has only a single amp value for the input even if it
has multiple "sources". Handle the situation in activate_path().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc_auto_is_dac_reachable() can be replaced fully with
is_reachable_path(). The only difference is the order of arguments.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... and rewrite the initialization of output paths as a generic
function that is applicable for both i/o directions.
The new flag, active, is introduced to each nid_path entry. This
indicates whether the given path is active, and it's used for checking
whether a certain widget can be turned off or changed when a path is
no longer used or newly enabled.
It's still used only in the output paths. More wider adaption for
input and loopback paths will be achieved in the later patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We are using only AUTOMUTE_MODE_PIN in patch_realtek.c and all others
have been already dropped. Let's remove the old superfluous codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new function snd_hda_codec_amp_init() (and the stereo variant)
initializes the amp value only once at the first access. If the amp
was already initialized or updated, this won't do anything more.
It's useful for initializing the input amps that are in the part of
the path but never used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For optimizing the verb executions, a new mechanism to cache the verbs
and amp update commands is introduced. With the new "write to cache
and flush" way, you can reduce the same verbs that have been written
multiple times.
When codec->cached_write flag is set, the further
snd_hda_codec_write_cache() and snd_hda_codec_amp_stereo() calls will
be performed only on the command or amp cache table, but not sent to
the hardware yet. Once after you call all commands and update amps,
call snd_hda_codec_resume_amp() and snd_hda_codec_resume_cache().
Then all cached writes and amp updates will be written to the
hardware, and the dirty flags are cleared.
In this implementation, the existing cache table is reused, so
actually no big code change is seen here. Each cache entry has a new
dirty flag now (so the cache key is now reduced to 31bit).
As a good side-effect by this change, snd_hda_codec_resume_*() will no
longer execute verbs that have been already issued during the resume
phase by checking the dirty flags.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for
audioformat mismatch] introduced the correction of parameters to be
set for sync EP. But since the new code assumes that the sync EP is
always paired with the data EP of another direction, it triggers Oops
when a device only with a single direction is used.
This patch adds a proper check of sync EP type and the presence of the
paired substream for avoiding the crash.
Reported-and-tested-by: Jens Axboe <axboe@kernel.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nothing terribly exciting here except for the DOUBLE_RANGE fix which
just hadn't worked before, nobody noticed due to lack of use.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=dgII
-----END PGP SIGNATURE-----
Merge tag 'asoc-fix-3.8-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.8
Nothing terribly exciting here except for the DOUBLE_RANGE fix which
just hadn't worked before, nobody noticed due to lack of use.
When initializing the output paths, we assumed the input amps have
almost two inputs blindly. It's not only generic but even incorrect
for some codecs like ALC268 & co. Also, the same assumption (two
sources) exists for the bind input-amp controls.
This patch changes the codes in these places to handle the input
connections in a more generic way.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For handling the analog-loopback paths more generically, check the amp
capabilities of the aa-mixer widget, and create only the appropriate
mixer elements.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve the parser of analog loopback paths and handle in a more
generic way. The following changes are included in this patch:
- Instead of assuming direct connections between pins and
the mixer widget, track the whole path between them. This fixes
some missing connections like ALC660.
- Introduce the path list for loopback paths like input and output
path lists. Currently it's not used for any real purposes, yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like the output paths, parse the whole paths for inputs as well
and store in a path list. For that purpose, rewrite the output parser
code to be generically usable.
The input path list is not referred at all in this patch. It'll be
used to replace the fixed adc/capsrc array in later patches for more
flexible input path selections.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, idx[i] and multi[i] indicate the attribute of the widget
path[i - 1]. This was just for simplifying the code in
__parse_output_path(), but this is rather confusing for later use.
It's more natural if both idx[i] and multi[i] point to the same widget
of path[i]. This patch changes to that way.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the output path initialization using the existing path
information instead of assuming the topology specific to Realtek
codecs. This is also implicitly a fix for some amp values on output
pins where the old parser missed (e.g. ALC260 output pins).
The same function alc_auto_set_output_and_unmute() can be used now for
the multi-io activation, since the output selection means nothing but
activating the given output path.
And, finally at this stage, we can get rid of alc_go_down_to_selector()
and other functions that are codec really specifically to Realtek
codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>