- spec->hp_detect has to be overridden in HDA_FIXUP_ACT_PARSE, not in
PRE_PARSE.
- Remove err == 0 check but return directly -EINVAL from
stac92xx_parse_auto_config()
- Set spec->default_polarity for 92HD71bxx
- Some code shuffles
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>
* test/hda-gen-parser:
ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecs
ALSA: hda - force different capture controls if amp caps differ
ALSA: hda - do not add non-existing Mic boost controls
ALSA: hda - initialize channel counts correctly
ALSA: hda - fix wrong adc_idx in generic parser
ALSA: hda - Check array bounds in get_input_path
ALSA: hda - Add prefer_hp_amp flag to hda_gen_spec
ALSA: hda - fix OOPS in hda_mark_cmd_cache_dirty
ALSA: hda - Check pincap while parsing the configuration
Otherwise no PCM will be built for codecs without analog I/O.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise setting the capture volume for amps will be weird and
inconsistent (it will try to set values outside the range of the
second amp based on capabilities of the first amp).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the input node does not have any volume capable input amp,
don't add such a control.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even a single DAC can output two channels, so the channel count
is twice the number of DACs.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We use knew->index for adc_idx when we create "Capture Volume" and
"Capture Switch", so use the same to retrieve adc_idx.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c:387:19: sparse: symbol 'ca0132_voicefx' was not declared. Should it be static?
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag to indicate whether HP amp is turned on as default for
speaker or line-outs, and enable this for ALC260 codec, as many
machines with this codec require the HP amp even for speakers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c: In function ‘ca0132_effects_set’:
sound/pci/hda/patch_ca0132.c:3391:2: warning: too many arguments for
format [-Wformat-extra-args]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Spotted by smatch,
sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: potential
null dereference 'dma_engine'. (kzalloc returns null)
sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: we
previously assumed 'dma_engine' could be null (see line 1857)
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c:1781 dspxfr_one_seg() info: why not
propagate 'status' from dsp_dma_stop() instead of (-5)?
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent update of ca0132 driver replaced the pinctl setup to the
direct write via snd_hda_codec_write() again. This should be covered
by snd_hda_set_pin_ctl() to be safer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit c3b4eea262.
Since the recent firmware loader code supports caching at S3/S4 by
itself, we don't have to handle f/w caching in the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Handle a potential dma_engine alloc error and fix the possible use of an
uninitialized status variable in dspxfr_one_seg(). Also correct the initial
sampling rate for Mic 1.
Update the module description.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the controls used for tuning the DSP effects.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the unsolicited response handler for incoming DSP responses and
jack detection reporting, and routines for reading the incoming DSP response.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the playback PCM open callback.
PCM stream setup and cleanup functions are added for use by PCM callbacks.
Delay stream cleanup if effects are on, to allow time for any effects tail to
finish.
Add the analog capture PCM callbacks.
Change the max channels of analog playback to 6.
Add two new PCMs: AMic2 and What-U-Hear.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the kcontrols for the DSP effects, playback and recording
source selection.
ca0132_is_vnode_effective() checks whether virtual node settings have
taken effect.
The control change helpers ca0132_pe_switch_set(), ca0132_voicefx_set()
and ca0132_cvoice_switch_set() are added to toggle playback / capture
DSP effects, ca0132_voicefx_info(), _get() and _put() are added for
input path DSP effect value access. The volume helpers are updated to
volume_info(), _get() and _set() to use the virtual nodes.
The redundant headphone and speaker switches and ct_extension function
are removed.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the framework to set effect parameters: ca0132_effects_set()
and ca0132_setup_defaults() are general functions for parameter setting and
initializing to default values. dspio_set_param() and dspio_set_uint_param()
are lower-level fns to simplify setting individual DSP parameters via an
SCP buffer transfer to the firmware.
The CA0132 chip parameter init code is added in ca0132_init_params().
In chipio_[write,read]_data(), the current chip address is auto-incremented
if no error has occurred.
ca0132_select_out() selects the current output. If autodetect is enabled,
use headphones (if jack detected) or speakers (if no jack).
ca0132_select_mic() selects the current mic in. If autodetect is enabled,
use exterior mic (if jack detected) or built-in mic (if no jack).
Init digital mic and switch between dmic and amic with ca0132_init_dmic(),
ca0132_set_dmic(). amic2 is initialized in ca0132_init_analog_mic2().
Finally, add verb tables for configuring DSP firmware.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds definitions and structs used for configuring DSP effects,
virtual nodes, effect tuning controls, and mixer control helpers.
The effect structs are also initialized.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When "alsactl restore" is performed on HDMI codecs, it tries to
restore the channel map value since the channel map controls are
writable. But hdmi_chmap_ctl_put() returns -EBADFD when no PCM stream
is assigned yet, and this results in an error message from alsactl.
Although the error is harmless, it's certainly ugly and can be
regarded as a regression.
As a workaround, this patch changes the return code in such a case to
be zero for making others happy. (A slight excuse is: when the chmap
is changed through the proper alsa-lib API, the PCM status is checked
there anyway, so we don't have to be too strict in the kernel side.)
Cc: <stable@vger.kernel.org> [v3.7+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of checking the codec SSID in find_mute_led_cfg() for HP Mini
110, set the proper spec->default_polairty in the fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI vendor ID check in find_mute_led_cfg() is now superfluous
because the function is called in the fixup table entries of HP
machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally all codecs in patch_sigmatel.c have been converted to use the
standard fixup helpers. This change also includes trivial cleanups
like the call of common setup for GPIO LED or the removal of unused
function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This one is rather a simple conversion. The fixups for Dell machines
are implemented by fixup functions in the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time, the only intrusive changes are for HP machines.
As the mute LED fixup and the bass speaker switch are required only
for HP machines, we can move these checks into the fixup entries; the
former is applied generically to all HP machines while the latter for
only certain models.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes (or rather often) BIOS sets the pin default configurations
obviously wrongly. Looking through these failures, one common pattern
is to enable some dead pins that are usually marked as speaker pins.
In such a case, we can skip them if the pins don't have the output
capability.
In this patch, add a check for the valid pin cap bit for each parsed
pin, and filter out when it's invalid.
The fix was originally suggested by Raymond Yau.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This conversion is a bit tricky. Since STAC927x may take two
different volume-knob initialization values depending on the model, a
new flag, spec->volknob_init, is introduced to indicate whether it's
the standard volume-knob initialization or not.
Also, Dell BIOS model is now directly mapped onto the fixup table
instead of parsing in the function. This resulted in a new model ref,
STAC_927X_DELL_BIOS_SPDIF, which is a chained entry.
Also, for reducing the fixups, virtual entries like
STAC_927X_DELL_DMIC and STAC_D965_VERBS are introduced.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather straightforward conversion, except for ones for Intel Mac.
As Intel Mac have only unique codec SSIDs, we need to remap the fixup
again for the codec SSID and call the new fixup there.
Also, we can reduce model enums like STAC_MACMINI, which are model
aliases for backward compatibility, since they can be pointed directly
via hda_model_fixup table.
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>
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>
So far, Realtek codec driver evaluates the NIDs for volume and mute
controls twice, once while parsing the DACs and evaluating the
assignment, and another while creating the mixer elements. This is
utterly redundant and even fragile, as it's assuming that the ctl
element evaluation is identical between both parsing DACs and creating
mixer elements.
This patch simplifies the code flow by doing the volume / mute
controls evaluation only once while parsing the DACs. The patch ended
up in larger changes than expected because of some cleanups became
mandatory.
As a gratis bonus, this patch also fixes some cases where the stereo
channels are used wrongly for mono amps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mono widget is always connected to the left channel, thus the left
channel amp value also should be referred for mono widgets instead of
the right channel.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As we parse the output paths more precisely now, we can use this path
list for parsing the widgets for volume and mute mixer controls.
The spec->vol_ctls[] and sw_ctls[] bitmasks are replaced with the
ctls[] in each output path instance.
Interestingly, this move alone automagically fixes some bugs that the
conflicting volume or mute NIDs weren't properly detected.
Also, by parsing the whole path, there are more chances to get a free
widget for volume/mute controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the output path parser to Realtek codec driver as we already have
in patch_via.c. The nid_path struct represents the complete output
path from a DAC to a pin. The alc_spec contains an array of these
paths, and a new path is added at each time when a new DAC is
assigned.
So far, this path list is used only in limited codes: namely in this
patch, only alc_is_dac_already_used() checks the list instead of dac
arrays in all possible outputs. In the later development, the path
list will be referred from more places, such as the mixer control
assignment / check, the mute/unmute of active routes, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the probing phase, create a list of all available DACs in the codec
and use it for checking the single DAC connections.
This list will be used in more other places in the later commits, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the helper function snd_hda_get_conn_index() instead of open
codes. This also improves the detection of some routes to DAC on
ALC260 (although the difference doesn't influence on the end
results of the mapping).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for a new codec ALC284, which is compatible with
ALC269. Also add more codec variants to handle the SSID check
properly.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the power_save_controller option to bint from bool so that user
can override the runtime PM capability bit and force to enable or
disable the runtime PM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 697c373e34.
The original patch was meant to remove clicking, but in fact caused even
more clicking instead.
Thanks to c4pp4 for doing most of the work with this bug.
BugLink: https://bugs.launchpad.net/bugs/886975
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a few bug reports that the runtime D3 results in the dead
HD-audio controller. It seems that the problem is in a deeper level
than the sound driver itself, so as a temporal solution, disable the
feature for these controllers again.
Reported-and-tested-by: Vincent Blut <vincent.debian@free.fr>
Reported-and-tested-by: Maurizio Avogadro <mavoga@gmail.com>
Cc: <stable@vger.kernel.org> [v3.7]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems useful, and power states are required for AFG nodes,
so I see no reason not to print it. As a bonus, also print the
AFG nid.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mute LED is in this case connected to the Mic1 VREF.
The machine also exposes the following string in BIOS:
"HP_Mute_LED_0_A", so if more machines are coming, it probably
makes sense to try to do something more generic, like for the
IDT codec.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1096789
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The vmaster hook sends 1 for enabled/unmuted and 0 for disabled/muted,
but "Mute-LED Mode" being "On" refers to the LED being on, not the
volume being on.
Therefore "On" and "Off" should be switched.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Resuming a switcheroo'd HDA controller hangs since the completion
is one-shot (thus works the first time). Fix by using completions
that explictly need rearming, so remain fired before.
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Haswell HDMI codec pins may report invalid connection list entries, which
will cause failure to play audio via HDMI or Display Port.
So this patch adds fixup for Haswell to workaround this hardware issue:
enable DP1.2 mode and override the pins' connection list entries with proper
value.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Xingchao Wang <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The workaround to force VREF50 for dallas/hp model with ALC861VD
was introduced in commit 8fdcb6fe42,
but it contained wrong pincap override bits.
This patch fixes to exclude VREF80 pincap bit correctly.
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that Realtek codecs (ALC260, etc) with input amps in
audio-input widgets don't handle the multiple individual input amps.
Thus we need to set codec->single_adc_amp flag for them in general.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've seen the broken HDMI *video* output on some machines with GM965,
and the debugging session pointed that the culprit is the disabled
audio output pins. Toggling these pins dynamically on demand caused
flickering of HDMI TV.
This patch changes the behavior to keep the pin ON constantly.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=51421
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the quirk entry for HP Pavilion dv7 in order to make the bass
speaker working.
Reported-and-tested-by: Tomas Pospisek <tpo2@sourcepole.ch>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The runtime_idle callback is the right place to check the suspend
capability, but currently we do it wrongly in the runtime_suspend
callback. This leads to a kernel error message like:
pci_pm_runtime_suspend(): azx_runtime_suspend+0x0/0x50 [snd_hda_intel] returns -11
and the runtime PM core would even repeat the attempts.
Reported-and-tested-by: Borislav Petkov <bp@alien8.de>
Cc: <stable@vger.kernel.org> [v3.7]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer Aspire One 522 has the infamous digital mic unit that needs the
phase inversion fixup for stereo.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=715737
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HD-audio driver artificially calls the suspend and the resume code
path in the VGA switcheroo state changes. When a machine goes to
suspend, it tries to suspend the device again, and it stalls at
snd_power_wait().
This patch adds checks whether the devices were already in (forced)
suspend in PM callbacks for avoiding the doubly suspend.
Reported-by: Daniel J Blueman <daniel@quora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the HD-audio controller is disabled (e.g. via vga switcheroo) but
the driver is still accessing it, it spews floods of "spurious
response" kernel messages. It's because CORB/RIRB WP reads 0xff, and
the driver tries to fill up until this number.
This patch changes the CORB/RIRB WP reads to word instead of byte, and
add the check of the read value. If it's 0xffff, the controller is
supposed to be disabled, so the further action will be skipped.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reducing the time on HDA link reset can help to reduce the driver loading
time. So we replace msleep with usleep_range to get more accurate time
control and change the value to a smaller one. And a 100ms timeout is set
for both entering and exiting the link reset.
Signed-off-by: Xingchao Wang <xingchao.wang@intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for new codec VT1808, which is similiar with VT1705CF.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for new codec VT1705CF.
When power on/off Audio output converter of VT1705CF, the stream tag
will be cleared. But driver caches the value. So when power on Audio
output converter, the update_conv_power_state() will restore the saved
stream tag of it.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the leftover __devinit* in comments.
They have been commented out because they couldn't fit with __dev*
although they should have matched.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since __devinit* have been removed completely, DELAYED_INIT*_MARK in
hda_intel.c became NOP. Let's rip them off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_HOTPLUG is going away as an option. As result the __dev*
markings will be going away.
Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.
Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When printing, use a prefix of the PCI domain, bus, device and function
as in other drivers, to differentiate multiple devices.
Important for reporting and debugging. A future step is to tidy this up with
dev_printk et al.
v2: Move conversion specifier into call site, preventing build issues
v3: Refactor for Takashi's for-next branch
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for vga switcheroo race in commit 128960a9 opened yet
another race. At the time the audio driver starts probing, user may
turn off D-GPU off. But at this moment, the audio driver still
doesn't register the vga switcheroo client, thus the switching isn't
notified. Then the hardware gets off out of sudden, resulting in
invalid reads and lots of "spurious response" error messages.
For solving this situation, the following changes have been done in
this patch:
- Move again vga switcheroo registration to the very early stage of
the probing; this also requires to set pci drvdata properly before
registration
- Introduce the completion to synchronize the driver probe at vga
switcheroo callbacks; this assures that the whole probing finished
before executing the callbacks
Reported-by: Daniel J Blueman <daniel@quora.org>
Tested-by: Daniel J Blueman <daniel@quora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current Realtek driver reconfigures the max PCM channels
dynamically according to the value of Channel Mode enum if the
multi-io retasking is available. It works fine for multi-io pins.
But when multiple speaker pins are available, the channels of speakers
also have to obey to the channel mode, which isn't nice.
(That is, when you select "2ch" in Channel Mode so that the line-in
and mic jack behave as input, you can't play surrounds properly from
the built-in speaker.)
This patch fixes the problem by taking the channel number for multiple
speakers into account in the channel-mode setup code.
Also it fixes the wrongly set up max_channels value in the case of
multi-io extension.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Very quiet release for ASoC really:
- Standardisation of the logging.
- DT and dmaengine support for Atmel.
- Support for Wolfson ADSP cores.
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090.
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Merge tag 'asoc-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.8
Very quiet release for ASoC really:
- Standardisation of the logging.
- DT and dmaengine support for Atmel.
- Support for Wolfson ADSP cores.
- New drivers for Freescale/iVeia P1022 and Maxim MAX98090.
The same type of code is being used in multiple places in various
codec drivers, so put it as a core library.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for introducing a protection to access
races of snd_array instances. Call snd_array_init() appropriately
at the initialization time and don't call it twice.
Also the allocations of codec-spec structs are cleaned up by helper
functions in patch_sigmatel.c and patch_analog.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tidy up and condense chipio_write_address|addx() functions.
Improve dspio_write_wait() to use jiffies for timeout calc.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add comments and descriptions to functions.
Bump dsp_free_ports() to below dsp_allocate_ports_format() to group the
alloc functions together for commenting.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the code needed to fetch the DSP binary image from the local
firmware install location and transfer it over to the chip using the new DSP
loader bus ops.
Actual DSP effect controls, parameters and mixers are to be included later.
- Add calls to new DSP loader system to transfer firmware to the hardware.
- Add chip read/write routines, DSP I/O, SCP packet format helper functions and
transfer DMA management.
- Add guard around DSP download to ensure loader config switch is enabled.
The general scheme for downloading the DSP is as follows:
1) If DSP firmware loader is enabled, ca0132_download_dsp() is called
to start the process.
2) Driver requests DSP image using request_firmware().
3) Driver sets up the streaming DMA for DSP image download
with dspload_image() and dspxfr_image(), which in turn calls the DSP loader
op snd_hda_codec_load_dsp_prepare() to ready the system.
4) DSP image will consist of 1 or more segments, each transferred in sequence
by a call to dspxfr_one_seg() and snd_hda_codec_load_dsp_trigger().
5) Once complete, the loader state is cleaned up with
snd_hda_codec_load_dsp_cleanup().
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pass DMA buffer pointers in calls to setup_bdle().
Add DSP loader callback routines to controller.
Add new DSP loader switch to Kconfig to turn off DSP firmware.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot this again... codec->in_pm is in #ifdef CONFIG_PM
Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal mic on MBP81 gives only the right channel, and the left
channel is static. Add a verb to fix the ADC2 channel mode to expand
mono right to stereo.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50781
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We forgot to apply the fixup verbs in cs_init().
But adding the fixup verbs will break mbp101 fixup that has been fixed
recently again, since the mbp101 fixup contains the wrong verbs to
override. So these bogus verbs must be removed, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new firmware code tries to re-read the formerly read firmware
files before suspend. Thus it's wiser to keep the "patch" firmware in
the driver for avoiding this unnecessary re-reading.
Of course, this will consume a bit of memory for unused stuff, but
the patch fw is supposed to be fairly small, so it's more benefit in
the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we keep the pin default config values anyway internally, we
don't have to set the values in the codec. This patch removes the
code writing the pincfg values.
As a gratis bonus, we can remove also the code restoring the original
pincfg values at PM resume or module free. This will give us more
benefit, as it can reduce the unnecessary power-up of codecs.
This won't change the driver functionality. The only difference would
be that the codec proc file will show the original pincfg values
instead of the actually referred values. The actually referred values
can be determined from sysfs *_pin_configs files.
(Also hda-emu was updated to follow this change.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The free callback is called at the state where no extra verbs are
executed, thus calling *_shutup() is useless, as it's checking the
shutdown flag. Remove such superfluous calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If this array is not cleared, the jack related code later might
fail to create "Internal Speaker Phantom Jack" on Dell Inspiron 3420 and
Dell Vostro 2420.
BugLink: https://bugs.launchpad.net/bugs/1076840
Cc: stable@vger.kernel.org (3.6+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We found a new codec ID 292, and that just a simple quirk would enable
sound output/input on this ALC292 chip.
BugLink: https://bugs.launchpad.net/bugs/1081466
Cc: stable@vger.kernel.org
Tested-by: Acelan Kao <acelan.kao@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a report that the runtime PM may make the codec the
unresponsive on AMD platforms. Since the feature has been tested only
on the recent Intel platforms, it's safer to limit the support to such
devices for now.
This patch adds a new DCAPS bit flag indicating the runtime PM
support, and mark it for Intel controllers.
Reported-and-tested-by: Julian Wollrath <jwollrath@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Acer Aspire AO756 has an analog built-in mic on nid 0x1b and an
external mic on nid 0x19. The BIOS doesn't set this up.
The mic detect on this Acer Aspire netbook and Acer C7 ChromeBook is
only valid when the headphone is plugged. The detect circuit relies on
the tip detect switch being closed on the jack. Tell hda_jack to ignore
the mic sense unless the headphones are plugged.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the concept of a "gated" jack. The gated jack's pin sense
is
only valid when the "gating" jack is plugged. This requires checking
the gating jack when the gated jack changes and re-checking the gated
jack when the gating jack is plugged/unplugged.
This allows handling of devices where the mic jack detect floats when
the headphone jack is unplugged.
[Rewritten for fixing the possible snd_array reallocation, covering
the missing callback calls and jack sync operations, as well as some
code cleanups -- tiwai]
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the bus reset is performed during the suspend/resume (including
the power-saving too), it calls snd_hda_suspend() and
snd_hda_resume() again, and deadlocks eventually.
For avoiding the recursive call, add a new flag indicating that the PM
is being performed, and don't go to the bus reset mode when it's on.
Reported-and-tested-by: Julian Wollrath <jwollrath@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is another variant of iMac 9,1 with a different codec SSID.
Reported-and-tested-by: Everaldo Canuto <everaldo.canuto@gmail.com>
Cc: <stable@vger.kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A closer look shows that the name is not even used and can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When 2.1 speakers are detected, use the corresponding channel map
instead of the standard map with front+rear surrounds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When two built-in speakers are found on the machine, we can suppose
it's rather a 2.1 speaker system with a bass output instead of
front/surround channels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc269_toggle_power_output() was only use in ALC269VB. I rename it to
alc269vb_toggle_power_output().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802
codec, correct the default configurations of speaker pins 0x24 and
0x33.
Reported-by: Massimo Del Fedele <max@veneto.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1802 codec provides the invalid connection lists of NID 0x24 and
0x33 containing the routes to a non-exist widget 0x3e. This confuses
the auto-parser. Fix it up in the driver by overriding these
connections.
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at
the point of the current line-out (i). When no valid path is found
for this output, this results in dac = 0, thus it creates a hole in
dac_nids[]. This confuses is_empty_dac() and trims the detected DAC
in later reference.
This patch fixes the bug by appending DAC properly to dac_nids[] in
via_auto_fill_adc_nids().
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some of the PantherPoint HDMI machines we currently enable, we're seeing
trouble with unsol events, i e detecting monitor presence, especially when
on battery and after suspend/resume.
BugLink: https://bugs.launchpad.net/bugs/1075882
Tested-by: Cyrus Lien <cyrus.lien@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Several bug reports suggest that the forcibly resetting IEC958 status
bits is required for AD codecs to get the SPDIF output working
properly after changing streams.
Original fix credit to Javeed Shaikh.
BugLink: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/359361
Reported-by: Robin Kreis <r.kreis@uni-bremen.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correctly enable the digital microphones with the right bits in the
right coeffecient registers on Cirrus CS4206/7 codecs. It also
prevents misconfiguring ADC1/2.
This fixes the digital mic on the Macbook Pro 10,1/Retina.
Based-on-patch-by: Alexander Stein <alexander.stein@systec-electronic.com>
Signed-off-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The audio chipset used in Teradici's Tera2 host cards is the same as that in
the 1200 host cards. This patch allows ALSA to recognize the Tera2 cards.
Signed-off-by: Lars R. Damerow <lars@pixar.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Starting audio or seeking in various music players causes
setup_dig_out_stream() to be called, which resets the SPDIF stream,
which caused one DAC (but not another) to make a clicking noise every
time.
This patch ensures the reset only happens when it needs to, which is
when the format changes, and makes the code a little more readable.
Signed-off-by: Laurence Darby <ldarby@tuffmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The BIOS on HP dv5 doesn't have the DMI string to guide the setup of
mute led GPIO and polarity. Associate this laptop with the hp-inv-led
model.
Signed-off-by: Gustavo Maciel Dias Vieira <gustavo@sagui.org>
Tested-by: Vinícius Angiolucci <angiolucci@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.
Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reuse code from clocksource to handle wall clock counter.
Since wrapparound occurs, the audio timestamp is reinitialized
to zero on a trigger. Synchronized linked devices will
start counting from same reference to avoid any drift.
Max buffer time is limited to 178 seconds to make sure
wall clock counter does not overflow
Wallclock timestamps are disabled on capture streams
until we figure out how to handle digital inputs.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variable stream is initialized but never used
otherwise, so remove the unused variable.
dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By some reason, Toshiba laptop doesn't like the EAPD turned up for the
headphone pin. Add a fix up code to force to turn down EAPD for NID
0x15.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=569991
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I have a Lenovo ThinkPad T430 and an UltraBase Series 3 docking
station.
Without this patch, if I plug my headphones into the jack on the
computer, everything works fine. The computer speakers mute and the
audio is played in the headphones. However, if I plug into the docking
station headphone jack the computer speakers are muted but there is no
audio in the headphones.
Addresses https://bugs.launchpad.net/bugs/1060372
Signed-off-by: Joseph Salisbury <joseph.salisbury@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will greatly reduce the frequency for detection errors in
those cases where the hardware is "flaky", i e, rapidly changing
between plugged and unplugged states even without user interaction.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even when CONFIG_SND_DEBUG is not enabled, we don't want to
return an arbitrary memory location when the channel count is
larger than we expected.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When both an SPDIF and an HDMI device are created on the same card
instance, multiple IEC958 controls are created with indices=0, 1, ...
But the alsa-lib configuration can't know which index corresponds
actually to which PCM device, and both the SPDIF and the HDMI
configurations point to the first IEC958 control wrongly.
This patch introduces a (hackish and ugly) workaround: the IEC958
controls for the SPDIF device are re-labeled with device=1 when HDMI
coexists. The device=1 corresponds to the actual PCM device for
SPDIF, so it's anyway a better representation. In future, HDMI
controls should be moved with the corresponding PCM device number,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If LPIB reports a pretty bad value, we can't trust such hardware for
calculating the PCM delay. Automatically turn off the delay counting
when such a problem is encountered.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=48911
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have a generic unsol mechanism, we can implement a generic
poll loop, which can be used for debugging, or if a codec's unsol
mechanism is broken.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec,
this patch modify patch_conexant.c to add Lemote specific code.
Both A1004 and A1205 use the same pin configurations, but A1004 need
to increase the default boost of internal mic.
Signed-off-by: Jie Chen <chenj@lemote.com>
Signed-off-by: Huacai Chen <chenhc@lemote.com>
Signed-off-by: Hongliang Tao <taohl@lemote.com>
Signed-off-by: Hua Yan <yanh@lemote.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Delay the registration of VGA switcheroo client to the end of the
probing. Otherwise a too quick switching may result in Oops during
probing.
Also add the check of the return value from snd_hda_lock_devices().
Reported-and-tested-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The white-list entries of position_fix for ASUS laptops have been
added just as a workaround for broken COMBO mode. Now the combo mode
itself is disabled, we can safely remove these entries.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This caused the internal speaker to mute itself because it was
present, which happened after powersave.
It was found on Dell XPS 15 (L502x), ALC665.
Reported-by: Da Fox <da.fox.mail@gmail.com>
Cc: stable@vger.kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that the COMBO position fix mode is rather more harmful,
and it got reverted (with the replacement of runtime->delay
calculation) recently. Hence we can get rid of AZX_DCAPS_POSFIX_COMBO
as well.
It's still possible to pass this mode via position_fix module option,
in case where this really helps on weird machines (who knows).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the transition to the generic fixup code, the call of
snd_hda_gen_init() and snd_hda_gen_free() was missing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [4b527b65 ALSA: hda - limit internal mic boost for Asus
X202E] introduced the use of auto-parser code, but it forgot to add
struct hda_gen_spec at the head of codec->spec which the auto-parser
assumes silently. Without this record, it may result in memory
corruption.
This patch adds the missing piece.
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell Latitude 5x30 and 6x30 series of machines all have
a single 4-pin headset jack. Enabling line in mode for such jack
is very confusing (you would only get mono input, and would have to
use non-standard adapters), so remove the option by default.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From what I can conclude all GPIO handling was removed in 2009.
Remove dead code remnants.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case there is one "Headphone Jack" and one "Dock Headphone Jack",
one of them will get an index, even though that is not needed.
This patch fixes that issue.
BugLink: https://bugs.launchpad.net/bugs/1060729
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the recent patch "ALSA: hda - use both input paths on Conexant
auto parser" suddenly we can have more than one "Mic Boost", this
happened on Acer Aspire One 722. Therefore we must add the possibility
to put an index on this "Mic Boost" just as we do for the other
"Mic Boost" earlier in the same function.
BugLink: https://bugs.launchpad.net/bugs/1059523
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There was a race condition when the system suspends while hda_power_work
is running in the work queue. If system suspend (snd_hda_suspend)
happens after the work queue releases power_lock but before it calls
hda_call_codec_suspend, codec_suspend runs with power_on=0, causing the
codec to power up for register reads, and hanging when it calls
cancel_delayed_work_sync from the running work queue.
The call chain from the work queue will look like this:
hda_power_work <<- power_on = 1, unlock, then power_on cleard by suspend
hda_call_codec_suspend
hda_set_power_state
snd_hda_codec_read
codec_exec_verb
snd_hda_power_up
snd_hda_power_save
__snd_hda_power_up
cancel_delayed_work_sync <<-- cancelling executing wq
Fix this by waiting for the work queue to finish before starting suspend
if suspend is not happening on the work queue.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define()
always after fixup"), alc_auto_parse_customize_define was moved after
detection of ALC271X.
The problem is that detection of ALC271X relies on spec->cdefine.platform_type,
and it's set on alc_auto_parse_customize_define.
Move the alc_auto_parse_customize_define and its required fixup setup
before the block doing the ALC271X and other codec setup.
BugLink: https://bugs.launchpad.net/bugs/1006690
Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo IdeaPad U310 has an internal mic where the right channel
is phase inverted.
Signed-off-by: Felix Kaechele <felix@fetzig.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For less duplication of code between codecs, and to make it easier
in the future to improve code for all codecs simultaneously.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moving towards less duplication of code between codecs - this patch
takes some of the common code of unsol event handling and makes it
generic.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.7
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
Tested with LPIB delay without any issues.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DMA Position in Buffer (DPIB) should be used for
ring buffer management, while LPIB register provides
information on the number of samples transfered on
the link. The difference between the two pieces of
information corresponds to hardware/DMA buffering.
This patch reports this difference in runtime->delay, and
removes the use of the COMBO mode on recent Intel hardware.
Credits to Takashi Iwai for an initial patch.
[rebased to for-next branch and replaced snd_printk() with
snd_printdd() by tiwai]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SSYNC bits are typically used to start multiple
streams synchronously. It makes sense to use them
for a single stream for a more predictable startup
sequence. The transfers only start once the DMA and
FIFOs are ready. This results in a better correlation
between timestamps and number of samples played.
Credits to Kar Leong Wang for suggesting this
improvement.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If headphone jack can't detect plug presence, and we have the jack in
the jack table, snd_hda_jack_detect will return the plug as always
present (as it'll be considered as a phantom jack). The problem is that
when this happens, line out pins will always be disabled, resulting in
no sound if there are no headphones connected.
This was reported as a no sound problem after suspend on
http://bugs.launchpad.net/bugs/1052499, since the bug doesn't manifests
on first initialization before the phantom jack is added, but on resume
we reexecute the initialization code, and via_hp_automute starts
reporting HP always present with the jack now on the table.
BugLink: https://bugs.launchpad.net/bugs/1052499
Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
Cc: <stable@vger.kernel.org> [v3.6+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Thinkpad W520 - and probably several other machines with
Conexant 506x chips - the Dock Mic and Mic are connected to the
same two selector nodes. This patch will make Dock Mic take one
selector node and Mic take the other, when possible.
Without the patch, both paths would take the first selector,
leading to the normal Mic's volume being controlled by
"Dock Mic Boost".
(On other machines, this could instead fixup similar problems between
Mic and Line In, for example.)
BugLink: https://bugs.launchpad.net/bugs/1037642
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard name (and what PulseAudio picks up) is "Dock Mic",
not "Docking Mic" or "Docking-Station".
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the input gain for the internal mic is set to its maximum level,
the background noise becomes so high - and any relevant signal clipped -
that the setting becomes unusable. It is better to limit the amplification.
BugLink: https://bugs.launchpad.net/bugs/1052460
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of new HP laptops have a LED for microphone (or recording) mute,
and it's controlled by GPIO pin 3.
Bind this with the capture switch to turn it on/off properly by the
mixer change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the default value of position_fix -1, and allow user passing
position_fix=0 explicitly to set the "auto" position-fix mode.
Otherwise the auto mode may be switched to others like COMBO of
VIACOMBO when the controller prefers it, thus user can't set the auto
mode any longer.
Also updated the documentation appropriately, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the definitions of CS420X_IMAC27_122 and CS420X_APPLE as
aliases, the rest enums are set to duplicated values unexpectedly.
Move the alias definitions at the end so that the enum values are
defined in the proper order.
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MacBook Pro 10,1 needs a few adjustments to make it working:
- more COEF verbs
- some pin config overrides to disable the unused pin (0x0d, 0x12),
and fix the internal mic (0x0e)
In addition, it uses GPIO 1 and 3 like other MacBooks.
The internal digital mic on the machine is still problematic: it seems
that only the right channel is used and the left is always static.
This looks like a hardware design, so we need to cope in the software
side somehow...
The primary information and test were brought from Daniel J Blueman.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_pick_fixup() didn't check the case where the device mask bits
are set, typically used for SND_PCI_QUIRK_VENDOR() entries. Fix this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally the bogus period at BDL head was introduced as a workaround
for the mismatching position update at the period boundary, typically
seen on dmix. However, for applications like PulseAudio that don't
require period wake ups, this workaround is just superfluous. Thus
better to disable it when no_period_wakeup is given in hw_params.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit c20c5a841c changed some chipsets to
default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of
POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including
repeated sounds on my Asus laptop.
My hardware is Cougar Point which the commit log of
c20c5a841c mentions as tested so POS_FIX_COMBO
probably works in general but apparently it doesn't on Asus K53E therefore the
need for the quirk.
Signed-off-by: Catalin Iacob <iacobcatalin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_codec_reset() calls restore_pincfgs() where the codec is
powered up again, which eventually tries to resume and initialize via
the callbacks of the codec. However, it's the place just after codec
free callback, thus no codec callbacks should be called after that.
On a codec like CS4206, it results in Oops due to the access in init
callback.
This patch fixes the issue by clearing the codec callbacks properly
after freeing codec.
Reported-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The purpose of this flag is unclear. If the problem is that some machines
have broken misc/NO_PRESENCE bits, they should be fixed by pin fixups.
In addition, this causes jack detection functionality to be flawed on
the M31EI, where there are two jacks without jack detection (which is
properly marked as NO_PRESENCE), but due to ignore_misc_bit, these
jacks are instead being reported as being present but always unplugged.
BugLink: https://bugs.launchpad.net/bugs/939161
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... instead of the standard fixed channel maps.
The generic HDMI is based on the audio infoframe, and its configuration
can be selected via CA bits. Thus we need a translation between the
CA index and the verbose channel map list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although HD-audio allows pair-wise channel configurations, only the
fixed channel positions are used in this version. In future, this can
be changed and allow user to modify the channel positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently the check for non-PCM stream state was added to the generic
HDMI driver code. But this check should be done rather to each pin
instead of each converter. Otherwise when a different converter is
assigned at the next open, the audio infoframe can be inconsistent
with the setup using the previous converter.
For fixing this issue, this patch moves the state of the current
non-PCM status from per_cvt to per_pin. (In addition an unused
argument cvt_nid is stripped from hdmi_setup_channel_mapping())
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For multiple speaker outs, the names were previously
"Speaker,0", "Speaker,1", "Center"/"LFE", "Speaker,3". This is
inconsistent, confusing, and is not picked up correctly by PulseAudio.
Instead use "Front", "Surround", "Center"/"LFE", "Side" which
is more standard.
BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1046734
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the commit [2faa3bf: ALSA: hda - Rewrite the mute-LED hook with
vmaster hook in patch_sigmatel.c], the former Master volume control
was converted to PCM. This was supposed to be covered by the vmaster
control. But due to the lack of "PCM" slave definition, this didn't
happen properly. The patch fixes the missing entry.
Reported-by: Andrew Shadura <bugzilla@tut.by>
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For HBR stream test, use straight channel mapping way.
when switched back to "speaker-test -c8", even the audio
infoframe is up-to-date, there should be correct channel mapping setup.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDMI channel remapping apparently effects HBR packets on Intel's chips.
For compressed non-PCM audio, use "straight-through" channel mapping.
For uncompressed multi-channel pcm audio, use normal channel mapping.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The array channel_allocations[] is an ordered list, add function to get
correct order by ca_index.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of pm_notify callback in snd_hda_codec_free() should be with
the check of the current state whether pm_notify(false) is called or
not, instead of codec->power_on check.
For improving the code readability and fixing this inconsistency,
codec->d3_stop_clk_ok is renamed to codec->pm_down_notified, and this
flag is set only when runtime PM down is called. The new name reflects
to a more direct purpose of the flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_HDA_POWER_SAVE is no longer an experimental feature and its
behavior can be well controlled via the default value and module
parameter. Let's just replace it with the standard CONFIG_PM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a codec provides its own set_power_state op, the D3-clock-stop
isn't checked correctly. And the recent changes for repeating the
state-setting operation isn't applied to such a codec, too.
This patch fixes these issues by moving the call of codec's own op to
the place where the generic power-set operation is done, and move the
power-state synchronization code out of
snd_hda_set_power_state_to_all() so that it can be called always at
the end of power-up/down sequence, and updates the D3 clock-stop flag
properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the HD-audio is removed, it leaves the refcounts when codecs are
powered up (usually yes) in the destructor. For fixing the unbalance,
and cleaning up the code mess, this patch changes the following:
- change pm_notify callback to take the explicit power on/off state,
- check of D3 stop-clock and keep_link_on flags is moved to the caller
side,
- call pm_notify callback in snd_hda_codec_new() and snd_hda_codec_free()
so that the refcounts are proprely updated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These codecs seem reporting EPSS but require longer delay for the
proper D3 transition. For example, D3_STOP_CLOCK_OK bit won't be set
correctly even after D3.
In this patch, codec->epss flag is overridden for avoid the
misbehavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
EPSS parameter should be static, so we can read it once and remember.
This also allows more easily to override the wrong EPSS capability
reported from a codec by changing the flag in the codec
initialization step.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use unsigned int to make clear that the codes required only for
modules will be reduced by the compiler optimization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch to support runtime PM introduced a bug:
Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk'
'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But
there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE.
This patch is to fix the bug.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Runtime PM can bring more power saving:
- When the controller is suspended, its parent device will also have a chance
to suspend.
- PCI subsystem can choose the lowest power state the controller can signal
wake up from. This state can be D3cold on platforms with ACPI PM support.
And runtime PM can provide a gerneral sysfs interface for a system policy
manager.
Runtime PM support is based on current HDA power saving implementation. The user
can enable runtime PM on platfroms that provide acceptable latency on transition
from D3 to D0.
Details:
- When both power saving and runtime PM are enabled:
-- If a codec supports 'stop-clock' in D3, it will request suspending the
controller after it enters D3 and request resuming the controller before
back to D0. Thus the controller will be suspended only when all codecs are
suspended and support stop-clock in D3.
-- User IO operations and HW wakeup signal can resume the controller back to
D0.
- If runtime PM is disabled, power saving just works as before.
- If power saving is disabled, the controller won't be suspended because the
power usage counter can never be 0.
More about 'stop-clock' feature:
If a codec can support targeted pass-through operations in D3 state when there
is no BCLK present on the link, it will set CLKSTOP flag in the supported power
states and report PS-ClkStopOk when entering D3 state. Please refer to HDA spec
section 7.3.3.10 Power state and 7.3.4.12 Supported Power State.
[Fixed CONFIG_PM_RUNTIME dependency in hda_intel.c by tiwai]
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling the jack sync in the init callback of each codec,
call it generically at initialization and resume. By calling it at
the last of resume sequence, a possible race between the jack sync and
the unsol event enablement in the current code will be closed, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes an issue with a machine where there were no speakers,
but GPIO0 had to be data=1 for the headphone to be functioning.
I'm not sure if we need a more advanced patch to solve all possible cases,
but if so, this patch would still provide a minor optimisation.
BugLink: https://bugs.launchpad.net/bugs/1040077
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the codec turn-on operation is canceled by the immediate
power-on, the driver left the power_transition flag as is.
This caused the persistent avoidance of power-save behavior.
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai