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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=2irt
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.12 (GNU/Linux)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=JSSt
-----END PGP SIGNATURE-----
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>
... by calling the newly introduced snd_hda_power_sync().
I had to reimplement a wheel for adding the trigger at changing the
parameter -- the parameter set ops is overwritten to pass the integer
parameter, then trigger the power-state sync.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new helper function snd_hda_power_sync() to trigger the
power-saving manually. It's an inline function call to
snd_hda_power_save() helper function.
Together with this addition, snd_hda_power_up*() and
snd_hda_power_down() functions are inlined to a call of the same
snd_hda_power_save() helper function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's possible that these amps are settable somehow, e g through
secret codec verbs, but for now, don't create the controls (as
they won't be working anyway, and cause errors in amixer).
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1038651
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of blindly initializing a volume knob widget, first check
that there actually is a volume knob widget.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a simple support of automute for the front HP jack to AD1882
stack model. Such an addition is basically an exception -- we really
want to avoid the static quirk codes, but AD1882 parser isn't still
ready for moving to the BIOS auto-parser yet. So, as a quick fix, I
merged it for now.
In near future, we really need the big clean up of patch_analog.c to
move on to the auto-parser...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace with a proper ifdef check of CONFIG_PM_SLEEP in hda_intel.c.
But other places in HD-audio driver are still marked with CONFIG_PM,
since these can be called for power-saving even without
CONFIG_PM_SLEEP.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Conexant devices (e g CX20590) have no mute capability on
their Beep widgets.
This patch makes sure we don't try setting mutes on those widgets.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As spec said, 1 indicates no copyright is asserted.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return value of snd_hda_param_read() is -1 for an error, otherwise
it's the supported power states of a codec.
The supported power states is a 32-bit value. Bit 31 will be set to 1
if the codec supports EPSS, thus making "sup" negative. And the bit
28:5 is reserved as "0".
So a negative value other than -1 shall be further checked.
Please refer to High-Definition spec 7.3.4.12 "Supported Power
States", thanks!
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When "Beep Playback Switch" had a different value on left and right
channels (such as muting left but not right, or vice versa), this
could result in the right channel being ignored.
This patch enables beep to be sounding from right channel only, and
also give correct result back to userspace (e g amixer).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH
Signed-off-by: James Ralston <james.d.ralston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For processing the firmware handling properly for built-in kernels,
implement an asynchronous firmware loading with
request_firmware_nowait(). This means that the codec probing is
deferred when the patch option is specified.
Tested-by: Thierry Reding <thierry.reding@avionic-design.de>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary work for the deferred probing for
request_firmware() errors at init.
This patch moves the call of request_firmware() to hda_intel.c, and
call it in the earlier stage of probing rather than
azx_probe_continue().
Tested-by: Thierry Reding <thierry.reding@avionic-design.de>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To turn off pin control for the pin was tested, and helped against
this issue.
BugLink: https://bugs.launchpad.net/bugs/1034779
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CA0132 driver had some codes to handle the S/PDIF I/O, but the actual
setups of pins and converters were missing. Now the pins are added.
Also, fixed a few points triggering invalid codec verbs and mixer
elements since the digital I/O audio widgets on CA0132 have no amp.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now with the workaround using codec->pcm_format_first flag, we can
clean up the home-baked codes in patch_ca0132.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduced a new flag to set up the PCM stream format at first before
the stream_id and channel tag. Some codecs (e.g. CA0132) seem
preferring this over stream_id -> format order.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The same ID is twice in the quirk table, so the second one is not used.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When CONFIG_SND_VERBOSE_PRINTK=y is set, the debug print in
hda_auto_parser.c looks really ugly like:
ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0
ALSA sound/pci/hda/hda_auto_parser.c:334 dig-out=0x12/0x0
ALSA sound/pci/hda/hda_auto_parser.c:335 inputs:
ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x11ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0x10
ALSA sound/pci/hda/hda_auto_parser.c:341
ALSA sound/pci/hda/hda_auto_parser.c:343 dig-in=0x13
Better to put one item at each line.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the auto model is the default, these quirks are redundant
and can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This computer is confirmed working with model=auto on kernel 3.2.
Also, parsing fails with hda-emu with the current model.
Cc: stable@kernel.org (3.2+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As with the ThinkPad Models X230 Tablet and T530 the X230 needs a qurik to
correctly set up the pins for the dock port.
Signed-off-by: Felix Kaechele <felix@fetzig.org>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit c4bfe94a causes a regression on some codecs at probing.
Since this was just a workaround to shut up a kernel warning, it'd be
better to revert and fix properly. So we ended up with re-adding the
cleanup callback.
Tested-and-reported-by: Matt Horan <matt@matthoran.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a model/fixup string "lenovo-dock", for Thinkpad T430s, to allow
sound in docking station.
Tested on Lenovo T430s with ThinkPad Mini Dock Plus Series 3
Cc: stable@kernel.org
Signed-off-by: Philipp A. Mohrenweiser <phiamo@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo Thinkpad T530 with ALC269VC codec has a dock port but BIOS
doesn't set up the pins properly. Enable the pins as well as on
Thinkpad X230 Tablet.
Reported-and-tested-by: Mario <anyc@hadiko.de>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The IDT codecs initializes the GPIO setup for mute LEDs via
snd_hda_sync_vmaster_hook(). This works in most cases except for the
very first call, which is called before PCM and control creations.
Thus before Master switch is set manually via alsactl, the mute LED
may show the wrong state, depending on the polarity.
Now it's fixed by calling the LED-status update function manually when
no vmaster is set yet.
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit a3e199732b made the LED working again on HP Mini 210 but
with a wrong polarity. This patch fixes the polarity for this
machine, and also introduce a new model string "hp-inv-led".
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=772923
Cc: <stable@vger.kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On recent kernels, Realtek codec parser tries to optimize the routing
aggressively and take the headphone output as primary at first. This
caused a regression on VAIO Z with ALC889, the silent output from the
speaker.
The problem seems that the speaker pin must be connected to the first
DAC (0x02) on this machine by some reason although the codec itself
advertises the flexible routing with any DACs.
This patch adds a fix-up for choosing the speaker pin as the primary
so that the right DAC is assigned on this device.
Reported-and-tested-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix to converter detaching timing in patch_hdmi.c
leads to a kernel WARNING due to a sanity check when the debug
option is set. Add a workaround by setting a dummy hinfo->nid.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The generic HDMI code detaches the converter from the stream when
unused, but it must be done rather in the close callback instead of
the cleanup callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on HP Mini 210 doesn't provide the proper "HP_Mute_LED" DMI
string, thus the driver doesn't initialize the GPIO, too. In the
earlier kernel, the driver falls back to GPIO1, but since 3.3 we've
stopped this due to other wrongly advertised machines.
For fixing this particular case, add a new model type to specify the
default polarity explicitly so that the fallback to GPIO1 is handled.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=772923
Cc: <stable@vger.kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a bug report about the silent output from the headphone on a
mobo with VT2021, and spotted out that this was because of the wrong
D3 state on the DAC for the headphone output. The bug is triggered by
the incomplete check for this DAC in set_widgets_power_state_vt1718S().
It checks only the connectivity of the primary output (0x27) but
doesn't consider the path from the headphone pin (0x28).
Now this patch fixes the problem by checking both pins for DAC 0x0b.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also add a model/fixup string "lenovo-dock", so that other Thinkpad
users will be able to test this fixup easily, to see if it enables
dock I/O for them as well.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1026953
Tested-by: John McCarron <john.mccarron@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Turn on the pin widget's PIN_OUT bit from playback prepare. The pin is
enabled in open, but is disabled in hdmi_init_pin which is called during
system resume. This causes a system suspend/resume during playback to
mute HDMI/DP. Enabling the pin in prepare instead of open allows calling
snd_pcm_prepare after a system resume to restore audio.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This merges the changes for converting to new PM ops for platform
and some other drivers.
Also move some header files to local places from the public
include/sound.
A recent commit made patch_alc268 call snd_hda_pick_fixup with
NULL quirk pointer. Make sure we do not reference that NULL pointer.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This codec has a separate dmic path (separate dmic only ADC),
and thus it looks mostly like ALC275.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1025377
Tested-by: Ray Chen <ray.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since some jack controls may be renamed as phantom jacks, the existing
check for index conflicts doesn't work because it simply compares the
name with the last used name, assuming that the controls with the same
name continue. Thus, it would result in the duplicated controls when
two or more phantom jacks with the very same type exist, and the
driver gives up with an error.
This patch fixes the problem by checking the index number conflicts
more intensively (but dumbly).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vendor ID 0x10de0051 is used by a yet-to-be-named GPU chip.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Acked-by: Andy Ritger <aritger@nvidia.com>
Reviewed-by: Daniel Dadap <ddadap@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
this patch add proper id for Haswell HDA Controller.
[Added AZX_DCAPS_POSFIX_COMBO flag by tiwai]
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the model parsers out of the way, we have no custom unsol
events to worry about, we can therefore simplify the code.
In addition, this fixes a bug on the ASUS TC710, which has only
a headphone jack and a mic jack, but no internal mic or speakers.
Therefore the unsol_event pointer was not set, and as a result,
the jack kcontrols were not correctly updated.
BugLink: https://bugs.launchpad.net/bugs/1021192
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call cancel_work_sync() when turning off the beep switch so that the
mute call is executed in a proper order.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's no longer necessary since beep_mode=2 option was dropped.
It can be checked simply via codec->beep != NULL.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move snd_hda_mixer_amp_switch_put_beep() to hda_beep.c as a clean up
to remove one more ifdef.
Also add the corresponding get callback to return consistently the
digital beep state independently from the mixer amp value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The beep_mode=2 option was introduced to make the beep mixer
controlling the beep input allocation/deallocation dynamically, so
that a user can switch between HD-audio codec digital beep and the
system beep only via mixer API. This was necessary because the
keyboard driver took only the first input beep instance at that time.
However, the recent keyboard driver already processes the multiple
input instances, thus there is no point to keep this mode.
Let's remove it.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.
Since we change vx222, vx_core and vxpocket have to be converted,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Windows use hidden register to control EAPD.
Linux use verb to control EAPD.
If windows reboot to Linux, it must change the EAPD control to verb
control.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This 3-pin jack was labeled "Headphone Jack", but it could also be
used as a mic jack just by switching "Input Source". Therefore we need
to call the jack something else, to make sure PulseAudio can use the
speaker together with the external mic. (PulseAudio might mute the
speaker if it detects a headphone being plugged in.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ASUS device has a single 3-pin jack that can either be a mic or
a headphone, but the pin does not have VREF capabilities. We've been
told by Realtek to instead enable VREF on pin 0x18 in that case.
BugLink: https://bugs.launchpad.net/bugs/1018262
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for power-map controls (commit b0791dda81) caused
regressions on some other HP laptops. They have fixed pins but these
pins are exposed as jack-detectable. Thus the driver tries to control
the power-map dynamically per jack detection where it never gets on.
This patch adds the check of connection and it assumes the no jack
detection is available for fixed pins no matter what pin capability
says.
BugLink: http://bugs.launchpad.net/bugs/1013183
Reported-by: Luis Henriques <luis.henriques@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This chip looks very similar to ALC269 and ALC27* variants. The bug reporter
has verified that sound was working after this patch had been applied.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1017017
Tested-by: Richard Crossley <richardcrossley@o2.co.uk>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For convenience, add "inv-dmic" model string for enabling the inverted
internal mic workaround to possible Realtek codecs, so far,
ALC882-variants, ALC262, ALC268, ALC269-variants, and ALC662-variants.
Also, the model strings for hardware inv-dmic workarounds,
"alc269-dmic" and "alc271-dmic", are added for ALC269(VA) and ALC271
codecs as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some laptops are equipped with ForteMedia digital mics that give the
differential input. With such devices, summing stereo streams into a
mono (like PulseAudio does) results in almost silence.
This patch provides a workaround for this bug by adding a new mixer
switch to turn on/off the right channel of digital mic, just like a
similar fix for Conexant codecs.
When the new switch "Inverted Internal Mic Capture Switch" is off and
the current input source is the digital mic, the right channel of the
recording stream is muted. When another input source is selected, the
right channel is restored.
Tested-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After cancel_delayed_work_sync returns, the power down work either never
started (power_on == 1) or finished (power_on == 0). In the former case
there is no need to power up again.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo Ideapad S205 has an internal mic where the right channel
is phase inverted.
BugLink: https://bugs.launchpad.net/bugs/884652
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec ALC272X is a special codec for some Dell machines, and its
detection got broken in the recent kernel because SSID check (required
by ALC272X check) was moved to the later point. Now we need to move
this codec check to the right place, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the recent code, the value shown there is a tag number, and it's no
longer same as the pin nid. Correct the message to avoid confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VIA codecs seem not returning the event tag in the unsolicited events,
thus the current code relying on the tag value doesn't work.
Since simple_hdmi stuff has only a single pin, we can use simply
snd_hda_jack_set_dirty_all() to activate the pin-detection
independently from the tag value.
Tested-by: Annie Liu <AnnieLiu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The simple_hdmi stuff is designed only for a single pin and a single
converter (thus a single PCM stream), and no need for loops.
Let's flatten the code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codes to initialize work struct or create a proc interface should
be called only once and never although it's called many times through
the init callback. Move that stuff into patch_generic_hdmi() so that
it's called only once.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes the regression introduced by the commit d0b1252d for
refactoring simple_hdmi*(). The pin NID wasn't assigned correctly.
Reported-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This addresses an issue encountered when a pcm is opened while
transitioning to low power state (codec->power_on == 1 &&
codec->power_transition == -1). Add snd_pcm_power_up_d3wait to
hda_codec. This function is used to power up from azx_open as opposed
to snd_hda_power_up used from codec_exec_verb. When powering up from
azx_open, wait for pending power downs to complete, avoiding the power
up continuing in parallel with the power down on the work queue.
The specific issue seen was with the CS4210 codec, it powers off the ADC
and DAC nid in its suspend handler. If it is re-opened before the
~100ms power down process completes, the ADC and DAC nid are initialized
while powered down and audio is lost until another suspend/resume cycle.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATI and Nvidia HDMI codecs have also the pin-detection capability,
so let's enable the jack-detecion for them, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The code refactoring using the same helper functions for sharing the
codes among ATI, Nvidia and VIA simple_hdmi* stuff. Except for that
spec->pcm_playback is no longer pointer, the functionality doesn't
change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the hotplug unsol event handling to simple_hdmi*().
It works on VIA VX900. If AMD or Nvidia chips support the
pin-detection similarly, it can be added easily, too.
Reported-by: Annie Liu <annieliu@viatech.com.cn>
Tested-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch activates the COMBO position_fix for recent Intel client chipsets.
COMBO mode is the recommended setting for Intel chipsets and eliminates HD
audio warnings in dmesg. This patch has been tested on Lynx Point, Panther
Point, and Cougar Pont.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PulseAudio sometimes have difficulties knowing that there is a
"Speaker" or "Internal Mic", if they have no individual volume
controls or selectors. As a result, only e g "Headphone" might
be created for a laptop, but no "Speaker".
To help out, create phantom jacks (that are always present,
at least for now) for "Speaker", "Internal Mic" etc, in case we
detect them.
The naming convention is e g "Speaker Phantom Jack".
In order not to pollute the /dev/input namespace with even more
devices, these are added to the kcontrols only, not the input devices.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the split to the auto-parser helper functions, the actual call
of init verbs was lost.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43366
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pin 0x1b was connected to the front panel connector, which according to
the HDA standard should contain a mic and a headphone. In this case,
the headphone was listed as "line out" by BIOS.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/993162
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI ID entries of Creative SoundCore3D HD-audio controllers should
be before the wildcard for vendor = Creative.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is patch supporting the CODECes of HD-Audio function of VIA GFX
cards which support HDMI.
For CODECes 0x9f80/0x9f81, which belong to VX900, since the hardware
is not fully compliant to HD-Audio 1.3, simple_i*() is adopted
temporarily.
Signed-off-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is patch supporting HD-Audio function of VIA GFX cards which
support HDMI.
Those are integrated graphics of chipsets VX900 and VX11 separately.
Signed-off-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When VGA_SWITCHEROO support is enabled hda_intel initialises the HDMI
audio device on the current VGA device. When it's not enabled it only
initialises the HDMI device on the default VGA adaptor, this means
secondary cards get no audio support which is very unhelpful for
multi-seat!
With this patch, when SUPPORT_VGA_SWITCHEROO is disabled hda_intel
initialises all HDMI audio devices, not just the default VGA.
[minor optimizations by tiwai]
Signed-off-by: Steven Newbury <steve@snewbury.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When VGA-switcheroo is built in but unused on systems with multiple
graphics cards, the initializations of non-default graphics cards are
skipped and never enabled (because the switcheroo is activated only
when the controller supports). The current behavior is for avoiding
the system lockup by accessing the disabled GPU, but due to the recent
change in VGA-switcheroo, it determines the state simply by checking
with the default VGA device. This is the culprit.
Now with the new vga_switcheroo_get_client_state(), we can know the
initial state of the bound GPU, thus can determine the initial audio
client state more correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
codec may reject power state transition requests(reporting PS-ERROR set),
in that case we re-issue a power state setting and check error bit again.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_param_read() return value -1 means error, others are responses
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
if EPSS supported, transition from D3 state to D0 state in less
than 10ms
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add function to check whether power states supported by specific
codec node.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
add more power states information:
- reset status
- clock stop ok
- power states error
Output like:
Power: setting=D0, actual=D0, Error, Clock-stop-OK, Setting-reset
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the second updates for 3.5-rc1. It's mainly for OMAP4 HDMI
updates and the device tree updates for OMAP, in addition to a couple
of PCM accuray improvement and Realtek ALC269VD codec support.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.18 (GNU/Linux)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=88ho
-----END PGP SIGNATURE-----
Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound update from Takashi Iwai:
"This is the second updates for 3.5-rc1. It's mainly for OMAP4 HDMI
updates and the device tree updates for OMAP, in addition to a couple
of PCM accuray improvement and Realtek ALC269VD codec support."
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits)
ALSA: hda/realtek - Add new codec support for ALC269VD
ALSA: core: group read of pointer, tstamp and jiffies
ASoC: OMAP: HDMI: Rename sound card source file
ASoC: OMAP: HDMI: Make sound card naming more generic
ASoC: OMAP: HDMI: Make build config options more generic
ASoC: OMAP: HDMI: Expand capabilities of the HDMI DAI
ASoC: OMAP: HDMI: Improve how the display state is verified
ASoC: OMAP: HDMI: Expand configuration of hw_params
ASoC: OMAP: HDMI: Use the DSS audio interface
ASoC: OMAP: HDMI: Create a structure for private data of the CPU DAI
ASoC: OMAP: HDMI: Change error values in HDMI CPU DAI
ASoC: OMAP: HDMI: Update the platform device names
ASoC: omap-abe-twl6040: Introduce driver data for runtime parameters
ASoC: omap-abe-twl6040: Move Digital Mic widget into dapm table
ASoC: omap-abe-twl6040: Keep only one snd_soc_dai_link structure
ASoC: omap-dmic: Add device tree bindings
ASoC: omap-mcpdm: Add device tree bindings
ASoC: omap-mcbsp: buffer size constraint only applies to playback stream
ASoC: omap-mcbsp: Use the common interrupt line if supported by the SoC
ASoC: omap-mcbsp: Remove unused FRAME dma_op_mode
...
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.18 (GNU/Linux)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=to5n
-----END PGP SIGNATURE-----
Merge tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull VGA-switcheroo audio client support for HD-audio from Takashi Iwai.
This depended on the recent drm pull.
* tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - unlock on error in azx_interrupt()
ALSA: hda - Support VGA-switcheroo
ALSA: hda - Export snd_hda_lock_devices()
ALSA: hda - Check the dead HDMI audio controller by vga-switcheroo
This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten
for the better support of "implicit feedback". If anything about USB
got broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up immediately
at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital links
between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid the accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal routing
through components with tight sequencing and formatting constraints
within their internal paths or where there are multiple components
connected with CPU managed DMA controllers inside the SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like digital
basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124, Texas
Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be
sent slightly later as partly depending on the changes of DRM.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v2.0.18 (GNU/Linux)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=+JSm
-----END PGP SIGNATURE-----
Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten for
the better support of "implicit feedback". If anything about USB got
broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up
immediately at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital
links between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal
routing through components with tight sequencing and formatting
constraints within their internal paths or where there are multiple
components connected with CPU managed DMA controllers inside the
SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like
digital basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
Texas Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be sent
slightly later as partly depending on the changes of DRM."
Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree). Also some
fairly trivial context conflicts in the imx and mcx soc drivers.
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: snd-usb: fix stream info output in /proc
ALSA: pcm - Add proper state checks to snd_pcm_drain()
ALSA: sh: Fix up namespace collision in sh_dac_audio.
ALSA: hda/realtek - Fix unused variable compile warning
ASoC: sh: fsi: enable chip specific data transfer mode
ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
ASoC: sh: fsi: use same format for IN/OUT
ASoC: sh: fsi: add fsi_version() and removed meaningless version check
ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
ASoC: tegra: Add machine driver for WM8753 codec
ALSA: hda - Fix possible races of accesses to connection list array
ASoC: OMAP: HDMI: Introduce codec
ARM: mx31_3ds: Add sound support
ASoC: imx-mc13783 cleanup
mx31moboard: Add sound support
ASoC: mc13783 codec cleanups
ASoC: add imx-mc13783 sound support
ASoC: Add mc13783 codec
mfd: mc13xxx: add codec platform data
ASoC: don't flip master of DT-instantiated DAI links
...
Pull trivial updates from Jiri Kosina:
"As usual, it's mostly typo fixes, redundant code elimination and some
documentation updates."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (57 commits)
edac, mips: don't change code that has been removed in edac/mips tree
xtensa: Change mail addresses of Hannes Weiner and Oskar Schirmer
lib: Change mail address of Oskar Schirmer
net: Change mail address of Oskar Schirmer
arm/m68k: Change mail address of Sebastian Hess
i2c: Change mail address of Oskar Schirmer
net: Fix tcp_build_and_update_options comment in struct tcp_sock
atomic64_32.h: fix parameter naming mismatch
Kconfig: replace "--- help ---" with "---help---"
c2port: fix bogus Kconfig "default no"
edac: Fix spelling errors.
qla1280: Remove redundant NULL check before release_firmware() call
remoteproc: remove redundant NULL check before release_firmware()
qla2xxx: Remove redundant NULL check before release_firmware() call.
aic94xx: Get rid of redundant NULL check before release_firmware() call
tehuti: delete redundant NULL check before release_firmware()
qlogic: get rid of a redundant test for NULL before call to release_firmware()
bna: remove redundant NULL test before release_firmware()
tg3: remove redundant NULL test before release_firmware() call
typhoon: get rid of redundant conditional before all to release_firmware()
...
Like the previous fixes for cache hash accesses, a protection over
accesses to the widget connection list array must be provided.
Together with this action, remove snd_hda_get_conn_list() which can be
always race, and replace it with either snd_hda_get_num_conns() or
snd_hda_get_connections() calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an spin_unlock() missing on this error path.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch just sets the codec probe_mask=0x101 value for the WinFast VP200 H
PCoIP card based on Teradici hardware matching the PCI subsystem vendor/device
IDs 3a21:040d. The user reported no codec detection issues without this
explicit codec configuration.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai