Codec creation and stream initialization can be shared between
hda_intel and hda platform drivers. Move it and the static functions
it depends on to hda_controller.c.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This code will be reused by an hda_platform driver as it has no PCI
dependencies. This allows update_rirb to be static as all users are
now in hda_controller.c.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This op will be used by hda_intel to do the position check. Takashi
wisely suggested adding this before moving the interrupt handler to
common HDA code. Having this callback prevents the need to move the
hda_intel specific delayed interrupt handling with the irq.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Share more code from hda_intel. This moves the link control and
initialization to hda_controller. The code will also be used by an
hda platform driver.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Combining the call to alloc_cmd_io with the allocate pages function
removes an extra interface between hda_intel and hda_controller.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is done to allow an HDA platform driver to reuse the code.
A few of the interfaces added to hda_controller will disappear in
following commits as their users are also moved to hda_controller.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moving the DSP loading functionality to hda_controller.c means that
the dsp lock doesn't need to be shared in hda_intel and
hda_controller. The forthcoming platform driver doesn't need the DSP
loading code, but sharing it doesn't hurt.
Tested on Chromebook Pixel's ca0132 that uses the DSP loader.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull allocation from first_init to a new function in hda_controller.c.
Short term this will allow the dsp loader to be moved as well. In
later commits it will allow the same allocation to be used by the
platform hda driver.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull the pcm_ops and the functions they use into a new hda_controller
file. This is done to allow for other hda implementations besides PCI
to use the same ops. The hda_controller file will house functionality
related to HDA but independent of the bus used to talk to the
controller.
This currently shares dsp locking across the two files. This will be
remedied in a following commit.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding this op allows the X86 specific mmap operation to help in
hda_intel without needing a CONFIG_X86 in future non-PCI hda drivers.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Break out the allocation of pages for DMA and PCM buffers to ops in
the chip structure. This is done to allow for architecture specific
work-arounds to be added. Currently mark_pages_wc is used by
hda_intel. This avoids needing to move that x86-specific code to a
common area shared with hda platform drivers.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Passing the max slots and power save arguments to codec_create will
allow for its reuse by an hda_platform driver. It makes the function
independent of the module params in hda_intel and ready to move to
hda_shared in a following commit.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keeping a pointer to the jackpoll_ms array in the chip will allow
azx_codec_create to be shared between hda_intel and hda_platform
drivers. Also modify get_jackpoll_ms to make the jackpoll_ms member
optional, this way a platform driver can leave it out if it's not
needed.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although the code was updated last year the "#if 0" surrounding it
dates back to the original git commit. The function will be moved to
a new file, no need to carry the dead code.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a PCI-only feature, but adding a callback for it in the chip
structure breaks the PCI dependency in the RIRB code allowing the
logic there to be re-used by the platform HDA driver.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This removes calls to get the device via PCI from other parts of the
code that will be able to be re-used by the platform driver.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will allow for a platform hda driver to use it as well. It
removes the dependency on the module param from hda_intel, which will
allow for azx_setup_periods to be shared.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The forthcoming platform hda driver needs to override the way
registers are read and written. In preparation for that, introduce a
reg_ops struct that can be implemented differently by the new driver.
Change the existing macros to use the new structure, and move them to
hda_priv.h where they will be accessible to both PCI and platform
drivers.
Start with register access, but later commits will add more ops that
differ between PCI and platform.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Later commits adding support for hda platform drivers will want to use
the same defines and structures. Put them in a place reachable by both
hda_intel and the new platform driver.
This is a mostly a direct copy with a few whitespace and comment
changes to make checkpatch happy.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the HDA spec, we must write 1 to bit 15 on a CORBRP
reset, read back 1, then write 0, then read back 0. This must be
done while the DMA is not running.
We accidentaly ended up writing back the 0 by using a writel
instead of a writew to CORBWP.
This caused occasional controller failure on Bay Trail hardware.
[replaced error messages with dev_err() by tiwai]
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The beep input device is registered via input_register_device(), but
this is called in snd_hda_attach_beep_device() where the sound devices
aren't registered yet. This leads to the binding to non-existing
object, thus results in failure. And, even if the binding worked
(against the PCI object), it's still racy; the input device appears
before the sound objects.
For fixing this, register the input device properly at dev_register
ops of the codec object it's bound with. Also, call
snd_hda_detach_beep_device() at dev_disconnection so that it's
detached at the right timing. As a bonus, since it's called in the
codec's ops, we can get rid of the further call from the other codec
drivers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many HP laptops with STAC codecs have a docking station port and BIOS
sets the pins for the input on the dock as a line in. Because the
generic parser doesn't handle a line in pin as auto-switchable, this
resulted in the manual capture source selection on these laptops.
However, from the usability POV, the automatic switching is easier.
This patch adds the line_in_auto_switch hint in the fixup function for
these laptops. Even if no dock port is present, this should be
harmless as the generic parser allows the auto-switching only in a
limited situation (all three pins are located in different
positions).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of the controller, the new codec object is assigned as a
parent for the hd-audio beep input devices, just like already done for
PCM and hwdep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert with dev_err() and co from snd_printk(), etc.
A couple of prints are difficult to convert with dev_err() so they are
converted to pr_err() at least.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert with dev_err() and co from snd_printk(), etc.
A couple of prints are difficult to convert with dev_err() so they are
converted to pr_err() at least.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert with dev_err() and co from snd_printk(), etc.
The debug prints are also reformatted to suit with dev_dbg().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert with dev_err() and co from snd_printk(), etc.
Some commented debug prints are also enabled as dev_dbg().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert with dev_err() and co from snd_printk(), etc.
All debug print macros have been replaced with dev_dbg(), too.
Also, added the missing definition of snd_azf3328_ctrl_inw().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() and co for messages from HD-audio controller and codec
drivers. The codec drivers are mostly bound with codec objects, so
some helper macros, codec_err(), codec_info(), etc, are provided.
They merely wrap the corresponding dev_xxx().
There are a few places still calling snd_printk() and its variants
as they are called without the codec or device context.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some sysfs attributes like init_pin_configs or vendor_name are really
basic and should be available no matter whether the codec driver is
re-configurable or not. Put them out of #ifdef
CONFIG_SND_HDA_RECONFIG and allow the read-only accesses.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have currently sysfs attributes for each hwdep, but basically these
should belong to the codec itself, per se. Let's add them to the
codec object while keeping them for hwdep as is for compatibility.
While we are at it, split the sysfs-related stuff into a separate
source file, hda_sysfs.c, and keep only the stuff necessary for hwdep
in hda_hwdep.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the HD-audio is treated individually in each codec driver, it's
more convenient to assign an own struct device to each codec object.
Then we'll be able to use dev_err() more easily for each codec, for
example.
For achieving it, this patch just creates an object "hdaudioCxDy".
It belongs to sound class instead of creating a new bus, just for
simplicity, at this stage. No pm ops is implemented in the device
struct level but currently it's merely a container. The PCM and hwdep
devices are now children of this codec device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now each snd_hda_codec instance is managed via the device chain, the
registration and release are done by its callback instead of calling
from bus.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For referring to a different object from sysfs ops, take hwdep
private_data as stored via dev_set_drvdata() at creating the device
object. In that way, the same sysfs ops can be used by different
device types.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling each time device_create_file(), create the groups
of sysfs attribute files at once in a normal way. Add a new helper
function, snd_get_device(), to return the associated device object,
so that we can handle the sysfs addition locally.
Since the sysfs file addition is done differently now,
snd_add_device_sysfs_file() helper function is removed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The GPIO line used for the mute LED control on Lenovo Yxx0 laptops is
cleared unexpectedly when the codec goes to D3, typically by
power-saving. For avoiding it, add a power filter in the fixup.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=16373
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Folio 13 may have a broken BIOS that doesn't set up the mute LED
GPIO properly, and the driver guesses it wrongly, too. Add a new
fixup entry for setting the GPIO pin statically for this laptop.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70991
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The front headphone and mic jackes on a HP desktop model (Vendor Id:
0x111d76c7 Subsystem Id: 0x103c2b17) can not work, the codec on this
machine has 8 physical ports, 6 of them are routed to rear jackes
and all of them work very well, while the remaining 2 ports are
routed to front headphone and mic jackes, but the corresponding
pin complex node are not defined correctly.
After apply this fix, the front audio jackes can work very well.
[trivial fix of enum definition by tiwai]
BugLink: https://bugs.launchpad.net/bugs/1282369
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: Gerald Yang <gerald.yang@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Incorrect ADC is picked in ca0132_capture_pcm_prepare(),
where it assumes multiple streams while there is one stream
per ADC. Note that ca0132_capture_pcm_cleanup() already does
the right thing.
The Chromebook Pixel has a microphone under the keyboard that
is attached to node id 0x8. Before this fix, recording would
always go to the main internal mic (node id 0x7).
Signed-off-by: Hsin-Yu Chao <hychao@chromium.org>
Reviewed-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a HDMI stream is opened with the same stream tag
as a following opened stream to ca0132, audio will be
heard from two ports simultaneously.
Fix this issue by change to use snd_hda_codec_setup_stream
and snd_hda_codec_cleanup_stream instead, so that an
inactive stream can be marked as 'dirty' when found
with a conflict stream tag, and then get purified.
Signed-off-by: Hsin-Yu Chao <hychao@chromium.org>
Reviewed-by: Chih-Chung Chang <chihchung@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we plug a 3-ring headset on the Dell machines (Vendor ID:
0x10ec0255, Subsystem ID: 0x10280657; Vendor ID: 0x10ec0255,
Subsystem ID: 0x1028065f), the headset mic can't be
detected, after apply this patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: Cyrus Lien <cyrus.lien@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Realtek codec driver contains some codes referring to the PCI
subdevice IDs, but most of them are optional, typically for checking
the codec name variants. Add NULL checks appropriately so that it can
work without PCI assignment.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The default parent device can be obtained directly via card object, so
we don't need to rely on pci->dev.parent. Since there is no access to
pci_dev, we can reduce the inclusion of linux/pci.h, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit d3c56568f4.
The reverted commit breaks audio through headphone line out on
the Acer TravelMate B113 (Type1Sku0) Notebook, my main work
machine. I don't know much about it but this fixes my problem.
Bisected and tested.
Fixes: d3c56568f4 ('ALSA: hda/realtek - Avoid invalid COEFs for ALC271X')
Cc: <stable@vger.kernel.org>
Tested-by: Martin Kepplinger <martink@posteo.de>
Signed-off-by: Martin Kepplinger <martink@posteo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last user of snd_hda_gen_spec_free() is patch_via.c, and we can
rewrite it safely with snd_hda_gen_free(), so that
snd_hda_gen_spec_free() can be a local function in hda_generic.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to alsa-info.sh outputs, all three entries with static
quirks have the correct pin configs, so it's safe to remove static
quirks. For now, turn the static quirks off via ifdef. The dead
codes will be removed in later release.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even after the fix for leftover kconfig handling (commit f8f1becf),
the current code still doesn't handle properly the builtin/module
mixup case between the core snd-hda-codec and other codec drivers.
For example, when CONFIG_SND_HDA_INTEL=y and
CONFIG_SND_HDA_CODEC_HDMI=m, it'll end up with an unresolved symbol
snd_hda_parse_hdmi_codec. This patch fixes the issue.
Now codec->parser points to the parser object *only* when a module
(either generic or HDMI parser) is loaded and bound. When a builtin
symbol is used, codec->parser still points to NULL. This is the
difference from the previous versions.
Fixes: f8f1becfa4 ('ALSA: hda - Fix leftover ifdef checks after modularization')
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the lengthy #if defined(XXX) || defined(XXX_MODULE) with the
new IS_ENABLED() macro.
The patch still doesn't cover all ifdefs. For example, the dependency
on CONFIG_GAMEPORT is still open-coded because this also has an extra
dependency on MODULE. Similarly, an open-coded ifdef in pcm_oss.c and
some sequencer-related stuff are left untouched.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unused function in pci/lx6464es/lx_core.c.
This eliminates the following warning in pci/lx6464es/lx_core.c:
sound/pci/lx6464es/lx_core.c:144:5: warning: no previous prototype for ‘lx_plx_mbox_read’ [-Wmissing-prototypes]
sound/pci/lx6464es/lx_core.c:172:5: warning: no previous prototype for ‘lx_plx_mbox_write’ [-Wmissing-prototypes]
sound/pci/lx6464es/lx_core.c:494:5: warning: no previous prototype for ‘lx_dsp_es_check_pipeline’ [-Wmissing-prototypes]
Signed-off-by: Rashika Kheria <rashika.kheria@gmail.com>
Reviewed-by: Josh Triplett <josh@joshtriplett.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Bug reporter report that the -mode4 makes the subwoofer work.
I have simplified the quirk a bit to avoid possible regressions
with the microphones.
BugLink: https://bugs.launchpad.net/bugs/871808
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just a small refactoring to make the next patch slightly simpler.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The very same fixup is needed to make the mic on Sony VAIO Pro 11
working as well as VAIO Pro 13 model.
Reported-and-tested-by: Hendrik-Jan Heins <hjheins@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code for controlling mic mute LED in patch_sigmatel.c
blindly assumes that there is a single capture switch. But, there can
be multiple multiple ones, and each of them flips the state, ended up
in an inconsistent state.
For fixing this problem, this patch adds kcontrol to be passed to the
hook function so that the callee can check which switch is being
accessed. In stac_capture_led_hook(), the state is checked as a
bitmask, and turns on the LED when all capture switches are off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the commit [595fe1b702c3: ALSA: hda - Make
CONFIG_SND_HDA_CODEC_* tristate], the kconfig variables for the
generic parser and codec drivers can be "m" instead of boolean, but
some codes are left unchanged to check only #ifdef
CONFIG_SND_HDA_CODEC_XXX, which is no longer true for modules.
This patch fixes them by replacing with IS_ENABLED() macros.
Fixes: 595fe1b702 ('ALSA: hda - Make CONFIG_SND_HDA_CODEC_* tristate')
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70161
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD1983 has flexible loopback routes and the generic parser would take
wrong path confusingly instead of taking individual paths via NID 0x0c
and 0x0d. For avoiding it, limit the connections at these widgets so
that the parser can think more straightforwardly. This fixes the
regression of the missing line-in loopback on Dell machine.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70011
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mac Pro 1,1 with ALC889A codec needs the VREF setup on NID 0x18 to
VREF50, in order to make the speaker working. The same fixup was
already needed for MacBook Air 1,1, so we can reuse it.
Reported-by: Nicolai Beuermann <mail@nico-beuermann.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mixer widget on AD1983 at NID 0x0e was missing in the commit
[f2f8be43c5c9: ALSA: hda - Add aamix NID to AD codecs].
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=70011
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've seen often problems after suspend/resume on Acer Aspire One
AO725 with ALC271X codec as reported in kernel bugzilla, and it turned
out that some COEFs doesn't work and triggers the codec communication
stall.
Since these magic COEF setups are specific to ALC269VB for some PLL
configurations, the machine works even without these manual
adjustment. So, let's simply avoid applying them for ALC271X.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Toshiba Satellite L40 with AD1986A codec requires the EAPD of NID 0x1b
to be constantly on, otherwise the output doesn't work.
Unlike most of other AD1986A machines, EAPD is correctly implemented
in HD-audio manner (that is, bit set = amp on), so we need to clear
the inv_eapd flag in the fixup, too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=67481
Cc: <stable@vger.kernel.org> [v3.11+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo Ideapad with ALC272 has a mute LED that is controlled via
GPIO1. Add a simple vmaster hook for it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=16373
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 384a48d715 "ALSA: hda: HDMI: Support codecs with fewer cvts
than pins" dynamically enabled each pin widget's PIN_OUT only when the
pin was actively in use. This was required on certain NVIDIA CODECs for
correct operation. Specifically, if multiple pin widgets each had their
mux input select the same audio converter widget and each pin widget had
PIN_OUT enabled, then only one of the pin widgets would actually receive
the audio, and often not the one the user wanted!
However, this apparently broke some Intel systems, and commit
6169b67361 "ALSA: hda - Always turn on pins for HDMI/DP" reverted the
dynamic setting of PIN_OUT. This in turn broke the afore-mentioned NVIDIA
CODECs.
This change supports either dynamic or static handling of PIN_OUT,
selected by a flag set up during CODEC initialization. This flag is
enabled for all recent NVIDIA GPUs.
Reported-by: Uosis <uosisl@gmail.com>
Cc: <stable@vger.kernel.org> # v3.13
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While looking into some spurious responses, I found that the addr value was
treated a bit inconsistent: values 8..0xf will be treated as codec 0 and
values 0..7 will be treated as no error regardless of whether there is a codec
there, or not.
With this patch, all non-existing codecs will be treated equally.
In addition, printing rp and wp could help figuring out if the wp value is
reported wrongly from the controller or if something else is wrong.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now all weird setups have been converted to fixups for the generic
parser, and we can disable the static quirks. This commit just turns
the build off. The bulky static quirk code still remains for a while,
in case we get an overlooked regression. It'll be removed at the next
kernel version.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both CX20549 and CX20551 codecs have a mixer widget and it can be
connected as the ADC source. Like AD and VIA codecs, enable the
add_stereo_mix_input flag for these codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
OLPC XO needs a few special handling. Now these are implemented as a
fixup to the generic parser.
Obviously, the DC BIAS mode had to be added manually. This is mainly
implemented in the mic_autoswitch hook, where the mic pins are
overwritten depending on the DC bias mode. This also required the
override of the mic boost control, since the mic boost is applied only
when the DC mode is disabled.
In addition, the mic pins must be set dynamically at recording time
because these also control the LED.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... by using snd_Hda_codec_update_cache() instead of *_write_cache().
Since all path elements should have been updated by this function,
we are safe to assume that the cache contents are consistent.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we plug a 3-ring headset on the Dell machine (Vendor ID:
0x10ec0255, Subsystem ID: 0x1028064d), the headset mic can't be
detected, after apply this patch, the headset mic can work well.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: Doro Wu <fan-cheng.wu@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This completes the hardware support for the Asus Xonar DG/DGX cards,
and makes them actually usable.
This is v4 of Roman's patch set with some small formatting changes.
Remove old SPI control functions, change anti-pop init
sequence, remove some garbage from structures. The 'Apply' functions
must be called at the mixer initialization, otherwise
mixer settings sometimes will not be applied at startup.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the 'put' function of the high-pass filter control to use the new
SPI functions.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
First of all, we should not touch the GPIOs. They are not
for selecting the capture source, but they seems just enable
the whole audio input curcuit. The 'put' function calls the
'apply' functions to change register values. Change the order
of capture sources.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modify the input_vol_* functions to use the new SPI routines,
There is a new applying function that will be called when
the capture source changed.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
I tried both variants: volume control and impedance selector.
In the first case one minus is that we can't change the
volume of multichannel output without additional software
volume control. However, I am using this variant for the
last three months and this seems good. All multichannel
speaker systems have internal amplifier with the
volume control included, but not all headphones have
this regulator. In the second case, my software volume
control does not save the value after reboot.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the order of elements in the output select control. This will
reduce the number of relay switches. Change 'put' function to call the
oxygen_update_dac_routing() function. Otherwise multichannel playback
does not work. Also there is a new function to apply settings, this
prevents from duplicating the code.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Actually CS4245 connected to the I2S channel 1 for
capture, not channel 2. Otherwise capturing and
playback does not work for CS4245.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Moving the mixer code away makes things easier. The mixer
will control the driver, so the functions of the
driver need to be non-static.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change the function to read the data from the new shadow buffer.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When selecting the audio output destinations (headphones,
FP headphones, multichannel output), the channel routing
should be changed depending on what destination selected.
Also unnecessary I2S channels are digitally muted. This
function called when the user selects the destination
in the ALSA mixer.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When selecting the audio sample rate for CS4245,
the MCLK divider should also be changed.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Change CS4245 initialization: different sequence and GPIO values,
according to datasheets and reverse-engineering information.
Change cleanup/resume/suspend functions, since they use
initialization.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add the new SPI write and read functions. The SPI read function
is used for creating initial registers dump and may be used for
debugging purposes. SPI operations are cached, so there is a new
function to manage the cache (shadow). I have to remove
the shift from the CS4245_SPI_* constants, since when
we are performing the reading, we need to shift by 8 instead
of 16.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add additional constants to the xonar_dg.h file:
capture and playback sources. Move GPIO_* constants and the
dg struct to the header file from the xonar_dg.c file.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Add some additional information in comments and my copyright.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
When the user switches the output from stereo to multichannel
or vice versa, the driver needs to update the channel routing.
Instead of creating additional subroutines, I better export existing
oxygen_update_dac_routing symbol from the oxygen mixer
and call this function. It calls model.adjust_dac_routing()
and my function does the work.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The Xonar DG/DGX driver needs this mask to mute unnecessary
channels.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Modify the oxygen_write_spi() function to use the newly
introduced oxygen_wait_spi() function. Change return value
from void to int, so it can return error codes. Older
drivers just ignore that return value, new drivers can
check this value. We need to wait AFTER
initiating the SPI transaction, otherwise read
operation will not work.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
The oxygen_wait_spi() function now performs waiting when the
SPI bus completes a transaction. Introduce the timeout error
checking and increase timeout to 200 from 40.
Signed-off-by: Roman Volkov <v1ron@mail.ru>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Processing coefficients are often a vital part of the codec's configuration,
so dumping them can be important. However, because they are undocumented and
secret, we do not want to enable this for all codecs by default.
Therefore instead add this as a debugging parameter.
I have prepared for codecs that want to enable this by default by the extra
dump_coef bitfield, but unsure if we want to do that as long as the
(unlikely, but still) race remains.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly to other Apple products, MBA 1,1 needs a specific quirk.
Pin 0x18 must be set to VREF_50 to have sound output. This was no
longer done since commit 1a97b7f, resulting in a mute built-in speaker.
This patch corrects the regression by creating a fixup for the MBA 1,1.
Fixes: 1a97b7f227 ("ALSA: hda/realtek - Remove the last static quirks for ALC882")
Cc: <stable@vger.kernel.org> [v3.4+]
Tested-by: Adrien Vergé <adrienverge@gmail.com>
Signed-off-by: Adrien Vergé <adrienverge@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS Zenbook UX31A has yet another problem -- softer output level than
others. According to the measurement, the peak output difference
between 31A and 31E is 5dB. As ALC269VB has a COEF for the class-D
pre-amp, let's apply it for +5dB.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dylan Reid