commit 6175ddf06b optimized the mem*io
functions that have been used to send commands to the device. these
optimizations somehow corrupted the communication with the lx6464es,
that resulted the device to be unusable with kernels after 2.6.33.
this patch emulates the memcpy_*_io functions via a loop to avoid these
problems.
Signed-off-by: Tim Blechmann <tim@klingt.org>
LKML-Reference: <4ECB5257.4040600@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
the command buffer is only accessed from one file, so we can declare the
specific functions as static in that file
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve the one-shot ELD repoll to up to 6 retries.
Up to now the 300ms looks sufficient for the test boxes. However
I'm a bit worried about how well it can fit the wider user base.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit c09403dcc5 has introduced a
regression: PCIe versions of RME MADI were no longer detected, because
the MADIface ID (0xd5) was used instead of the correct 0xd2.
This commit fixes the problem.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The multiple headphone pins are usually handled as copied from the same
source, not as individual channels like front and surround. Thus it'd
be more correct to avoid the channel suffix for "Headphone" pin labels
in snd_hda_get_pin_label() but give an index number instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
desc->size is supposed to be a le16 type. On a big endian system the
current code will set ->size to zero. We fixed a similar bug
on the next line but missed this one.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
spec->cur_adc isn't set until cs_capture_pcm_prepare() is called although
the driver tries to select the connection at init time and at auto-mic
switch. This results in the access to the widget NID 0, which is
obviously invalid, also a wrong capture source.
This patch fixes the issue by issuing the connect-select verb conditionally
at appropriate places.
Reported-and-tested-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function now gives more unique names for the output pins by adding
some prefix and suffix for the location and the channels. Otherwise, it
can pass the index number.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise the jack kctls will report invalid values until the jack
is re-plugged.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These two functions are being used by the codec-idt and codec-hdmi
modules, so they need to be exported properly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now move the helper function for creating and reporting the jack-detection
to the common place. The driver that needs this functionality should
select CONFIG_SND_KCTL_JACK kconfig.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Manage the tags assigned for unsolicited events dynamically together
with the jack-detection routines. Basically this is almost same as what
we've done in patch_sigmatel.c. Assign the new tag number for each new
unsol event, associate with the given NID and the action type, etc.
With this change, now all pins looked over in snd_hda_jack_add_kctls()
are actually enabled for detection now even if the pins aren't used for
jack-retasking by the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create kcontrols for pin jack-detections, which work similarly like
jack-input layer. Each control will notify when the jack is plugged or
unplugged, and also user can read the value at any time via the normal
control API.
The control elements are created with iface=CARD, so that they won't
appear in the mixer apps.
So far, only the pins that enabled the jack-detection are registered.
For covering all pins, the transition of the common unsol-tag handling
would be needed. Stay tuned.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a table containing the pins and their jack-detection states
for avoiding the unnecessary verbs to check the pin status at each time.
When the unsol event is enabled via snd_hda_jack_detect_enable(), it
automatically adds the given NID to the table. Then the driver supposes
that the codec driver will set the dirty flag appropariately when an
unsolicited event is invoked for that pin.
The behavior for reading other pins that aren't registered in the table
doesn't change. Only the pins assigned to the table are cached, so far.
In near futre, this table can be extended to use the central place for
the unsolicited events of all pins, etc, and eventually include the
jack-detect kcontrols that replace the current input-jack stuff.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a new helper function snd_hda_get_pin_label() for getting a label
string for both input and output pins. hda_get_input_pin_label() is
obsoleted by this function, and the callers are replaced appropriately
now by this patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Intel HDMI chips (ironlake at least) are found to have ~250ms delay
between the ELD_Valid=1 hotplug event is send and the ELD buffer becomes
actually readable. During the time the ELD buffer is mysteriously all 0.
Fix it by scheduling a delayed work to re-read ELD buffer after 300ms.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
memset(eld) clears eld->proc_entry which will leak the struct
snd_info_entry when unloading module.
Fix it by
- memset only the fields before eld->eld_buffer
- set eld->eld_valid to true _after_ all eld fields have been filled
Cc: <stable@kernel.org>
Cc: Pierre-louis Bossart <pierre-louis.bossart@intel.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Translate ALC880 medion-rim static configs to the auto-parser with the
additional GPIO2 verb and COEF setup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ALC882 macpro and imac24 static configs can be transferred to the
auto-parser with the additional GPIO setup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit [c3e837bb: ALSA: hda/realtek - Rewrite ALC882 acer-aspire-*
models with the auto-parser], the check of the model option got removed
mistakenly. Re-added the board_config check again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the pin or the DAC doesn't support the stereo, create a mono
control instead of creating a stereo control blindly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up and fix pwr_nids for 92HD71 / 73 / 83 family codecs; remove
pwr_mapping which was incorrect.
The original pwr_nids support of 92HD83xxx was incorrect and never
actually worked before. Now we should have things working correctly
without having to hack by DID anymore.
It is also not necessary to explicitly turn on all the pins near the
beginning of patch_stac92hd83xxx() now, the pins will go though
initialization properly.
Tested on 92HD66 / 71 / 73 / 75 / 83 / 89 / 91 demo boards.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we can move the big acer-aspire-* static quirks to the auto-paresr
with some additional pin-configs and verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far the driver creates the multi-io jacks only when a single output
jack, i.e. no multiple speakers are assigned. This patch adds the
similar multi-io detection even with multiple speakers are assigned
primarily, so that 5.1-speakers + HP/mic/LI combination can work.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We disabled the check of NO_PRESENCE bit of the default pin-config
in commit f4419172 temporarily. One problem was that the first
implementation was wrong -- the bit after the shift must be checked.
However, this would still give many regressions on machines with broken
BIOS. They set this bit wrongly even on active pins.
A workaround is to check whether all pins contain this bit. As far as
I've checked, broken BIOSen set this bit on all pins, no matter whether
active or not. In such a case, the driver should ignore this bit check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a virtual mater control is created, the driver looks for slave
elements from the assigned card instance. But this may include the
elements of other codecs when multiple codecs are on the same HD-audio
bus. This works at the first time, but it'll give Oops when it's once
freed and re-created via reconfig sysfs.
This patch changes the element-look-up strategy to limit only to the
mixer elements of the same codec.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Detection code improved by PCI SSID usage. VM optimization
now enabled only for known devcices (skip host devices forwarded
to VM by VT-d or same kind of technology).
For debug/troubleshooting purposes optimization can be
forced (on/off) by module parameter: "inside_vm" (boolean).
Known devices (PCI SSID):
1af4:1100: Reserved for KVM devices. Note this is not yet
implemented for KVM's ICH/AC'97 emulation.
1ab8:xxxx: Parallels ICH/AC'97 emulated sound.
[ fixed a minor coding-style issue by tiwai]
Signed-off-by: Konstantin Ozerkov <kozerkov@parallels.com>
Signed-off-by: Denis V. Lunev <den@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is refactoring patch: preparation for add improved detection code.
Now all detection code placed in one place.
Signed-off-by: Konstantin Ozerkov <kozerkov@parallels.com>
Signed-off-by: Denis V. Lunev <den@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we're touching the desktop static configs for ALC88x codecs.
These are mostly OK with the auto-parser, but some models need careful
handling; ALC889 intel mobo requires the COEF setup, and W2JC needs
GPIO1 and COEF.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All ALC882 targa-* models can be replaced with the auto-parser just
with the additional GPIO3 setup. And it's generically applied to all
MSI boards unless other quirks are present.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove all ALC882 static configurations for all Lenovo and Haier-w66
quirks. They work fine with the auto-parser now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These models work fine with the auto-parser with the additional COEF
setup. The iMac 7,1 (106b:3200) also uses the same quirk, so remove it
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not only PCI SSIDs but also look through codec SSIDs for fix-up table
entries. MacBook tend to give the same PCI SSID but unique codec SSIDs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS EEE1601 works almost fine with the auto-parser but the static
configuration has a certain specific COEF verb. Add this to the fix-up
list so that we can drop the whole EEE1601 static config from
alc882_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the not working internal mic on Dell Vostro 3500 laptop by introducing the
new model dell-vostro-3500.
Signed-off-by: Julian Wollrath <jwollrath@web.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the bug reporter, model=auto is needed to make the
internal microphone work.
BugLink: https://bugs.launchpad.net/bugs/819699
Reported-by: Andrej (agno01)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Providing a pincfg fix for VAIO-TT with ALC889 codec to work with the
auto-parser, and drop the static configuration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that model=ultra is supported well by the auto-parser, we can get rid
of the whole alc262_quirks.c and its related codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A machine like Q1-ultra which has only a single HP but no mic-jack, we
can re-task the headhpone as an external mic jack. This was done formerly
in ALC262 model=ultra quirk, and now the auto-parser supports this mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both entries for ALC262 model=hippo work well with the auto-parser and
the default BIOS setup. No static configs are needed, so drop them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This model (actually BenQ Joybook) works fine with the default auto-parser
and the BIOS setup. Just drop the static quirks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It works well with the auto-parse and the default BIOS setup when an
additional COEF setup (for benq) is used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The conversion from ALC262 model=benq and model=benq-t31 static configs
to auto-parser requires the manual COEF setups for corresponding models.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The static quirks for ALC262 Lenovo 3000 can be covered by the auto-
parser with a fixup of the mic-pin to VREF50 and the additional COEF
verb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This laptop works fine with the current auto-parser and the BIOS setup,
so let's drop the static configuration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the auto-parser for ALC262 model=toshiba-rx1 with the fixed pin-
configs. The BIOS table seems incorrect, so many pin entries are
overwritten to match with the former quirk.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a Realtek codec has a matrix-style capture-source selection, we
need to scan all connections instead of only imux items. Otherwise some
input might be kept unmuted. Although the corresponding input must be
dead so there should be no input from it, it's still safer to mute the
route completely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently merged 228cf79376 looks a bit hackish while it is not.
The change was quite simple. In a virtualized environment the
patch unhacks old kludge introduced for old broken AC97 hardware.
This patch adds proper comment to "unkludge" code.
Signed-off-by: Denis V. Lunev <den@openvz.org>
Signed-off-by: Konstantin Ozerkov <kozerkov@parallels.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'modsplit-Oct31_2011' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux: (230 commits)
Revert "tracing: Include module.h in define_trace.h"
irq: don't put module.h into irq.h for tracking irqgen modules.
bluetooth: macroize two small inlines to avoid module.h
ip_vs.h: fix implicit use of module_get/module_put from module.h
nf_conntrack.h: fix up fallout from implicit moduleparam.h presence
include: replace linux/module.h with "struct module" wherever possible
include: convert various register fcns to macros to avoid include chaining
crypto.h: remove unused crypto_tfm_alg_modname() inline
uwb.h: fix implicit use of asm/page.h for PAGE_SIZE
pm_runtime.h: explicitly requires notifier.h
linux/dmaengine.h: fix implicit use of bitmap.h and asm/page.h
miscdevice.h: fix up implicit use of lists and types
stop_machine.h: fix implicit use of smp.h for smp_processor_id
of: fix implicit use of errno.h in include/linux/of.h
of_platform.h: delete needless include <linux/module.h>
acpi: remove module.h include from platform/aclinux.h
miscdevice.h: delete unnecessary inclusion of module.h
device_cgroup.h: delete needless include <linux/module.h>
net: sch_generic remove redundant use of <linux/module.h>
net: inet_timewait_sock doesnt need <linux/module.h>
...
Fix up trivial conflicts (other header files, and removal of the ab3550 mfd driver) in
- drivers/media/dvb/frontends/dibx000_common.c
- drivers/media/video/{mt9m111.c,ov6650.c}
- drivers/mfd/ab3550-core.c
- include/linux/dmaengine.h
The implementation on commit [08a1f5eb: ALSA: hda - Check NO_PRESENCE
pincfg default bit] seems like a mis-interpretation of specification.
The spec gives the reversed bit definition. But, following the spec
also causes to change so many existing device configurations, thus we
can't change it so easily for now. For 3.2-rc1, it's safer to revert
this check (actually this patch comments out the code).
We may re-introduced the fixed version once after the wider test-case
coverages are done.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The regression-fix in 3.1 for the check of DMA-position validity caused
yet another regression for CA0110. As usual, this hardware seems working
only with LPIB properly. Adding the appropriate driver-caps bit to force
LPIB fixes the problem.
Reported-and-tested-by: Andres Freund <andres@anarazel.de>
Cc: <stable@kernel.org> [v3.1]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The 3.1 kernel has a regression for ALC861 codec where no sound output
is heard with the default setup. It's because the amps in DACs aren't
properly unmuted while the output mixers are assigned only to pins.
This patch fixes the missing initialization of DACs when no mixer is
assigned to them.
Tested-by: Andrea Iob <andrea_iob@yahoo.it>
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some BIOS report invalid pins as digital output pins. The driver checks
the connection but it doesn't do it fully correctly, and it leaves some
undefined value as the audio-out widget, which makes the driver spewing
warnings. This patch fixes the issue.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=727348
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revise stac92xx_parse_auto_config to automatically scan for digital input
and output converters.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the driver finds multiple ADCs, it tries to create an alternative
capture PCM stream. However, these secondary ADCs might be useless or
in uncontrolled paths in some cases, e.g. when auto-mic or dynamic
ADC-switching is enabled. Also, when only a single capture source is
available, the multi-streams don't make sense, too.
With this patch, the driver checks such condition and skips the alt
stream appropriately.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These codecs have SPDIF-in, which is new to the 92HD83xxx compatible
families, so a bit of logic is added to support them.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The power-widget control in patch_stac92hd83xxx() never worked properly,
thus it's safer to turn it off as default for now.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio spec defines a bit in pin default configuration for indicating
that the pin isn't used for jack-detection although the codec is capable
of it. Better to check this bit as well in jack_is_detectable() helper
function.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a line in the firmware file is larger than the given buffer size (and
so the firmware file size), size is set to a value larger than the actual
buffer size. This results in an overflow in the buffer passed.
Signed-off-by: Alexander Stein <alexander.stein@systec-electronic.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These aren't modules, but they do make use of these macros, so
they will need export.h to get that definition. Previously,
they got it via the implicit module.h inclusion.
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
Lots of sound drivers were getting module.h via the implicit presence
of it in <linux/device.h> but we are going to clean that up. So
fix up those users now.
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
The implicit presence of module.h lured several users into
incorrectly thinking that they only needed/used modparam.h
but once we clean up the module.h presence, these will show
up as build failures, so fix 'em now.
Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
v3: detection code is x86 and KVM specific, hide it under ifdef
v2: add detection for virtual environments (KVM and Parallels)
This patch is intended to improve performance in virtualized environments
like Parallels Desktop or KVM/VirtualBox/QEMU (virtual ICH/AC97 audio).
I/O access is very time-expensive operation in virtual world: VCPU
can be rescheduled and in the worst case we get more than 10ms delay on
each I/O access.
In the virtual environment loop exit rule
(old_civ == current_civ && old_picb == current_picb) is never satisfied,
because old_picb is never the same as current_picb due to delay inspired
by reading current_civ. As a result loop ended by timeout and we get 10x
more I/O operations.
Experimental data from Prallels Desktop 7, RHEL6 guest (I/O ops per
second):
Original code:
In Port Counter Callback
f014 41550 fffff00000179d00 ac97_bm_read_civ+0x000
f018 41387 fffff0000017a580 ac97_bm_read_picb+0x000
With patch:
In Port Counter Callback
f014 4090 fffff00000179d00 ac97_bm_read_civ+0x000
f018 1964 fffff0000017a580 ac97_bm_read_picb+0x000
Signed-off-by: Konstantin Ozerkov <kozerkov@parallels.com>
Signed-off-by: Denis V. Lunev <den@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From the Windows INF file, we know the firmware ranges for all RME
cards. For PCIe, a single revision ID per device (RayDAT, MADI, AIO,
AES) is used. Contrary, the older PCI versions use ranges, that is,
one revision ID per firmware version.
Instead of listing all possible revisions individually, match the range.
This commit enables all MADI and AES PCI versions ever shipped.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDSP_VERSION_BIT has to be ORed with HDSP_S_LOAD. This fixes the detection
of at least some RME RPM boxes.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SNDRV_HDSPM_IOCTL_GET_STATUS is supposed to query the current card
status, so we have to return what we receive on the MADI wire (RX), not
what we transmit (TX) to others. The latter is a config item to be
queried via SNDRV_HDSPM_IOCTL_GET_CONFIG.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that Conexant CX20549 chip handle only a single input-amp even
though the audio-input widget has multiple sources. This has been never
clear, and I implemented in the current way based on the debug information
I got at the early time -- the device reacts individual input-amp values
for different sources. This is true for another Conexant codec, but it's
not applied to CX20549 actually.
This patch changes the auto-parser code to handle a single input-amp
per audio-in widget for CX20549. After applying this, you'll see only a
single "Capture" volume control instead of separate "Mic" or "Line"
captures when the device is set up to use a single ADC.
We haven't tested 20551 and 20561 codecs yet. If these show the similar
behavior like 20549, they need to set spec->single_adc_amp=1, too.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the old Conexant chips (5045, 5047, 5051 and 5066), a single EAPD
may handle both headphone and speaker outputs while it's assigned only
to one of them. Turning off dynamically leads to the unexpected silent
output in such a configuration with the auto-mute function.
Since it's difficult to know how the EAPD is handled in the actual h/w
implementation, better to keep EAPD on while running for such codecs.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC260 has multiple mixer widgets connected to the shared DAC, but the
driver currently doesn't check this possibility and ignores when the DAC
is shared with others. This resulted in the silent output from some
routes because of lack of the amp setup.
This patch adds the workaround for it by checking the route even with the
shared DAC, but also checking the conflict with the existing control for
the very same widget NID.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=726812
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The association numbers of surround/CLFE speaker pins aren't correctly
mapped by the auto-parser. This patch fixes the CLFE speaker pin to the
right assoc value (from 3 to 1).
Tested-by: Nika Topolchanskaya <nanodesuu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When 5.1 or more headphone or speaker pins are provided, the parser still
takes as is without fixing the order of channel mapping, which leads in
the unexpected strange channel order by surround outputs.
This patch fixes the issue by applying the same fix-up not only to
line_out_pins[] but also hp_pins[] and speaker_pins[].
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The revision 0x100300 was found for ALC662. It seems to work well
with patch_alc662.
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/877373
Tested-by: Shengyao Xue <Shengyao.xue@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Acked-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a device has multiple speakers and still has the auto-mute support,
the driver copies line_outs[] to speaker_outs[]. And then it tries to
assign DACs for both. This ended up with the assignment only to the
primary DAC to all speakers.
This patch fixes the situation by checking the duplicated LO/SPK case
appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is patch for Conexant codec of Intel HDA driver, adding new quirk
for Lenovo Thinkpad T520 and W520. Conexant autodetection works fine for
T520 (similar subsystem ID is used also in W520 model) and detects more
mixer features compared to generic (fallback) Lenovo quirk with
hardcoded options in Conexant codec.
Patch was activelly tested with Linux 3.0.4, 3.0.6 and 3.0.7 without any
problems.
Signed-off-by: Daniel Suchy <danny@danysek.cz>
Cc: <stable@kernel.org> [3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous fix for the position-buffer check gives yet another
regression on a Dell laptop. The safest fix right now is to add a
static quirk for this device (and better to apply it for stable
kernels too).
Reported-by: Éric Piel <Eric.Piel@tremplin-utc.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The COEF #0 value represents a sort of device id, so it's supposedly
constant while operation. Better to use the cached value instead of
reading it at each time from the performance POV.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a static table for detecting the codec renames.
Also clean up the error paths in each patch_*() function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This typo caused headphone pins not to be initialized correctly.
BugLink: https://bugs.launchpad.net/bugs/871582
Reported-by: Effenberg
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The purpose of this patch is to remove a section of "bad" code that
assigns the last DAC to ports E or F in order to support notebooks
with docking in earlier days, around ALSA 1.0.19 - 21. This is not
necessary now and actually breaks some configurations that use these
ports as other devices. This have been tested on several different
configurations to make sure that it is working for different combinations.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit ef18beded8 introduced a
mechanism to assign the previously used slot for the next reopen of a
PCM stream. But the PCM device number isn't always unique (it may
have multiple substreams), and also the code doesn't check the stream
direction, thus both playback and capture streams share the same
device number.
For avoiding this conflict, make a unique key for each substream and
store/check this value at reopening.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the speaker outputs are more than the headphone outputs, it implies
that the system has surround speakers while the headphones are only for
monitoring the front. In such a case, it's better to put speakers as
the primary outputs so that the driver can build up and keep the
surround setup. Otherwise the system will pick up the headphone as
primary, and offers less channels than the speakers do support.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since hda_proc.c is now the only user of snd_print_pcm_rates(), better to
put it back locally to hda_proc.c and revert to the old style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SAD sampling rate information reported in
/proc/asound/cardX/eldX is incorrect due to a mismatch
between HDA and HDMI frequencies. Add new routine to provide
relevant values.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar to Line Out, these constants form the base for future
patches enabling input jack reporting for Line in jacks.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If we run out of DACs when trying to assign a DAC to a secondary
headphone, prefer the DAC of the first headphone to the primary
(usually line out) DAC.
BugLink: http://bugs.launchpad.net/bugs/845275
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Sigmatel/IDT parser should have the same naming convention
for input jacks as the other codecs have.
BugLink: http://bugs.launchpad.net/bugs/859704
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Applications may want to read ELD information to
understand what codecs are supported on the HDMI
receiver and handle the a-v delay for better lip-sync.
ELD information is exposed in a device-specific
IFACE_PCM kcontrol. Tested both with amixer and
PulseAudio; with a corresponding patch passthrough modes
are enabled automagically.
ELD control size is set to zero in case of errors or
wrong configurations. No notifications are implemented
for now, it is expected that jack detection is used to
reconfigure the audio outputs.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit a810364a04
ALSA: hda - Handle -1 as invalid position, too
caused a regression on some machines that require the position-buffer
instead of LPIB, e.g. resulting in noises with mic recording with
PulseAudio.
This patch fixes the detection by delaying the test at the timing as
same as 3.0, i.e. doing the position check only when requested in
azx_position_ok().
Reported-and-tested-by: Rocko Requin <rockorequin@hotmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the ugly real world, there area really broken devices that don't set
codec SSID correctly. In such a case, the ID can be random, thus the
patching won't work reliably.
For applying the patch forcibly to such a device, the driver will skip
the vendor and/or subsystem ID checks when zero or a negative number is
given in [codec] section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new option "snoop" for the traffic control of the HD-audio
controller chip. When set to 0, the non-snooping mode is used with
the traffic control bit is set in each stream control register.
This may allow better operations in the low power mode, but the actual
implementation is depending pretty much on the chipset.
As already implemented, more or less each chipset has own snoop-control
register bit. Now this setup refers to the snoop option, too.
Also, a new VIA chipset may require the non-snooping mode when set so
in BIOS. In such a case, the option value is overridden.
As default, it's still set to snoop=1 for keeping the same behavior as
before. In near future, it'll be set to 0 as default after checking
it works in every system well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since really_cleanup_stream() is called from both purity_inactive_streams()
and hda_cleanup_all_streams(), the verbs to clear the PCM channel and
format may be called multiple times unnecessarily.
This patch adds checks to skip these unneeded verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The spec->autocfg.line_out_pins[] may contain the same pins as hp_pins[]
depending on the configuration. When they are identical, detecting the
line_jack_present flag screws up the auto-mute because alc_line_automute()
is called unconditionally at initialization while it won't be triggered
by unsol events, thus the old line_jack_present flag is kept for the
whole run.
For fixing this buggy behavior, the driver needs to check whether the
line-outs are really individual, and skip if same as headphone jacks.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone pin is assigned as primary output to line_out_pins[],
the automatic HP-pin assignment by ASSID must be suppressed. Otherwise
a wrong pin might be assigned to the headphone and breaks the auto-mute.
Reference: https://bugzilla.novell.com/show_bug.cgi?id=716104
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
This patch is necessary to make internal speakers work on this chip.
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/854468
Tested-by: Alex Wolfson <alex.wolfson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add "AD198x Headphone" playback device for independent headphone playback
while playing 7.1 surround using rear panel audio jacks.
- Remove "6stack-dig-fp" model since "Headphone Playback Volume" control using
DAC0 instead of DAC1 (HDA_FRONT) was already added to all models.
- Add "Independent HP" switch to enable/disable this playback device.
When the switch is OFF, headphone use "copy front" mode to get the front
channel as the green jack.
When the switch is ON, you can play stereo sound through "AD198x Headphone"
device to headphone while playing 7.1 surround sound through "AD198x Analog"
device.
The switch cannot be changed when either "AD198x Headphone" or "AD198X Analog"
is open.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Smatch has a new check for Rosenberg type information leaks where
structs are copied to the user with uninitialized stack data in them.
The status struct has a hole in it, and on some paths not all the
members were initialized.
struct hdspm_status {
unsigned char card_type; /* 0 1 */
/* XXX 3 bytes hole, try to pack */
enum hdspm_syncsource autosync_source; /* 4 4 */
long long unsigned int card_clock; /* 8 8 */
The hdspm_version struct had holes in it as well.
struct hdspm_version {
unsigned char card_type; /* 0 1 */
char cardname[20]; /* 1 20 */
/* XXX 3 bytes hole, try to pack */
unsigned int serial; /* 24 4 */
short unsigned int firmware_rev; /* 28 2 */
/* XXX 2 bytes hole, try to pack */
int addons; /* 32 4 */
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9676001559
("ALSA: fm801: add error handling if auto-detect fails") seems to
break systems that were previously working without a tuner.
As a bonus, this should fix init and cleanup for the case where the
tuner is explicitly disabled.
Reported-and-tested-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9676001559
("ALSA: fm801: add error handling if auto-detect fails") added
incorrect error handling.
Once we have successfully called snd_device_new(), the cleanup
function fm801_free() will automatically be called by snd_card_free()
and we must *not* also call fm801_free() directly.
Reported-by: Hor Jiun Shyong <jiunshyong@gmail.com>
References: http://bugs.debian.org/641946
Signed-off-by: Ben Hutchings <ben@decadent.org.uk>
Cc: stable@kernel.org [v3.0+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit [e58aa3d2: genirq: Run irq handlers with interrupts disabled],
We run all interrupt handlers with interrupts disabled
and we even check and yell when an interrupt handler
returns with interrupts enabled (see commit [b738a50a:
genirq: Warn when handler enables interrupts]).
So now this flag is a NOOP and can be removed.
Signed-off-by: Yong Zhang <yong.zhang0@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since modern HDMI cards often have more than one output pin and thus
input device, we need to know which one has actually been plugged in.
This patch adds a name hint that indicates which PCM device is connected
to which pin.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Increase readability and understandability in the automute code.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCM rule to allow disabling the PCM playback SRC.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCM rule to allow disabling the PCM playback SRC.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the PCM rules to allow disabling the PCM playback and capture SRCs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The installation of the minimum period size constraint in the PCM open
callbacks was not checked for errors. Add this check, and move the call
to the beginning of the function to avoid having to do any cleanups in
the error case.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the system has only the headphone and the line-out jacks without
speakers, the current auto-mute code doesn't work. It's because the
spec->automute_lines flag is wrongly referred in update_speakers().
This flag must be meaningless when spec->automute_hp_lo isn't set, thus
they should be always coupled.
The patch fixes the problem and add a comment to indicate the
relationship briefly.
BugLink: http://bugs.launchpad.net/bugs/851697
Reported-by: David Henningsson <david.henningsson@canonical.com>
Tested-By: Jayne Han <jayne.han@canonical.com>
Cc: stable@kernel.org (3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes "Surround Speaker Playback Volume" being cut off.
(Commit b4dabfc452 was probably meant to fix this, but it fixed
only the "Switch" name, not the "Volume" name.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The semantics of snd_mpu401_uart_new()'s interrupt parameters are
somewhat counterintuitive: To prevent the function from allocating its
own interrupt, either the irq number must be invalid, or the irq_flags
parameter must be zero. At the same time, the irq parameter being
invalid specifies that the mpu401 code has to work without an interrupt
allocated by the caller. This implies that, if there is an interrupt
and it is allocated by the caller, the irq parameter must be set to
a valid-looking number which then isn't actually used.
With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value,
which forces us to handle the parameters differently.
This patch introduces a new flag MPU401_INFO_IRQ_HOOK for when the
device interrupt is handled by the caller, and makes the allocation of
the interrupt to depend only on the irq parameter. As suggested by
Takashi, the irq_flags parameter was dropped because, when used, it had
the constant value IRQF_DISABLED.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recursive search of widget connections in snd_hda_get_conn_index()
must be terminated at the pin and the audio-out widgets. Otherwise
you'll get "too deep connection" warnings unnecessarily.
Reported-by: Francis Moreau <francis.moro@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- use DAC0 instead of DAC1 for Port-A Headphone
- assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp"
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
YDSXGR_LEGACYOUTVOL is a Playback Volume control for OPL3 FM Synth.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headphones has stopped working for the original reported (a regression
compared to 2.6.38). This is because Speaker and Headphones share the
same DAC, in which case no Headphones volume control was created.
This patch fixes so that both Speaker and Headphones volume
controls are created in such scenario.
BugLink: http://bugs.launchpad.net/bugs/817943
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the multi-io jacks are available, parse them first and assign DACs
before parsing speakers and headphones. This allows a better chance of
surround I/O in some desktops and laptops with limited DACs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 23c09b0090
ALSA: hda - Support multiple speakers by Realtek auto-parser
changes the return value from alc_get_line_out_pfx(), and it breaks
the center/LFE mixer split check. The caller must test with a string
"CLFE" now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the pincfg table to patch_conexant.c for fixing up the extra
pin-configuration for auto-parser. As an example, Lenovo X200 model is
replaced with this new mechanism. (This also fixes the wrong mixer
elements for docking-station I/O in the previous model quirk
automagically.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are references in the code to 256 sources, so I tested it with 256 aplays,
of which the first and last with real data and the rest playing /dev/zero .
Also increase amount of page tables, so the default aplay size works.
Signed-off-by: Maarten Lankhorst <m.b.lankhorst@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly like ALC662 asus-mode* models, rewrite the laptop-amic and
dmic models with the static pin-config tables.
Now we can get rid of all alc269_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Re-implement the asus-mode[1-8] quirks with the pin-config tables.
They are provided in case where BIOS is broken on the device, so it's
not enabled in PCI SSID lookup table. User needs to specify it via model
option explicitly if the driver doesn't work with the BIOS setup as is.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For supporting both the multiple headphones and the multiple speakers,
add the new field in struct hda_multi_out, and evaluate in the standard
setup functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let's remove the rest of ALC861 and ALC861-VD quirks.
If any breakage is found, it can be fixed easily via the pin-config
table update.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the cleanup by commit 6727b12669,
the specific setups for dallas and hp models, using VREF50 for mic pins,
were lost. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... and add a new bit-flags argument to specify the behavior of the
function. The older function is kept as is (as a wrapper).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple headphones or speakers are assigned but no individual
DACs are available, the driver should take the first HP/SPK DAC instead
of another primary output. The patch adds a bit-flag to dac field of
struct pin_dac_pair indicating that it's a slave DAC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal states, jack_present and line_jack_present should be
updated upon unsolicited events even if no automute is set.
Otherwise the wrong state is referred when the automute behavior is
changed by the mixer control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone or speaker output has no own DAC, initialize the path
using the primary DAC. Otherwise the path won't be set properly and
can result in the silence.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_get_conn_index() returns a negative value while the current code
stores it in an unsigned int. It must be stored in a signed integer.
Reported-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently HD-audio driver shows the all error ELD byte as an error
in the kernel message. This is annoying when the video driver doesn't
set the correct ELD from the beginning. e.g. radeon sends a zero-byte
data, but we still check ELD with the fixed 128 byte as a workaround
for some broken devices, it spews 128-times errors.
For avoiding this, the driver aborts reading when the first byte is
invalid. In such a case, the whole data is certainly invalid.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_hda_parse_pin_def_config(), we checked the associated number
of speaker pins and accepts only one number exclusively. But many BIOS
seem to give different assoc number for surround speakers, thus we'd
better to accept all speaker pins no matter which assoc number, and sort
like done for the headphone pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of multiple speakers by Realtek auto-parser.
When all speaker pins have individual DACs, create each speaker volume
control. Otherwise, create a bind-volume control for all speaker outs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new parser may use "PCM" volume, but it was missing the vmaster
slave list, thus "Master" volume didn't control it.
Reference: https://bugzilla.kernel.org/show_bug.cgi?id=41342
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement new fixup entries for Quanta FL1 and Fujitsu Lifebook
specific COEF and pin configurations. Removed the model entries
from alc269_quirks.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the duplicated creation of capture-mixer elements for some static
ALC268 configurations. The capture mixers must be put to cap_mixer field
instead of mixers array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Refactoring the code using snd_pcm_hw_constraint_pow2() helper function.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AES32 supports the non-standard 128kHZ, and this is enabled only when
SNDRV_PCM_RATE_KNOT is set in hw.rates field.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Xonar Essence ST/STX, the connector J14 has been confirmed to be
a digital input, so enable it in the driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC260 HP models work with the BIOS auto-parser. Let's cut them off.
Also move alc260_hp_master_*() to alc262_quirks.c as these are still
referred from there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/826081
The original reporter needs 'Headphone Jack Sense' enabled to have
audible audio, so add his PCI SSID to the whitelist.
Reported-and-tested-by: Muhammad Khurram Khan
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Newer RME cards like RayDAT and AIO support 32 samples per period. This
value is encoded as {1,1,1} in the HDSP_LatencyMask bits in the control
register.
Since {1,1,1} is also the representation for 8192 samples/period on
older RME cards, we have to special case 32 samples and 32768 bytes
according to the actual card.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, hdspm_decode_latency is called several times, violating the
DRY principle. Given that we need to distinguish between old and new
cards when decoding the latency bits in the control register, introduce
hdspm_get_latency() to provide the required functionality.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On newer RME cards like RayDAT and AIO, the 8192 samples per period size
are no longer supported. Instead, setting all three bits of
HDSP_LatencyMask to one ({1,1,1}) now corresponds to 32 samples per
period.
To make this more obvious to future developers, let's reorder the array
according to their bit representation, starting at 64 ({0,0,0}) up to
4096 ({1,1,0}) and finally 32 ({1,1,1}).
Note that this patch doesn't change semantics.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On newer RME cards like RayDAT and AIO, the lower bound is 32 samples
per period in contrast to 64 samples as seen on older cards.
We hence lower period_bytes_min to 32 * 4. Four bytes per sample.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Older RME cards like MADI and AES support period sizes of 8192 samples.
The original hdspm driver already featured this value, apparently, it
was lost during the rewrite.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_azf3328_dbgcallenter is called at the very beginning of the function,
so it could be useful to call snd_azf3328_dbgcallleave at all exit points.
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In commit 45eebda7, it add new function stac_vrefout_set, but it
is only used in code between CONFIG_SND_HDA_POWER_SAVE macro, so
add the macro to avoid such warning:
sound/pci/hda/patch_sigmatel.c:676:12: warning: 'stac_vrefout_set' defined but not used
Signed-off-by: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_TEA575X is enabled by RADIO_SF16FMR2, but the latter one is
no PCI device. Since tea575x-tuner itself is independent from the board
bus type, the config should be moved out of SND_PCI dependency.
Reported-by: Randy Dunlap <rdunlap@xenotime.net>
Acked-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use kzalloc rather than kmalloc followed by memset with 0
This considers some simple cases that are common and easy to validate
Note in particular that there are no ...s in the rule, so all of the
matched code has to be contiguous
The semantic patch that makes this output is available
in scripts/coccinelle/api/alloc/kzalloc-simple.cocci.
More information about semantic patching is available at
http://coccinelle.lip6.fr/
Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_via.c:2087: warning: 'dac' may be used uninitialized in this function
Signed-off-by: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put the exception checks for io_type switch() for possible mistakes in
future. Also this shuts up annoying compile warnings.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new parameter to disable rounding of buffer/period sizes to
multiples of 128 bytes. This is more efficient in terms of memory
access but isn't required by the HDA spec and prevents users from
specifying exact period/buffer sizes. For example for 44.1kHz, a
period size set to 20ms will be rounded to 19.59ms.
Tested and enabled on Intel HDA controllers. Option is disabled by
default for other controllers.
Tested-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It works fine with auto-parser and now the digital mic workaround was
implemented in auto-parser fixup, let's drop the static model quirks for
these models.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The digital-mic unit on ASUS Eee PC gives PDM signals instead of the
normal stereo PCM, thus you can't record a mono stream from the stereo
stream as is; the summed stereo signal results in almost zero level, and
you'll hear only soft noise.
As a workaround, use ALC269-specific COEF to manipulate the dmic route
for mono, like used for ALC271x. This is implemented as a fix-up, thus
it works only with model=auto or without REALTEK_QUIRKS Kconfig.
Reported-and-tested-by: Pavel Roskin <proski@gnu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoids assigning possibly invalid address to pa, even if it
is never dereferenced.
Correct error response to reflect request object/function ids.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We leak the memory allocated to 'firmware' when we fail to
release_firmware() after a kmalloc() failure in hpi_dsp_code_open().
This patch should take care of the leak.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (430 commits)
[media] ir-mce_kbd-decoder: include module.h for its facilities
[media] ov5642: include module.h for its facilities
[media] em28xx: Fix DVB-C maxsize for em2884
[media] tda18271c2dd: Fix saw filter configuration for DVB-C @6MHz
[media] v4l: mt9v032: Fix Bayer pattern
[media] V4L: mt9m111: rewrite set_pixfmt
[media] V4L: mt9m111: fix missing return value check mt9m111_reg_clear
[media] V4L: initial driver for ov5642 CMOS sensor
[media] V4L: sh_mobile_ceu_camera: fix Oops when USERPTR mapping fails
[media] V4L: soc-camera: remove soc-camera bus and devices on it
[media] V4L: soc-camera: un-export the soc-camera bus
[media] V4L: sh_mobile_csi2: switch away from using the soc-camera bus notifier
[media] V4L: add media bus configuration subdev operations
[media] V4L: soc-camera: group struct field initialisations together
[media] V4L: soc-camera: remove now unused soc-camera specific PM hooks
[media] V4L: pxa-camera: switch to using standard PM hooks
[media] NetUP Dual DVB-T/C CI RF: force card hardware revision by module param
[media] Don't OOPS if videobuf_dvb_get_frontend return NULL
[media] NetUP Dual DVB-T/C CI RF: load firmware according card revision
[media] omap3isp: Support configurable HS/VS polarities
...
Fix up conflicts:
- arch/arm/mach-omap2/board-rx51-peripherals.c:
cleanup regulator supply definitions in mach-omap2
vs
OMAP3: RX-51: define vdds_csib regulator supply
- drivers/staging/tm6000/tm6000-alsa.c (trivial)
Apparently, there are multiple old firmware revisions in the wild for
the PCI RME MADI cards. Just add them to the list of supported devices
and treat them like their modern counterparts.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In slave mode, the card can only detect the base frequency (32..48kHz)
on the MADI link (exception: 96k frames), so the real external sample
rate is this base frequency multiplied by 1, 2 or 4 depending on the
speed mode.
This patch enables 64..192kHz sample rates in clock slave mode, which
failed before due to an alleged sample rate mismatch between the MADI
link (e.g., 48kHz) and the application in DS/QS mode (e.g., 96kHz,
192kHz).
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When running in slave mode (no clock master), there is no way to
determine the real wirespeed on the MADI link (single/double/quad
speed). Like physical gear, simply provide the user with a tristate
switch to select the appropriate format.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
sound: oss: rename local change_bits to avoid powerpc bitsops.h definition
ALSA: hda - Fix duplicated DAC assignments for Realtek
ALSA: asihpi - off by one in asihpi_hpi_ioctl()
ALSA: hda - Fix Oops with Realtek quirks with NULL adc_nids
ALSA: asihpi - bug fix pa use before init.
ALSA: hda - Add support for vref-out based mute LED control on IDT codecs
Convert radio-sf16fmr2 to use generic TEA575x implementation. Most of the
driver code goes away as SF16-FMR2 is basically just a TEA5757 tuner
connected to ISA bus.
The card can optionally be equipped with PT2254A volume control (equivalent
of TC9154AP) - the volume setting is completely reworked (with balance control
added) and tested.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Copying hp_pins and speaker_pins from line_out_pins may confuse the
parser, and it can lead to duplicated initializations for the same pin
with a wrong DAC assignment. The problem appears in 3.0 kernel code.
Cc: <stable@kernel.org> (for 3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"adapter" is used as an array index in the adapters[] array so
the off by one would make us read past the end.
1c073b6797 "ALSA: asihpi - Remove spurious adapter index check"
reverted Dan Rosenberg's check that would have prevented the
overflow here.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Somce quirk models don't set adc_nids but let the parser filling it.
But the recent code has unnecessary NULL-checks of spec->input_mux,
and it resulted in NULL dereferences.
This patch fixes that regression.
Reported-and-tested-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes bug introduced by 1c073b67.
Also declare pa local to block in which it is used.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch also registers all necessary callbacks to support mute LED
only when such control is enabled. And it keeps codec AFG in D0 or D1
state all the time when aggressive power managemnt is enabled for vref-out
control (and mute LED) work correctly.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows us to move duplicated code in <asm/atomic.h>
(atomic_inc_not_zero() for now) to <linux/atomic.h>
Signed-off-by: Arun Sharma <asharma@fb.com>
Reviewed-by: Eric Dumazet <eric.dumazet@gmail.com>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: David Miller <davem@davemloft.net>
Cc: Eric Dumazet <eric.dumazet@gmail.com>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This update includes the changes necessary for supporting the
CS421x family of codecs. Previously this file only supported
the CS420x family of codecs.
This file also contains init verbs to correct several issues in
the CS421x hardware.
Behavior between the CS421x and CS420x codec families is similar,
so several functions have been reused with "if" statements to
determine which codec family (CS421x or CS420x) is present.
Also, this file will be updated sometime in the near future in
order to add support for a system using CS421x that requires
mono mix on the speaker output only.
[Fix const usages and adaption for new APIs by tiwai]
Signed-off-by: Tim Howe <tim.howe@cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The procedure for codec D-state change may have exceptional cases
depending on the codec chip, such as a longer delay or suppressing D3.
This patch adds a new codec ops, set_power_state() to override the system
default function. For ease of porting, snd_hda_codec_set_power_to_all()
helper function is extracted from the default set_power_state() function.
As an example, the Conexant codec-specific delay is removed from the
default routine but moved to patch_conexant.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new ops, post_suspend(), which is called after suspend() ops is
performed. This is called only in the case of the real PM suspend, and
the codec driver can use this for further changing of D-state or
clearing the LED, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It makes little sense to enable power-saving without PM.
This removes SND_HDA_NEEDS_RESUME define so that we can use CONFIG_PM
in all places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds checking of mute state on all outputs besides just
speakers to calculate the master mute state for mute led support.
It also renames and splits the function that does it for better code
clarity.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Codec state is not restored immediately on resume but on the first
access when power-save is enabled. That leads to an invalid mute led
state after resume until either sound is played or some control is
changed. This patch adds a possibility for a vendor specific patch to
restore codec state immediately after resume if required. And it adds
code to restore IDT codecs state immediately on resume on HP systems
with mute led support.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (43 commits)
fs: Merge split strings
treewide: fix potentially dangerous trailing ';' in #defined values/expressions
uwb: Fix misspelling of neighbourhood in comment
net, netfilter: Remove redundant goto in ebt_ulog_packet
trivial: don't touch files that are removed in the staging tree
lib/vsprintf: replace link to Draft by final RFC number
doc: Kconfig: `to be' -> `be'
doc: Kconfig: Typo: square -> squared
doc: Konfig: Documentation/power/{pm => apm-acpi}.txt
drivers/net: static should be at beginning of declaration
drivers/media: static should be at beginning of declaration
drivers/i2c: static should be at beginning of declaration
XTENSA: static should be at beginning of declaration
SH: static should be at beginning of declaration
MIPS: static should be at beginning of declaration
ARM: static should be at beginning of declaration
rcu: treewide: Do not use rcu_read_lock_held when calling rcu_dereference_check
Update my e-mail address
PCIe ASPM: forcedly -> forcibly
gma500: push through device driver tree
...
Fix up trivial conflicts:
- arch/arm/mach-ep93xx/dma-m2p.c (deleted)
- drivers/gpio/gpio-ep93xx.c (renamed and context nearby)
- drivers/net/r8169.c (just context changes)
Formatting a PCM name is useful for module debug too.
Add snd_prefix when making function public.
[minor coding-style fixes by tiwai]
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a regression in the DAC filling code in patch_realtek.c. The already
filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0,
thus always pointed to the first DAC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HPI Version is used to check for firmware compatibility.
This version will accept 4.08.xx released firmware,
and will also accept 4.09.xx beta firmware
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add names corresponding to new HPI node types.
Shorten some names so that constructed names don't overflow the
maximum name length.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because mutex is used in adapter struct defined here.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Subsystem requests don't have or need a valid adapter index.
The adapter index is already checked further on, before it is used to index
the adapters array. (Reverts 4a122c10f)
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Work towards moving the function into alsa common header.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loader API has been revised so that OS specific data is kept
local to hpidspcd.c, and the public API is unchanged across OSes.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some cobranet control data would not fit in an original HPI message.
Now that HPI is able to transfer larger messages, this special handling
is no longer required.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow for up to 256 bytes of extra data on top of standard hpi
request and response sizes.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Having a 'request message' makes more sense than a 'message message'
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes non-working indep-HP control on VT1708* codecs.
The problems are that via_independent_hp_put() wasn't fixed to follow
the recent change of three HP paths, and hp_indep_path didn't contain
the amp nids of mixer elements.
Together with the fixes, a few code clean-ups are done.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the behavior of independent-HP enum switch. Now
instead of returning a busy error, the driver switches dynamically the
stream of the HP (and shared) DACs according to the current mode.
The logic is similar like the dual-mic ADC switch, but a bit more
complicated because of the presence of shared DAC.
Together with the change, a mutex is introduced to protect against the
possible races for the indep-HP mode setting.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the dynamic control of analog-loopback for VIA codecs.
When the loopback is enabled, the inputs from line-ins and mics are
mixed with the front DAC, and sent to the front outputs. The very same
input is routed to the headhpones and speakers in loopback mode.
However, since the loopback mix can't take other than the front DAC,
there is no longer individual volume controls for headphones and
speakers. Once when the loopback control is off, these volumes take
effect.
Since the individual volumes are more desired in general use caess, the
loopback mode is set to off as default for now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit dd203fa97b (ALSA: virtuoso: remove non-working controls on
Essence ST Deluxe) made it impossible to adjust the volume after the
driver initialized it to muted.
Ensure that those DACs that can be accessed with I2C are initialized
to the same volume that is the reset default of the DAC without I2C.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.38+ <stable@kernel.org>
BugLink: https://bugs.launchpad.net/bugs/774895
The original reporter states that his volume keys do not change the
desired Master and PCM mixer elements together, so apply the hp+mute led
quirk for his PCI SSID.
Reported-by: Jeffrey Finkelstein
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the rewrite, the check of spec->need_dac_fix and the corresponding
num_dacs change was dropped from the channel-mode control.
This patch re-adds it, and also enables need_dac_fix for ALC880 as default,
as this feature was originally introduced to fix h/w bugs of this chip.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit is a fix up for commit acfa634f.
commit acfa634f7e
Author: Takashi Iwai <tiwai@suse.de>
Date: Tue Jul 12 17:27:46 2011 +0200
ALSA: hda - Add Kconfig for the default buffer size
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a Kconfig entry to specify the default buffer size.
Distros using PulseAudio can choose a larger value here.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1718S and co have a secret connection from DAC to AA-mix, which
doesn't appear in the connection list obtained from the h/w.
Currently the driver fixes the connection index locally at init, but
now we can expose it statically via snd_hda_override_connections()
so that this conection can be checked better by the parser in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the codec proc outputs, read the raw connections instead of the
cached connection list, i.e. proc files contain only raw values.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a function to add/modify the connection-list cache entry.
It'll be useful to fix a buggy hardware result.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some machines seem to use EAPD control of the unused pin for controlling
the overall EAPD. Since the driver currently doesn't check the EAPD of
unused pins, the EAPD isn't enabled. For avoiding such a problem, turn
all extra EAPDs on as default.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For smart51 pins, we need to preserve the input pin-control bits at
auto-mute controls instead of overwriting zero or pin-out-only.
Otherwise the VREF won't be set properly when smart51 is disabled
again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When Independent-HP mode is changed for VIA, the driver needs to
re-issue the auto-mute check so that the line-out pins are set properly
without influence of HP pin state.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the line-jack is plugged/unplugged, the driver must check also
the headphone jack state in addition to the line-out jack. Currently
it checks only the line-out state and ignores the headphone.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of checking the model quirk, use a fixup table for workaround
of 44kHz-fixed PCM for Lenovo IdeaPad with ALC269.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's harmless but annyoing.
sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now all alc*_parse_auto_config() do almost same thing except for the
NID list to ignore and the PINs for SSID-check, we can merge all these
to a single function. A good amount of code reduction.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One more code reduction. This codec has less DACs, thus the wiring
to DAC can't be filled uniquely for all output pins, i.e. some outputs
share the same volume control.
Except for that, all seems working fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge more auto-parser code in patch_realtek.c, now for ALC861.
The topology of this codec is pretty simple, and can be parsed well
by the current starndard parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
query_amp_caps() may return non-zero if the amp cap isn't supported
by the codec. Thus one needs to check widget-caps first, then check
the corresponding amp-caps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A regression fix from commit 21268961d3
ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs
The auto-mic wasn't detected properly when no ADC-switch is needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, to create Independent HP
control.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, there're only two DACs. So smart51
control shouldn't be created.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, the original activate_output_path()
function can't initialize output and hp path correctly, since mixers connected to
output pin widgets are not considered. So modify the activate_output_path()
function to satisify this kind of codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put the all static quirk codes out of patch_realtek.c, split into the
file for each codec model. For controlling the build of quirk codes,
a new Kconfig, CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS is introduced.
By setting this off, all quirk codes won't be built, thus you can save
lots of memory.
The codes in patch_realtek.c are also shuffled and more comments are
given, but the contents aren't changed. This is just a refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the auto-parser and the auto-mic handling codes to
allow more flexible dynamic ADC-switching with Realtek codecs.
In the new code, the following strategy is taken:
- When a cap-src can't handle all input-sources, either skip it, or
switch to the ADC-switching mode. In ADC-switching mode, like the
former dual-ADC mode for ALC275, it changes ADC on the fly according
to the current input source.
- When auto-mic is possible, always assign imux. If the mic pins are
set statically via a quirk, rebuild imux according to the pins.
In the auto-mic mode, the driver always changes the imux (although
the imux isn't exposed as a mixer element).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of assigning each default hda_pcm_stream pointers, do NULL-checks
and assign default values in alc_build_pcms().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only different implmentation was alc880_auto_init_input_src(),
and now it covers this variant, and we can use the single function
for all codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now with the new code for looking for ADCs and MUXs, we can replace
the whole ADC assignment with the parsed results.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All alc*_auto_init_analog_input() calls are identical, so let's use
the same function more clearly without aliases.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Parse ADCs and cap-srcs in alc_auto_create_input_ctls() by itself
instead of passing explicitly from the caller. By this change, all
alc*_auto_create_input_ctls() can be unified to the same calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the same common code for auto-parsing the output paths and their
initializations, based on the existing ALC662 code, which is smarter
than the old ALC880/2 code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple inputs are present on the mixer widget (typically a DAC
and a loopback), mute/unmute both inputs with the corresponding mixer
element.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In alc662_auto_fill_dac_nids(), the HP and speaker DACs aren't parsed
when the corresponding pins aren't fixed with single DACs.
Now check these DACs even for non-fixed pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the dual-adc switching mode is active in Realtek auto-parser,
we need to couple all ADCs as a single capture-volume. Currently, the
volume control changes only the first ADC, thus others may remain silent.
This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of ALC269VC codec.
Also delete the unnecessary codec_variant type enum list:
now only three variants (ALC269VA ALC269VB ALC269VC) are needed.
In addition, added some aliases:
- Add ALC269VB alias name ALC277
- Add ALC269VC alias name ALC259 ALC281X
- Add ALC269VC for Lenovo device 0x21f3 name ALC3202
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we now route the front DAC via aa-mix widget, adding the aa-mix
to surrounds will result in a mix-up of both front and surround PCM
signals. For avoiding this, the aa-mix routes have to be disabled
for surround paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the individual DAC is available for the headphone output, the driver
should create the DAC for its volume control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, the multi-channel path should be like following:
DAC 0-->Mixer 9(index 5)-->Mixer 0(index 1)-->Front Pin;
DAC 1-->Mixer 1(index 0)-->Surround Pin;
DAC 2-->C/LFE Pin;
DAC 3-->Mixer 2(index 0)-->Side Pin;
But current code built Surround and Side path through index 1 of
Mixer 1 and 2. So Adjusting Surround and Side channel amplifier is
invalid. This patch fixes the issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, Mixer 9 doesn't expose the connection to DAC 0. So when
building up a 'PCM Playback' amplifier control, it will fail since
getting DAC 0 index of Mixer 9 returned -1. So I added a dac_mixer_idx
to indicated the actual index of DAC 0 to Mixer 9. Following is the
patch and next mail is another.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unmute DAC on front speaker path when Independent HP is enabled.
When to enable Independent HP, the front speaker won't output any sound
for VT1708, VT1708B, VT1708S and VT1702.
I find the via_independent_hp_put() routine will mute DAC 0 path in Mixer 0.
For these codecs, when using Independent HP, there could have two
independent streams, one is from DAC0-->Mixer0-->Front Pin, the other is
from DAC3-->GainSW3-->Side Pin.
So I added a check for DAC-->Mixer path in activate_output_path().
If current path is DAC-->Mixer, no need to mute DAC index in Mixer.
In fact, to change connection of Headphone pin or Mux connected with HP
is enough.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the re-implementation of Independent-HP mode in the
case where the DAC is shared between HP and side-channel streams.
Now the driver tries to parse the output-path using the pre-parsed
side-channel DAC for the independent HP output, too.
When a playback PCM stream is opened with this shared mode, the
Independent-HP mixer switch can't be changed for avoiding the conflict,
thus it returns -EBUSY error.
One remaining unintuitive issue is that the DAC volume is still
controlled as "Side" volume although it's shared by both independent-HP
and side streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The char can be unsigned on some architectures. Since the code checks
the negative values, they should be declared as signed char explicitly.
sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type
sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrongly converted short values:
sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type
sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I noticed that the last character of the ELD monitor name is lost,
this fixes the issue.
This fix should be confirming to the HDA spec, and works together with
the DRM part of the ELD patch.
The HDA spec does not mention that Monitor_Name_String is an '\0'
ending string, and it allows NML to be 1, which is only valid when MNL
does not count the possible ending '\0'.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move codec init verb which is only applicatable for VT1708.
I've found the root cause that jack plugged in can't be detected.
The verb in vt1708_init_verbs is used to power down jack detect circuit.
This verb is only applicable to VT1708. vt1708 didn't implement jack
detect function in hardware, so we should shut down this function to
avoid noise. But for other codecs, hardware implement jack detect
function. If sending this verb during initialization, jack detect will
be invalid. So I move this verb from via_parse_auto_config() to
patch_vt1708().
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create snd_hda_get_conn_index() helper function for obtaining the
connection index of the widget. Replaced the similar codes used in
several codec-drivers with this common helper.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to enable unsolicited response to line-out pins.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we're not using the new auto parser as a fallback yet,
add it manually as a quirk.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant ID 506c was found on Acer Aspire 3830TG. As users report
no playback, sending to stable should be safe.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/783582
Reported-by: andROOM
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use static inline for dummy function to fix the warnings like below
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_init’:
sound/pci/hda/patch_sigmatel.c:4387:3: warning: statement with no effect
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_resume’:
sound/pci/hda/patch_sigmatel.c:4927:3: warning: statement with no effect
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More similar fixes like previous commits: handle the exceptional case
like ALC267 where no volume amp is found in ADC widget but in the
capsrc widget instead.
Also minor checks for avoiding possible erros: no connection-select
when the pin has a single selection, and add beep verbs only when the
0x1d is used for beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC259 seems to provide an invalid widget capability for the input-src
selector widget. The widget shows the input-amp while it's a selector,
and this confuses the current ALC882 initialization code that is used
for ALC259, too. For fixing this, check the amp capability and handle
the connection selection individually.
Also, ALC259 has no mute bit in DAC volume, so we need to initialize
it as ZERO instead of MUTE.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 and compatible codecs have the output volume in DACs, thus we
can't use the ALC880's code as is. Fixed by checking the amp caps and
picking up the right widget for initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Look through the known NIDs that may have EAPD capabilities and turn
on/off them appropriately instead of checking the individual vendor ids.
This will also avoid the forgotten entries of newly added codec ids
in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the regression introduced by the commit
1f0f4b8036
ALSA: hda - Reduce static init verbs for Realtek auto-parsers
The input amps of mixer widgets should be unmuted as default (as
usually they have no assigned mixer switches).
More fixes in this commit are, however, for ALC260: ALC260 codec can
have multiple output mixers connnected to a single DAC althouh the
driver didn't pick up them properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Realtek codecs have the output pins hardwired with certain DACs.
These DACs have to be assigned at first and assign the rest for
multi-DAC pins so that all DACs can be assigned properly.
Without such an optimization, speaker outputs may be assigned to the
same DAC as the headphone or others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In alc_auto_add_multi_channel_mode(), when the primary HP workaround
is enabled, it re-initializes the DAC list but calls alc662's function
in a fixed way. This isn't pretty suitable for other codecs, of course.
Now we call it with fill_dac function pointer so that the proper
function can be called at that point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using fixed init verbs, initialize DACs, ADCs and mixers
more dynamically for Realtek auto-parsers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper function snd_hda_get_conn_list().
Unlike snd_hda_get_connections(), this function doesn't copy the
connection-list but gives the raw pointer for the cached list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
each device has a unique mac address, which can be used to distinguish
multiple devices in the same machine. we therefore include the full mac
address in the device shortname and the last 6 bytes in the device id.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the sound/ directory there are two files (flagged by 'make
versioncheck'); sound/pci/asihpi/asihpi.c and
sound/soc/codecs/wm8991.c that include linux/version.h although they
don't need it. This patch removes the unneeded includes.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change alc_get_line_out_pfx() in patch_realtek.c to provide the channel
specific name and assign the index so that each caller doesn't have to
set the channel name by itself.
Also, check the multi-io case with the primary hp-out; for the multi-io
channels, assign the channel name instead of "Headphone" with indices.
This makes the mixer names more intuitive and reduces confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some BIOS set up the pin config wrongly as line-out although it's
supposed to be a speaker out. In most cases, though, we can judge
the validity by checking the connection type -- when it's FIXED,
mostly it's an invalid line-out but a speaker.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even if the machine has no line-out but only HP-out, try to detect the
multi-io. It'll allow more possibilities for 5.1 outputs on laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS lists the internal speaker as an internal line-out. Change to
internal speaker + model=auto for better auto-mute capabilities.
BugLink: http://bugs.launchpad.net/bugs/754964
Reported-by: Marc Legris <marc.legris@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some VIA codecs like VT1702 provide the input-route only to specific
ADCs such as digital-mic inputs. These routes aren't covered by the
normal primary ADC, and for now, user had to open the capture stream
assigned to that special ADC manually for using such inputs.
This patch implements a way to switch the current ADC dynamically per
the input-source selection in such a case. When this workaround is
activated, the driver provides only one capture stream and one input-
source control but with the full possible inputs. The driver switches
the ADC to be used (or being used) according to the input-source on the
fly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When smart51 mode is enabled, auto-mute these surround outputs
as well as the primary line-out. Also this patch includes minor
clean-ups.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unify the VT1709 10ch and 6ch parsers, as well as VT1708B 8ch and 4ch
parsers. They have no difference now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codecs like VT1708 needs more complicated routing using the mixer
widget rather than the simple selector widgets.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The surround/CLFE/side DACs on VT1708B and co have no amp but the
connected selector widgets have the amp instead. Fix the parser to
check these selector widgets for the possible mixer controls as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the check of the multiple loopback-mixer, which gave sometimes
a wrong index assigned to an element even for different names, e.g.
Mic and Front Mic. Now check the label properly for avoid duplication.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input jacks assigned as the smart51 outputs must be in the same
stack, either rear, front or other. Also, prefer line-in as the surround
to mic-in.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to create playback volume control if pin has amplifier capability
but not DAC.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the order of the output-path list in a way from the DAC to the
target pin. Also now the list include the target pin, too.
Together with this format change, simplify the arguments of
parse_output_path() function, and fix the initialization in
via_auto_init_output().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop "Capture" prefix from the mic-boost names.
Otherwise some control names can overflow the max name length.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create patch_ca0132.c, to add support for devices featuring the
Creative CA0132 HD-audio codec.
This driver implements :-
* 1 playback subdevice to headphone and speaker
* 2 capture subdevices:
i - Mic-in
ii- Line-in
* mixer device
Advanced DSP features are not yet included.
Developed and maintained by Creative Labs, Inc.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a master volume and mute control of playback for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When switch HP independent mode, mute/unmute connctions of mixer which is
connected to headphone for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some invalid initial verbs and correct some wrong initial verbs
for VT1718S codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER.
This keeps headphone automute and microphone input from operating
on at least one laptop from Opti Systems.
Without the override, the BIOS parser does a fine job setting the
card up and everything works.
Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com>
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reporter, who is running kernel 2.6.38, reports that
he needs to set model=auto for the headphone output to work
correctly.
BugLink: http://bugs.launchpad.net/bugs/761022
Cc: stable@kernel.org (v2.6.38+)
Reported-by: Jo
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the existing aa-loop list for simplifying the check for analog
low-current mode. Also fix the stream count test for playback streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Issue the init verbs of unsolicited events dynamically from the parsed
results for VIA codecs. Also, consolidate the unsol handlers for HP
and line-out mutes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly like the previous commit, initialize the input-paths dynamically
from the parsed results instead of the fixed array for VIA codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of fixed array for each codec type, initialize the output path
dynamically from the parsed results.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix races in handling of HP DAC and independent streams for VIA codecs.
Also, allow the HP output path without front-DAC, and removed
unnecessary activation of HP mixer elements.
This also removes the handling of shared side/HP stream; it's anyway
implemented in a broken way, so we need to re-implement the feature
later...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of ignoring the invalid pin configuration, return the error.
This will avoid unexpected crash, anyway.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create capture-related mixer elements dynamically from the parsed
ADCs and input-pins instead of fixed values for each codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using the secondary substream, create an individual PCM
stream for HP-independent PCM. Otherwise it's difficult to handle
different channel numbers with multi-channel stream in the sam PCM
stream structure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VIA codecs, we shouldn't create a substream for independent HP mode,
when no individual HP DAC is found.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Parse the output-paths more dynamically, i.e. traverse the paths
from each output pin instead of fixed assignment for each codec.
Now all codecs are using the same output parser code.
The smart51 setup doesn't work with this change, and will be fixed
in the next commits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mute the outputs via pin-controls instead of amps for the auto-mute
handling. This makes our life easier as it avoids conflict of the states
between the mixer elements and the auto-mute toggles.
With this change, we can use vmaster for the master control easily now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The jack-detect control should be created at the time of build_controls
callback instead of calling snd_hda_add_ctls() at the tree-parsing time.
For that, copy the control to the temporary array like other cases.
Also, fixed typos of vt1708_jack_detect in all places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of giving the fixed ADC list, parse the widgets and fill in
ADCs dynamically.
Also, probe the stereo-mixer input more dynamically, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently VIA driver controls the power-state of each pin per jack
detection. But, it means that the power-state mismatch may occur when
the machine doesn't give the proper jack-detection.
For avoiding this problem, a new control element "Dynamic Power-Control"
is provided so that user can turn on/off the pin-power control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have changed the position_fix default for ATI and AMD
to be LPIB (see commit 50e3bbf989), we can remove the quirks that
were added for ATI chipsets.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The via driver spews warnigs like
hda-codec: no NID for mapping control Independent HP:0:0
with some codecs because snd_hda_add_nid() is called with nid=0.
This patch fixes it by skipping the call when no corresponding widget
is found.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since commit f2b3614cef (Don't check DMA time-out too shortly),
drivers need no longer restrict their PCM period length to be shorter
than 10 seconds.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a reworked patch from Creative to change the PLL code to address
unreliable 44100Hz initialization.
Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialise model-specific DAC and ADC parts.
Add controls for output and mic source selection.
Rename some mixer controls according to ControlNames.txt.
Remove Playback switches for Line-in and IEC958-in - these
were controlling the input mute/unmute which affected
capture too. Use the capture switches to control the
input mute/unmute instead - it's less confusing.
Initialise the WM8775 to invert the left-right clock
to swap the left and right channels of the mic and aux
input.
Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If given a -1 cmd parameter then make_exec_verb() returns -1 without
setting the res output value.
Prior to this change snd_hda_codec_read() assumed that make_exec_verb()
unconditionally set res regardless of the cmd value.
This change explicitly checks the make_exec_verb() return value before
consuming the potentially unset res value.
Signed-off-by: Greg Thelen <gthelen@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using static inline functions can reduce compilation messages
and macro misuse.
sound/pci/hda/patch_conexant.c: In function ‘patch_cxt5045’:
sound/pci/hda/patch_conexant.c:1232:3: warning: statement with no effect
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-mute setup for Acer Aspire-one with ALC268 was set wrongly
during the clean-up of auto-mute function. Fixed now.
Tested-by: Borislav Petkov <bp@alien8.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the necessary details to support the PCIe version of
E-MU's 0404 card.
From comparing the PCBs it seems the PCIe version just added a PCIe
chipset and left all other components pretty much in place.
For anyone intrigued to take a look at the PCB there are pictures I took
at <http://babelmonkeys.de/~florob/E-MU%200404/>.
Signed-off-by: Florian Zeitz <florob@babelmonkeys.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI version of the RME HDSP MADI card uses 0xcf as revision ID. Just
add this to the list of supported cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When using Word Clock on RME MADI cards, AutoSync mode was alternating
betweeen MADI and WC due to a typo: AutoSync is indicated in the second
status register (status2), not the first one (status).
While the proc output was always correct, the reported WC frequency to
ALSA was unstable as mentioned in
http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006723.html
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the MIDI part, we need to acquire (and release) the hmidi->lock,
access to the global hdspm structure is serialized through
hmidi->hdspm->lock instead.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The name argument of request_irq() appears in /proc/interrupts, and
it's quite ugly when the name entry contains a space or special letters.
In general, it's simpler and more readable when the module name appears
there, so let's replace all entries with KBUILD_MODNAME.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/761171
The original reporter needs the model=auto quirk for his internal
speakers to be audible in the latest daily snapshot, so add an entry in
the quirk table for his PCI SSID.
A trivially different version of this patch using the model=asus quirk
should be applied to the 2.6.38 and 2.6.39 stable kernels. We don't use
the asus quirk in 3.0-rc2, because 3.0-rc2's autoparser is much
improved.
Reported-and-tested-by: tomdeering7
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The convention for pci_driver.name entry in kernel drivers seem to be
the module name or equivalent ones. But, so far, almost all PCI sound
drivers use more verbose name like "ABC Xyz (12)", and these are fairly
confusing when appearing as a file name.
This patch converts the all pci_driver.name entries in sound/pci/* to
use KBUILD_MODNAME for more unified appearance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Reatlek model quirks use master_mute bool switch for controlling
the master-mute of outputs. For these cases, the initialization of HP
pins/amps were forgotten during the transition to the common automute
helper function in 3.0 development time, and resulted in the muted HP
output as default.
This patch fixes the issue by adjusting the HP output explicitly with
master_mute switch.
Tested-by: Michal Hocko <mhocko@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The tag number was forgotten to be fixed after cleaning up the model
quirks for ALC262 fujitsu and lenovo-3000 models.
Tested-by: Michal Hocko <mhocko@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SSYNC register was once defined as 0x34-37 in the old Intel datasheet,
but corrected later to 0x38-3b. For fixing the register usage, a new
bit-flag is introduced for indicating the old ICH SSYNC register, and
ICH* PCI entries are added explicitly to enable this quirk.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some HP laptops with AD1981 have SPDIF connections, but currently the
driver disables it statically. Better to check the pin default config
to judge whether to enable or disable the SPDIF.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of checking the azx_dev index with a fixed number (4), check
the stream direction of the assigned substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When reading from the position-buffer results in -1, handle as it's
invalid and falls back to LPIB mode as well as 0.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/792712
The original reporter states that sound from the internal speakers is
inaudible until using the model=auto quirk. This symptom is due to an
existing quirk mask for 0x102802b* that uses the model=dell quirk. To
limit the possible regressions, leave the existing quirk mask but add
a higher priority specific mask for the reporter's PCI SSID.
Reported-and-tested-by: rodni hipp
Cc: <stable@kernel.org> [2.6.38+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The general concept of this change is to create a PCM device for each
pin widget instead of each converter widget. Whenever a PCM is opened,
a converter is dynamically selected to drive that pin based on those
available for muxing into the pin.
The one thing this model doesn't support is a single PCM/converter
sending audio to multiple pin widgets at once.
Note that this means that a struct hda_pcm_stream's nid variable is
set to 0 except between a stream's open and cleanup calls. The dynamic
de-assignment of converters to PCMs occurs within cleanup, not close,
in order for it to co-incide with when controller stream IDs are
cleaned up from converters.
While the PCM for a pin is not open, the pin is disabled (its widget
control's PIN_OUT bit is cleared) so that if the currently routed
converter is used to drive a different PCM/pin, that audio does not
leak out over a disabled pin.
We use the recently added SPDIF virtualization feature in order to
create SPDIF controls for each pin widget instead of each converter
widget, so that state is specific to a PCM.
In order to support this, a number of more mechanical changes are made:
* s/nid/pin_nid/ or s/nid/cvt_nid/ in many places in order to make it
clear exactly what the code is dealing with.
* We now have per_pin and per_cvt arrays in hdmi_spec to store relevant
data. In particular, we store a converter's capabilities in the per_cvt
entry, rather than relying on a combination of codec_pcm_pars and
the struct hda_pcm_stream.
* ELD-related workarounds were removed from hdmi_channel_allocation
into hdmi_instrinsic in order to simplifiy infoframe calculations and
remove HW dependencies.
* Various functions only apply to a single pin, since there is now
only 1 pin per PCM. For example, hdmi_setup_infoframe,
hdmi_setup_stream.
* hdmi_add_pin and hdmi_add_cvt are more oriented at pure codec parsing
and data retrieval, rather than determining which pins/converters
are to be used for creating PCMs.
This is quite a large change; it may be appropriate to simply read the
result of the patch rather than the diffs. Some small parts of the change
might be separable into different patches, but I think the bulk of the
change will probably always be one large patch. Hopefully the change
isn't too opaque!
This has been tested on:
* NVIDIA GeForce 400 series discrete graphics card. This model has the
classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM
audio to a PC monitor that supports audio.
* NVIDIA GeForce 520 discrete graphics card. This model is the new
1 codec n converters m pins m>n model. Tested stereo PCM audio to a
PC monitor that supports audio.
* NVIDIA GeForce 400 series laptop graphics chip. This model has the
classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM,
multi-channel PCM, and AC3 pass-through to an AV receiver.
* Intel Ibex Peak laptop. This model is the new 1 codec n converters m
pins m>n model. Tested stereo PCM, multi-channel PCM, and AC3 pass-
through to an AV receiver.
Note that I'm not familiar at all with AC3 pass-through. Hence, I may
not have covered all possible mechanisms that are applicable here. I do
know that my receiver definitely received AC3, not decoded PCM. I tested
with mplayer's "-afm hwac3" and/or "-af lavcac3enc" options, and alsa a
WAV file that I believe has AC3 content rather than PCM.
I also tested:
* Play a stream
* Mute while playing
* Stop stream
* Play some other streams to re-assign the converter to a different
pin, PCM, set of SPDIF controls, ... hence hopefully triggering
cleanup for the original PCM.
* Unmute original stream while not playing
* Play a stream on the original pin/PCM.
This was to test SPDIF control virtualization.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change won't store an entire hda_pcm_stream just to represent
the capabilities of a codec; a custom data-structure will be used. To
ease that transition, modify hdmi_eld_update_pcm_info to expect the
hda_pcm_stream to be pre-initialized with the codec's capabilities, and
to update those capabilities in-place based on the ELD.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change will significantly rework the generic implementation
in order to support codecs with a different number of pins and
converters. Isolate the more custom codec variants from this change by
duplicating the small portions of generic code they share. This
simplifies the later rework of that previously shared code, since we
don't have to consider the more custom codecs, and also prevents
support for those codecs from regressing.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SPDIF output controls apply to converter widgets. A future change
will create a PCM device per pin widget, and hence a set of SPDIF output
controls per pin widget, for certain HDMI codecs. To support this, we
need the ability to virtualize the SPDIF output controls. Specifically:
* Controls can be "unassigned" from real hardware when a converter is
not used for the PCM the control was created for.
* Control puts only write to hardware when they are assigned.
* Controls can be "assigned" to real hardware when a converter is picked
to support output for a particular PCM.
* When a converter is assigned, the hardware is updated to the cached
configuration.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, the data that backs the kcontrols created by
snd_hda_create_spdif_out_ctls is stored directly in struct hda_codec. When
multiple sets of these controls are stored, they will all manipulate the
same data, causing confusion. Instead, store an array of this data, one
copy per converter, to isolate the controls.
This patch would cause a behavioural change in the case where
snd_hda_create_spdif_out_ctls was called multiple times for a single codec.
As best I can tell, this is never the case for any codec.
This will be relevant at least for some HDMI audio codecs, such as the
NVIDIA GeForce 520 and Intel Ibex Peak. A future change will modify the
driver's handling of those codecs to create multiple PCMs per codec. Note
that this issue isn't affected by whether one creates a PCM-per-converter
or PCM-per-pin; there are multiple of both within a single codec in both
of those codecs.
Note that those codecs don't currently create multiple PCMs for the codec
due to the default HW mux state of all pins being to point at the same
converter, hence there is only a single converter routed to any pin, and
hence only a single PCM.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's perfectly valid for an ELD to contain no SADs. This simply means that
only basic audio is supoprted.
In this case, we still want to limit a PCM's capabilities based on the ELD.
History:
* Originally, ELD application was limited solely by sad_count>0, which
was used to check that an ELD had been read.
* Later, eld_valid was added to the conditions to satisfy.
This change removes the original sad_count>0 check, which when squashed
with the above two changes ends up replacing if (sad_count) with
if (eld_valid).
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the original code if auto detect failed and tea575x_tuner == 4
then we copy bogus information to chip->tea.card. I've changed the
autodetect code to cleanup and return -ENODEV on error instead.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ad198x_power_eapd(), wrong pin NIDs are used for controlling EAPD for
HP and Front outputs of AD1988/AD1989. These are actually same with the
ones for AD1984 & co, port-A is 0x11 and port-D 0x12.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 9477c58e33 ("ALSA: hda - Reorganize controller quriks with bit
flags") changed the driver type compares into various quirk bits.
However, the check for AZX_DCAPS_NO_TCSEL got reverted: instead of
clearing TCSEL for chipsets that have that standard capability, it
cleared then when the NO_TCSEL bit was set.
This can lead to noise and repeated sounds - a weird "echo" behavior.
As the comment just above says: "Ensuring these bits are 0 clears
playback static on some HD Audio codecs". Which is definitely true at
least on my Core i5 Westmere system.
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Introduce bit-flags indicating the necessary controller quirks, and
set them in pci driver_data field. This simplifies the checks in the
driver code and avoids the pci-id lookup in different places.
Also, this patch adds the PCI ID entry for AMD Hudson. AMD Hudson
requires a similar workaround like ATI SB while other generic ATI and
AMD controllers don't need but some ATI-HDMI quirks. So, we need a
different entry for Hudson.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the wrong usage of snd_printdd() for debug prints of input
entries. It should be snd_printd() like others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tim Blechmann