This patch adds two fields to unsolicited response, according to spec HDA040-A:
- Device Entry (bit 20:15)
- Inactive (bit 2)
and show the info in debug message.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is only to allow codec proc file to expose devices list/select info
for Haswell codec pins.
Since Haswell Gfx driver cannot support DP1.2 MST now, so all pins' device list
is empty, meaning no pin is multi-streaming capaple.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a display codec supports multi-stream transport on the pins, the pin's
device list length and device entries will be exposed to codec proc file.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds flags and routines to get device list & selection info on
a pin.
To support Display Port 1.2 multi-stream transport (MST) over single DP port,
a pin can support multiple devices. Please refer to HD-A spec Document Change
Notificaton HDA040-A.
A display audio codec can set flag "dp_mst" in its patch, indicating its pins
can support MST. But at runtime, a pin may not be multi-streaming capable and
report the device list is empty, depending on Gfx driver configuration.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using 0x%# emits 0x0x. Only one is necessary.
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
As reported by kbuild test robot <fengguang.wu@intel.com>:
warning: (SND_ES1968_RADIO && SND_FM801_TEA575X_BOOL) selects RADIO_TEA575X which has unmet direct dependencies (MEDIA_SUPPORT && RADIO_ADAPTERS && VIDEO_V4L2)
That happens because a radio driver is selected, without selecting the
RADIO_ADAPTERS menu.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
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Merge tag 'asoc-v3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.12
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
Without the dynamic minor assignment, HDMI codec may have less PCM
instances than the number of pins, which eventually leads to Oops.
Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixing warning message:
sound/pci/rme96.c: In function ‘snd_rme96_resume’:
sound/pci/rme96.c:2418:19: warning: ignoring return value of ‘pci_enable_device’, declared with attribute warn_unused_result [-Wunused-result]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The headphone automute on this machine triggers annoying pop noises.
It seems that only the first DAC can be used, the secondary DAC always
results in this problem. This patch disables the secondary DAC with
a few additional workarounds.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed ALC283 D3 to D0 and D0 to D3 Headphone pop noise.
The previous fix [c5177c86: ALSA: hda - Fix the noise after suspend on
ALC283 codec] doesn't work sufficiently for some laptops.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without proper power management handling, the first use
of a Digi96/8 anytime after a suspend / resume cycle will
start playback with distortions.
v3: Abort if vmalloc() of suspend buffers fail, but do not
leak memory in that case.
[fixed wrong memory leak fix again -- tiwai]
Signed-off-by: Knut Petersen <Knut_Petersen@t-online.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch doesn't change functionality, it only improves readability
and fixes a copy&paste error in a comment.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use enum hdspm_ltc_format's fps_30 (corresponds to 4) instead of 30,
Other case branches return 1, 2 or 3 respectively, so 30 obviously is
wrong.
Since SNDRV_HDSPM_IOCTL_GET_LTC had never been working due to a
copy&paste error in hdspm.h, this change doesn't break userspace.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another entry, just use the existing fixup for this machine, too.
Reported-by: "Nathanael D. Noblet" <nathanael@gnat.ca>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If compiled without CONFIG_SND_HDA_I915, the audio driver cannot
request power well. However, if the power well is on for other
reasons, maybe audio can still work. Therefore, do not skip the
card completely if compiled without CONFIG_SND_HDA_I915.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The higher mic boosts (on internal mic) are so noisy they're unusable
in practice.
BugLink: https://bugs.launchpad.net/bugs/1213820
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move tea575x from sound/i2c/other to drivers/media/radio
Includes Kconfig changes by Hans Verkuil.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
Move include/sound/tea575x-tuner.h to include/media/tea575x.h and update files that include it.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <m.chehab@samsung.com>
This just cleans up the table, no functional changes.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal mic boost is so noisy on boosts 2 and 3 so they are
unusable in practice.
BugLink: https://bugs.launchpad.net/bugs/1213055
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Gateway LT27 needs a fixup for the inverted digital mic.
Reported-by: "Nathanael D. Noblet" <nathanael@gnat.ca>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The hardware does support synchronized start/pause/stop of pcm streams,
so there is no reason not to add that feature after more than ten years.
Some minor coding style / white space fixes in the surroundings of the
changes.
Signed-off-by: Knut Petersen <Knut_Petersen@t-online.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current generic parser code assumes that always a pin widget
controls the mute for an output blindly although it might be a
different widget in the middle. Instead of the fixed assumption,
check each parsed path and just pick up the right widget that has been
already defined as a mute control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-mute using the amp currently works only for a single amp on a
pin. Make it working also with HDA_CTL_BIND_MUTE type, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've added a fake mute control (setting the amp volume to zero) for
CX5051 at commit [3868137e: ALSA: hda - Add a fake mute feature], but
this feature was overlooked in the generic parser implementation. Now
the driver lacks of mute controls on these codecs.
The fix is just to check both AC_AMPCAP_MUTE and AC_AMPCAP_MIN_MUTE
bits in each place checking the amp capabilities.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59001
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct the pins for a line-in and a headphone on LG LW25 laptop with
ALC880 codec. Other pins seem fine.
Reported-and-tested-by: Joonas Saarinen <jonskunator@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_hda_jack_set_gating_jack() call didn't work when
auto_{mute,mic} is suppressed because (1) am_entry is
not filled with nid of the mic pin. (2) The jacks are not
created (by snd_hda_jack_detect_enable_callback) before the
snd_hda_jack_set_gating_jack call.
Now we use the first input pin nid directly, and create the jack if it
doesn't exist yet.
Signed-off-by: Chih-Chung Chang <chihchung@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A fixup for Apple Mac Mini was lost during the adaption to the generic
parser because the fallback for the generic ID 8384:7680 was dropped,
and it resulted in the silence output (and maybe other problems).
Unfortunately, just adding the missing subsystem ID wasn't enough, in
this case. The subsystem ID of this machine is 0000:0100 (what Apple
thought...?), and since snd_hda_pick_fixup() doesn't take the vendor
id zero into account, the driver ignored this entry. Now it's fixed
to regard the vendor id zero as a valid value.
Reported-and-tested-by: Linus Torvalds <torvalds@linux-foundation.org>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VAIO-Z laptops need to use the specific DAC for the speaker output
by some unknown reason although the codec itself supports the flexible
connection. So we implemented a workaround by a new flag,
no_primary_hp, for assigning the speaker pin first.
This worked until 3.8 kernel, but it got broken because the driver
learned for a better multi-io pin mapping, and not it can assign two
mic pins for multi-io. Since the multi-io requires to be the primary
output, the hp and two mic pins are assigned in prior to the speaker
in the end.
Although the machine has two mic pins, one of them is used as a noise-
canceling headphone, thus it's no real retaskable mic jack. Thus, at
best, we can disable the multi-io assignment and make the parser
behavior back to the state before the multi-io.
This patch adds again a new flag, no_multi_io, to indicate that the
device has no multi-io capability, and set it in the fixup for
VAIO-Z. The no_multi_io flag itself can be used generically, added
via a helper line, too.
Reported-by: Tormen <my.nl.abos@gmail.com>
Reported-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current fixup for dell-bios model with STAC9228 codec contains the
override of pin 0x0c for analog mic. But this is actually just adding
a bogus pin and confuses the parser. Better to remove it for the
auto-mic switching.
Meanwhile, for a possible regression, keep the old configuration as
model=dell-bios-amic, so that people can test it again quickly.
Tested on Dell 1420n laptop.
Reported-and-tested-by: Eric Shattow <lucent@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With runtime power save feature enabled, Headphone hotplug
event will not be detected while controller/codec in D3. HDA has
feature WAKEEN to let codec wake up system if controller is in D3 or
system in S3.(HDA Spec 4.5.9.2/3). Codec can send out INT or wake up
controller depending on whether CIE or GIE enabled.(Figure 4, Interupt
structure).
The controller must be in RESET mode after enter runtime-suspend, otherwise
it will not be waken up even if codec send out wake-up event. And STATESTS
will be cleared after controller brought out of RESET mode.
This patch only enable WAKEEN for runtime-suspend(Controller D3) mode,
not for system S3 mode. with tool "evtest", Headphone hotplug events
could be cought and reported successfully.
[fixed an unused variable warning by tiwai]
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With jackpoll_interval != 0, it's used to poll jack event periodically
in a delayed work. if it's 0, give the caller chance to probe jack status
but will not restart the delayed work.
In the next patch which enable WAKEEN feature, HDA controller was able to wake
up system when it's in D3, it's useful to detect Jack hotplug event and notify
userspace. By default the jackpoll_interval=0, this patch let jack poll once
without starting the delayed work.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clearing jackpoll_interval before calling cancel_delayed_work_sync(),
otherwise the work will be triggered again and cause impact in
hda_jackpoll_work(). The next patch will poll jack once even with
jackpoll_interval=0.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Register STATESTS is 16-bit length, use correct API for read/write.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the power state of ALC283 codec goes to D3, it gives a noise via
headphone output. This is because the driver tries to clear all pins
via snd_hda_shutup_pins(). Setting the mic pin to zero triggers such
a noise.
Define a new shutup call specific to this codec and control the pins
there more precisely. Also, add the power-save enable/disable
sequences in the resume and the new shutup calls.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling snd_hda_shutup_pins() unconditionally, allow it be
called in spec->shutup callback. In this way, we can avoid calling
this function if it causes a problem like we see in the next patch
following this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usage of strict_strto*() is not preferred, because
strict_strto*() is obsolete. Thus, kstrto*() should be
used.
Signed-off-by: Jingoo Han <jg1.han@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_jack_detect() function returns a boolean value for a jack
plugged in or not, but it also returns always true when the
corresponding pin is phantom (i.e. fixed). This is OK in most cases,
but it makes the generic parser misbehaving about the auto-mute or
auto-mic switching, e.g. when one of headphone pins is a fixed.
Namely, the driver decides whether to mute the speaker or not, just
depending on the headphone plug state: if one of the headphone jacks
is seen as active, then the speaker is muted. Thus this will result
always in the muted speaker output.
So, the problem is the function returns a boolean, after all, although
we need to think of "phantom" jack. Now a new function,
snd_hda_jack_detect_state() is introduced to return these tristates.
The generic parser uses this function for checking the headphone or
mic jack states.
Meanwhile, the behavior of snd_hda_jack_detect() is kept as is, for
keeping compatibility in other driver codes.
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The quirk for Dell laptops with STAC9228 overrides the pin default
config of NID 0x0f to the value with AC_DEFCFG_MISC_NO_PRESENCE bit
on. I'm not quite sure why this was done so, but can guess that this
was introduced for avoiding this to be muted by another headphone
plug. Now, after transition to the generic parser, this workaround
rather causes a problem (notably as unexpected speaker mutes) because
the pin is seen as if it's always plugged in.
Since the generic parser can handle multiple headphone plugging
gracefully, we can get rid of this override now.
Reported-and-tested-by: Eric Shattow <lucent@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The EAPD GPIO is dynamically turned on/off for some machines with
Sigmatel codecs, but this didn't work as expected, and it resulted in
spontaneous lost of speaker outputs per HP plugging or power-saving.
This patch fixes the bug by simply including spec->eapd_mask into
spec->gpio_mask and spec->gpio_data bits.
Reported-and-tested-by: Eric Shattow <lucent@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This Conexant codec has a single jack that can be used as either
headphone or mic (but not headset). The existing hp_mic functionality
does not apply here, because the mic and the HP are on separate pins.
Hence make a lighter version of what has been earlier done for Realtek
codecs.
BugLink: https://bugs.launchpad.net/bugs/1198030
Tested-by: Franz Hsieh <franz.hsieh@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vendor ID 0x10de0060 is used by a yet-to-be-named GPU chip.
Reviewed-by: Andy Ritger <aritger@nvidia.com>
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull drm updates from Dave Airlie:
"Okay this is the big one, I was stalled on the fbdev pull req as I
stupidly let fbdev guys merge a patch I required to fix a warning with
some patches I had, they ended up merging the patch from the wrong
place, but the warning should be fixed. In future I'll just take the
patch myself!
Outside drm:
There are some snd changes for the HDMI audio interactions on haswell,
they've been acked for inclusion via my tree. This relies on the
wound/wait tree from Ingo which is already merged.
Major changes:
AMD finally released the dynamic power management code for all their
GPUs from r600->present day, this is great, off by default for now but
also a huge amount of code, in fact it is most of this pull request.
Since it landed there has been a lot of community testing and Alex has
sent a lot of fixes for any bugs found so far. I suspect radeon might
now be the biggest kernel driver ever :-P p.s. radeon.dpm=1 to enable
dynamic powermanagement for anyone.
New drivers:
Renesas r-car display unit.
Other highlights:
- core: GEM CMA prime support, use new w/w mutexs for TTM
reservations, cursor hotspot, doc updates
- dvo chips: chrontel 7010B support
- i915: Haswell (fbc, ips, vecs, watermarks, audio powerwell),
Valleyview (enabled by default, rc6), lots of pll reworking, 30bpp
support (this time for sure)
- nouveau: async buffer object deletion, context/register init
updates, kernel vp2 engine support, GF117 support, GK110 accel
support (with external nvidia ucode), context cleanups.
- exynos: memory leak fixes, Add S3C64XX SoC series support, device
tree updates, common clock framework support,
- qxl: cursor hotspot support, multi-monitor support, suspend/resume
support
- mgag200: hw cursor support, g200 mode limiting
- shmobile: prime support
- tegra: fixes mostly
I've been banging on this quite a lot due to the size of it, and it
seems to okay on everything I've tested it on."
* 'drm-next' of git://people.freedesktop.org/~airlied/linux: (811 commits)
drm/radeon/dpm: implement vblank_too_short callback for si
drm/radeon/dpm: implement vblank_too_short callback for cayman
drm/radeon/dpm: implement vblank_too_short callback for btc
drm/radeon/dpm: implement vblank_too_short callback for evergreen
drm/radeon/dpm: implement vblank_too_short callback for 7xx
drm/radeon/dpm: add checks against vblank time
drm/radeon/dpm: add helper to calculate vblank time
drm/radeon: remove stray line in old pm code
drm/radeon/dpm: fix display_gap programming on rv7xx
drm/nvc0/gr: fix gpc firmware regression
drm/nouveau: fix minor thinko causing bo moves to not be async on kepler
drm/radeon/dpm: implement force performance level for TN
drm/radeon/dpm: implement force performance level for ON/LN
drm/radeon/dpm: implement force performance level for SI
drm/radeon/dpm: implement force performance level for cayman
drm/radeon/dpm: implement force performance levels for 7xx/eg/btc
drm/radeon/dpm: add infrastructure to force performance levels
drm/radeon: fix surface setup on r1xx
drm/radeon: add support for 3d perf states on older asics
drm/radeon: set default clocks for SI when DPM is disabled
...
ALC5505 DSP is enabled even though we don't use the features yet at
all. This results in the unnecessarily high power consumption, more
than 100mV higher. Until we implement the DSP support, better to
bypass DSP for saving more power.
Reported-by: Mengdong Lin <mengdong.lin@intel.com>
[Patch modified by Mengdong to cal alc5505_dsp_init() with extra
acl5505_dsp_halt().]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also use snd_ctl_enum_info() to fill the autosync text fields on AES32
and MADI cards (only users of snd_hdspm_info_autosync_ref).
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also use snd_ctl_enum_info() to fill the autosync enumerated controls.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use snd_ctl_enum_info() to fill most of the enumerated controls. More
non-trivial occurrences will follow in separate commits.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch finally enables TCO support on RME AES(32) cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new ALSA control to read the external sample rate from
userspace on RME AES(32) cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch refactors the code to query the external sample rate and its
translation into the corresponding enum into a helper function to
prevent future code duplication.
A later commit will make use of this new helper function.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also report TCO status and Sync-In via /proc/ on AES(32) cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables the user to select "TCO" and "Sync In" as a preferred
sync reference on RME AES(32) cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support to read the TCO sample rate in
hdspm_external_sample_rate() on RME AES(32) cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As mentioned in the comment, the AES32 cards must not set the format
bit, since it is used to indicate the preferred sync setting instead.
We hence simply skip the corresponding part in the hw_params function.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds AES32 specific code to hdspm_get_tco_sample_rate to
query the TCO sample rate.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds AES32 specific code to hdspm_get_wc_sample_rate() to
query the wordclock frequency.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch only introduces prototype declarations, no real change. The
functions themselves are already present.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Helper function to return the AES sample rate class. This class needs to
be translated via HDSPM_bit2freq() to get the more common
representation.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide the text for the two new clock options "TCO" and "Sync In" on
AES32 cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDSPM_AUTOSYNC_REF macro is only implemented for MADI and AES32
cards, so it doesn't make sense to call it on AIO boards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch does nothing, it's sole intent is to clean up the code.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDSPM_tco_lock and HDSPM_tcoLock were too close, so the previous code
didn't honour the difference between the two.
Let's be more verbose and use HDSPM_tcoLockMadi for MADI cards,
HDSPM_tcoLockAes for AES(32) and fix the code that makes use of both.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch separates the TCO bits from snd_hdspm_proc_read_madi(), so
the new function can later be shared between MADI and AES32 cards.
It's essentially only moving code around, no new functionality.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a left-over mistake from old code, the correct register offset is
provided in kcontrol->private_value, not in the index.
Cf. RayDAT case, where it has already been corrected.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AIO cards allow to use AEB (Analogue Expansion Boards) to add four
input and/or output channels.
This patch adds the necessary code to detect and enable the additional
I/O channels.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch uses the newly introduced HDSPM_CONTROL_TRISTATE functions to
create and expose the following ALSA controls:
- Gain selection for Input, Output and Phones (HiGain, +4dBu, -10dbV)
- S/PDIF Input select (Coaxial, Optical, Internal)
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AIO cards offer at least four individual settings options with three
states each. Those settings are represented as two bits in the settings
register with the following meaning:
0*some_base_bit --> Option value 0
1*some_base_bit --> Option value 1
2*some_base_bit --> Option value 2
3*some_base_bit --> mask to select the two involved bits
This patch adds a generic ALSA control macro for such a value-to-bit
pattern mapping. It will be used in a later commit to expose four new
controls.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ENUMERATED_CTL_INFO is a macro, so the binary code is generated multiple
times. To avoid code duplication, refactor the involved functionality
into a function and make ENUMERATED_CTL_INFO a call to this function.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds the following ALSA controls:
- S/PDIF Out Optical to switch S/PDIF Out from coaxial to optical
- S/PDIF Out Professional to send the Pro bit in the output stream
- ADAT-Internal to enable ADAT/TDIF Expansion Board (AEB/TEB)
- XLR Breakout Cable if analogue I/O uses the XLR breakout cable
- WCK48 to force WordClock to the 32-48kHz range (single speed) if
the card is operating at higher frequencies
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds new ALSA controls to send single-speed WordClock and
S/PDIF-Professional on RME RayDAT cards.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
hdspm_set_system_clock_mode() is almost a one-by-one copy of
hdspm_set_toggle_setting(). To improve code quality, remove the
duplication.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDSPM_TOGGLE_SETTING functions alter the control_register on older
cards. On newer cards (AIO/RayDAT), they have to operate on the
settings_register instead.
This patch augments the existing functions to work with AIO/RayDAT, too.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
RME RayDAT and AIO cards are new designs with different register
settings. Since we need to distinguish them from older cards multiple
times in the driver, refactor the code into a separate helper function.
No functional change intended.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver did not support all possible configurations. These defines
will be used by later commits to add the missing functionality.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally all the static quirks in patch_analog.c are reduced by this
patch. As machines with AD1986A codec are all old and often their
BIOS are buggy, we need to keep at least a few static pin conifgs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull trivial tree updates from Jiri Kosina:
"The usual stuff from trivial tree"
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (34 commits)
treewide: relase -> release
Documentation/cgroups/memory.txt: fix stat file documentation
sysctl/net.txt: delete reference to obsolete 2.4.x kernel
spinlock_api_smp.h: fix preprocessor comments
treewide: Fix typo in printk
doc: device tree: clarify stuff in usage-model.txt.
open firmware: "/aliasas" -> "/aliases"
md: bcache: Fixed a typo with the word 'arithmetic'
irq/generic-chip: fix a few kernel-doc entries
frv: Convert use of typedef ctl_table to struct ctl_table
sgi: xpc: Convert use of typedef ctl_table to struct ctl_table
doc: clk: Fix incorrect wording
Documentation/arm/IXP4xx fix a typo
Documentation/networking/ieee802154 fix a typo
Documentation/DocBook/media/v4l fix a typo
Documentation/video4linux/si476x.txt fix a typo
Documentation/virtual/kvm/api.txt fix a typo
Documentation/early-userspace/README fix a typo
Documentation/video4linux/soc-camera.txt fix a typo
lguest: fix CONFIG_PAE -> CONFIG_x86_PAE in comment
...
BIOS on Samsung R55, M55 and M50 provide the proper pin-configs,
so we can remove the corresponding static quirk entries gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For removing static quirks for AD1988 variants, a new fixup defining
the 6stack pinconfig has been added for the buggy BIOS. Other than
that, we can cut off straightforwardly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the necessary device-specific fixups for Thinkpad and HP devices
have been already ported, we can remove all static quirks for AD1884,
AD1984, AD1884A and AD1984A codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now the generic parser can work stably enough, we can get rid of the
static quirks. Let's start from AD1882.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AD1884 HP laptop/mobile quirks control GPIO1 bit as the primary
mute as well. Add the similar control to ad1884 fixup for auto
parser, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ad1884_fixup_hp_eapd() tries to set the NID for controlling the
speaker EAPD from the pin configuration. But the current code can't
work expectedly since it sets spec->eapd_nid before calling the
generic parser where the autocfg pins are set up.
This patch changes the function to set spec->eapd_nid after the
generic parser call while it sets vmaster hook unconditionally. The
spec->eapd_nid check is moved in the hook function itself instead.
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The char arrays with size 44 are for the name string of
snd_ctl_elem_id. Define the constant and replace the raw numbers with
it for clarifying better.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
add_control_with_pfx() in hda_generic.c assumes a shorter name string
for the control element, and this resulted in the truncation of the
long but valid string like "Headphone Surround Switch" in the middle.
This patch aligns the max size to the actual limit of snd_ctl_elem_id,
44.
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add two more machines that need quirks for headset mics to work.
Tested-by: Shawn Wang <shawn.wang@canonical.com>
BugLink: https://bugs.launchpad.net/bugs/1195636
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This quirk is needed for the headset mic to work on this Dell
machine.
BugLink: https://bugs.launchpad.net/bugs/1195597
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the power-saving control for ALC5505 DSP on some
Realtek codecs.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Tested-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Last 3.11 feature pull. I have a few odds bits and pieces and fixes in my
queue, I'll sort them out later on to see what's for 3.11-fixes and what's
for 3.12. But nothing to hold this here up imo.
Highlights:
- more hangcheck work from Mika and Chris to prepare for arb robustness
- trickle feed fixes from Ville
- first parts of the shared pch pll rework, with some basic hw state
readout and cross-checking (this shuts up the confused pch pll refcount
WARN that Linus just recently forwarded)
- Haswell audio power well support from Wang Xingchao (alsa bits acked by
Takashi)
- some cleanups and asserts sprinkling around the plane/gamma enabling
sequence from Ville
- more gtt refactoring from Ben
- clear up the adjusted->mode vs. pixel clock vs. port clock confusion
- 30bpp support, this time for real hopefully
* tag 'drm-intel-next-2013-06-18' of git://people.freedesktop.org/~danvet/drm-intel: (97 commits)
drm/i915: remove a superflous semi-colon
drm/i915: Kill useless "Enable panel fitter" comments
drm/i915: Remove extra "ring" from error message
drm/i915: simplify the reduced clock handling for pch plls
drm/i915: stop killing pfit on i9xx
drm/i915: explicitly set up PIPECONF (and gamma table) on haswell
drm/i915: set up PIPECONF explicitly for i9xx/vlv platforms
drm/i915: set up PIPECONF explicitly on ilk-ivb
drm/i915: find guilty batch buffer on ring resets
drm/i915: store ring hangcheck action
drm/i915: add batch bo to i915_add_request()
drm/i915: change i915_add_request to macro
drm/i915: add i915_gem_context_get_hang_stats()
drm/i915: add struct i915_ctx_hang_stats
drm/i915: Try harder to disable trickle feed on VLV
drm/i915: fix up pch pll enabling for pixel multipliers
drm/i915: hw state readout and cross-checking for shared dplls
drm/i915: WARN on lack of shared dpll
drm/i915: split up intel_modeset_check_state
drm/i915: extract readout_hw_state from setup_hw_state
...
Conflicts:
drivers/gpu/drm/i915/intel_display.c
drivers/gpu/drm/i915/intel_fb.c
drivers/gpu/drm/i915/intel_sdvo.c
Use standard PM state macros PCI_Dx instead of numeric 0/1/2..
Signed-off-by: Yijing Wang <wangyijing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As some of ALC269 quirks use the inverted dmic feature, we need to
call alc_inv_dmic_sync() in the resume callback like in alc_resume(),
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The stac_resume() is exactly what the default resume code does, so
we don't have to define and use it doubly. Let's cut it off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hdmi driver may not receive intrinsic event from gfx side when
it's in runtime suspend mode. There's no ELD info when exit from
runtime suspend. This patch avoid missing ELD info.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is a cleanup to the previous patch "reset hda link during system/
runtime suspend".
In this patch
- azx_enter_link_reset() and azx_exit_link_reset() are defined for entering and
exiting the link reset respectively. azx_link_reset() is no longer used and
replaced by azx_enter_link_reset().
- azx_reset() reuses the above two new functions for a link reset cycle
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pin's connection list may change dynamically with hot-plug event
on Intel Haswell chip. Users would see connections be "0" in codec#.
when play audio on this pin, software driver choose converter from cache
connections. So add "In-driver connection" info to avoid confuse when
raw connections are different with cache connection.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got bug report wrt many machines with VT1708 (e.g. IBM POS
machines) showing the broken auto-mute behavior. It turned out that
the problem is that the pin control values of the speaker and line-out
pins are completely ignored. As a workaround, let's use the newly
introduced feature of the generic parser, to control the mute via amp
on pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag, auto_mute_via_amp, to determine the behavior of the
headphone / line-out auto-mute. When this flag is set, the generic
driver mutes the speaker and line outputs via the amp mute of each
pin, instead of changing the pin control values.
This is introduced for devices that don't work expectedly with the pin
control values; for example, some devices are known to keep enabling
the speaker outputs no matter which pin control values are set on the
speaker pins.
The driver doesn't check actually whether the pins have the output amp
caps, but assumes that the proper mixer (mute) controls are created on
all these pins. If not the case, you can't use this flag for your
device.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If all the codecs report ClkStopOK (OK to stop bus clock) after being put to
D3, this patch will reset the HDA link before the controller is put to D3.
So the link will be in reset during system or runtime suspend, the bus clock
stops and the codecs are in D3(ClkStop) state.
This may help to reduce power consumption by dozens of mW on some peripheral
hda codecs.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the X220, this quirk was added to support docking station,
so enable the fixup instead.
According to Jan, the generic parser works equal or better
than the current parser. This was tested under a 3.9 kernel.
Reported-by: Jan Alexander Steffens <jan.steffens@gmail.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some VIA codecs like VT1708S have Mic boost amps in the mic pins but
they aren't exposed in the capability bits. In the past driver code,
we override the pin caps and create mic boost controls forcibly.
While transition to the generic parser, we lost the mic boost controls
although the pin caps are still overridden, because the generic parser
code checks the widget caps, too.
So this patch adds a new helper function to allow the override of the
given widget capability bits, and makes VIA codecs driver to add the
missing input-amp capability bit.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59861
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Intel Haswell HDMI codecs, the pins choose converter 0 by default.
This would cause conflict when playing audio on unused pins,the pin with
physical device connected would get audio data too.
i.e. Pin 0/1/2 default choose converter 0, pin 1 has HDMI monitor connected.
when play audio on Pin 0 or pin 2, pin 1 could get audio data too.
This patch configure unused pins to choose different converter.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a selection to a converter MUX is changed in hdmi_pcm_open(), it
should be cached so that the given connection can be restored properly
at PM resume. We need just to replace the corresponding
snd_hda_codec_write() call with snd_hda_codec_write_cache().
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Haswell converters maybe in wrong power state before usage.
i.e. only converter 0 is in D0, converter 1/2 are in D3.
When pin choose converter 1/2, there's no audio output, this
cause dependency when playing differnt stream on pins.
AUD_PWRST ConvertorA_Widget_Power_State_Current D0
AUD_PWRST ConvertorA_Widget_Power_State_Requsted D0
AUD_PWRST ConvertorB_Widget_Power_State_Current D3
AUD_PWRST ConvertorB_Widget_Power_State_Requested D3
AUD_PWRST ConvC_Widget_PwrSt_Curr D3
AUD_PWRST ConvC_Widget_PwrSt_Req D3
This patch check converter's power state and set D0 if it's in D3 mode.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The refactoring by commit 9040d102 introduced the new function
snd_hda_check_power_state(). This function is supposed to return true
if the state already reached to the target state, but it actually
returns false for that. An utterly stupid typo while copy & paste.
Fortunately this didn't influence on much behavior because powering up
AFG usually powers up the child widgets, too. But the finer power
control must have been broken by this bug.
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These headset jacks keep coming in on more and more platforms, and
it's possible I don't catch them all. Make it easier to test and
verify by making models.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* for-linus: (635 commits)
ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310
ALSA: hda - Fix pin configurations for MacBook Air 4,2
ALSA: usb-audio: work around Android accessory firmware bug
ALSA: hda - Headset mic support for three more machines
Linux 3.10-rc6
smp.h: Use local_irq_{save,restore}() in !SMP version of on_each_cpu().
powerpc: Fix missing/delayed calls to irq_work
powerpc: Fix emulation of illegal instructions on PowerNV platform
powerpc: Fix stack overflow crash in resume_kernel when ftracing
snd_pcm_link(): fix a leak...
use can_lookup() instead of direct checks of ->i_op->lookup
move exit_task_namespaces() outside of exit_notify()
fput: task_work_add() can fail if the caller has passed exit_task_work()
xfs: don't shutdown log recovery on validation errors
xfs: ensure btree root split sets blkno correctly
xfs: fix implicit padding in directory and attr CRC formats
xfs: don't emit v5 superblock warnings on write
mei: me: clear interrupts on the resume path
mei: nfc: fix nfc device freeing
mei: init: Flush scheduled work before resetting the device
...
MacBook Air 4,2 requires the whole default pin configuration table to
be overridden by the driver, as usual, as Apple's machines don't set
up properly after boot. Otherwise mic won't work, and other ill
effect may happen.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59381
Reported-and-tested-by: Peter John Hartman <peterjohnhartman@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
They need these quirks to have headset mic support.
BugLink: https://bugs.launchpad.net/bugs/1189363
Tested-by: Shawn Wang <shawn.wang@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For Intel Haswell chip, HDA controller and codec have
power well dependency from GPU side. This patch added support
to request/release power well in audio driver. Power save
feature should be enabled to get runtime power saving.
There's deadlock when request_module(i915) in azx_probe.
It looks like:
device_lock(audio pci device) -> azx_probe -> module_request
(or symbol_request) -> modprobe (userspace) -> i915 init ->
drm_pci_init -> pci_register_driver -> bus_add_driver -> driver_attach ->
which in turn tries all locks on pci bus, and when it tries the one on the
audio device, it will deadlock.
This patch introduce a work to store remaining probe stuff, and let
request_module run in safe work context.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Liam Girdwood <liam.r.girdwood@intel.com>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
This is a preliminary work for the upcoming Haswell HDMI audio fixes.
azx_first_init() function can be safely called after the f/w loader,
since the f/w loader doesn't require the sound hardware initialization
beforehand. Moving it into azx_probe_continue() cleans up the code
flow a bit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
The device can support runtime PM no matter whether it support
signal wakeup or not. For some chips like Haswell which doesnot
support PME by default, this patch let haswell Display HD-A controller
enter runtime suspend, and bring more power saving whith power-well
feature enabled.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
When a codec is powered off, some systems don't respond properly after
D3 FG transition, while the driver still expects the response and
tries to fall back to different modes (polling and single-cmd). When
the fallback happens, the driver stays in that mode, and falling back
to the single-cmd mode means it'll loose the unsol event handling,
too.
The unresponsiveness at D3 isn't too serious, thus this fallback is
mostly superfluous. We can gracefully ignore the error there so that
the driver keeps the normal operation mode.
This patch adds a new bit flag for codec read/write, set in the power
transition stage, which is notified to the controller driver via a new
bus->no_response_fallback flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_codec_read(), snd_hda_codec_write() & co take the argument
"direct" that indicates whether the given NID is a direct reference or
an indirect reference. However, the indirect reference is practically
unimplemented and never exists. And moreover, we don't need the
indication of indirect reference at this high level, as NID can be
represented in 16bit values at this point.
Meanwhile, there are some cases where it'd be nice to give some
operational options to these functions. So, we can reuse this
argument as a new bit flag! Pretty frugal, eh?
All callers so far pass zero to this argument, thus there is no
behavior change by this replacement.
The real usage of this new bit option will be added in the following
patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently HDMI codec driver sets the hard dependency (reverse
selection) on CONFIG_SND_DYNAMIC_MINORS because the recent codecs may
support more than two PCMs. But, this doesn't mean that we need
always this option, since there can be a single PCM stream even with
the modern codecs.
This patch drops the hard dependency again but give more sensible
error message when no enough PCMs are available due to the lack of
this option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* for-linus: (778 commits)
ALSA: usb-audio - Fix invalid volume resolution on Logitech HD webcam c270
ALSA: usb-audio - Apply Logitech QuickCam Pro 9000 quirk only to audio iface
ALSA: hda/via - Clean up duplicated codes
ALSA: hda/via - Fix wrongly cleared pins after suspend on VT1802
ALSA: hda - Add keep_eapd_on flag to generic parser
ALSA: hda - Allow setting automute/automic hooks after parsing
ALSA: hda/via - Disable broken dynamic power control
ALSA: usb-audio: fix Roland/Cakewalk UM-3G support
ALSA: hda - Add headset quirk for two Dell machines
ALSA: hda - add dock support for Thinkpad T431s
ALSA: sis7019: fix error return code in sis_chip_create()
ASoC: cs42l52: fix default value for MASTERA_VOL.
ASoC: wm8994: check for array index returned
ASoC: wm8994: Fix reporting of accessory removal on WM8958
ASoC: wm8994: use the correct pointer to get the control value
Linux 3.10-rc3
ipc/sem.c: Fix missing wakeups in do_smart_update_queue()
score: remove redundant kcore_list entries
ASoC: wm5110: Correct DSP4R Mixer control name
ARC: lazy dcache flush broke gdb in non-aliasing configs
...
This fixes the internal and external mic on Ordissimo EVE2, also known
as Malata PC-B1303.
We still don't know how to detect mic jack like Realtek's windows
driver.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit was written in the way to make the backport to
3.9.y easier, and left the duplicated open codes intentionally.
Now let's clean up the duplicated codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VIA driver has a special suspend handling only for VT1802 to reduce
the pop noise. During the transition to the generic parser, the
behavior of snd_hda_set_pin_ctl() was also changed to modify the
cached values, too. And this caused a regression where the pin is
still cleared even after the resume (including the resume from power
save), resulting in the silent output.
The fix is simply to replace snd_hda_set_pin_ctl() with the explicit
call of snd_hda_codec_write() again.
Reported-by: Alex Riesen <raa.lkml@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1802 codec seems to reset EAPD of other pins in the hardware level,
and this was another reason of the silent headphone output on some
machines. As a workaround, introduce a new flag indicating to keep
the EPAD on to the generic parser, and set it in patch_via.c.
Reported-by: Alex Riesen <raa.lkml@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codec drivers (VIA codecs and some Realtek fixups) set the
automute and automic hooks after calling
snd_hda_gen_parse_auto_config(). In the current code, the hook
pointers are referred only in snd_hda_gen_parse_auto_config() and
passed to snd_hda_jack_detect_enable_callback(), thus changing the
hook values won't change the actually called callbacks properly.
This patch fixes this bug by setting the static functions as the
primary callback functions for the jack detection, and let them
calling the appropriate hooks dynamically.
Cc: <stable@vger.kernel.org> [v3.9]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the transition to the generic parser, the actual routes used
there don't match always with the assumed static paths in some
set_widgets_power_state callbacks. This results in the wrong power
setup in the end. As a temporary workaround, we need to disable the
calls together with the non-functional dynamic power control enum.
Reported-by: Alex Riesen <raa.lkml@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly to MADI, WordClock can also be at SingleSpeed while the card
is actually working at twice or four times this rate. If so, multiply
the base rate accordingly.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the DoubleSpeed or QuadSpeed bit is set, the SingleSpeed frequency
has to be multiplied accordingly. Since this functionality will be
required at least twice, refactor it into a separate function.
The second reference to the newly introduced hdspm_rate_multiplier()
will be in a separate commit.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This quirk is required for the headset mic to work on these
two machines.
BugLink: https://bugs.launchpad.net/bugs/1186170
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for TEA5757 tuner on MediaForte M56VAP sound+modem+radio card.
The GPIO connection type is automatically detected (like snd-fm801 driver).
Also add a safety subsystem vendor check to skip radio detection if vendor
differs from ESS (so we don't touch GPIOs on laptops).
Tested with SF64-PCE2 and M56VAP cards.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a model/fixup string "lenovo-dock", for Thinkpad T431s, to allow sound in docking station.
Tested on Lenovo T431s with ThinkPad Mini Dock Plus Series 3
Signed-off-by: Ebben Aries <earies@dscp.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix to return a negative error code in the pci_set_dma_mask() error
handling case instead of 0, as done elsewhere in this function.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As drvdata is cleared to NULL at probe failure or at removal by the
driver core, we don't have to call pci_set_drvdata(pci, NULL) any
longer in each driver.
The only remaining pci_set_drvdata(NULL) is in azx_firmware_cb() in
hda_intel.c. Since this function itself releases the card instance,
we need to clear drvdata here as well, so that it won't be released
doubly in the remove callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The newer HP laptops have SSID 103c:20xx and 103c:21xx, and these
usually have the mic-mute LED on Fn-F8. Let's enable it, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add HD Audio Device PCI ID for the Intel BayTrail platform.
Signed-off-by: Chew, Chiau Ee <chiau.ee.chew@intel.com>
Signed-off-by: Artem Bityutskiy <artem.bityutskiy@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an inactive path is powered down with spec->power_down_unused
flag, we should check the activity of each widget in the path whether
it's still referred from any active path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"This contains small fixes since the previous pull request:
- A few regression fixes and small updates of HD-audio
- Yet another fix for Haswell HDMI audio
- A copule of trivial fixes in ASoC McASP, DPAM and WM8994"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
Revert "ALSA: hda - Don't set up active streams twice"
ALSA: Add comment for control TLV API
ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs
ALSA: HDA: Fix Oops caused by dereference NULL pointer
ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()
ALSA: mips/hal2: Remove redundant platform_set_drvdata()
ALSA: hda - Fix 3.9 regression of EAPD init on Conexant codecs
sound: Fix make allmodconfig on MIPS
ALSA: hda - Fix system panic when DMA > 40 bits for Nvidia audio controllers
ALSA: atmel: Remove redundant platform_set_drvdata()
ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.
ASoC: wm8994: missing break in wm8994_aif3_hw_params()
ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format
ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare
This reverts commit affdb62b81.
The commit introduced a regression with AD codecs where the stream is
always clean up. Since the patch is just a minor optimization and
reverting the commit fixes the issue, let's just revert it.
Reported-and-tested-by: Michael Burian <michael.burian@sbg.at>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a revised patch based on Mengdong Lin's fix patch, which is a
supplement to a previous patch [1611a9c9: ALSA: hda - Add fixup for
Haswell to enable all pin and convertor widgets].
Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets
when the HD-A controller changes state from D3 to D0. So when the
controller resumes after a system or runtime suspend, these widgets
are disabled and programming these widgets to D0 will cause H/W error
and codec will not respond.
In addition, we found out that some BIOS disables the pins at S3
although it shows up at boot. This confuses the driver utterly, and
the hardware falls into the fatal communication error like the above.
So in this patch, we apply intel_haswell_enable_all_pins() not only as
a fixup to a certain device (with 8086:2010) but to all Haswell
machines. The codec driver basically assumes that all pins are
exposed, so it's anyway better to see them from the beginning. Even
if all pins and converters are shown by this call, there should be no
regression in practice: the pin default configurations are still kept,
thus the disabled pins are handled as disabled by the driver
properly.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interrupt handler azx_interrupt will call azx_update_rirb,
which may call snd_hda_queue_unsol_event, snd_hda_queue_unsol_event
will dereference chip->bus pointer.
The problem is we alloc chip->bus in azx_codec_create
which will be called after we enable IRQ and enable unsolicited
event in azx_probe.
This will cause Oops due dereference NULL pointer. I meet it, good luck:)
[Rearranged the NULL check before the tracepoint and added another
NULL check of bus->workq -- tiwai]
Signed-off-by: Wang YanQing <udknight@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The older Conexant codecs have up to two EAPDs and these are supposed
to be rather statically turned on. The new generic parser code
assumes the dynamic on/off per path usage, thus it resulted in the
silent output on some machines.
This patch fixes the problem by simply assuming the static EAPD on for
such old Conexant codecs as we did until 3.8 kernel.
Reported-and-tested-by: Christopher K. <c.krooss@gmail.com>
Cc: <stable@vger.kernel.org> [v3.9]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone mic
and headset mic support, jack_modes hint consolidation, proper beep
attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone
mic and headset mic support, jack_modes hint consolidation, proper
beep attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
ALSA: pcm_format_to_bits strong-typed conversion
ALSA: compress: fix the states to check for allowing read
ALSA: hda - Move Thinkpad X220 to use auto parser
ALSA: USB: adjust for changed 3.8 USB API
ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
sound: oss/dmabuf: use dma_map_single
ALSA: ali5451: use mdelay instead of large udelay constants
ALSA: hda - Add the support for ALC286 codec
ALSA: usb-audio: USB quirk for Yamaha THR10C
ALSA: usb-audio: USB quirk for Yamaha THR5A
ALSA: usb-audio: USB quirk for Yamaha THR10
ALSA: usb-audio: Fix autopm error during probing
ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
ALSA: sound kconfig typo
ALSA: emu10k1: Fix dock firmware loading
ASoC: ux500: forward declare msp_i2s_platform_data
ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
...
The audio driver mistakenly allows 64 bit addresses to be created for
the audio driver on Nvidia GPUs. Unfortunately, the hardware normally
only supports up to 40 bits of DMA. This can cause system panics as
well as misdirected data when the address is > 40 bits as the upper
part the address is truncated.
Signed-off-by: Mike Travis <travis@sgi.com>
Reviewed-by: Mike Habeck <habeck@sgi.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Asihpi formats are not supported or invalid, and their mapping to
ALSA format is set to -1.
Before performing the format conversion into ALSA bitwise formats,
add a consistency check for the requested format, as done in
snd_card_asihpi_playback_formats().
Compile tested only.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.
Change such conversions to use this function and silence sparse
warnings.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This enables better volume controls than the current model parser.
Also, because the original quirk for X220 was added to fix
docking station support, add the TP410 fixup instead.
Reported-by: Willian Jon McCann <william.jon.mccann@gmail.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ARM cannot handle udelay for more than 2 miliseconds, so we
should use mdelay instead for those.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [b209c4df: ALSA: emu10k1: cache emu1010 firmware] broke the
firmware loading of the dock, just (mistakenly) ignoring a different
firmware for docks on some models. This patch revives them again.
Bugzilla: https://bugs.archlinux.org/task/34865
Reported-and-tested-by: Tobias Powalowski <tobias.powalowski@googlemail.com>
Cc: <stable@vger.kernel.org> [v3.8+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These are being reported as being so noisy at high mic boost levels,
so they are unusable in practice.
Therefore artificially limit the boosts.
BugLink: https://bugs.launchpad.net/bugs/1089795
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When pin default configs are overridden via patch option, these are
evaluated before fixups are applied. Since some fixups change the
whole codec trees and/or add pins dynamically, this sanity check might
not pass when pins aren't present at the time the function is called.
We may reorder the execution, but an easier fix is simply to disable
this sanity check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix to return a negative error code from the error handling
case instead of 0, as returned elsewhere in this function.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The flag bus->shutdown implies that the control elements might have
been already destroyed. When a codec is resumed at this state and
tries to call vmaster hook (e.g. in snd_hda_gen_init()), it would
refer to a non-existing object, resulting in Oops in the end.
This patch just adds a check of the flag in the caller side for
avoiding such a crash.
Though, the best would be to clear hook->sw_kctl by the destructor of
the corresponding ctl element, but vmaster uses its own private_free,
it can't be done easily. So let it be for a while.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When graphics initializes the HDMI chip, sometimes this leads to
pins going into D3 and right channel being muted. If the audio driver
finishes initialization before the graphic driver does, this situation
becomes permanent.
This is a workaround that checks for this situation and corrects it on
playback prepare. It has been verified working on at least one machine.
BugLink: https://bugs.launchpad.net/bugs/1167270
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When setting up the aamix output paths, use the primary DAC instead of
the individual DAC for each output as default. Otherwise multiple
DACs will be turned on for a single aamix widget, which results in
doubly or more volumes, because the duplicated signals will be sent
through all these DACs for a single stream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When we have a loopback mixer control, this should manage the state
whether the output paths include the aamix or not. But the current
code blindly initializes the output paths with aamix = true, thus the
aamix is enabled unless the loopback mixer control is changed.
Also, update_aamix_paths() called by the loopback mixer control put
callback invokes snd_hda_activate_path() with aamix = true even for
disabling the mixing. This leaves the aamix path even though the
loopback control is turned off.
This patch fixes these issues:
- Introduced aamix_default() helper to indicate whether with_aamix is
true or false as default
- Fix the argument in update_aamix_paths() for disabling loopback
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For capture, the delay through the codec contributes to the time stamp
of the sample recorded at the A to D. Rename the codec time stamp
function appropriately.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
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Merge tag 'asoc-v3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.10
A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:
- Add a new component object type which will form the basis of moving
to a more generic handling of SoC and off-SoC components, contributed
by Kuninori Morimoto.
- A fairly large set of cleanups for the dmaengine integration from
Lars-Peter Clausen, starting to move towards being able to have a
generic driver based on the library.
- Performance optimisations to DAPM from Ryo Tsutsui.
- Support for mixer control sharing in DAPM from Stephen Warren.
- Multiplatform ARM cleanups from Arnd Bergmann.
- New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.
With this patch, a TRRS headset mic cannot be successfully detected
on the Asus X101CH, and we can also distinguish between headphone
and headset automatically.
Buglink: https://bugs.launchpad.net/bugs/1169138
Co-authored-by: Kailang <kailang@realtek.com>
Tested-by: Luis Henriques <luis.henriques@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some machines, there is a headset jack that can support both
headphone, headsets (of both CTIA and OMTP type) and mic-in.
On other machines, the headset jack supports headphone, headsets
(both CTIA and OMTP), but not mic-in.
This patch implements that functionality as different capture sources.
Buglink: https://bugs.launchpad.net/bugs/1169143
Tested-by: David Chen <david.chen@canonical.com>
Co-authored-by: Kailang <kailang@realtek.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have a flag for headphone mics, we can use that flag
in the jack creation instead of creating the jack manually.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I never liked that we move our speaker and hp pins to line out
if there are not any line outs; but now that we do,
add some convenience functions to find hp and speaker pins even
if they have been moved.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows a specific mic to get the "Headphone Mic" name, in addition
to the existing "Headset Mic" name.
Also, it allows for a special mark: if the sequence number is set
to 0xc, that's an indication to prefer it for headset mic, and if it's
set to 0xd, that's an indication to prefer it for headphone mic.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For playback add the codec-side delay to the timestamp, for capture
subtract it. This brings the timestamps in line with the time that
was recently added to the delay reporting.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct pin configs for the Acer AC700. Most importantly indicate
that SPDIF is connected, it routes to HDMI out.
Similar to Aspire models, chain in the DMIC fixup and allow it to be
applied to this codec (ALC269VB) as well.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The DSP in the CA0132 codec adds a variable latency to audio depending
on what processing is being done. Add a new patch op to return that
latency for capture and playback streams. The latency is determined
by which blocks are enabled and knowing how much latency is added by
each block.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new codec PCM ops, get_delay(), to obtain the codec/stream-
specific PCM delay count. When it's NULL, nothing changes.
This new feature was requested for CA0132, which has significant
delays in the path depending on the running DSP code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
changed is not initialized in path_power_down_sync, but it is expected
to be false in case no change happened in the loop. So set it to
false.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 6ab317419c.
The commit [6ab317419c: ALSA: hda - Allow power_save_controller option
override DCAPS] changed the behavior of power_save_controller so that
it can override the driver capability. This assumed that this option
is rarely changed dynamically unlike power_save option. Too naive.
It turned out that the user-space power-management tool tries to set
power_save_controller option to 1 together with power_save option
without knowing what's actually doing. This enabled forcibly the
runtime PM of the controller, which is known to be broken om many
chips thus disabled as default.
So, the only sane fix is to revert this commit again. It was intended
to ease debugging/testing for runtime PM enablement, but obviously we
need another way for it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171
Reported-and-tested-by: Nikita Tsukanov <keks9n@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename "Digitial In" to "Digital In". This function is only used for
proc output, so should not cause any problems to change.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Added the device ID to the modalias list and assinged ALC662 patches
for it
* Added 4 port support for the device ID 0671 in alc662_parse_auto_config
Signed-off-by: Rainer Koenig <Rainer.Koenig@ts.fujitsu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch let ELD debug message show 'pin_eld->monitor_present' which reflects
the real pin response to verb GET_PIN_SENSE.
'eld->monitor_present' should not be used here because 'eld' is a temp
structure now and so its "monitor_present" is not set.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0.
Otherwise it will be returned uninitialized as non-zero after ELD info is got
successfully. Thus hdmi_present_sense() will always assume ELD info is invalid
by mistake, and /proc file system cannot show the proper ELD info.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: stable@vger.kernel.org
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Turing on the headphone amp interferes with the impedance measurement
used to detect a TRRS style headset microphone. Delay the HP turn on
until 500ms after the jack is detected, allowing the mic detection
state machine to run to completion.
Signed-off-by: Chih-Chung Chang <chihchung@chromium.org>
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the independent HP reduced the availability of the
side surround output, because there are only 4 DACs for 7.1 and a HP
outputs. Adjust the badness tables for VIA so that 7.1 outputs are
activated for the cost of missing independent HP.
Once when we implement the dynamic DAC switching to multiple outputs,
this conflicts will be eased in future...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The lack of independent HP mode shouldn't be too bad, but currently
its badness is set a bit too high. Let's lower it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard badness values don't seem to fit to all preferences.
Some configuration prefer the side output over the headphone, some
want the speaker over the surround, etc.
This patch moves the badness table pointers into hda_gen_spec, so that
the codec driver can override them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge back for-linus branch for the badness table adjustment for VIA codecs
* for-linus:
ALSA: hda - Fix DAC assignment for independent HP
ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
ALSA: hda - Fix typo in checking IEC958 emphasis bit
ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
ALSA: snd-usb: mixer: propagate errors up the call chain
ALSA: usb: Parse UAC2 extension unit like for UAC1
ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
The generic parser should evaluate the availability of the independent
HP when specified. Otherwise a DAC without the direct connection to
the corresponding pin may be assigned for the HP, but the driver
doesn't check it at all. The problem was actually seen on some
machines with VT1708s or equivalent codec, where DAC0 is assigned to
HP although it can be connected only via aamix.
This patch adds the badness evaluation for the independent HP to make
it working properly.
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have a "Headset Mic" name, let's use it for some devices
we know for sure has a headset mic jack.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headset mic jacks, i e TRRS style jacks with Headphone Left,
Headphone Right, Mic and GND signals, are becoming increasingly
common and are now being shipped by several manufacturers.
Unfortunately, the HDA specification does not give us any hint
of whether a Mic pin belongs to such a jack or not, but it would
still be helpful for the user to know (especially if there is one
TRS Mic jack and one TRRS headset jack).
This new fixup causes the first (non-dock, non-internal) mic to
be a headset mic jack. The algorithm can be later refined if needed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current DSP loader code abuses snd_hda_lock_devices() for ensuring
the DSP loader not conflicting with the other normal operations. But
this trick obviously doesn't work for the PM resume since the streams
are kept opened there where snd_hda_lock_devices() returns -EBUSY.
That means we need another lock mechanism instead of abuse.
This patch provides the new lock state to azx_dev. Theoretically it's
possible that the DSP loader conflicts with the stream that has been
already assigned for another PCM. If it's running, the DSP loader
should simply fail. If not -- it's the case for PM resume --, we
should assign this stream temporarily to the DSP loader, and take it
back to the PCM after finishing DSP loading. If the PCM is operated
during the DSP loading, it should get an error, too.
Reported-and-tested-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a typo in convert_to_spdif_status() about checking the
emphasis IEC958 status bit. It should check the given value instead
of the resultant value.
Reported-by: Martin Weishart <martin.weishart@telosalliance.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot to update spec->gpio_data in the automute hook, so it will be
overridden at the init sequence, thus the machine is still silent when
no headphone jack is plugged at boot time.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new HP desktop machines have Realtek codecs and their LEDs are
controlled via GPIO as for many laptop models. Add similar hooks as
well as in patch_sigmatel.c for controlling LEDs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The verb to set up the digital beep via AC_VERB_SET_DIGI_CONVERT_2
should be executed at resume as well. Use the cached write for it
being performed automatically at resume.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While playing the digital beep tone, the codec shouldn't be turned
off. This patch adds proper snd_hda_power_up()/down() calls at each
time when the beep is played or off.
Also, this fixes automatically an unnecessary codec power-up at
detaching the beep device when the beep isn't being played.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling snd_hda_attach_beep_device() and
snd_hda_detach_beep_device() in each codec driver, move them to the
generic parser. The codec driver just needs to set spec->beep_nid for
activating the digital beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The argument passed to snd_hda_attach_beep_device() is a widget NID
while spec->beep_amp holds the composed value for amp controls.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the transition to the generic parser, the hook to the codec
specific automute function was forgotten. This resulted in the silent
output on some MacBooks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset
and check it in ca0132_download_dsp().
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using the dspload_is_loaded() function, check the dsp_state
that is kept in the spec. The dspload_is_loaded() function returns
true if the DSP transfer was never started. This false-positive leads
to multiple second delays when ca0132_setup_efaults() times out on
each write.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If dspload_image() fails, it was ignored and dspload_wait_loaded() was
still called. dsp_loaded should never be set to true in this case,
skip it. The check in dspload_wait_loaded() return true if the DSP is
loaded or if it never started.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If there are no internal speakers, we should not turn the eapd switch
off, because it might be necessary to keep high for Headphone.
BugLink: https://bugs.launchpad.net/bugs/1155016
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the driver doesn't power down the widget at going down to D3
when the widget node has an EAPD capability and EAPD is actually set
on all codecs unless codec->power_filter is set explicitly.
This caused a problem on some Conexant codecs, leading to click
noises, and we set it as NULL there. But it is very unlikely that the
problem hits only these codecs.
Looking back at the development history, this workaround for EAPD was
introduced just for some laptops with STAC9200 codec, then we applied
it blindly for all. Now, since it's revealed to have an ill effect,
we should disable this workaround per default and apply only for the
known requiring systems.
The EAPD workaround is implemented now as snd_hda_codec_eapd_power_filter(),
and this has to be set explicitly by the codec driver when needed.
As of now, only patch_stac9200() sets this one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the dynamic array allocations for pins, converters and PCM arrays
instead of the fixed size arrays. The modern HDMI codecs get more and
more pins, and we don't know the sensitive limit.
Most of the patch are spent for the straight conversions from the
fixed array access to snd_array helpers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the connection list expansion in hda_codec.c and hda_proc.c, the
value returned from snd_hda_get_num_raw_conns() is used as the array
size to store the connection list. However, the function returns
simply a raw value of the AC_PAR_CONNLIST_LEN parameter, and the
widget list with ranges isn't considered there. Thus it may return a
smaller size than the actual list, which results in -ENOSPC in
snd_hda_get_raw_conections().
This patch fixes the bug by parsing the connection list correctly also
for snd_hda_get_num_raw_conns().
Reported-and-tested-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dereference should be moved below the NULL test.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
script/kernel-doc reports the following type of warnings (when run in verbose
mode):
Warning(sound/core/init.c:152): No description found for return value of
'snd_card_create'
To fix that:
- add missing descriptions of function return values
- use "Return:" sections to describe those return values
Along the way:
- complete some descriptions
- fix some typos
Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Expose the newly added TCO LTC and sync check functions to userspace.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds new ALSA controls to query the LTC state from userspace.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch prepares snd_hdspm_get_sync_check() to also check the TCO
sync state. The added feature will be exposed to the user in a later
commit.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Considerably shorten the code by using a macro. Though this won't lower
the binary size, it makes the source more readable.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For 96kHz, MADI allows to multiplex the samples (SMUX) or to use a
dedicated 96K mode. The RME cards default to 96K mode, but since not all
external MADI equipment supports this, provide a switch to users that
changes the on-wire protocol to SMUX.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When using the additional Time Code Option module in slave mode or the
SYNC-In wordclock connector, the sample rate needs to be returned by
hdspm_external_sample_rate().
Since this sample rate may contain any value with 1Hz granularity, we
need to round it to a common rate as done by the OSX driver.
[Fixed missing function declarations by tiwai]
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit introduces hdspm_get_pll_freq() to avoid code duplication.
Reading the sample rate from the DDS register will be required by
upcoming code.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a headphone pin is set up as a shared hp/mic pin, we rather want
to keep it as a headphone primarily as default, but the driver
overrides it always as a mic pin, just because the input controls are
created after outputs. Add a check of pin NID and skip the
re-initialization of pinctl for such a shared hp/mic pin.
Reported-by: Jonathan Woithe <jwoithe@just42.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the headphone mic jack enum control is created (via explicitly
specification by user), it doesn't make much sense to change the I/O
direction dynamically per capture source change, since the I/O
direction is rather controlled over the enum ctl.
This also reduces the implicit dependency between the capture source
and the hp mic jack enum ctls, which might confuse a program accessing
the whole control elements at once like alsactl.
In addition, this patch introduces update_hp_automute_hook() function
to call the proper hook function. It's just to remove the open codes
in multiple places in hda_generic.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no big merit to distinguish these two hints. Instead, just
have a single flag, add_jack_modes, for creating the jack mode enum
ctls for both I/O directions.
The hint string parser code is left and translated as add_jack_modes
just for keeping compatibility.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commits added the capability to change the pin control of
hp/mic shared jack, but it actually didn't work as expected when the
value is changed from the output to the input, since I forgot to reset
the pin I/O bit in that case. This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to let user test the known workaround more easily, give a few
known fixups for ALC260 to the model strings so that it can be passed
via the module option.
Also, move the unusual setups found in FSC S7020 fixup into a special
model, fujitsu-jwse, Jonathan Woithe Special Edition.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a headphone jack is configured as a shared hp/mic jack, the jack
mode enum needs to handle both input and output directions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch improves the generic parser code to allow to set up the
headphone jack as a mic input. User can enable this feature by giving
hp_mic hint string.
The former shared hp/mic feature for the single built-in mic is still
retained. This detection can be disabled now via hp_mic_detect hint
string, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set card->private_data in snd_ice1712_create for fixing NULL
dereference in snd_ice1712_remove().
Signed-off-by: Sean Connor <sconnor004@allyinics.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the zero check `hda_frame_size_words == 0' before the modulus
`buffer_size_words % hda_frame_size_words'.
Also remove the redundant null check `buffer_addx == NULL'.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the new control cannot be created, this function will return to avoid
snd_hda_ctl_add dereferencing a NULL control pointer.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the SPDIF control array cannot be reallocated, the function
will return to avoid dereferencing a NULL pointer.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 57e5c63007 "emu10k1: allow to
disable the SRC" force hardware use only one rate (48000 hz).
EMU 0404/1010/1616 have support two hardware sampling rates (44100 and
48000 hz). This patch add check if we have EMU 0404/1010/1616 and
choose correct sample rate to restrict.
Signed-off-by: Mihail Zenkov <mihail.zenkov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This expands the regression fix from
d28215996b.
The firmware also needs to be loaded when it was already cached.
Signed-off-by: Florian Zeitz <florob@babelmonkeys.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes initialization of the AK4114 chip so spdif capture is working properly.
Worked out together with Pavel Hofman.
Signed-off-by: Jonas Petersen <jnsptrsn1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Along with a clean up commit [e9f66d9b9: ALSA: pci: clean up using
module_pci_driver()], bt87x driver lost the functionality of load_all
parameter. This patch does a partial revert of the commit only for
bt87x.c to recover it.
Reported-by: Clemens Ladisch <cladisch@googlemail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ELD validity can change during the lifetime of a presence detect,
so we need to be able to listen for changes on the ELD control.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because the eld buffer can be simultaneously accessed from both
workqueue context (updating) and process context (kcontrol read),
we need to protect it with a mutex to guarantee consistency.
To avoid holding the mutex while reading the ELD info from the
codec, we introduce a temporary eld buffer.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For better readability, the information that is parsed out of the
ELD data is now put into a separate parsed_hdmi_eld struct.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previously, it was possible to read the eld data of the previous
monitor connected. This should not be allowed.
Also refactor the function slightly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, eld_valid is never set to false, except at kernel module
load time. This patch makes sure that eld is no longer valid when
the cable is (hot-)unplugged.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the recent update, Fujitsu S7020 laptop with ALC260 codec lost the
speaker output, no matter how the amps and the pins are set. After a
long debugging session, we found out that the default codec init code
is harmful for this machine, and we have to reset it to
ALC_INIT_NONE.
Reported-and-tested-by: Jonathan Woithe <jwoithe@just42.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These two machines have no mute LED string in BIOS.
BugLink: https://bugs.launchpad.net/bugs/1128934
Tested-by: Tammy Yang <tammy.yang@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This chip needs the speaker pin to go to D3 to avoid clicks,
so default_power_filter does not work here.
This was found on Thinkpad R61i/T61i but I guess it applies to
the entire chip. If not, quirks should be set for at least
PCI SSID 17aa:20ac.
Thanks to c4pp4 for testing.
BugLink: https://bugs.launchpad.net/bugs/886975
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a regression report wrt the IRQ issue related with the
power-save on a Dell machine, and disabling runtime PM works around.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=53441
Cc: <stable@vger.kernel.org> [v3.7+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current badness value used for the missing multi-io seems too
weak, and the multi-io tends to be skipped for desktop configurations
when no enough DACs are available. It's because the total badness of
the multi-io becomes often larger than the badness with assigning an
individual DAC to a headphone jack. This is good for one side, but it
seems that the surround outputs are more demanded by that.
This patch increases the badness value for the missing multi-io
slightly so that the multi-io would be preferred than the individual
headphone DAC if they conflict. Through the tests with hda-emu,
mostly only desktop configurations with ALC662/663 and CMI codecs are
affected by this change, and all look reasonable.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [dcda58061: ALSA: hda - Add workaround for conflicting
IEC958 controls] introduced a workaround for cards that have both
SPDIF and HDMI devices for giving device=1 to SPDIF control elements.
It turned out, however, that this workaround doesn't work well -
- The workaround checks only conflicts in a single codec, but SPDIF
and HDMI are provided by multiple codecs in many cases, and
- ALSA mixer abstraction doesn't care about the device number in ctl
elements, thus you'll get errors from amixer such as
% amixer scontrols -c 0
ALSA lib simple_none.c:1551:(simple_add1) helem (MIXER,'IEC958
Playback Switch',0,1,0) appears twice or more
amixer: Mixer hw:0 load error: Invalid argument
This patch fixes the previous broken workaround. Instead of changing
the device number of SPDIF ctl elements, shift the element indices of
such controls up to 16. Also, the conflict check is performed over
all codecs found on the bus.
HDMI devices will be put to dev=0,index=0 as before. Only the
conflicting SPDIF device is moved to a different place. The new place
of SPDIF device is supposed by the updated alsa-lib HDA-Intel.conf,
respectively.
Reported-by: Stephan Raue <stephan@openelec.tv>
Reported-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org> [v3.8]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The chip address is 32bit long but INVALID_CHIP_ADDRESS is defined as
an unsigned long. This makes dsp_chip_to_dsp_addx() misbehaving on
64bit architectures. Fix the INVALID_CHIP_ADDRESS definition to be
32bit.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The problem addressed by this fixup is not specific to Vaio Z, affecting
some Vaio all-in-one desktop PCs too. Update the code comments accordingly.
Signed-off-by: Fernando Luis Vazquez Cao <fernando@oss.ntt.co.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Vaio all-in-one desktop PCs (for example VGC-LN51JGB) are affected by
the same issue that caused Vaio Z laptops to become silent: the speaker pin
must be connected to the first DAC even though the codec itself advertises
flexible routing through any of the DACs.
Use the no-primary-hp fixup for choosing the speaker pin as the primary so
that the right DAC is assigned on this device.
Cc: stable@vger.kernel.org
Signed-off-by: Fernando Luis Vazquez Cao <fernando@oss.ntt.co.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_ali_pointer function is called with local
interrupts disabled. However it seems very strange to
reenable them in such way.
Found by Linux Driver Verification project (linuxtesting.org).
Signed-off-by: Denis Efremov <yefremov.denis@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the other code in this driver and similar
code in rme96 it seems, that spin_lock_irq in
snd_rme32_capture_close function should be paired
with spin_unlock_irq.
Found by Linux Driver Verification project (linuxtesting.org).
Signed-off-by: Denis Efremov <yefremov.denis@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes a few obvious bugs in DSP loader stuff:
- Fix possible memory leaks in the error path
- Avoid double-free calls in dma_reset()
- Properly set/unset WC bits for DMA buffers
- Add missing error status checks
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Base the DSP firmware transfer and communication timeouts on jiffy values.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the HD Audio Device IDs for the Intel Wellsburg PCH
Signed-off-by: James Ralston <james.d.ralston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A Haswell test machine showed that the invalid connection list, but
this time it has only a single pin on the codec, thus the former fixup
code doesn't work as it assumes the three pins blindly.
This patch splits the former fixup code to two parts:
- Enable eDP 1.2 for Haswell codec
- Fix the connection list of pins on Haswell codec;
the converter list is recorded dynamically in hdmi_add_cvt(), and
applied in hdmi_add_pin()
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Haswell machines support more than one display outputs (HDMI or DP),
but its BIOS may not enable the codec's 2nd and 3rd pin and output cvt widgets.
This patch implements a board-specific fixup for Intel Haswell Machines:
If the hidden pins are not enabled by BIOS, the driver will enable them
and call common code to update the codec tree.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A codec may allow software to hide some unused pin/cvt widgets.
Sometimes BIOS does not enable the hidden widgets properly although they are
needed for the board. Thus the driver need to enable them as a board-specific
fixup and the whole tree will change.
This patch implements a common code for rereading codec widgets. So the fixup
code can call it after enabling the hidden widgets.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we set the max number of connections to be 32, but there
seems codec that gives longer connection lists like AD1988, and we see
errors in proc output and else. (Though, in the case of AD1988, it's
a list of all codecs connected to a single vendor widget, so this must
be something fishy, but it's still valid from the h/w design POV.)
This patch tries to remove this restriction. For efficiency, we still
use the fixed size array in the parser, but takes a dynamic array when
the size is reported to be greater than that.
Now the fixed array size is found only in patch_hdmi.c, but it should
be fine, as the codec itself can't support so many pins.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix these two compile errors on s390 which does not have HAS_IOPORT
nor GENERIC_HARDIRQS:
sound/pci/lx6464es/lx6464es.c: In function ‘snd_lx6464es_free’:
sound/pci/lx6464es/lx6464es.c:565:2: error: implicit declaration of function ‘ioport_unmap’
sound/soc/codecs/wm8903.c: In function ‘wm8903_set_pdata_irq_trigger’:
sound/soc/codecs/wm8903.c:1954:9: error: implicit declaration of function ‘irq_get_irq_data’
Signed-off-by: Heiko Carstens <heiko.carstens@de.ibm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c: In function ‘ca0132_is_vnode_effective’:
sound/pci/hda/patch_ca0132.c:3331:15: warning: ‘nid’ may be used uninitialized in this function [-Wmaybe-uninitialized]
sound/pci/hda/patch_ca0132.c:4345:13: warning: ‘ca0132_download_dsp’ defined but not used [-Wunused-function]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loopback list is referred by the VIA codec driver no matter
whether CONFIG_PM is set or not, thus we need to enable it always.
Otherwise it gets compile errors.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The wm->regs[] array has WM8766_REG_COUNT (16) elements not
WM8766_REG_RESET (31).
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's mostly harmless to apply it for new models even if they have no
mic mute LED (just toggling an unused GPIO pin).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These days, GUIs such as Gnome sound settings want to be able to
show the correct jack status even when no streams are currently
running. I doubt this gives any measurable difference in power,
but if it does, the "Jack Detect" control can still be used to
turn polling off.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The VT1708 has no unsol event capability, and polling is set using
the "Jack Detect" alsamixer control. In order not to create
phantom Jack controls, temporary enable jackpoll during build_controls.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... to be less confusing for the update path.
This new kconfig will choose CONFIG_SND_HDA_DSP_LOADER, which is
basically a device-independent feature in hda_intel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit d45e6889ee ("ALSA: hda - Provide
the proper channel mapping for generic HDMI driver") added support for
custom channel maps in the HDA HDMI driver. Due to a mistake in an
'if' condition the custom map is always used even when no such map has
been set. This causes incorrect channel mapping for multichannel audio
by default.
Pass per_pin->chmap_set to hdmi_setup_channel_mapping() as a parameter
so that it can use it for detecting if a custom map has been set instead
of checking if map is NULL (which is never the case).
Reported-by: Staffan Lindberg <pike@xbmc.org>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the driver detects and invalid ELD, it gives an open error.
But it forgot to release the assigned pin, converter and spdif ctls
before returning.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new PCI ID 0x0a0c for Haswell ULT platform.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For non-snoop mode, we fiddle with the page attributes of CORB/RIRB
and the position buffer, but also the ring buffers. The problem is
that the current code blindly assumes that the buffer is contiguous.
However, the ring buffers may be SG-buffers, thus a wrong vmapped
address is passed there, leading to Oops.
This patch fixes the handling for SG-buffers.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=800701
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we use LPIB forcibly for both playback and capture for
Poulsbo and Oaktrail devices, and this seems rather problematic.
The recent fix for LPIB delay count seems working well with these
devices, so let's enable it instead.
Reported-by: Martin Weishart <martin.weishart@telosalliance.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes a regression of the external mic not working on
HP Probook 4520s.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]
For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros. This should have the same effect but shut up warnings like
above.
But since we had already put ifdefs, changing to inline functions
would trigger compile errors. So, such ifdefs is removed in this
patch.
In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too. For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.
Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a better power filter hook for powering down unused
widgets in the generic parser.
The feature is enabled by setting hda_gen_spec.power_down_unused
flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put the power state synchronization at the end of the parsing of
codec. This is necessary when the power filter is changed during the
codec probe. Since the first power-up sequence is performed without
the special filter, all widgets are supposed to be ON at this point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a hook to struct hda_codec for filtering the target power state of
each widget when powering up/down. The current hackish EAPD check is
implemented as the default hook pointer, too.
This allows codec drivers to implement own power filter. In the
upcoming changes, the generic parser will have the better power filter
based on the active paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AC_VERB_GET_POWER_STATE returns the combined bits of the actual state
and the target state. Thus, comparing the obtained value directly
with the target value can't work. The value has to be shifted and
masked properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The arguments to call is_active_nid() in activate_amp() were swapped,
and this resulted in the muted amp on some SPDIF output pins.
Also, the index to be passed to is_active_nid() must be idx_to_check.
Otherwise it checks the wrong connection in the case of implicit aamix
connection paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using the new chained_before flag, we can correct the headphone jack
detection capability easily over the existing ALC880 6stack model
(which disables the jack detection intentionally for compatibility
reason).
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=901846
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A Packard-Bell desktop machine gives no proper pin configuration from
BIOS. It's almost equivalent with the 6stack+fp standard config, just
take the existing fixup.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=901846
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes we want to call a fixup after applying other existing
fixups, but currently the fixup chain mechanism allows only the call
the others after the target fixup. This patch adds a new flag,
chained_before, to struct hda_fixup, for allowing the chained call
before the current execution.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD1988 family and AD1882 codecs have another mixer widget (0x21)
between the analog-loopback mixer widget (0x20) and the actual
outputs. Due to this hole, the analog-loopbacks aren't sent properly
to the output pins.
As a band-aid fix, introduce another fields holding the aamix merge
path, and activate it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [26a6cb6c: ALSA: hda - Implement a poll loop for jacks as a
module parameter] introduced the polling jack detection code, but it
also moved the call of snd_hda_jack_set_dirty_all() in the resume path
after resume/init ops call. This caused a regression when the jack
state has been changed during power-down (e.g. in the power save
mode). Since the driver doesn't probe the new jack state but keeps
using the cached value due to no dirty flag, the pin state remains
also as if the jack is still plugged.
The fix is simply moving snd_hda_jack_set_dirty_all() to the original
position.
Reported-by: Manolo Díaz <diaz.manolo@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fixup function is called multiple times before parsing the pins,
so snd_BUG_ON() hits when loaded. Move it to the proper place in the
if block.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a merge of really big changes: the generic parser is heavily
enhanced for handling all cases, based on the former Realtek codec
driver code. And all codec drivers except for a few ones (CA0132,
HDMI and modem) have been converted to use the new generic driver.
Conflicts:
sound/pci/hda/patch_realtek.c
Now all AD codecs have the proper BIOS auto-parser, and we can make
it for default, finally. (AD1988 already did it because it had the
auto-parser.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As done for patch_conexant.c, put ifdef ENABLE_AD_STATIC_QUIRKS for
preparing t odrop the static quirk codes in patch_analog.c.
The whole static quirk code can be omitted by commenting out
ENABLE_AD_STATIC_QUIRKS define now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD codecs have strange implementations for choosing the SPDIF-output
mux source: the digital audio out widget may take the sources from
multiple connections, where 0x01 indicates it's a PCM while others
point ADCs. It's obviously invalid in the HD-audio spec POV, but it's
somehow convincing, too. And, to make things more complex, AD1988A
and AD1882 have deeper connection routes that aren't expressed
correctly.
In this patch, the SPDIF mux control is implemented in two ways:
- For easier one like AD1981, AD1983, AD1884 and AD1984, where the
SPDIF audio out widget takes just two or three sources, we can
simply implement via the normal input_mux and connection verb
calls.
- For the complex routes like AD1988A (but not AD1988B) or AD1882, we
prepare "faked" paths represented statically, and switch the paths
using these static ones, instead of parsing the routes from the
widget tree.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since both snd_hda_codec_flush_amp_cache() and
snd_hda_codec_flush_cmd_cache() are called usually at the same time,
we can simply combine them to a single function,
snd_hda_codec_flush_cache().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture volume put callback may call the node selection change,
and its actual call won't be triggered unless flushed. In general,
we always need to call both snd_hda_codec_flush_amp_cache() and
snd_hda_codec_flush_cmd_cache() at the same place...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both the HP auto-mute and the independent HP mode conflict with each
other. Make HP auto-mute disabled (only for the affected HP jack)
during the driver is in HP independent mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It'd be better to give another name to the secondary (alt) analog PCM
stream, which is dedicated for the independent HP out and extra
inputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The aamix NIDs are also missing for AD codecs. All AD codecs seem to
have a (more or less) working aamix widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IDT codecs have analog-loopback mixer widgets, but we haven't cared
about it, so far. Let's set them. This will avoid also possible
wrong routes for the input paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch eventually fixes two issues:
- Handle the case where the primary output is a headphone and can have
independent HP mode;
so far we checked only the case where the headphone is the secondary
output.
- Fix the conflict of HP independent mode and aamix mode;
when switched to aamix mode, the DAC might be also switched to
another widget shared with other outputs. Then even if we disable
the DAC for the original output, it doesn't change -- because the
active route is from another (shared) DAC to HP pin through aamix.
So, in such a case, we have to prohibit the switch to aamix for HP
routes.
This fixes issues appearing on VT codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many codecs provide routes to multiple output pins through an aamix
widget, but most of them do it only from a single DAC. However, the
current generic parser checks only the aamix paths from the original
(directly bound) DACs through aamix NID, and miss the path:
primary DAC -> aamix -> target out pin
This patch adds a more check for the routes like the above.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a patch couldn't be resolved in try_assign_dacs() although the
target DAC is expected, we forgot to add a proper badness value but
continued.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since fill_and_eval_dacs() may be called repeatedly with different
configurations, setting pinctls at each time there isn't optimal.
We can set it better only once after deciding the output configuration
in parse_output_paths().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Print the information of outputs in a bit more details and concisely
in a single place instead of printing the path at each time when
detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant CX20551 codec has a mixer in NID 0x19 and a few outputs have
to take the input through this widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Looking through the whole definitions, some fields have inappropriate
array sizes, especially about the capture. The array assigned to each
input (pin) should have HDA_MAX_NUM_INPUTS entries while the array
assigned to each ADC should have AUTO_CFG_MAX_INS entries.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch "ALSA: hda - fix wrong adc_idx in generic parser" fixed the
adc_idx for the capture volume and capture switch controls. But also
modified the adc_idx retrieval for the capture source controls
wrongly. As multiple capture source controls are created in a single
shot with counts > 1, the id.index doesn't contain the real value.
The real index has to be taken via snd_ctl_get_ioffidx() as in the
original code.
This patch reverts the fixes partially to recover from the
regression.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If there's one each of HDMI and SPDIF, we should not add an index
on the one that comes second.
[slight code refactoring by tiwai]
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just stumbled over this one while reading the code.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I found a codec configuration which had six inputs, so the max of
five was not appropriate.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some BIOS version of FSC Lifebook S7110 laptop seems to give a wrong
default pin config for NID 0x15, which confuses the parser. Give a
fixup to correct the value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although I commented that boost volumes would be added only for
line-in and mic pins in the source code, the actual code excludes but
for mic-in. Fix it to accept the line-ins, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two hooks in hda_gen_spec, cap_sync_hook and capture_switch_hook, play
very similar roles. The only differences are that the former is
called more often (e.g. at init or switching capsrc) while the latter
can take an on/off argument.
As a more generic implementation, consolidate these two hooks, and
pass snd_ctl_elem_value pointer as the second argument. If the
secondary argument is non-NULL, it can take the on/off value, so the
caller handles it like the former capture_switch_hook. If it's NULL,
it's called in the init or capsrc switch case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a standard capture switch without multiple binding is used, the
call for capture_switch_hook isn't called properly. Replace the put
ops to add the hook call in that case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This machine also has the "HP_Mute_LED_0_A" string in DMI information.
Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1096789
Tested-by: Tammy Yang <tammy.yang@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current generic parser code, we look for the (mic) boost
controls only on input pins. But many codecs assign the boost volume
to a widget connected to each input pin instead of the input amp of
the pin itself.
In this patch, the parser tries to look through more widgets connected
to the pin and find a boost amp.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an amp in the activation path is associated with mixer controls,
activate_amp() tries to skip the initialization. It's good, but only
if the mixer really initializes both mute and volume. Otherwise,
either the mute of the volume is left uninitialized.
This patch adds this missing check and properly initialize the
partially controlled amps in an activation path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places creating the labels and indices of kctls for
each input pin in the current generic parser code. This is redundant
and makes harder to maintain. Let's create the labels and indices at
once and keep them in hda_gen_spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the imux table entries can be a subset of autocfg.input table,
the indices of these aren't always same. For passing the proper index
value of autocfg.input at creating input ctl labels (via
snd_hda_autocfg_input_label()), keep the corresponding autocfg.input
idx value in the index field of each imux item, which isn't used in
the generic driver.
Also, this makes easier to check the invalid imux pin for stereo mix.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally we reached here. All codecs driver (except for CA0132, which
has really device-specific requirements) have been converted to use
the generic parser.
This patch appears bigger than others since it also involves with the
code shuffling, but mostly the cut-off of parser codes and adapt to
the generic parser flags. Most of fixup codecs haven't been changed
but just removed a few unnecessary codes.
The only missing stuff is the SPDIF mux control. It'll be added again
later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* test/hda-gen-parser:
ALSA: hda - Improve naming rule for primary output
ALSA: hda - Add PCM capture hook to hda_gen_spec
ALSA: hda - Record all detected ADCs in hda_gen_spec
ALSA: hda - Move vmaster TLV parsing to snd_hda_gen_parse_auto_config()
ALSA: hda - Add input jack mode enum controls to generic parser
ALSA: hda - Give more comments to hda_gen_spec flags
ALSA: hda - Add suppress_auto_mute flag to hda_gen_spec
ALSA: hda - Record the current speaker / LO mute status in hda_gen_spec
ALSA: hda - Properly call automute/switch hooks at init
When the volume or mute control of the primary output is shared with
other (headphone or speaker) outputs, we shouldn't name it as a
specific output type but rather name it with the channel name or a
generic name like "PCM".
Also, this check should be performed individually for the volume and
the mute controls because some codecs may have shared volumes but
separate mute controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the generic parser reduces the ADC list, copy the list of the
all detected ADCs and keep it.
This list can be later referred by the codec driver for finer power
controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add vmaster_tlv[] to hda_gen_spec and store the suggested TLV data
in snd_hda_gen_parse_auto_config(). This allows the codec driver to
correct the TLV data (e.g. mute capability) before actually creating
vmaster instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like the jack mode enum ctls for output jacks, add the support
for similar enum ctls for input pins to control the bias Vref.
The new controls will be added when spec->add_in_jack_modes is set
either by the codec driver or by a hint string.
Note that ground and 100% vrefs are excluded from the list for
simplicity, currently. We may add a new flag to allow them, too.
But I guess it's easier to put a value override in the pinfix in such
a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- spec->hp_detect has to be overridden in HDA_FIXUP_ACT_PARSE, not in
PRE_PARSE.
- Remove err == 0 check but return directly -EINVAL from
stac92xx_parse_auto_config()
- Set spec->default_polarity for 92HD71bxx
- Some code shuffles
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A new flag to skip the auto-mute handling in the generic parser, just
like suppress_auto_mic flag. It has to be set before calling
snd_hda_gen_parse_auto_config().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* test/hda-gen-parser:
ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecs
ALSA: hda - force different capture controls if amp caps differ
ALSA: hda - do not add non-existing Mic boost controls
ALSA: hda - initialize channel counts correctly
ALSA: hda - fix wrong adc_idx in generic parser
ALSA: hda - Check array bounds in get_input_path
ALSA: hda - Add prefer_hp_amp flag to hda_gen_spec
ALSA: hda - fix OOPS in hda_mark_cmd_cache_dirty
ALSA: hda - Check pincap while parsing the configuration
Otherwise no PCM will be built for codecs without analog I/O.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise setting the capture volume for amps will be weird and
inconsistent (it will try to set values outside the range of the
second amp based on capabilities of the first amp).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the input node does not have any volume capable input amp,
don't add such a control.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even a single DAC can output two channels, so the channel count
is twice the number of DACs.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We use knew->index for adc_idx when we create "Capture Volume" and
"Capture Switch", so use the same to retrieve adc_idx.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c:387:19: sparse: symbol 'ca0132_voicefx' was not declared. Should it be static?
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With HDSP_TOGGLE_SETTING in place, these functions are no longer
required. Removing them makes the code DRY and considerably shorter.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDSP_TOGGLE_SETTING and its corresponding functions allow to change
settings in the control register. Instead of using the specialised
functions, use the generic code to make the code DRY.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver contains multiple similar functions that change only a single
bit in the control register, only the bit position varies.
This patch implements a generic function to toggle a certain bit
position that will be used to replace the old code.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current iobox detection code reportedly fails for various users, so
simply do what the Win32 driver does instead.
Patch originally by Karl Grill <kgrill@chello.at> and then modified to
comply with kernel coding guidelines + current HEAD.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag to indicate whether HP amp is turned on as default for
speaker or line-outs, and enable this for ALC260 codec, as many
machines with this codec require the HP amp even for speakers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c: In function ‘ca0132_effects_set’:
sound/pci/hda/patch_ca0132.c:3391:2: warning: too many arguments for
format [-Wformat-extra-args]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Spotted by smatch,
sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: potential
null dereference 'dma_engine'. (kzalloc returns null)
sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: we
previously assumed 'dma_engine' could be null (see line 1857)
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c:1781 dspxfr_one_seg() info: why not
propagate 'status' from dsp_dma_stop() instead of (-5)?
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent update of ca0132 driver replaced the pinctl setup to the
direct write via snd_hda_codec_write() again. This should be covered
by snd_hda_set_pin_ctl() to be safer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit c3b4eea262.
Since the recent firmware loader code supports caching at S3/S4 by
itself, we don't have to handle f/w caching in the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Handle a potential dma_engine alloc error and fix the possible use of an
uninitialized status variable in dspxfr_one_seg(). Also correct the initial
sampling rate for Mic 1.
Update the module description.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the controls used for tuning the DSP effects.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the unsolicited response handler for incoming DSP responses and
jack detection reporting, and routines for reading the incoming DSP response.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the playback PCM open callback.
PCM stream setup and cleanup functions are added for use by PCM callbacks.
Delay stream cleanup if effects are on, to allow time for any effects tail to
finish.
Add the analog capture PCM callbacks.
Change the max channels of analog playback to 6.
Add two new PCMs: AMic2 and What-U-Hear.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the kcontrols for the DSP effects, playback and recording
source selection.
ca0132_is_vnode_effective() checks whether virtual node settings have
taken effect.
The control change helpers ca0132_pe_switch_set(), ca0132_voicefx_set()
and ca0132_cvoice_switch_set() are added to toggle playback / capture
DSP effects, ca0132_voicefx_info(), _get() and _put() are added for
input path DSP effect value access. The volume helpers are updated to
volume_info(), _get() and _set() to use the virtual nodes.
The redundant headphone and speaker switches and ct_extension function
are removed.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the framework to set effect parameters: ca0132_effects_set()
and ca0132_setup_defaults() are general functions for parameter setting and
initializing to default values. dspio_set_param() and dspio_set_uint_param()
are lower-level fns to simplify setting individual DSP parameters via an
SCP buffer transfer to the firmware.
The CA0132 chip parameter init code is added in ca0132_init_params().
In chipio_[write,read]_data(), the current chip address is auto-incremented
if no error has occurred.
ca0132_select_out() selects the current output. If autodetect is enabled,
use headphones (if jack detected) or speakers (if no jack).
ca0132_select_mic() selects the current mic in. If autodetect is enabled,
use exterior mic (if jack detected) or built-in mic (if no jack).
Init digital mic and switch between dmic and amic with ca0132_init_dmic(),
ca0132_set_dmic(). amic2 is initialized in ca0132_init_analog_mic2().
Finally, add verb tables for configuring DSP firmware.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds definitions and structs used for configuring DSP effects,
virtual nodes, effect tuning controls, and mixer control helpers.
The effect structs are also initialized.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When "alsactl restore" is performed on HDMI codecs, it tries to
restore the channel map value since the channel map controls are
writable. But hdmi_chmap_ctl_put() returns -EBADFD when no PCM stream
is assigned yet, and this results in an error message from alsactl.
Although the error is harmless, it's certainly ugly and can be
regarded as a regression.
As a workaround, this patch changes the return code in such a case to
be zero for making others happy. (A slight excuse is: when the chmap
is changed through the proper alsa-lib API, the PCM status is checked
there anyway, so we don't have to be too strict in the kernel side.)
Cc: <stable@vger.kernel.org> [v3.7+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of checking the codec SSID in find_mute_led_cfg() for HP Mini
110, set the proper spec->default_polairty in the fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI vendor ID check in find_mute_led_cfg() is now superfluous
because the function is called in the fixup table entries of HP
machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally all codecs in patch_sigmatel.c have been converted to use the
standard fixup helpers. This change also includes trivial cleanups
like the call of common setup for GPIO LED or the removal of unused
function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This one is rather a simple conversion. The fixups for Dell machines
are implemented by fixup functions in the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time, the only intrusive changes are for HP machines.
As the mute LED fixup and the bass speaker switch are required only
for HP machines, we can move these checks into the fixup entries; the
former is applied generically to all HP machines while the latter for
only certain models.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes (or rather often) BIOS sets the pin default configurations
obviously wrongly. Looking through these failures, one common pattern
is to enable some dead pins that are usually marked as speaker pins.
In such a case, we can skip them if the pins don't have the output
capability.
In this patch, add a check for the valid pin cap bit for each parsed
pin, and filter out when it's invalid.
The fix was originally suggested by Raymond Yau.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This conversion is a bit tricky. Since STAC927x may take two
different volume-knob initialization values depending on the model, a
new flag, spec->volknob_init, is introduced to indicate whether it's
the standard volume-knob initialization or not.
Also, Dell BIOS model is now directly mapped onto the fixup table
instead of parsing in the function. This resulted in a new model ref,
STAC_927X_DELL_BIOS_SPDIF, which is a chained entry.
Also, for reducing the fixups, virtual entries like
STAC_927X_DELL_DMIC and STAC_D965_VERBS are introduced.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather straightforward conversion, except for ones for Intel Mac.
As Intel Mac have only unique codec SSIDs, we need to remap the fixup
again for the codec SSID and call the new fixup there.
Also, we can reduce model enums like STAC_MACMINI, which are model
aliases for backward compatibility, since they can be pointed directly
via hda_model_fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another step forward. As all features for VIA codecs have been
implemented in the generic driver, we can move on to migrate the VIA
codec parser, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the support for the generic auto-parser to AD codec
driver. For AD1988, the old code is replaced simply with the new
generic parser. For other codecs, new model "auto" is added and
directed to use the generic parser.
No fixup codes have been implemented yet as of now. Eventually we'd
replace each static quirk with the generic parser + fixup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just shuffle the codes and add ifdefs for testing to drop the static
quirk codes from patch_conexant.c.
By commenting out ENABLE_CXT_STATIC_QUIRKS define at the beginning of
the file, you can disable the whole static codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time, the target is Cirrus codec. Its parser is a subset of
generic parser, so we can migrate fully with it now.
The only tricky part is the handling of SPDIF automute.
Cirrus driver sets the SPDIF out plug over the headphone. As a
workaround, set spec->gen.master_mute for toggling the headphone (and
other) mute.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CA0110 codec is a fairly straightforward hardware implementation,
and we can use the generic parser almost as is.
Just set spec->multi_cap_vol flag to follow the current behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the old parser code for C-Media auto-parser with the latest
generic parser. For compatibility reason, the static bindings are
still left, but they could be cleaned up in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pincfgs, init_verbs and hints set by sysfs or patch might be
changed dynamically on the fly, thus we need to protect it.
Add a simple protection via a mutex.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As David Henningsson recently suggested, some HP laptops use an unused
mic pin for controlling a mute LED, and this information is provided
via DMI string "HP_Mute_LED_X_Y" string. This patch adds the generic
support for such cases, as we've already done in patch_sigmatel.c.
This is applied generically to all devices with ID 0x103c.
But as we don't know whether the device 103c:1586 really contains
HP_Mute_LED_X_Y DMI string, still keep the static setup for this
device using the mic2 pin 0x19.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some fixups such as setting the flags influencing on the parser
behavior should be applied before actually parsing the tree.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Try to recover from the regression: set the HP amp for the speaker and
add the hp jack mode enum as default.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the enum controls for changing the headphone amp bits of output
jacks, such as "Headphone Jack Mode". This feature isn't enabled as
default, so far, unless spec->add_out_jack_modes flag is set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a multi-io jack is switched to another direction, call the
automute and autoswitch update functions, as this jack won't be used
as the headphone or the mic jack that may turn off others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new fixup type, HDA_FIXUP_PINCTLS, for overriding the pinctl
values of the given pins. It takes the same array of struct pintbl
like HDA_FIXUP_PINS, but each entry contains the pinctl value instead
of the pin default config value.
This patch also replaces the corresponding codes in patch_realtek.c.
Without this change, the direct call of verbs may be overridden again
by the later call of pinctl restoration by the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now the whole codebase has been changed from the earlier kernels, it
makes little sense to keep these aliases. Simply replace with the
official names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a jack is retasked as a different direction (e.g. a mic jack is
used as a CLFE output), such a jack shouldn't be counted as the target
for the automatic jack switching. Skip the automute or the autoswitch
when the current pinctl direction is different from what we suppose.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new pin target accessors for managing the current pinctl
values in the generic parser. The pinctl values of all active pins
are once determined at the initialization phase, and stored via
snd_hda_codec_set_pin_target(). This will be referred again in the
codec init or resume phase to set the actual pinctl.
This value is kept while the auto-mute. When a line-out or a speaker
pin is muted by auto-mute, the driver simply disables the pin, but it
doesn't touch the cached pinctl target value. Upon unmute, this value
is used to restore the original pinctl in return.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check more strictly about the validity of pinctl values in
snd_hda_set_pin_ctl() and correct the wrong bits automatically.
Also provide the helper function to correct pinctl bits to codec
drivers.
This automatically fixes the invalid pinctl writes that are found in
a few Realtek fixups for NID 0x0f amp like ASUS A6Rp.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We already have the list of whole pin widgets and there is an unused
space in the list; let's use it for caching the current pinctl value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a DAC is reassigned from surrounds to front or ADCs are reduced
due to incomplete imux, we clear the path indices but the path
instances remain as is. Since the paths might be still referred
through the whole path list parsing (e.g. is_active_nid()), we should
clear these path instances as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since some codecs can choose the aamix as a capture source, we should
support it as well. When spec->add_stereo_mix_input flag is set, the
parser checks the availability of aamix as the input source, and adds
the paths automatically when possible.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current parser code, the input_paths[] may become inconsistent
when some of detected ADCs are dropped due to incomplete inputs, since
the driver rearranges only adc_nids[] but doesn't touch input_paths[].
This patch fixes the issue, and also it optimizes the reachability
checks by simply referring to the parsed input_paths[] instead of
calling is_reachable() again for each connection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of handling special cases in the caller side, give a proper
name string "Headphone Mic" from hda_get_autocfg_input_label() when
the headhpone jack pin is specified as an input.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture paths shouldn't contain the analog loopback mixer.
Pass a proper argument to exclude the aamix NID.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag spec->suppress_mic_auto_switch for codecs that don't
support unsol events properly like VT1708.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the default config value shows the connection AC_JACK_PORT_BOTH,
it's better to handle it as a speaker pin. This makes the behavior
consistent in snd_hda_get_pin_label() and snd_hda_parse_pin_defcfg().
There are only few old machines showing this attribute, and all of
them are actually fixed speaker pins, as far as I know.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit modifies the definition of snd_hda_parse_nid_path()
slightly, now with_aa_mix argument is changed to anchor_nid, so that
it can handle any NID generically as an anchor point to include or
exclude.
The with_aa_mix field in struct nid_path is removed again by this
change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The multi-io channels can vary not only from 1 to 6 but also may vary
from 6 to 8 or such. At the same time, there are more speaker pins
available than the primary output pins. So, we need three variables
to check: the minimum channel counts for primary outputs, the current
channel counts for primary outputs, and the minimum channel counts for
all outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of looking through paths with the dac -> pin connection at
each time, just pass the already parsed path index to
assign_out_path_ctls(). This simplifies the code a bit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The path indices must be reset at each evaluation of DAC assignment.
Otherwise the badness value will be wrongly calculated and mixers may
be inconsistently assigned.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Let is_jack_detectable() return true when the jack polling is enabled
for the codec. VT1708 uses the polling explicitly so we need to allow
it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new hook which is called at each PCM playback ops.
It can be used to control the codec-specific power-saving feature in
each codec driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bind-volume workaround was introduced for simplifying the mixer
abstraction in the case where one or more pins of multiple outputs
lack of individual volume controls. This was essentially the case
like Acer Aspire 5935, which has 5.1 speakers and 5.1 (multi-io)
jacks although there are 5 DACs, so some of them must share a DAC.
However, the recent code rewrite changed the DAC assignment policy to
share with the same channel instead of binding to the front, thus
binding the volumes for all channels makes little sense now, rather
it's confusing. So in this patch, the ugly workaround is finally
dropped and simply create the volume control corresponding to the
parsed path position.
For dual headphones or 2.1 speakers with a shared volume control, it's
anyway bound to the same DAC if needed, so this change shouldn't bring
any practical difference.
And, as a good bonus, we can cut off the whole code handling the bind
volume elements.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When 5.1 or more multiple speakers with found but not enough DACs are
available, it's better to bind such pins to the DACs of the primary
outputs with the same channels rather than binding to the first DAC
(i.e. the front channel). For the cases with two speaker pins, it's
rather regarded as front + bass combination, thus it's more practical
to still bind to the front, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... like "Speaker Surround Playback Switch".
This fix had been already applied to patch_conexant.c but was
forgotten in other places, then migrated to hda_generic.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For codecs that have individual routes going through a loopback mixer
(like VIA codecs), we need to provide an explicit switch to choose
whether the output goes through mixer or directly from DAC.
This patch adds the check for such paths and creates "Loopback Mixing"
enum control when available.
It won't influence on codecs like Realtek or others where the loopback
mixer is connected independently from the primary output routes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The output paths including aamix should be parsed only for the first
output. The surround paths including aamix must be wrong, since it
would mix all streams, i.e. all channels would be mixed into a single
and multiplexed again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call the path activation for the digital input pin properly, not only
setting the pin control. Also add spec->digin_path for keeping the
path index.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of search for the path with the certain route at each time,
keep the path index for each output and loopback, and just use it when
referred.
In this implementation, the path index number begins with one, not
zero (although I've been writing in C over decades). It's just to
make the check for uninitialized values easier.
So far, the input paths aren't handled with indices yet, but still
picked up via snd_hda_get_nid_path() at each time.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When speakers are chosen as the the primary output during evaluation,
we did some tricks to assign the possible multi-io jacks with a
certain offset value to multi_out dacs. This was a workaround for the
case with multiple speakers like Acer Aspire. But this is quite ugly
at the same time and the resultant code is hard to understand. More
badly, it works wrongly for 2.1 speakers like Apple iMac91.
In this patch, instead of fiddling with the offset to multi_out dacs,
simply add a certain badness number if headphone(s) + multi-ios are
possible. This simplify the code a bit, and it's more robust.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the requested path has been already added, return the existing path
instance instead of adding a duplicated instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the paths are created in map_singles(), we don't have to
re-create new paths in try_assign_dacs(). Just evaluate the badness
and skip to the next item.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set path->active flag at the path creation time and let the paths
initialized according to the current path->active state in
set_output_and_unmute(). This allows to modify the active flag of
some output paths dynamically, e.g. switching the front output route
with or without aamix like patch_via.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
activate_amp() in the generic parser checks whether the given NID is
included in any active paths and skips it if found. This was a
workaround for avoiding disabling the widgets in the active paths when
one path is disabled, thus it shouldn't be applied to the case for
path activation. Due to this wrong check, some analog loopback paths
haven't been initialized correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Manage the connection list cache using linked lists instead of
snd_array, and revive snd_hda_get_conn_list() again, so that we don't
have to keep the expanded values locally.
This will reduce the stack usage by recursive call of
snd_hda_get_conn_index() or parse_nid_path() of the generic parser.
The list management doesn't include any mutex protection, thus the
caller needs to take care of race appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another broken hardware workaround: there are hardware indicating
the inverted jack detection bit result.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the new flag, codec->inv_eapd, indicating that the EAPD
implementation is inverted.
There are always broken hardware in the world.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar like the implementation in patch_analog.c and patch_via.c,
the generic parser can provide the independent HP PCM stream now.
It's enabled when spec->indep_hp is set by the caller while parsing.
Currently no dynamic PCM switching as in patch_via.c is implemented
yet. The control returns -EBUSY when the value is changed during PCM
operations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the path including the loopback mixer widget in the primary
output channel as an alternative in the generic codec parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a better debug print code to show the new assigned paths in
generic parser. It appears only with CONFIG_SND_DEBUG_VERBOSE=y.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's never used in the generic parser. It was there from the old
Realtek code, which has been dropped quite ago, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a PCM name string is generated from the chip name, it might
become strange like "CX20549 (Venice) Analog". In this patch, the
parser tries to drop the invalid words like "(Venice)" in the PCM name
string. Also, when the name string is given beforehand by the caller,
respect it and use it as is.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There were some old codes that look not stable enough, which was
derived from the old Realtek code. The initialization for primary
output in init_multi_out() needs to consider the case of shared DAC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For preliminary works to migrate the generic parser for Conexant
codecs: the same function is ported to hda_generic.c.
But now it looks through the jack detect table so that it can cover
better.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a flag to indicate whether the vmaster mute hook enum is exposed
or not. Conexant codecs may want not to expose the control depending
on the model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Old codecs like AD1986A tend to have long paths as they were just made
to be the way like AC97. The current max depth 5 can be too short for
such devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The access to a cache array element could be invalid outside the
mutex, so copy locally for the later references.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dirty entry has to be checked at the beginning in the loop, not in
the inner loop for channels. This caused a regression that the right
channel isn't properly written.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bound capture volume and switch controls use the cached amp
updates, but it's missing the flushing at the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The inverted dmic fix overwrites the right channel amp value, but it
would work only when the amp values have been already actually
written. Put snd_hda_codec_resume_amp() before the amp write for
flushing caches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add an overflow check of CORB in HD-audio controller and codec drivers
so that flood of sequential writes would work properly.
In the controller side, add a check of CORB read-pointer to make
returning -EAGAIN when it's full. Meanwhile in the codec side, when
-EAGAIN error is received, it retries the write after flushing the
pending verbs (calling get_response() essentially does it).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These functions are supposed to be called at finishing the cached
sequential writes, so clear the flag properly for lazy developers who
often forget details.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When verbs or amps are actually written to the hardware, we can clear
dirty flag so that the later snd_hda_codec_resume_*() calls can skip
these verbs / amps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [2e9bf24: ALSA: hda_codec: Check for invalid zero
connections] trims the whole connection list when an invalid value is
reported by the hardware. But some codecs (at least AD1986A) may give
a zero NID in the middle of the connection list, so dropping the whole
list isn't good for such cases.
In this patch, as a workaround, allow a single zero NID in the read
connection list. If it hits zero twice, it's handled as an error, so
that we can avoid "too many connections" errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In general we prefer "Capture Source" to "Input Source".
The latter was chosen in many places just because "Capture Source"
label doesn't work well with the current alsa-lib mixer abstraction
when multiple instances are present. But when we know that there is a
single input-source element, we can safely choose "Capture Source"
label.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The next migration step is to use the common code in generic driver
for Realtek driver. This is no drastic change and there should be no
real functional changes, as the generic parser code comes from Realtek
driver originally.
As Realtek driver requires the generic parser code, it needs a
reverse-selection of CONFIG_SND_HDA_GENERIC kconfig.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These handlers are supposed to be called externally from the codec
drivers once when they need to handle own jack events.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When no controls are assigned in the parser (e.g. no analog path),
spec->kctls.list is still NULL. We need to check it before passing to
snd_hda_add_new_ctls().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some cases, we want to manipulate the auto_pin_cfg table before
passing to snd_hda_gen_parse_auto_config() (e.g. Realtek SSID check
code fiddles with the headphone pin). Also passing ignore_pins just
for snd_hda_parse_pin_defcfg() isn't good.
In this patch, snd_hda_gen_parse_auto_config() is changed to receive
the auto_pin_cfg table to be parsed. The passed auto_pin_cfg table
must have been initialized (typically by calling
snd_hda_gen_parse_auto_config()) beforehand by the caller.
Also together with this change, spec->parse_flags is also removed.
Since this was referred only at the place calling
snd_hda_parse_pin_defcfg(), no longer needed to be kept in spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally the whole generic parser code in Realtek driver is moved into
hda_generic.c so that it can be used for generic codec driver.
The old dumb generic driver is replaced. Yay.
The future plan is to adapt this generic parser for other codecs,
i.e. the codec driver calls the exported functions in generic driver
but adds some codec-specific fixes and setups.
As of this commit, the complete driver code is still duplicated in
Realtek codec driver. The big code reduction will come from now on.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch extends the capability of the auto-mic feature.
Instead of limiting the automatic input-source selection only to the
mics (internal, external and dock mics), allow it for generic inputs,
e.g. switching between the rear line-in and the front mic.
The logic is to check the attribute and location of input pins, and
enable the automatic selection feature only if all such pins are in
different locations (e.g. internal, front, rear, etc) and line-in or
mic pins. That is, if multiple input pins are assigned to a single
location, the feature isn't enabled because we don't know the
priority.
(You may wonder why this restriction doesn't exist for the headphone
automute. The reason is that the output case is different from the
input: the input source is an exclusive selection while the output
can be multiplexed.)
Note that, for avoiding regressions, the line-in auto switching
feature isn't activated as default. It has to be set explicitly via
spec->line_in_auto_switch flag in a fixup code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put INPUT_PIN_ATTR_FRONT after INPUT_PIN_ATTR_REAR, and define
INPUT_PIN_ATTR_LAST to point to the last element.
This is a preliminary work for cleaning up Realtek auto-mic parser.
In the auto-mic implementation, the front panel is preferred over the
rear panel. By arranging the attr definitions like in this commit, we
can simply use sort() for figuring out the priority order.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit passed an utterly wrong value for checking the
split inv dmic pin. This patch fixes it and also tries to remove
inv_dmic_split_idx field.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the fixup code is used commonly, it's worth to move it to the
common place, struct hda_codec, instead of keeping in hda_gen_spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To make the parser more generic, a few codes to handle the inverted
stereo dmic in a way Conexant parser does is added in this patch.
The caller should set spec->inv_dmic_split flag appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far we create only "Capture Volume" and "Capture Switch" controls
for binding all possible amps, but we'd prefer creating individual
capture volume and switch controls per input in some cases
(e.g. conexant parser does it).
Add a new flag, spec->multi_cap_vol, to follow that policy.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge a few functions that have been split due to historical reasons
to single functions. Splitting too much (and placing too far away)
actually worsens the readability.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few more cases where we can assign "Master" mixer element
safely, e.g. when a single DAC is used in the whole output paths.
Also, when vmaster hook is present, avoid "Master" but assign "PCM"
instead. Otherwise vmaster hook won't work properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... so that the fixup just needs to set the hook function in
FIXUP_ACT_PROBE. This will make easier to port for other codecs,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new field to indicate the possible pin NID for alternative vref
setup for the shared hp/mic. Although 0x18 is valid for all Realtek
codecs, it'll be different on other vendor's codecs.
Also, drop the sanity check in update_shared_mic_hp() since the
reference pin is set explicitly in the caller side.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the function more generic for both input and output directions,
and returns the assigned path pointer. The argument order is changed
to follow the standard (from, to) way.
Now this new function is used for analog input and loopback path
parser codes, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amps will be initialized via activate_path(), thus it's
superfluous to set in alc_auto_init_analog_input().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some fields from struct alc_spec, and clean up the usage.
Namely,
- spec->input_mux becomes a single element, private_imux[] is removed
- spec->adc_nids becomes an array by itself, and private_adc_nids[]
gets removed, too
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we reached to the final big piece of parser rewrite: the input
paths. While the old parser code assumes the more-or-less direct and
similar connections from input pin to ADC, the new code handles the
complete input paths. The capture source is switched by simple calls
of activate_path() function.
The parsing of capture volume and capture switches is, however, not
fully generalized. It assumes that amps are available in the vicinity
of ADCs (in three depth). This isn't perfect but it should cover all
codecs I know of.
Also, this commit removes some NID mapping of capture-related controls
temporarily for simplicity. It'll be restored in later commits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now we have a complete list of loopback paths, thus we can initialize
the paths more completely based on it, instead of assuming a direct
connection from pin to mixer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't forget to take boost volumes into account in the managed path
list. Since it's an additional volume, we need to extend the ctls[]
array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The paths used for multi-io haven't been initialized properly, so
far. It's usually no big matter because the pins are set to input as
default, but it's still cleaner to initialize the paths properly.
Now with the path active/inactive check, we can do it easily.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The pin widget has only a single amp value for the input even if it
has multiple "sources". Handle the situation in activate_path().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc_auto_is_dac_reachable() can be replaced fully with
is_reachable_path(). The only difference is the order of arguments.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... and rewrite the initialization of output paths as a generic
function that is applicable for both i/o directions.
The new flag, active, is introduced to each nid_path entry. This
indicates whether the given path is active, and it's used for checking
whether a certain widget can be turned off or changed when a path is
no longer used or newly enabled.
It's still used only in the output paths. More wider adaption for
input and loopback paths will be achieved in the later patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We are using only AUTOMUTE_MODE_PIN in patch_realtek.c and all others
have been already dropped. Let's remove the old superfluous codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new function snd_hda_codec_amp_init() (and the stereo variant)
initializes the amp value only once at the first access. If the amp
was already initialized or updated, this won't do anything more.
It's useful for initializing the input amps that are in the part of
the path but never used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For optimizing the verb executions, a new mechanism to cache the verbs
and amp update commands is introduced. With the new "write to cache
and flush" way, you can reduce the same verbs that have been written
multiple times.
When codec->cached_write flag is set, the further
snd_hda_codec_write_cache() and snd_hda_codec_amp_stereo() calls will
be performed only on the command or amp cache table, but not sent to
the hardware yet. Once after you call all commands and update amps,
call snd_hda_codec_resume_amp() and snd_hda_codec_resume_cache().
Then all cached writes and amp updates will be written to the
hardware, and the dirty flags are cleared.
In this implementation, the existing cache table is reused, so
actually no big code change is seen here. Each cache entry has a new
dirty flag now (so the cache key is now reduced to 31bit).
As a good side-effect by this change, snd_hda_codec_resume_*() will no
longer execute verbs that have been already issued during the resume
phase by checking the dirty flags.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When initializing the output paths, we assumed the input amps have
almost two inputs blindly. It's not only generic but even incorrect
for some codecs like ALC268 & co. Also, the same assumption (two
sources) exists for the bind input-amp controls.
This patch changes the codes in these places to handle the input
connections in a more generic way.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For handling the analog-loopback paths more generically, check the amp
capabilities of the aa-mixer widget, and create only the appropriate
mixer elements.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve the parser of analog loopback paths and handle in a more
generic way. The following changes are included in this patch:
- Instead of assuming direct connections between pins and
the mixer widget, track the whole path between them. This fixes
some missing connections like ALC660.
- Introduce the path list for loopback paths like input and output
path lists. Currently it's not used for any real purposes, yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like the output paths, parse the whole paths for inputs as well
and store in a path list. For that purpose, rewrite the output parser
code to be generically usable.
The input path list is not referred at all in this patch. It'll be
used to replace the fixed adc/capsrc array in later patches for more
flexible input path selections.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, idx[i] and multi[i] indicate the attribute of the widget
path[i - 1]. This was just for simplifying the code in
__parse_output_path(), but this is rather confusing for later use.
It's more natural if both idx[i] and multi[i] point to the same widget
of path[i]. This patch changes to that way.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify the output path initialization using the existing path
information instead of assuming the topology specific to Realtek
codecs. This is also implicitly a fix for some amp values on output
pins where the old parser missed (e.g. ALC260 output pins).
The same function alc_auto_set_output_and_unmute() can be used now for
the multi-io activation, since the output selection means nothing but
activating the given output path.
And, finally at this stage, we can get rid of alc_go_down_to_selector()
and other functions that are codec really specifically to Realtek
codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, Realtek codec driver evaluates the NIDs for volume and mute
controls twice, once while parsing the DACs and evaluating the
assignment, and another while creating the mixer elements. This is
utterly redundant and even fragile, as it's assuming that the ctl
element evaluation is identical between both parsing DACs and creating
mixer elements.
This patch simplifies the code flow by doing the volume / mute
controls evaluation only once while parsing the DACs. The patch ended
up in larger changes than expected because of some cleanups became
mandatory.
As a gratis bonus, this patch also fixes some cases where the stereo
channels are used wrongly for mono amps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mono widget is always connected to the left channel, thus the left
channel amp value also should be referred for mono widgets instead of
the right channel.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As we parse the output paths more precisely now, we can use this path
list for parsing the widgets for volume and mute mixer controls.
The spec->vol_ctls[] and sw_ctls[] bitmasks are replaced with the
ctls[] in each output path instance.
Interestingly, this move alone automagically fixes some bugs that the
conflicting volume or mute NIDs weren't properly detected.
Also, by parsing the whole path, there are more chances to get a free
widget for volume/mute controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the output path parser to Realtek codec driver as we already have
in patch_via.c. The nid_path struct represents the complete output
path from a DAC to a pin. The alc_spec contains an array of these
paths, and a new path is added at each time when a new DAC is
assigned.
So far, this path list is used only in limited codes: namely in this
patch, only alc_is_dac_already_used() checks the list instead of dac
arrays in all possible outputs. In the later development, the path
list will be referred from more places, such as the mixer control
assignment / check, the mute/unmute of active routes, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the probing phase, create a list of all available DACs in the codec
and use it for checking the single DAC connections.
This list will be used in more other places in the later commits, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mengdong Lin