If LPIB reports a pretty bad value, we can't trust such hardware for
calculating the PCM delay. Automatically turn off the delay counting
when such a problem is encountered.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=48911
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have a generic unsol mechanism, we can implement a generic
poll loop, which can be used for debugging, or if a codec's unsol
mechanism is broken.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec,
this patch modify patch_conexant.c to add Lemote specific code.
Both A1004 and A1205 use the same pin configurations, but A1004 need
to increase the default boost of internal mic.
Signed-off-by: Jie Chen <chenj@lemote.com>
Signed-off-by: Huacai Chen <chenhc@lemote.com>
Signed-off-by: Hongliang Tao <taohl@lemote.com>
Signed-off-by: Hua Yan <yanh@lemote.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Delay the registration of VGA switcheroo client to the end of the
probing. Otherwise a too quick switching may result in Oops during
probing.
Also add the check of the return value from snd_hda_lock_devices().
Reported-and-tested-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The white-list entries of position_fix for ASUS laptops have been
added just as a workaround for broken COMBO mode. Now the combo mode
itself is disabled, we can safely remove these entries.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This caused the internal speaker to mute itself because it was
present, which happened after powersave.
It was found on Dell XPS 15 (L502x), ALC665.
Reported-by: Da Fox <da.fox.mail@gmail.com>
Cc: stable@vger.kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that the COMBO position fix mode is rather more harmful,
and it got reverted (with the replacement of runtime->delay
calculation) recently. Hence we can get rid of AZX_DCAPS_POSFIX_COMBO
as well.
It's still possible to pass this mode via position_fix module option,
in case where this really helps on weird machines (who knows).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the transition to the generic fixup code, the call of
snd_hda_gen_init() and snd_hda_gen_free() was missing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [4b527b65 ALSA: hda - limit internal mic boost for Asus
X202E] introduced the use of auto-parser code, but it forgot to add
struct hda_gen_spec at the head of codec->spec which the auto-parser
assumes silently. Without this record, it may result in memory
corruption.
This patch adds the missing piece.
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell Latitude 5x30 and 6x30 series of machines all have
a single 4-pin headset jack. Enabling line in mode for such jack
is very confusing (you would only get mono input, and would have to
use non-standard adapters), so remove the option by default.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From what I can conclude all GPIO handling was removed in 2009.
Remove dead code remnants.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case there is one "Headphone Jack" and one "Dock Headphone Jack",
one of them will get an index, even though that is not needed.
This patch fixes that issue.
BugLink: https://bugs.launchpad.net/bugs/1060729
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the recent patch "ALSA: hda - use both input paths on Conexant
auto parser" suddenly we can have more than one "Mic Boost", this
happened on Acer Aspire One 722. Therefore we must add the possibility
to put an index on this "Mic Boost" just as we do for the other
"Mic Boost" earlier in the same function.
BugLink: https://bugs.launchpad.net/bugs/1059523
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There was a race condition when the system suspends while hda_power_work
is running in the work queue. If system suspend (snd_hda_suspend)
happens after the work queue releases power_lock but before it calls
hda_call_codec_suspend, codec_suspend runs with power_on=0, causing the
codec to power up for register reads, and hanging when it calls
cancel_delayed_work_sync from the running work queue.
The call chain from the work queue will look like this:
hda_power_work <<- power_on = 1, unlock, then power_on cleard by suspend
hda_call_codec_suspend
hda_set_power_state
snd_hda_codec_read
codec_exec_verb
snd_hda_power_up
snd_hda_power_save
__snd_hda_power_up
cancel_delayed_work_sync <<-- cancelling executing wq
Fix this by waiting for the work queue to finish before starting suspend
if suspend is not happening on the work queue.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define()
always after fixup"), alc_auto_parse_customize_define was moved after
detection of ALC271X.
The problem is that detection of ALC271X relies on spec->cdefine.platform_type,
and it's set on alc_auto_parse_customize_define.
Move the alc_auto_parse_customize_define and its required fixup setup
before the block doing the ALC271X and other codec setup.
BugLink: https://bugs.launchpad.net/bugs/1006690
Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo IdeaPad U310 has an internal mic where the right channel
is phase inverted.
Signed-off-by: Felix Kaechele <felix@fetzig.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For less duplication of code between codecs, and to make it easier
in the future to improve code for all codecs simultaneously.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Moving towards less duplication of code between codecs - this patch
takes some of the common code of unsol event handling and makes it
generic.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
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Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.7
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
Tested with LPIB delay without any issues.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DMA Position in Buffer (DPIB) should be used for
ring buffer management, while LPIB register provides
information on the number of samples transfered on
the link. The difference between the two pieces of
information corresponds to hardware/DMA buffering.
This patch reports this difference in runtime->delay, and
removes the use of the COMBO mode on recent Intel hardware.
Credits to Takashi Iwai for an initial patch.
[rebased to for-next branch and replaced snd_printk() with
snd_printdd() by tiwai]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SSYNC bits are typically used to start multiple
streams synchronously. It makes sense to use them
for a single stream for a more predictable startup
sequence. The transfers only start once the DMA and
FIFOs are ready. This results in a better correlation
between timestamps and number of samples played.
Credits to Kar Leong Wang for suggesting this
improvement.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If headphone jack can't detect plug presence, and we have the jack in
the jack table, snd_hda_jack_detect will return the plug as always
present (as it'll be considered as a phantom jack). The problem is that
when this happens, line out pins will always be disabled, resulting in
no sound if there are no headphones connected.
This was reported as a no sound problem after suspend on
http://bugs.launchpad.net/bugs/1052499, since the bug doesn't manifests
on first initialization before the phantom jack is added, but on resume
we reexecute the initialization code, and via_hp_automute starts
reporting HP always present with the jack now on the table.
BugLink: https://bugs.launchpad.net/bugs/1052499
Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
Cc: <stable@vger.kernel.org> [v3.6+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Thinkpad W520 - and probably several other machines with
Conexant 506x chips - the Dock Mic and Mic are connected to the
same two selector nodes. This patch will make Dock Mic take one
selector node and Mic take the other, when possible.
Without the patch, both paths would take the first selector,
leading to the normal Mic's volume being controlled by
"Dock Mic Boost".
(On other machines, this could instead fixup similar problems between
Mic and Line In, for example.)
BugLink: https://bugs.launchpad.net/bugs/1037642
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard name (and what PulseAudio picks up) is "Dock Mic",
not "Docking Mic" or "Docking-Station".
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the input gain for the internal mic is set to its maximum level,
the background noise becomes so high - and any relevant signal clipped -
that the setting becomes unusable. It is better to limit the amplification.
BugLink: https://bugs.launchpad.net/bugs/1052460
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some of new HP laptops have a LED for microphone (or recording) mute,
and it's controlled by GPIO pin 3.
Bind this with the capture switch to turn it on/off properly by the
mixer change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the default value of position_fix -1, and allow user passing
position_fix=0 explicitly to set the "auto" position-fix mode.
Otherwise the auto mode may be switched to others like COMBO of
VIACOMBO when the controller prefers it, thus user can't set the auto
mode any longer.
Also updated the documentation appropriately, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the definitions of CS420X_IMAC27_122 and CS420X_APPLE as
aliases, the rest enums are set to duplicated values unexpectedly.
Move the alias definitions at the end so that the enum values are
defined in the proper order.
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MacBook Pro 10,1 needs a few adjustments to make it working:
- more COEF verbs
- some pin config overrides to disable the unused pin (0x0d, 0x12),
and fix the internal mic (0x0e)
In addition, it uses GPIO 1 and 3 like other MacBooks.
The internal digital mic on the machine is still problematic: it seems
that only the right channel is used and the left is always static.
This looks like a hardware design, so we need to cope in the software
side somehow...
The primary information and test were brought from Daniel J Blueman.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_pick_fixup() didn't check the case where the device mask bits
are set, typically used for SND_PCI_QUIRK_VENDOR() entries. Fix this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Originally the bogus period at BDL head was introduced as a workaround
for the mismatching position update at the period boundary, typically
seen on dmix. However, for applications like PulseAudio that don't
require period wake ups, this workaround is just superfluous. Thus
better to disable it when no_period_wakeup is given in hw_params.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit c20c5a841c changed some chipsets to
default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of
POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including
repeated sounds on my Asus laptop.
My hardware is Cougar Point which the commit log of
c20c5a841c mentions as tested so POS_FIX_COMBO
probably works in general but apparently it doesn't on Asus K53E therefore the
need for the quirk.
Signed-off-by: Catalin Iacob <iacobcatalin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_codec_reset() calls restore_pincfgs() where the codec is
powered up again, which eventually tries to resume and initialize via
the callbacks of the codec. However, it's the place just after codec
free callback, thus no codec callbacks should be called after that.
On a codec like CS4206, it results in Oops due to the access in init
callback.
This patch fixes the issue by clearing the codec callbacks properly
after freeing codec.
Reported-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The purpose of this flag is unclear. If the problem is that some machines
have broken misc/NO_PRESENCE bits, they should be fixed by pin fixups.
In addition, this causes jack detection functionality to be flawed on
the M31EI, where there are two jacks without jack detection (which is
properly marked as NO_PRESENCE), but due to ignore_misc_bit, these
jacks are instead being reported as being present but always unplugged.
BugLink: https://bugs.launchpad.net/bugs/939161
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... instead of the standard fixed channel maps.
The generic HDMI is based on the audio infoframe, and its configuration
can be selected via CA bits. Thus we need a translation between the
CA index and the verbose channel map list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although HD-audio allows pair-wise channel configurations, only the
fixed channel positions are used in this version. In future, this can
be changed and allow user to modify the channel positions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently the check for non-PCM stream state was added to the generic
HDMI driver code. But this check should be done rather to each pin
instead of each converter. Otherwise when a different converter is
assigned at the next open, the audio infoframe can be inconsistent
with the setup using the previous converter.
For fixing this issue, this patch moves the state of the current
non-PCM status from per_cvt to per_pin. (In addition an unused
argument cvt_nid is stripped from hdmi_setup_channel_mapping())
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For multiple speaker outs, the names were previously
"Speaker,0", "Speaker,1", "Center"/"LFE", "Speaker,3". This is
inconsistent, confusing, and is not picked up correctly by PulseAudio.
Instead use "Front", "Surround", "Center"/"LFE", "Side" which
is more standard.
BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1046734
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the commit [2faa3bf: ALSA: hda - Rewrite the mute-LED hook with
vmaster hook in patch_sigmatel.c], the former Master volume control
was converted to PCM. This was supposed to be covered by the vmaster
control. But due to the lack of "PCM" slave definition, this didn't
happen properly. The patch fixes the missing entry.
Reported-by: Andrew Shadura <bugzilla@tut.by>
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For HBR stream test, use straight channel mapping way.
when switched back to "speaker-test -c8", even the audio
infoframe is up-to-date, there should be correct channel mapping setup.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDMI channel remapping apparently effects HBR packets on Intel's chips.
For compressed non-PCM audio, use "straight-through" channel mapping.
For uncompressed multi-channel pcm audio, use normal channel mapping.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The array channel_allocations[] is an ordered list, add function to get
correct order by ca_index.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call of pm_notify callback in snd_hda_codec_free() should be with
the check of the current state whether pm_notify(false) is called or
not, instead of codec->power_on check.
For improving the code readability and fixing this inconsistency,
codec->d3_stop_clk_ok is renamed to codec->pm_down_notified, and this
flag is set only when runtime PM down is called. The new name reflects
to a more direct purpose of the flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_HDA_POWER_SAVE is no longer an experimental feature and its
behavior can be well controlled via the default value and module
parameter. Let's just replace it with the standard CONFIG_PM.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a codec provides its own set_power_state op, the D3-clock-stop
isn't checked correctly. And the recent changes for repeating the
state-setting operation isn't applied to such a codec, too.
This patch fixes these issues by moving the call of codec's own op to
the place where the generic power-set operation is done, and move the
power-state synchronization code out of
snd_hda_set_power_state_to_all() so that it can be called always at
the end of power-up/down sequence, and updates the D3 clock-stop flag
properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the HD-audio is removed, it leaves the refcounts when codecs are
powered up (usually yes) in the destructor. For fixing the unbalance,
and cleaning up the code mess, this patch changes the following:
- change pm_notify callback to take the explicit power on/off state,
- check of D3 stop-clock and keep_link_on flags is moved to the caller
side,
- call pm_notify callback in snd_hda_codec_new() and snd_hda_codec_free()
so that the refcounts are proprely updated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These codecs seem reporting EPSS but require longer delay for the
proper D3 transition. For example, D3_STOP_CLOCK_OK bit won't be set
correctly even after D3.
In this patch, codec->epss flag is overridden for avoid the
misbehavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
EPSS parameter should be static, so we can read it once and remember.
This also allows more easily to override the wrong EPSS capability
reported from a codec by changing the flag in the codec
initialization step.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use unsigned int to make clear that the codes required only for
modules will be reduced by the compiler optimization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch to support runtime PM introduced a bug:
Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk'
'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But
there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE.
This patch is to fix the bug.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Runtime PM can bring more power saving:
- When the controller is suspended, its parent device will also have a chance
to suspend.
- PCI subsystem can choose the lowest power state the controller can signal
wake up from. This state can be D3cold on platforms with ACPI PM support.
And runtime PM can provide a gerneral sysfs interface for a system policy
manager.
Runtime PM support is based on current HDA power saving implementation. The user
can enable runtime PM on platfroms that provide acceptable latency on transition
from D3 to D0.
Details:
- When both power saving and runtime PM are enabled:
-- If a codec supports 'stop-clock' in D3, it will request suspending the
controller after it enters D3 and request resuming the controller before
back to D0. Thus the controller will be suspended only when all codecs are
suspended and support stop-clock in D3.
-- User IO operations and HW wakeup signal can resume the controller back to
D0.
- If runtime PM is disabled, power saving just works as before.
- If power saving is disabled, the controller won't be suspended because the
power usage counter can never be 0.
More about 'stop-clock' feature:
If a codec can support targeted pass-through operations in D3 state when there
is no BCLK present on the link, it will set CLKSTOP flag in the supported power
states and report PS-ClkStopOk when entering D3 state. Please refer to HDA spec
section 7.3.3.10 Power state and 7.3.4.12 Supported Power State.
[Fixed CONFIG_PM_RUNTIME dependency in hda_intel.c by tiwai]
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling the jack sync in the init callback of each codec,
call it generically at initialization and resume. By calling it at
the last of resume sequence, a possible race between the jack sync and
the unsol event enablement in the current code will be closed, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes an issue with a machine where there were no speakers,
but GPIO0 had to be data=1 for the headphone to be functioning.
I'm not sure if we need a more advanced patch to solve all possible cases,
but if so, this patch would still provide a minor optimisation.
BugLink: https://bugs.launchpad.net/bugs/1040077
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the codec turn-on operation is canceled by the immediate
power-on, the driver left the power_transition flag as is.
This caused the persistent avoidance of power-save behavior.
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... by calling the newly introduced snd_hda_power_sync().
I had to reimplement a wheel for adding the trigger at changing the
parameter -- the parameter set ops is overwritten to pass the integer
parameter, then trigger the power-state sync.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new helper function snd_hda_power_sync() to trigger the
power-saving manually. It's an inline function call to
snd_hda_power_save() helper function.
Together with this addition, snd_hda_power_up*() and
snd_hda_power_down() functions are inlined to a call of the same
snd_hda_power_save() helper function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's possible that these amps are settable somehow, e g through
secret codec verbs, but for now, don't create the controls (as
they won't be working anyway, and cause errors in amixer).
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1038651
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of blindly initializing a volume knob widget, first check
that there actually is a volume knob widget.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a simple support of automute for the front HP jack to AD1882
stack model. Such an addition is basically an exception -- we really
want to avoid the static quirk codes, but AD1882 parser isn't still
ready for moving to the BIOS auto-parser yet. So, as a quick fix, I
merged it for now.
In near future, we really need the big clean up of patch_analog.c to
move on to the auto-parser...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace with a proper ifdef check of CONFIG_PM_SLEEP in hda_intel.c.
But other places in HD-audio driver are still marked with CONFIG_PM,
since these can be called for power-saving even without
CONFIG_PM_SLEEP.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Conexant devices (e g CX20590) have no mute capability on
their Beep widgets.
This patch makes sure we don't try setting mutes on those widgets.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As spec said, 1 indicates no copyright is asserted.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The return value of snd_hda_param_read() is -1 for an error, otherwise
it's the supported power states of a codec.
The supported power states is a 32-bit value. Bit 31 will be set to 1
if the codec supports EPSS, thus making "sup" negative. And the bit
28:5 is reserved as "0".
So a negative value other than -1 shall be further checked.
Please refer to High-Definition spec 7.3.4.12 "Supported Power
States", thanks!
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When "Beep Playback Switch" had a different value on left and right
channels (such as muting left but not right, or vice versa), this
could result in the right channel being ignored.
This patch enables beep to be sounding from right channel only, and
also give correct result back to userspace (e g amixer).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH
Signed-off-by: James Ralston <james.d.ralston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For processing the firmware handling properly for built-in kernels,
implement an asynchronous firmware loading with
request_firmware_nowait(). This means that the codec probing is
deferred when the patch option is specified.
Tested-by: Thierry Reding <thierry.reding@avionic-design.de>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary work for the deferred probing for
request_firmware() errors at init.
This patch moves the call of request_firmware() to hda_intel.c, and
call it in the earlier stage of probing rather than
azx_probe_continue().
Tested-by: Thierry Reding <thierry.reding@avionic-design.de>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To turn off pin control for the pin was tested, and helped against
this issue.
BugLink: https://bugs.launchpad.net/bugs/1034779
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CA0132 driver had some codes to handle the S/PDIF I/O, but the actual
setups of pins and converters were missing. Now the pins are added.
Also, fixed a few points triggering invalid codec verbs and mixer
elements since the digital I/O audio widgets on CA0132 have no amp.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now with the workaround using codec->pcm_format_first flag, we can
clean up the home-baked codes in patch_ca0132.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduced a new flag to set up the PCM stream format at first before
the stream_id and channel tag. Some codecs (e.g. CA0132) seem
preferring this over stream_id -> format order.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The same ID is twice in the quirk table, so the second one is not used.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When CONFIG_SND_VERBOSE_PRINTK=y is set, the debug print in
hda_auto_parser.c looks really ugly like:
ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0
ALSA sound/pci/hda/hda_auto_parser.c:334 dig-out=0x12/0x0
ALSA sound/pci/hda/hda_auto_parser.c:335 inputs:
ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x11ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0x10
ALSA sound/pci/hda/hda_auto_parser.c:341
ALSA sound/pci/hda/hda_auto_parser.c:343 dig-in=0x13
Better to put one item at each line.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that the auto model is the default, these quirks are redundant
and can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This computer is confirmed working with model=auto on kernel 3.2.
Also, parsing fails with hda-emu with the current model.
Cc: stable@kernel.org (3.2+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As with the ThinkPad Models X230 Tablet and T530 the X230 needs a qurik to
correctly set up the pins for the dock port.
Signed-off-by: Felix Kaechele <felix@fetzig.org>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit c4bfe94a causes a regression on some codecs at probing.
Since this was just a workaround to shut up a kernel warning, it'd be
better to revert and fix properly. So we ended up with re-adding the
cleanup callback.
Tested-and-reported-by: Matt Horan <matt@matthoran.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a model/fixup string "lenovo-dock", for Thinkpad T430s, to allow
sound in docking station.
Tested on Lenovo T430s with ThinkPad Mini Dock Plus Series 3
Cc: stable@kernel.org
Signed-off-by: Philipp A. Mohrenweiser <phiamo@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lenovo Thinkpad T530 with ALC269VC codec has a dock port but BIOS
doesn't set up the pins properly. Enable the pins as well as on
Thinkpad X230 Tablet.
Reported-and-tested-by: Mario <anyc@hadiko.de>
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The IDT codecs initializes the GPIO setup for mute LEDs via
snd_hda_sync_vmaster_hook(). This works in most cases except for the
very first call, which is called before PCM and control creations.
Thus before Master switch is set manually via alsactl, the mute LED
may show the wrong state, depending on the polarity.
Now it's fixed by calling the LED-status update function manually when
no vmaster is set yet.
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit a3e199732b made the LED working again on HP Mini 210 but
with a wrong polarity. This patch fixes the polarity for this
machine, and also introduce a new model string "hp-inv-led".
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=772923
Cc: <stable@vger.kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On recent kernels, Realtek codec parser tries to optimize the routing
aggressively and take the headphone output as primary at first. This
caused a regression on VAIO Z with ALC889, the silent output from the
speaker.
The problem seems that the speaker pin must be connected to the first
DAC (0x02) on this machine by some reason although the codec itself
advertises the flexible routing with any DACs.
This patch adds a fix-up for choosing the speaker pin as the primary
so that the right DAC is assigned on this device.
Reported-and-tested-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix to converter detaching timing in patch_hdmi.c
leads to a kernel WARNING due to a sanity check when the debug
option is set. Add a workaround by setting a dummy hinfo->nid.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The generic HDMI code detaches the converter from the stream when
unused, but it must be done rather in the close callback instead of
the cleanup callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on HP Mini 210 doesn't provide the proper "HP_Mute_LED" DMI
string, thus the driver doesn't initialize the GPIO, too. In the
earlier kernel, the driver falls back to GPIO1, but since 3.3 we've
stopped this due to other wrongly advertised machines.
For fixing this particular case, add a new model type to specify the
default polarity explicitly so that the fallback to GPIO1 is handled.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=772923
Cc: <stable@vger.kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got a bug report about the silent output from the headphone on a
mobo with VT2021, and spotted out that this was because of the wrong
D3 state on the DAC for the headphone output. The bug is triggered by
the incomplete check for this DAC in set_widgets_power_state_vt1718S().
It checks only the connectivity of the primary output (0x27) but
doesn't consider the path from the headphone pin (0x28).
Now this patch fixes the problem by checking both pins for DAC 0x0b.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also add a model/fixup string "lenovo-dock", so that other Thinkpad
users will be able to test this fixup easily, to see if it enables
dock I/O for them as well.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1026953
Tested-by: John McCarron <john.mccarron@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Turn on the pin widget's PIN_OUT bit from playback prepare. The pin is
enabled in open, but is disabled in hdmi_init_pin which is called during
system resume. This causes a system suspend/resume during playback to
mute HDMI/DP. Enabling the pin in prepare instead of open allows calling
snd_pcm_prepare after a system resume to restore audio.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This merges the changes for converting to new PM ops for platform
and some other drivers.
Also move some header files to local places from the public
include/sound.
A recent commit made patch_alc268 call snd_hda_pick_fixup with
NULL quirk pointer. Make sure we do not reference that NULL pointer.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This codec has a separate dmic path (separate dmic only ADC),
and thus it looks mostly like ALC275.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1025377
Tested-by: Ray Chen <ray.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since some jack controls may be renamed as phantom jacks, the existing
check for index conflicts doesn't work because it simply compares the
name with the last used name, assuming that the controls with the same
name continue. Thus, it would result in the duplicated controls when
two or more phantom jacks with the very same type exist, and the
driver gives up with an error.
This patch fixes the problem by checking the index number conflicts
more intensively (but dumbly).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vendor ID 0x10de0051 is used by a yet-to-be-named GPU chip.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Acked-by: Andy Ritger <aritger@nvidia.com>
Reviewed-by: Daniel Dadap <ddadap@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
this patch add proper id for Haswell HDA Controller.
[Added AZX_DCAPS_POSFIX_COMBO flag by tiwai]
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the model parsers out of the way, we have no custom unsol
events to worry about, we can therefore simplify the code.
In addition, this fixes a bug on the ASUS TC710, which has only
a headphone jack and a mic jack, but no internal mic or speakers.
Therefore the unsol_event pointer was not set, and as a result,
the jack kcontrols were not correctly updated.
BugLink: https://bugs.launchpad.net/bugs/1021192
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call cancel_work_sync() when turning off the beep switch so that the
mute call is executed in a proper order.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's no longer necessary since beep_mode=2 option was dropped.
It can be checked simply via codec->beep != NULL.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move snd_hda_mixer_amp_switch_put_beep() to hda_beep.c as a clean up
to remove one more ifdef.
Also add the corresponding get callback to return consistently the
digital beep state independently from the mixer amp value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The beep_mode=2 option was introduced to make the beep mixer
controlling the beep input allocation/deallocation dynamically, so
that a user can switch between HD-audio codec digital beep and the
system beep only via mixer API. This was necessary because the
keyboard driver took only the first input beep instance at that time.
However, the recent keyboard driver already processes the multiple
input instances, thus there is no point to keep this mode.
Let's remove it.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Straightforward conversion to the new pm_ops from the legacy
suspend/resume ops.
Since we change vx222, vx_core and vxpocket have to be converted,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Windows use hidden register to control EAPD.
Linux use verb to control EAPD.
If windows reboot to Linux, it must change the EAPD control to verb
control.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This 3-pin jack was labeled "Headphone Jack", but it could also be
used as a mic jack just by switching "Input Source". Therefore we need
to call the jack something else, to make sure PulseAudio can use the
speaker together with the external mic. (PulseAudio might mute the
speaker if it detects a headphone being plugged in.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ASUS device has a single 3-pin jack that can either be a mic or
a headphone, but the pin does not have VREF capabilities. We've been
told by Realtek to instead enable VREF on pin 0x18 in that case.
BugLink: https://bugs.launchpad.net/bugs/1018262
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for power-map controls (commit b0791dda81) caused
regressions on some other HP laptops. They have fixed pins but these
pins are exposed as jack-detectable. Thus the driver tries to control
the power-map dynamically per jack detection where it never gets on.
This patch adds the check of connection and it assumes the no jack
detection is available for fixed pins no matter what pin capability
says.
BugLink: http://bugs.launchpad.net/bugs/1013183
Reported-by: Luis Henriques <luis.henriques@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This chip looks very similar to ALC269 and ALC27* variants. The bug reporter
has verified that sound was working after this patch had been applied.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1017017
Tested-by: Richard Crossley <richardcrossley@o2.co.uk>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For convenience, add "inv-dmic" model string for enabling the inverted
internal mic workaround to possible Realtek codecs, so far,
ALC882-variants, ALC262, ALC268, ALC269-variants, and ALC662-variants.
Also, the model strings for hardware inv-dmic workarounds,
"alc269-dmic" and "alc271-dmic", are added for ALC269(VA) and ALC271
codecs as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some laptops are equipped with ForteMedia digital mics that give the
differential input. With such devices, summing stereo streams into a
mono (like PulseAudio does) results in almost silence.
This patch provides a workaround for this bug by adding a new mixer
switch to turn on/off the right channel of digital mic, just like a
similar fix for Conexant codecs.
When the new switch "Inverted Internal Mic Capture Switch" is off and
the current input source is the digital mic, the right channel of the
recording stream is muted. When another input source is selected, the
right channel is restored.
Tested-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After cancel_delayed_work_sync returns, the power down work either never
started (power_on == 1) or finished (power_on == 0). In the former case
there is no need to power up again.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Lenovo Ideapad S205 has an internal mic where the right channel
is phase inverted.
BugLink: https://bugs.launchpad.net/bugs/884652
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec ALC272X is a special codec for some Dell machines, and its
detection got broken in the recent kernel because SSID check (required
by ALC272X check) was moved to the later point. Now we need to move
this codec check to the right place, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the recent code, the value shown there is a tag number, and it's no
longer same as the pin nid. Correct the message to avoid confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VIA codecs seem not returning the event tag in the unsolicited events,
thus the current code relying on the tag value doesn't work.
Since simple_hdmi stuff has only a single pin, we can use simply
snd_hda_jack_set_dirty_all() to activate the pin-detection
independently from the tag value.
Tested-by: Annie Liu <AnnieLiu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The simple_hdmi stuff is designed only for a single pin and a single
converter (thus a single PCM stream), and no need for loops.
Let's flatten the code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codes to initialize work struct or create a proc interface should
be called only once and never although it's called many times through
the init callback. Move that stuff into patch_generic_hdmi() so that
it's called only once.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This fixes the regression introduced by the commit d0b1252d for
refactoring simple_hdmi*(). The pin NID wasn't assigned correctly.
Reported-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This addresses an issue encountered when a pcm is opened while
transitioning to low power state (codec->power_on == 1 &&
codec->power_transition == -1). Add snd_pcm_power_up_d3wait to
hda_codec. This function is used to power up from azx_open as opposed
to snd_hda_power_up used from codec_exec_verb. When powering up from
azx_open, wait for pending power downs to complete, avoiding the power
up continuing in parallel with the power down on the work queue.
The specific issue seen was with the CS4210 codec, it powers off the ADC
and DAC nid in its suspend handler. If it is re-opened before the
~100ms power down process completes, the ADC and DAC nid are initialized
while powered down and audio is lost until another suspend/resume cycle.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATI and Nvidia HDMI codecs have also the pin-detection capability,
so let's enable the jack-detecion for them, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The code refactoring using the same helper functions for sharing the
codes among ATI, Nvidia and VIA simple_hdmi* stuff. Except for that
spec->pcm_playback is no longer pointer, the functionality doesn't
change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the hotplug unsol event handling to simple_hdmi*().
It works on VIA VX900. If AMD or Nvidia chips support the
pin-detection similarly, it can be added easily, too.
Reported-by: Annie Liu <annieliu@viatech.com.cn>
Tested-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch activates the COMBO position_fix for recent Intel client chipsets.
COMBO mode is the recommended setting for Intel chipsets and eliminates HD
audio warnings in dmesg. This patch has been tested on Lynx Point, Panther
Point, and Cougar Pont.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PulseAudio sometimes have difficulties knowing that there is a
"Speaker" or "Internal Mic", if they have no individual volume
controls or selectors. As a result, only e g "Headphone" might
be created for a laptop, but no "Speaker".
To help out, create phantom jacks (that are always present,
at least for now) for "Speaker", "Internal Mic" etc, in case we
detect them.
The naming convention is e g "Speaker Phantom Jack".
In order not to pollute the /dev/input namespace with even more
devices, these are added to the kcontrols only, not the input devices.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the split to the auto-parser helper functions, the actual call
of init verbs was lost.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43366
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pin 0x1b was connected to the front panel connector, which according to
the HDA standard should contain a mic and a headphone. In this case,
the headphone was listed as "line out" by BIOS.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/993162
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI ID entries of Creative SoundCore3D HD-audio controllers should
be before the wildcard for vendor = Creative.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is patch supporting the CODECes of HD-Audio function of VIA GFX
cards which support HDMI.
For CODECes 0x9f80/0x9f81, which belong to VX900, since the hardware
is not fully compliant to HD-Audio 1.3, simple_i*() is adopted
temporarily.
Signed-off-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is patch supporting HD-Audio function of VIA GFX cards which
support HDMI.
Those are integrated graphics of chipsets VX900 and VX11 separately.
Signed-off-by: Annie Liu <annieliu@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When VGA_SWITCHEROO support is enabled hda_intel initialises the HDMI
audio device on the current VGA device. When it's not enabled it only
initialises the HDMI device on the default VGA adaptor, this means
secondary cards get no audio support which is very unhelpful for
multi-seat!
With this patch, when SUPPORT_VGA_SWITCHEROO is disabled hda_intel
initialises all HDMI audio devices, not just the default VGA.
[minor optimizations by tiwai]
Signed-off-by: Steven Newbury <steve@snewbury.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When VGA-switcheroo is built in but unused on systems with multiple
graphics cards, the initializations of non-default graphics cards are
skipped and never enabled (because the switcheroo is activated only
when the controller supports). The current behavior is for avoiding
the system lockup by accessing the disabled GPU, but due to the recent
change in VGA-switcheroo, it determines the state simply by checking
with the default VGA device. This is the culprit.
Now with the new vga_switcheroo_get_client_state(), we can know the
initial state of the bound GPU, thus can determine the initial audio
client state more correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
codec may reject power state transition requests(reporting PS-ERROR set),
in that case we re-issue a power state setting and check error bit again.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_hda_param_read() return value -1 means error, others are responses
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
if EPSS supported, transition from D3 state to D0 state in less
than 10ms
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add function to check whether power states supported by specific
codec node.
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
add more power states information:
- reset status
- clock stop ok
- power states error
Output like:
Power: setting=D0, actual=D0, Error, Clock-stop-OK, Setting-reset
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the second updates for 3.5-rc1. It's mainly for OMAP4 HDMI
updates and the device tree updates for OMAP, in addition to a couple
of PCM accuray improvement and Realtek ALC269VD codec support.
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Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound update from Takashi Iwai:
"This is the second updates for 3.5-rc1. It's mainly for OMAP4 HDMI
updates and the device tree updates for OMAP, in addition to a couple
of PCM accuray improvement and Realtek ALC269VD codec support."
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (21 commits)
ALSA: hda/realtek - Add new codec support for ALC269VD
ALSA: core: group read of pointer, tstamp and jiffies
ASoC: OMAP: HDMI: Rename sound card source file
ASoC: OMAP: HDMI: Make sound card naming more generic
ASoC: OMAP: HDMI: Make build config options more generic
ASoC: OMAP: HDMI: Expand capabilities of the HDMI DAI
ASoC: OMAP: HDMI: Improve how the display state is verified
ASoC: OMAP: HDMI: Expand configuration of hw_params
ASoC: OMAP: HDMI: Use the DSS audio interface
ASoC: OMAP: HDMI: Create a structure for private data of the CPU DAI
ASoC: OMAP: HDMI: Change error values in HDMI CPU DAI
ASoC: OMAP: HDMI: Update the platform device names
ASoC: omap-abe-twl6040: Introduce driver data for runtime parameters
ASoC: omap-abe-twl6040: Move Digital Mic widget into dapm table
ASoC: omap-abe-twl6040: Keep only one snd_soc_dai_link structure
ASoC: omap-dmic: Add device tree bindings
ASoC: omap-mcpdm: Add device tree bindings
ASoC: omap-mcbsp: buffer size constraint only applies to playback stream
ASoC: omap-mcbsp: Use the common interrupt line if supported by the SoC
ASoC: omap-mcbsp: Remove unused FRAME dma_op_mode
...
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Merge tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull VGA-switcheroo audio client support for HD-audio from Takashi Iwai.
This depended on the recent drm pull.
* tag 'hda-switcheroo' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - unlock on error in azx_interrupt()
ALSA: hda - Support VGA-switcheroo
ALSA: hda - Export snd_hda_lock_devices()
ALSA: hda - Check the dead HDMI audio controller by vga-switcheroo
This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten
for the better support of "implicit feedback". If anything about USB
got broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up immediately
at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital links
between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid the accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal routing
through components with tight sequencing and formatting constraints
within their internal paths or where there are multiple components
connected with CPU managed DMA controllers inside the SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like digital
basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124, Texas
Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be
sent slightly later as partly depending on the changes of DRM.
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Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This is the first big chunk for 3.5 merges of sound stuff.
There are a few big changes in different areas. First off, the
streaming logic of USB-audio endpoints has been largely rewritten for
the better support of "implicit feedback". If anything about USB got
broken, this change has to be checked.
For HD-audio, the resume procedure was changed; instead of delaying
the resume of the hardware until the first use, now waking up
immediately at resume. This is for buggy BIOS.
For ASoC, dynamic PCM support and the improved support for digital
links between off-SoC devices are major framework changes.
Some highlights are below:
* HD-audio
- Avoid accesses of invalid pin-control bits that may stall the codec
- V-ref setup cleanups
- Fix the races in power-saving code
- Fix the races in codec cache hashes and connection lists
- Split some common codes for BIOS auto-parser to hda_auto_parser.c
- Changed the PM resume code to wake up immediately for buggy BIOS
- Creative SoundCore3D support
- Add Conexant CX20751/2/3/4 codec support
* ASoC
- Dynamic PCM support, allowing support for SoCs with internal
routing through components with tight sequencing and formatting
constraints within their internal paths or where there are multiple
components connected with CPU managed DMA controllers inside the
SoC.
- Greatly improved support for direct digital links between off-SoC
devices, providing a much simpler way of connecting things like
digital basebands to CODECs.
- Much more fine grained and robust locking, cleaning up some of the
confusion that crept in with multi-component.
- CPU support for nVidia Tegra 30 I2S and audio hub controllers and
ST-Ericsson MSP I2S controolers
- New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
Texas Instruments LM49453.
- Some regmap changes needed by the Tegra I2S driver.
- mc13783 audio support.
* Misc
- Rewrite with module_pci_driver()
- Xonar DGX support for snd-oxygen
- Improvement of packet handling in snd-firewire driver
- New USB-endpoint streaming logic
- Enhanced M-audio FTU quirks and relevant cleanups
- Increment the support of OSS devices to 256
- snd-aloop accuracy improvement
There are a few more pending changes for 3.5, but they will be sent
slightly later as partly depending on the changes of DRM."
Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree). Also some
fairly trivial context conflicts in the imx and mcx soc drivers.
* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
ALSA: snd-usb: fix stream info output in /proc
ALSA: pcm - Add proper state checks to snd_pcm_drain()
ALSA: sh: Fix up namespace collision in sh_dac_audio.
ALSA: hda/realtek - Fix unused variable compile warning
ASoC: sh: fsi: enable chip specific data transfer mode
ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
ASoC: sh: fsi: use same format for IN/OUT
ASoC: sh: fsi: add fsi_version() and removed meaningless version check
ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
ASoC: tegra: Add machine driver for WM8753 codec
ALSA: hda - Fix possible races of accesses to connection list array
ASoC: OMAP: HDMI: Introduce codec
ARM: mx31_3ds: Add sound support
ASoC: imx-mc13783 cleanup
mx31moboard: Add sound support
ASoC: mc13783 codec cleanups
ASoC: add imx-mc13783 sound support
ASoC: Add mc13783 codec
mfd: mc13xxx: add codec platform data
ASoC: don't flip master of DT-instantiated DAI links
...
Pull trivial updates from Jiri Kosina:
"As usual, it's mostly typo fixes, redundant code elimination and some
documentation updates."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (57 commits)
edac, mips: don't change code that has been removed in edac/mips tree
xtensa: Change mail addresses of Hannes Weiner and Oskar Schirmer
lib: Change mail address of Oskar Schirmer
net: Change mail address of Oskar Schirmer
arm/m68k: Change mail address of Sebastian Hess
i2c: Change mail address of Oskar Schirmer
net: Fix tcp_build_and_update_options comment in struct tcp_sock
atomic64_32.h: fix parameter naming mismatch
Kconfig: replace "--- help ---" with "---help---"
c2port: fix bogus Kconfig "default no"
edac: Fix spelling errors.
qla1280: Remove redundant NULL check before release_firmware() call
remoteproc: remove redundant NULL check before release_firmware()
qla2xxx: Remove redundant NULL check before release_firmware() call.
aic94xx: Get rid of redundant NULL check before release_firmware() call
tehuti: delete redundant NULL check before release_firmware()
qlogic: get rid of a redundant test for NULL before call to release_firmware()
bna: remove redundant NULL test before release_firmware()
tg3: remove redundant NULL test before release_firmware() call
typhoon: get rid of redundant conditional before all to release_firmware()
...
Like the previous fixes for cache hash accesses, a protection over
accesses to the widget connection list array must be provided.
Together with this action, remove snd_hda_get_conn_list() which can be
always race, and replace it with either snd_hda_get_num_conns() or
snd_hda_get_connections() calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an spin_unlock() missing on this error path.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch just sets the codec probe_mask=0x101 value for the WinFast VP200 H
PCoIP card based on Teradici hardware matching the PCI subsystem vendor/device
IDs 3a21:040d. The user reported no codec detection issues without this
explicit codec configuration.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BIOS on some HP laptops don't set the speaker-pins as fixed but expose
as jacks, and this confuses the driver as if these pins are
jack-detectable. As a result, the machine doesn't get sounds from
speakers because the driver prepares the power-map update via jack
unsol events which never come up in reality. The bug was introduced
in some time in 3.2 for enabling the power-mapping feature.
This patch fixes the problem by replacing the check of the persistent
power-map bits with a proper is_jack_detectable() call.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43240
Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support for VGA-switcheroo in the HD-audio controller side.
When the graphics controller is disabled, the HD-audio driver also delays
the initialization until it's activated by VGA-switcheroo.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43155
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's a preliminary work for the vga-switcher support.
Export the function to do pseudo-lock for the sound card to be used
in other places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a discrete-GPU is disabled by the VGA switcheroo, the
corresponding HD-audio controller for HDMI output is also disabled.
Such a dead controller still appears in the PCI device list, but you
can't access properly any longer (even calling pci_read_config_*()
triggers Oops!) which leads the stall of the whole communication of
the driver.
This patch adds a check of graphics controller at the probe time to
see whether it's disabled by vga-switcheroo. If disabled, skip the
whole initialization of this controller.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43155
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that the FLOAT format on CS4206 results in simple
noises, which implies that this is no right format as is.
Since CS4206 is the only codec supporting the float, let's disable it
until we find the correct format.
Reported-and-tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp and caps hashes aren't protected properly for concurrent
accesses. Protect them via a new mutex now.
But it can't be so simple as originally thought: since the update of a
hash table entry itself might trigger the power-up sequence which
again accesses the hash table, we can't cover the whole function
simply via mutex. Thus the update part has to be split from the mutex
and revalidated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The controller is compatible with HD-audio 1.0a with some specific
restrictions.
- The BDLE entries can't be over 4k boundary
- No position-buffer and no MSI
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the power_transition up/down state instead of boolean bit, so
that the power-up sequence can cancel the pending power-down work
properly. Also, by moving cancel_delayed_work_sync() before the
actual power-up sequence, make sure that the delayed power-down is
completed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the recent commit, the resume procedure is always performed at
the resume time. This makes the pre_resume hack for VREF mute LED on
some HP laptops superfluous. As this is the only user of pre_resume
(and there is no user of post_suspend) ops, let's kill them again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an IRQ for some reason gets lost, we wait up to a second using
udelay, which is CPU intensive. This patch improves the situation by
waiting about 30 ms in the CPU intensive mode, then stepping down to
using msleep(2) instead. In essence, we trade some granularity in
exchange for less CPU consumption when the waiting time is a bit longer.
As a result, PulseAudio should no longer be killed by the kernel
for taking up to much RT-prio CPU time. At least not for *this* reason.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Tested-by: Arun Raghavan <arun.raghavan@collabora.co.uk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the fixup helper functions in patch_realtek.c to hda_auto_parser.c
so that they can be used in other codec drivers like patch_conexant.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some power-saving states have been left unchanged in
snd_hda_codec_reset(), and this is a potential danger because the
function may be called in various situations including the continuous
operation after that call.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a fix for the problem in commit 785f857d1c, the pop noise
issue on some machines with ALC269. The problem was the uninitialized
state after the resume due to the delayed resume of the codec chips.
In that commit, we tried to fix by forcibly putting the codec to D3 at
suspend. But, this still also leaves the uninitialized state after
resume, and it _might_ be still problematic with some BIOS. Since the
commit turned out to regress another issues, we reverted it in the
end.
Now, in this fix, try to fix by turning on the codec immediately at
the resume path. We need to take care of the power-saving in this
case. When the device is woken up at the power-saved state, it should
go power-saving again after the resume.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 785f857d1c.
The commit causes a problem with the wrong D3 state after suspend
because the call of hda_set_power_state() involves with the power-up
sequence, which changes the power_count, and this confuses the resume
sequence that checks the power_count as well.
Originally, this go-to-D3 sequence should be a simple task without the
power-up sequence. But, it'd need some proper sanity checks in the
case of power-saved state, so it's not too easy to write now in the
3.4-rc cycle.
In short, the safest option now is to revert this affecting commit.
Of course, we need to clean up and robustify the power-saving code
better for 3.5 kernel.
Reported-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The call for alc_auto_parse_customize_define() must be done after the
fixup pre-probe initialization. Otherwise SKU_IGNORE fixup won't work
properly (e.g. HP RP5800 with ALC662 codec).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer Aspire 5739G requires the same fix-up for 4930G to support the
surround / bass speakers.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=43180
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some old codecs like ALC880 seem to give a bogus pin capability value 0
occasionally. This breaks the new sanity check in snd_hda_set_pin_ctl().
Skip the sanity checks in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper function to guess the default VREF pin control bits
for mic in. This can be used to set the pin control value safely
matching with the actual pin capabilities.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For setting the pin-control values more safely to match with the
actual pin capability bits, a copule of new helper functions,
snd_hda_set_pin_ctl() and snd_hda_set_pin_ctl_cache(), are
introduced. These are simple replacement of the codec verb write with
AC_VERB_SET_PIN_WIDGET but do more sanity checks and filter out
superfluous pin-control bits if they don't fit with the corresponding
pin capabilities.
Some codecs are screwed up or ignore the command when such a wrong bit
is set. These helpers will avoid such secret errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the reporter, external mic starts to work if the
laptop-dmic model is used. According to BIOS pin config, all
pins are consistent with the alc269vb_laptop_dmic fixup, except
for the external mic, which is not present.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/950490
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge fixes for Thinkpad docking-station regressions for 3.3 kernels
back to 3.4. These were committed in that branch to make the stable
merging easier.
Conflicts:
sound/pci/hda/patch_conexant.c
ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the
docking-station ports, but BIOS doesn't initialize for these pins.
Thus, like the former X200, we need to set up the pins manually in the
driver.
The odd part is that the same PCI SSID is used for X200 and T400, thus
we need to prepare individual fixup tables for cx5051 and others.
Bugzilla entries:
https://bugzilla.redhat.com/show_bug.cgi?id=808559https://bugzilla.redhat.com/show_bug.cgi?id=806217https://bugzilla.redhat.com/show_bug.cgi?id=810697
Reported-by: Josh Boyer <jwboyer@redhat.com>
Reported-by: Jens Taprogge <jens.taprogge@taprogge.org>
Tested-by: Jens Taprogge <jens.taprogge@taprogge.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some output pins on Conexant chips have no HP control bit, but the
auto-parser initializes these pins unconditionally with PIN_HP.
Check the pin-capability and avoid the HP bit if not supported.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While refactoring the mute-LED handling for HP laptops, I messed up
the polarity check in a wrong way. The red (or the mute-LED if any)
should appear in the muted state, corresponding to GPIO on.
Reported-by: Mikko Vinni <mmvinni@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Through the transition to the auto-parser, the support for
Quanta/Gericom KN1 got broken. There are two problems behind it:
- This machine doesn't like the default COEF setup for ALC260 we take
now as default
- BIOS doesn't set the pins correctly at all; especially the machine
uses only the pin 0x0f for both headphone and speaker
This patch adds the fixup as a workaround for these issues.
Reported-and-tested-by: Uros Vampl <mobile.leecher@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the
'for (;;)' loop, if the 'badness' value returned from
fill_and_eval_dacs() is negative, then we'll return from the function
without freeing the memory we allocated for 'best_cfg', thus leaking.
Fix the leak by kfree()'ing the memory when badness is negative.
While I was there I also noticed some trailing whitespace in the
function that I removed (along with all other trailing whitespace in
the file) - it didn't seem worth-while to do that as two patches, so I
hope it's OK that I just did it all as one patch.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A user reported that setting model=imac24 used to allow sound to work on their
Mac Pro 5,1 machine. Commit 5671087ffa "Move ALC885 macpro and imac24 models
to auto-parser" removed this model option. All Mac machines are now explicitly
handled with a quirk and the auto-parser. This adds a quirk for the device
found on the Mac Pro 5,1 machines.
This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559
[sorted the new entry in the ID number order by tiwai]
Reported-by: Gabriel Somlo <somlo@cmu.edu>
Signed-off-by: Josh Boyer <jwboyer@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's compatible with 8930G.
Using the same fixup gives the proper 5.1 sound back.
Reported-and-tested-by: Dany Martineau <dany.luc.martineau@gmail.com>
Cc: <stable@kernel.org> [v3.3+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add GPIO1 setup explicitly for Acer Aspire 493x & co.
This could be set by alc_auto_init_amp(), but it's safer to set it
more explicitly in the fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since there are still many Acer models that might not be covered by
the current fixup table, let's add back a few typical model names so
that user can test the fixup without recompiling.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make sure we don't dereference the "quirk" pointer when it is null.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also remove two warnings when CONFIG_SND_DEBUG is not set:
sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’:
sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable]
sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable]
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge with latest Linus' tree, as I have incoming patches
that fix code that is newer than current HEAD of for-next.
Conflicts:
drivers/net/ethernet/realtek/r8169.c
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
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Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: fixes for 3.4
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
name pins consistently (MIC1/LINE1/HP-OUT/CD) on all controls
affecting those pins.
remove duplicate SET_AMP_GAIN_MUTE to 0x17/index 0 and 0x17/index 1
really select MIC1, not Mixer out for recording
"Mixer out" for recording is not a "pin", adjust comment
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CX20549 (ctx5045) doesn't accept data on index 1 for output pins,
as shown in the following hda-var transaction:
$ hda-verb /dev/snd/hwC0D0 0x10 set_amp_gain 0xb126
nid = 0x10, verb = 0x300, param = 0xb126
value = 0x0
$ hda-verb /dev/snd/hwC0D0 0x10 get_amp_gain 0x8001
nid = 0x10, verb = 0xb00, param = 0x8001
value = 0x0
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ID used for detection of the BenQ R55E actually identifies the
Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop
series. Schematics on the internet clearly indicate that the "Port C"
(analog input connected to record source #4 and mixer input #4) is
unconnected.
Playing an audio CD through analog playback (using cdplay from cdtools)
produces no sound, even with the mixer input labelled "CD" enabled, and
the volume control in the CD drive set to maximum. This indicates the
connection is really not present.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "input converter" widget of the CX20459 has only one input amplifier,
expose that one as "Capture Volume/Capture Switch". The actual record
source selection is already exposed through the separately installed
input mux.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This includes renaming "Line In" to line, also in the mixer settings.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CX20549 has only one single input amp on it's input converter
widget. Fix printing of values in the codec file in /proc/asound.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The internal mic input is phase inverted on one channel.
To avoid people in userspace summing the channels together
and get zero result, use a separate mixer control for the
inverted channel.
BugLink: https://bugs.launchpad.net/bugs/903853
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes for 3.4-rc1, including
- mic-recording regression fix for Realtek codec
- clean-up of dmaengine parameter mess
- WM8894 calibration tweak
- minor fixes for asihpi and some bool module parms
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes for 3.4-rc1, including
- mic-recording regression fix for Realtek codec
- clean-up of dmaengine parameter mess
- WM8894 calibration tweak
- minor fixes for asihpi and some bool module parms"
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: fix isa/opti9xx module param type
sound: fix oss/msnd_pinnacle module param type
ALSA: asihpi - fix return type of hpios_locked_mem_alloc()
ASoC: dmaengine_pcm: use dmaengine cyclic wrapper
ASoC: Add extra parameter to device_prep_dma_cyclic
ALSA: hda/realtek - Fix ADC assignment with a shared HP/Mic pin
ASoC: wm8994: Update WM8994 DCS calibration
This patch updates Jonathan Woithe's contact details across the kernel tree.
Signed-off-by: Jonathan Woithe <jwoithe@just42.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Here is the first big update chunk of sound stuff for 3.4-rc1.
In the common sound infrastructure, there are a few changes for
dynamic PCM support (used in ASoC) and a few clean-ups. Majority of
changes are found, as usual, in HD-audio and ASoC.
Some highlights of HD-audio changes:
- All the long-standing static quirk codes for Realtek codec were
finally removed by fixing and extending the Realtek auto-parser.
- The mute-LED control is standardized over all HD-audio codec
drivers using the extended vmaster hook.
- The vmaster slave mixer elements are initialized to 0dB as default
so that the user won't be annoyed by the silent output after
updates, e.g. due to the additions of new elements.
- Other many fix-ups for the misc HD-audio devices.
In the ASoC side, this is a very active release, including a quite a
few framework enhancements. Some highlights:
- Support for widgets not associated with a CODEC, an important part
of the dynamic PCM framework.
- A library factoring out the common code shared by dmaengine based
DMA drivers contributed by Lars-Peter Clausen. This will save a lot
of code and make it much easier to deploy enhancements to
dmaengine.
- Support for binary controls, used for providing runtime
configuration of algorithm coefficients.
- A new DAPM widget type for regulator supplies allowing drivers for
devices that can power down unused supplies while active to do
without any per-driver code.
- DAPM widgets for DAIs, initially giving a speed boost for playback
startup and shutdown and also the basis for CODEC<->CODEC DAI link
support.
- Support for specifying the number of significant bits on audio
interfaces, useful for allowing applications to know how much effort
to put into generating data for a larger sample format.
- Conversion of the FSI driver used on some SH processors to
DMAEngine.
- Conversion of EP93xx drivers to DMAEngine.
- New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
WM2200.
- Move audmux driver from arc/arm to sound/soc
- McBSP move from arch/ to sound/ and updates
Also, a few small updates and fixes for other drivers like au88x0,
ymfpci, USB 6fire, USB usx2yaudio are included.
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Merge tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull updates of sound stuff from Takashi Iwai:
"Here is the first big update chunk of sound stuff for 3.4-rc1.
In the common sound infrastructure, there are a few changes for
dynamic PCM support (used in ASoC) and a few clean-ups. Majority of
changes are found, as usual, in HD-audio and ASoC.
Some highlights of HD-audio changes:
- All the long-standing static quirk codes for Realtek codec were
finally removed by fixing and extending the Realtek auto-parser.
- The mute-LED control is standardized over all HD-audio codec
drivers using the extended vmaster hook.
- The vmaster slave mixer elements are initialized to 0dB as default
so that the user won't be annoyed by the silent output after
updates, e.g. due to the additions of new elements.
- Other many fix-ups for the misc HD-audio devices.
In the ASoC side, this is a very active release, including a quite a
few framework enhancements. Some highlights:
- Support for widgets not associated with a CODEC, an important part
of the dynamic PCM framework.
- A library factoring out the common code shared by dmaengine based
DMA drivers contributed by Lars-Peter Clausen. This will save a
lot of code and make it much easier to deploy enhancements to
dmaengine.
- Support for binary controls, used for providing runtime
configuration of algorithm coefficients.
- A new DAPM widget type for regulator supplies allowing drivers for
devices that can power down unused supplies while active to do
without any per-driver code.
- DAPM widgets for DAIs, initially giving a speed boost for playback
startup and shutdown and also the basis for CODEC<->CODEC DAI link
support.
- Support for specifying the number of significant bits on audio
interfaces, useful for allowing applications to know how much
effort to put into generating data for a larger sample format.
- Conversion of the FSI driver used on some SH processors to
DMAEngine.
- Conversion of EP93xx drivers to DMAEngine.
- New CODEC drivers for Maxim MAX9768 and Wolfson Microelectronics
WM2200.
- Move audmux driver from arc/arm to sound/soc
- McBSP move from arch/ to sound/ and updates
Also, a few small updates and fixes for other drivers like au88x0,
ymfpci, USB 6fire, USB usx2yaudio are included."
* tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (446 commits)
ASoC: wm8994: Provide VMID mode control and fix default sequence
ASoC: wm8994: Add missing break in resume
ASoC: wm_hubs: Don't actively manage LINEOUT_VMID_BUF
ASoC: pxa-ssp: atomically set stream active masks
ASoC: fsl: p1022ds: tell the WM8776 codec driver that it's the master
ASoC: Samsung: Added to support mono recording
ALSA: hda - Fix build with CONFIG_PM=n
ALSA: au88x0 - Avoid possible Oops at unbinding
ALSA: usb-audio - Fix build error by consitification of rate list
ASoC: core: Fix obscure leak of runtime array
ALSA: pcm - Avoid GFP_ATOMIC in snd_pcm_link()
ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list
ASoC: wm8996: Add 44.1kHz support
ALSA: hda - Fix build of patch_sigmatel.c without CONFIG_SND_HDA_POWER_SAVE
ASoC: mx27vis-aic32x4: Convert it to platform driver
ALSA: hda - fix printing of high HDMI sample rates
ALSA: ymfpci - Fix legacy registers on S3/S4 resume
ALSA: control - Fixe a trailing white space error
ALSA: hda - Add expose_enum_ctl flag to snd_hda_add_vmaster_hook()
ALSA: hda - Add "Mute-LED Mode" enum control
...
The recent Realtek driver tries to assign an extra input via the
headphone plug when only a single input source is found. The code
worked on Samsung Q1, but it broke ASUS 1015 where the mic is a
digital-mic and only a specific ADC works.
This patch fixes the assignment of ADC in the shared mic/hp case.
Instead of assuming the single ADC at first, reduce the ADCs after
trying to assign both mic and HP pins.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42973
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull trivial tree from Jiri Kosina:
"It's indeed trivial -- mostly documentation updates and a bunch of
typo fixes from Masanari.
There are also several linux/version.h include removals from Jesper."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (101 commits)
kcore: fix spelling in read_kcore() comment
constify struct pci_dev * in obvious cases
Revert "char: Fix typo in viotape.c"
init: fix wording error in mm_init comment
usb: gadget: Kconfig: fix typo for 'different'
Revert "power, max8998: Include linux/module.h just once in drivers/power/max8998_charger.c"
writeback: fix fn name in writeback_inodes_sb_nr_if_idle() comment header
writeback: fix typo in the writeback_control comment
Documentation: Fix multiple typo in Documentation
tpm_tis: fix tis_lock with respect to RCU
Revert "media: Fix typo in mixer_drv.c and hdmi_drv.c"
Doc: Update numastat.txt
qla4xxx: Add missing spaces to error messages
compiler.h: Fix typo
security: struct security_operations kerneldoc fix
Documentation: broken URL in libata.tmpl
Documentation: broken URL in filesystems.tmpl
mtd: simplify return logic in do_map_probe()
mm: fix comment typo of truncate_inode_pages_range
power: bq27x00: Fix typos in comment
...
A previous commit af65cbf296 (ALSA: hdmi: fix printout of SAD sampling
rates) fixed the sample rates shown in /proc/asound/cardX/eldY and
kernel log to not be entirely wrong. However, a missing rate from the
array added in the patch causes HDMI rates 88.2 kHz, 96 kHz, 176.4 kHz,
and 192 kHz to be shown as 96 kHz, 176.4 kHz, 192 kHz, and 384 kHz,
respectively.
Fix the reporting by adding the ALSA rate 64 kHz into the conversion
array between 48 kHz and 88.2 kHz.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai