Treat both negative and zero return values from clk_round_rate()
as errors.  This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.

Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error; all other values will be
considered valid rates.  The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.
Signed-off-by: NPaul Walmsley <pwalmsley@nvidia.com>
Acked-by: NHans-Christian Egtvedt <egtvedt@samfundet.no>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Treat both negative and zero return values from clk_round_rate()
as errors.  This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.

Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error.  All other values will be
considered valid rates.  The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.
Signed-off-by: NPaul Walmsley <pwalmsley@nvidia.com>
Acked-by: NHans-Christian Egtvedt <egtvedt@samfundet.no>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Since there are more HD-audio compatible codecs, move the definitions
of HD-audio verbs into common header location, include/sound, so that
it can be included cleanly from other drivers than HD-audio driver.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

AD and VIA codecs had stereo mixer input enabled as default before
moving to the generic parser, and people think the lack of such a
regression.  In this patch, the stereo mixer input is added back to
the input selection if no auto-mic is available, and if it's not
disabled explicitly via hint.  This should satisfy most of demands,
i.e. stereo mix on desktop machines like what it worked before, and it
still keeps the new auto-mic feature on laptops.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Sometimes the hardware reports LPIB being advanced than POSBUF.
When this happens, the driver adjusts to a positive value by adding
the buffer size.  Then the driver detects it as an error (greater than
the period size), and stops the LPIB delay account from this point
on.

When I took a close look at these conditions, the values shown are all
very small numbers, and it'd be better to just ignore these values
instead of discontinuing the LPIB delay correction.

In this patch, the driver checks a negative delay value and ignores if
it's a significantly small error.  Currently the threshold is set to
64 frames, but could be smaller.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The loopback mixing paths aren't initialized correctly at init
callback.  Mostly this is harmless as codecs usually set the mute
state as default, but we still should make sure.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

We have blindly assumed that all valid configurations should have
either analog or digital playback, but there can be capture-only
configurations.  The parser shouldn't escape in such a case.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch skips the default depop delay before D3 for Haswell (10 ms) and
Valleyview2 (100 ms) display codec, to reduce codec suspend time.

The analog part of display audio is implemented in the external display. Some
displays have weak pop noise while others not when suspending, no matter there
is the default delay or not.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This simplifies lots of codes indeed.
Tested-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Even if the CONFIG_PM explicity is undefined, let's convert to the
modern PM ops.
Signed-off-by: NUlf Hansson <ulf.hansson@linaro.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch sets a 0ms depop delay in fixup funtion 'alc_fixup_no_depop_delay'.
And Realteck ALC262 applies this on Intel Baytrail BayleyBay platform to reduce
codec suspend time.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Reviewed-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

HD-audio devices tend to take long time for finishing the whole
probing procedure.  In this patch, the time-consuming part of the
probing procedure, the codec probe and the rest initializations, are
moved in the work, so that they can be done asynchronously in parallel
with probes of other devices.

Since we already have this mechanism in the driver code for the
firmware and i915 request_symbol() stuff, we just need to enable it
always; the resultant patch even reduces more lines, which is an
additional bonus.

Credit goes to David Henningsson, who suggested this workaround.
Reported-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When the probe of snd-hda-intel driver is deferred due to f/w loading
or the nested module loading, complete_all() should be also delayed
until the initialization really finished.  Otherwise, vga-switcheroo
client would start switching before the actual init is done.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

It seems that EAPD on NID 0x16 is the only control over all outputs on
HP machines with AD1984A while turning EAPD on NID 0x12 breaks the
output.  Thus we need to avoid fiddling EAPD on NID.  As a quick
workaround, just set own_eapd_ctrl flag for the wrong EAPD, then
implement finer EAPD controls.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66321
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The recent kernels got regressions on ASUS W7J with ALC660 codec where
no sound comes out.  After a long debugging session, we found out that
setting the pin control on the unused NID 0x10 is mandatory for the
outputs.  And, it was found out that another magic of NID 0x0f that is
required for other ASUS laptops isn't needed on this machine.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66081Reported-and-tested-by: NAndrey Lipaev <lipaev@mail.ru>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds 'depop_delay' to struct hda_codec, to indicate a depop delay
time in ms when power-down, in function set_power_state() to D3.

Default value is -1, for a default delay time.
Machine fixup can set a suitable value according to the codec chip and HW audio
design.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch defines a fixup 'alc_fixup_no_depop_delay', which sets alc_spec flag
'no_depop_delay' to indicate skipping delay in Realtek codec suspend/resume.

And Intel Baytrail BayleyBay board applies this fixup to reduce driver
suspend/resume time.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch defines a flag "no_depop_delay" in alc_spec. If this flag is set,
delay in alc_eapd_shutup and alc_resume will be skipped.

Machine-specific fixup can set this flag to reduce suspend/resume time, if
the codec and hardware analog design can avoid pop noise without this delay.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Dell assigned alias name for more codecs.
ALC3220 ALC3221 ALC3223 ALC3226 ALC3234 ALC3661.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The array limits are supposed to be in units of u32 instead of in bytes.
The current code has a potential array overflow.

Fixes: c614475b ('ALSA: dice: add a proc file to show device information')
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This machine also has mono output if run through DAC node 0x03.

Cc: stable@vger.kernel.org (v3.10+)
BugLink: https://bugs.launchpad.net/bugs/1256212Tested-by: NDavid Chen <david.chen@canonical.com>
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

As the previous commit 1f0bbf03 added the pin config for the bass
speaker, this patch adds the corresponding LFE-only channel map on
ASUS ET2700.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a fixup entry for the missing bass speaker pin 0x16 on ASUS ET2700
AiO desktop.  The channel map will be added in the next patch, so that
this can be backported easily to stable kernels.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This both devices need limit for internal dmic.

[cosmetic change; renamed fixup name by tiwai]
Signed-off-by: NOleksij Rempel <linux@rempel-privat.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The current generic parser assumes blindly that the volume and mute
amps are found in the aamix node itself.  But on some codecs,
typically Analog Devices ones, the aamix amps are separately
implemented in each leaf node of the aamix node, and the current
driver can't establish the correct amp controls.  This is a regression
compared with the previous static quirks.

This patch extends the search for the amps to the leaf nodes for
allowing the aamix controls again on such codecs.
In this implementation, I didn't code to loop through the whole paths,
since usually one depth should suffice, and we can't search too
deeply, as it may result in the conflicting control assignments.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65641
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When the hp mic pin has no VREF bits, the driver forgot to set PIN_IN
bit.  Spotted during debugging old MacBook Airs.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When a headphone jack is configurable as input, the generic parser
tries to make it retaskable as Headphone Mic.  The switching can be
done smoothly if Capture Source control exists (i.e. there is another
input source).  Or when user explicitly enables the creation of jack
mode controls, "Headhpone Mic Jack Mode" will be created accordingly.

However, if the headphone mic is the only input source, we have to
create "Headphone Mic Jack Mode" control because there is no capture
source selection.  Otherwise, the generic parser assumes that the
input is constantly enabled, thus the headphone is permanently set
as input.  This situation happens on the old MacBook Airs where no
input is supported properly, for example.

This patch fixes the problem: now "Headphone Mic Jack Mode" is created
when such an input selection isn't possible.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65681
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

To reduce driver resume time, this patch resumes the codecs in parallel
if there are multiple codecs on the bus.

- The PM workqueue of bus is also used to parallel resuming multiple codecs.
- The work item 'pm_work' is renamed to 'suspend_work' to parallel suspending
  codecs.
- Add a work item 'resume_work' to parallel resuming codecs.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The time to suspend a single codec may be several hundreds of ms. When runtime
power saving is disabled, driver suspend time can be long especially when there
are more than one codec on the bus.

To reduce driver suspend time, this patch creates a work queue for the bus, and
suspends the codecs in parallel if there are multiple codecs on the bus.

[fixed cosmetic issues by tiwai]
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

For Wireless USB audio devices, use multiple isoc packets per URB for
inbound endpoints with a datainterval < 5.  This allows the WUSB host
controller to take advantage of bursting to service endpoints whose
logical polling interval is less than the 4ms minimum polling interval
limit in WUSB.
Signed-off-by: NThomas Pugliese <thomas.pugliese@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Most Thinkpad Edge series laptops use conexant codec, so far although
the codecs have different minor Vendor Id and minor Subsystem Id,
they all belong to the cxt5066 family, this change can make the
mute/mic-mute LEDs support more generic among cxt_5066 family.

This design refers to the similar solution for the realtek codec
ALC269 family in the patch_realtek.c.

Cc: Alex Hung <alex.hung@canonical.com>
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Acked-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

According to Mengdong, we shouldn't enable runtime PM when a codec
doesn't support EPSS, based on the experiences on Windows.
We have already this check in HDMI codec drivers, but now apply it in
general in hda_codec.c.

Credit goes to Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Now we fixed the long-standing bugs of runtime PM, let's enable
Panther Point again.  The runtime PM was disabled in the HDMI codec
driver due to the S3 issue, and this should have been fixed now.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

No functional change.

- codec->pm_down_notified flag is renamed to codec->pm_up_notified
  flag (and takes the reversed meaning), which is managed solely in
  hda_call_pm_notify() now.

- The explicit call of hda_call_pm_notify() is moved into
  hda_keep_power_on().

- Removed a redundant call of hda_call_pm_notify() in
  snd_hda_power_up(), as it's called in hda_call_codec_resume()
  anyway.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If a codec contains only the digital outputs, it's very likely a
HDMI/DP codec, which isn't supported by the generic parser but via
HDMI codec parser code.  Detect such a case and bind with the proper
parser object if available.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>