The helper functions snd_hda_check_board_config() and
snd_hda_check_board_codec_sid_config() are no longer used since the
transition to the generic parser and all quirks have been replaced
with fixups.  Let's kill these dead codes.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

On some HP laptops, the mute led is controlled by codec gpio.

When some machine resume from s3/s4, the codec gpio data will be
cleared to 0 by BIOS:
Before suspend:
  IO[3]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0
After resume:
  IO[3]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0

To skip the AFG node to enter D3 can't fix this problem.

A workaround is to restore the gpio data when the system resume
back from s3/s4. It is safe even on the machines without this
problem.

BugLink: https://bugs.launchpad.net/bugs/1358116Tested-by: NFranz Hsieh <franz.hsieh@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ALC269 & co have many vendor-specific setups with COEF verbs.
However, some verbs seem specific to some codec versions and they
result in the codec stalling.  Typically, such a case can be avoided
by checking the return value from reading a COEF.  If the return value
is -1, it implies that the COEF is invalid, thus it shouldn't be
written.

This patch adds the invalid COEF checks in appropriate places
accessing ALC269 and its variants.  The patch actually fixes the
resume problem on Acer AO725 laptop.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181Tested-by: NFrancesco Muzio <muziofg@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Conexnat HD-audio driver has a workaround for cx5051 (aka CX20561)
chip to add fake mute controls to each amp (commit 3868137e).  This
implies the minimum-as-mute TLV bit in TLV for each corresponding
control.  Meanwhile we build the virtual master from these, but the
TLV bit is missing, even though the slaves have it.

This patch simply adds the missing TLV_DB_SCALE_MUTE bit for vmaster,
as already done in patch_sigmatel.c.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This mode is unsupported, as the DMA controller can't do zero-padding
of samples.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-by: NJohannes Stezenbach <js@sig21.net>
Signed-off-by: NMark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org

This reverts commit a603c8ee.

fsl_asoc_xlate_tdm_slot_mask() is different with snd_soc_xlate_tdm_slot_mask().
fsl_asoc_xlate_tdm_slot_mask() will set the enabled bit to 0, disabled bit
to 1. snd_soc_xlate_tdm_slot_mask() will set the enabled bit to 1, disabled
bit to 0.
For esai when the bit value is 1, the slot is enabled, when the bit value is 0,
the slot is disabled. If using fsl_asoc_xlate_tdm_slot_mask(), the esai will
work abnormally. So revert this patch, make the esai use default function.
Signed-off-by: NShengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: NNicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

The implicit BLCK divider setting was broken by "ASoC: mcasp: don't
override bclk divider if it was provided by the machine"-patch. After
the BCLK divider is implicitly set for the first time the
mcasp->bclk_div gets a non zero value and the implicit setting is
"turned off".
Signed-off-by: NJyri Sarha <jsarha@ti.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

TDM slot length was set same as word length, regardless of the value
received in set_tdm_slot. This patch sets the TDM slot length correctly
as received in set_tdm_slot DAI callback
Signed-off-by: NNikesh Oswal <Nikesh.Oswal@wolfsonmicro.com>
Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

We should prefer `struct pci_device_id` over `DEFINE_PCI_DEVICE_TABLE` to
meet kernel coding style guidelines.  This issue was reported by checkpatch.

A simplified version of the semantic patch that makes this change is as
follows (http://coccinelle.lip6.fr/):

// <smpl>

@@
identifier i;
declarer name DEFINE_PCI_DEVICE_TABLE;
initializer z;
@@

- DEFINE_PCI_DEVICE_TABLE(i)
+ const struct pci_device_id i[]
= z;

// </smpl>

[bhelgaas: add semantic patch]
Signed-off-by: NBenoit Taine <benoit.taine@lip6.fr>
Signed-off-by: NBjorn Helgaas <bhelgaas@google.com>

The source type should come before the direction specifier according to
ControlNames.txt.
Signed-off-by: NMark Brown <broonie@linaro.org>

If soc_dapm_read() fails, reg_val will be uninitialized, and bogus
values will be written later:

sound/soc/soc-dapm.c: In function 'snd_soc_dapm_get_enum_double':
sound/soc/soc-dapm.c:2862:15: warning: 'reg_val' may be used uninitialized in this function [-Wmaybe-uninitialized]
  unsigned int reg_val, val;
               ^

Return early on error to fix this.

Introduced by commit ce0fc93a ("ASoC:
Add DAPM support at the component level").
Signed-off-by: NGeert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: NMark Brown <broonie@linaro.org>
Acked-by: NLars-Peter Clausen <lars@metafoo.de>

There is no need to restore and restart PCM streams in case ADSP didn't
reach reset and power off state during system suspend/resume cycle. In that
case stream is still active but paused and firmware doesn't allow allocating
a new stream before paused stream is freed.

ADSP remains active in case suspend sequence didn't go to suspend_late
stage. This can happen when either suspend sequence is aborted by a wakeup
or by letting only devices suspend by "echo devices >/sys/power/pm_test".

Currently stream restoring fails in these suspend cases. Fix this by adding
a flag that indicates is complete stream reinitialization needed or is it
enough to resume paused stream. Flag is set when we know that ADSP reached
suspend_late.

Initial fix to this issue came from Fang Yang. I modified it a little and
forward ported it to top of two other suspend/resume patches from me.
Signed-off-by: NJarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: NBorun Fu <borun.fu@intel.com>
Cc: yang fang <yang.a.fang@intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Remove sst_byt_pcm_dev_resume() and move waiting of firmware boot into
sst_byt_pcm_dev_resume_early(). Now suspend_late and resume_early phases are
in sync with each other so that we know that ADSP was put into reset and was
unpowered after suspend_late and is ready to resume IO after resume_early
during resume stage in sst_byt_pcm_trigger().
Signed-off-by: NJarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: NBorun Fu <borun.fu@intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Merge DSP reset and cleanup sequence in sst_byt_pcm_dev_suspend_noirq()
into sst_byt_pcm_dev_suspend_late(). First their order was wrong by first
unloading firmware modules in suspend_late and then taking DSP into reset
in suspend_noirq. Second ACPI has put device into OFF state already during
suspend_late so trying to reset the DSP is a no-op at suspend_noirq stage.

Fix these by moving DSP reset and cleanup into
sst_byt_pcm_dev_suspend_late() before firmware unloading.
Signed-off-by: NJarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: NBorun Fu <borun.fu@intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

CA0132 driver tries to reload the firmware at resume.  Usually this
works since the firmware loader core caches the firmware contents by
itself.  However, if the driver failed to load the firmwares
(e.g. missing files), reloading the firmware at resume goes through
the actual file loading code path, and triggers a kernel WARNING like:

 WARNING: CPU: 10 PID:11371 at drivers/base/firmware_class.c:1105 _request_firmware+0x9ab/0x9d0()

For avoiding this situation, this patch makes CA0132 skipping the f/w
loading at resume when it failed at probe time.
Reported-and-tested-by: NJanek Kozicki <cosurgi@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If nid 0x15 (Headphone Playback Switch) is in D3 and headphones are
plugged in when the laptop reboots, a pop noise is generated.
Prevent this by keeping nid 0x15 in D0 when headphones are plugged in.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611Signed-off-by: NGabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If the laptop is powered on with a jack plugged in, independently on what
is plugged, the jack is treated as a microphone jack.

Initialize the capture source so that by default jacks are treated as
headphones jacks. This will also prevent pop noises on boot in case
headphones are plugged in since setting/unsetting mic-in as input source
causes a pop noise.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611Signed-off-by: NGabriele Mazzotta <gabriele.mzt@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The BOSS ME-25 turns out not to have any useful descriptors in its MIDI
interface, so its needs a quirk entry after all.
Reported-and-tested-by: NKees van Veen <kees.vanveen@gmail.com>
Fixes: 8e5ced83 ("ALSA: usb-audio: remove superfluous Roland quirks")
Cc: <stable@vger.kernel.org>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

CMI8888 codec chip has a boost amp (only) on the headphone pin, and
this confuses the generic parser, which tends to pick up the most
outside amp.  This results in the wrong volume setup, as the driver
complains like:
  hda_codec: Mismatching dB step for vmaster slave (-100!=1000)

For avoiding this problem, rule out the amp on NID 0x10 and create
"Headphone Amp" volume control manually instead.

Note that this patch still doesn't fix all problems yet.  The sound
output from the line out seems still too low.  It will be fixed in
another patch (hopefully).
Reported-and-tested-by: NVincent Lejeune <vljn@ovi.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ASUS Phoebus with CMI8888 HD-audio chip (PCI id 13f6:5011) doesn't
work with HD-audio driver as is because of some weird nature.  For
making DMA properly working, we need to disable MSI.  The position
report buffer doesn't work, thus we need to force reading LPIB
instead.  And yet, the codec CORB/RIRB communication gives errors
unless we disable the snooping (caching).

In this patch, all these workarounds are added as a quirk for the
device.  The HD-audio *codec* chip needs yet another workaround, but
it'll be provided in the succeeding patch.
Reported-and-tested-by: NVincent Lejeune <vljn@ovi.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

It will be recording voice delay for resume back recording for Headset Mic.
This alc286 will quickly open Headset Mic, to prevent avoid recording files are missing.
The issue was fixed. This is follow ALC286 programing guide.

[fix build error, add static and renamed the function by tiwai]
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Update the initial Baytrail ADSP firmware file name with the one that is now
in linux-firmware.git. Please see linux-firmware.git commit 7551a3a78453
("fw_sst_0f28: Add firmware for Intel Baytrail SST DSP").
Signed-off-by: NJarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

The Linux kernel coding style guidelines suggest not using typedefs
for structure types. This patch gets rid of the typedefs for wanc_info and
wavnc_port_info.

A simplified version of the Coccinelle semantic patch that finds the case is:

@tn@
identifier i;
type td;
@@

-typedef
 struct i { ... }
-td
 ;

@@
type tn.td;
identifier tn.i;
@@

-td
+ struct i
Signed-off-by: NHimangi Saraogi <himangi774@gmail.com>
Acked-by: NJulia Lawall <julia.lawall@lip6.fr>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The Linux kernel coding style guidelines suggest not using typedefs
for structure types. This patch gets rid of the typedef for uart401_devc.

The following Coccinelle semantic patch detects the case.

@tn@
identifier i;
type td;
@@

-typedef
 struct i { ... }
-td
 ;

@@
type tn.td;
identifier tn.i;
@@

-td
+ struct i
Signed-off-by: NHimangi Saraogi <himangi774@gmail.com>
Acked-by: NJulia Lawall <julia.lawall@lip6.fr>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NAdam Goode <agoode@google.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

sound/usb/card.c registers USB suspend and resume but did not previously
kill the input URBs. This means that USB MIDI devices left open across
suspend/resume had non-functional input (output still usually worked,
but it looks like that is another issue). Before this change, we would
get ESHUTDOWN for each of the input URBs at suspend time, killing input.
Signed-off-by: NAdam Goode <agoode@google.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The Linux kernel coding style guidelines suggest not using typedefs
for structure types. This patch gets rid of the typedefs for pss_mixerdata
and pss_confdata.

The following Coccinelle semantic patch is used to make the change.

@tn@
identifier i;
type td;
@@

-typedef
 struct i { ... }
-td
 ;

@@
type tn.td;
identifier tn.i;
@@

-td
+ struct i
Signed-off-by: NHimangi Saraogi <himangi774@gmail.com>
Acked-by: NJulia Lawall <julia.lawall@lip6.fr>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This typedef is unnecessary and should just be removed as they are
never used.

The following Coccinelle semantic patch detects the case.

@tn@
identifier i;
type td;
@@

-typedef
 struct i { ... }
-td
 ;

@@
type tn.td;
identifier tn.i;
@@

-td
+ struct i
Signed-off-by: NHimangi Saraogi <himangi774@gmail.com>
Acked-by: NJulia Lawall <julia.lawall@lip6.fr>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Use %d for loop counter and %X for device capabilities. This is a
supplemental patch for Hans Wennborg's patch.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Since we pass the port number through file private data for debugfs we cast
it to and from a pointer so use uintptr_t in order to ensure that the
types are compatible, avoiding warnings on 64 bit platforms where pointers
are 64 bit and unsigned integers 32 bit.
Signed-off-by: NMark Brown <broonie@linaro.org>

In order to be able to use simple-card with McASP on TI SoC based boards we
need to be able to select the McASP via menuconfig.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

sound/soc/fsl/fsl_asrc.c:250 fsl_asrc_config_pair()
	warn: variable dereferenced before check 'config' (see line 243)

git remote add next git://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git
git remote update next
git checkout 3117bb31
vim +/config +250 sound/soc/fsl/fsl_asrc.c

  237   */
  238  static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair)
  239  {
  240   struct asrc_config *config = pair->config;
  241   struct fsl_asrc *asrc_priv = pair->asrc_priv;
  242   enum asrc_pair_index index = pair->index;
 @243   u32 inrate = config->input_sample_rate, indiv;
  244   u32 outrate = config->output_sample_rate, outdiv;
  245   bool ideal = config->inclk == INCLK_NONE;
  246   u32 clk_index[2], div[2];
  247   int in, out, channels;
  248   struct clk *clk;
  249
 @250   if (!config) {
  251           pair_err("invalid pair config\n");
  252           return -EINVAL;
  253   }
Reported-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NNicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

The patch 3117bb31: "ASoC: fsl_asrc: Add ASRC ASoC CPU DAI and
platform drivers" from Jul 29, 2014, leads to the following Smatch
complaint:

sound/soc/fsl/fsl_asrc_dma.c:304 fsl_asrc_dma_shutdown()
warn: variable dereferenced before check 'pair' (see line 302)

sound/soc/fsl/fsl_asrc_dma.c
301          struct fsl_asrc_pair *pair = runtime->private_data;
302          struct fsl_asrc *asrc_priv = pair->asrc_priv;
                                          ^^^^^^^^^^^^^^^
                                            Dereference.

303
304          if (pair && asrc_priv->pair[pair->index] == pair)
                 ^^^^
                Check.

305                  asrc_priv->pair[pair->index] = NULL;
306

So we just let the driver check pair before using it.
Reported-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NNicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

Negating ADAU1977_BLOCK_POWER_SAI_LDO_EN creates an unsigned long constant
with all bits set which on 64 bit architectures needs to be truncated to
an unsigned int, generating a warning. Add an explicit cast since we know
this is OK.
Signed-off-by: NMark Brown <broonie@linaro.org>
Acked-by: NLars-Peter Clausen <lars@metafoo.de>

Just add the PCI ID for the STX II.  It appears to work the same as the
STX, except for the addition of the not-yet-supported daughterboard.
Tested-by: NMario <fugazzi99@gmail.com>
Tested-by: Ncorubba <corubba@gmx.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NHans Wennborg <hans@hanshq.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NHans Wennborg <hans@hanshq.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds codec ID (0x80862883) and module alias for Braswell
display codec.
Signed-off-by: NLibin Yang <libin.yang@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add HD Audio Device PCI ID for the Intel Braswell platform.
It is an HDA Intel PCH controller.

AZX_DCAPS_ALIGN_BUFSIZE is not necessary for this controller.
Signed-off-by: NLibin Yang <libin.yang@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Original patch fixed the original problem, but the sound was far too low
for most users. This patch references a compare matrix to allow the
volume levels to act normally. I personally tested this patch myself,
and volume levels returned to normal. Please see this discussion for
more details: https://bugzilla.kernel.org/show_bug.cgi?id=65251Signed-off-by: NPaul S McSpadden <fisch602@gmail.com>
Cc: <stable@vger.kernel.org> [v3.14+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>