This is a quick fix for the below two issues when building spdif as modules.

1) If modprobing modules in order: (Step 1) snd-soc-fsl-spdif -> (Step 2)
snd-soc-imx-spdif -> (Step 3) snd-soc-spdif-tx/rx, we will fail to create
imx-spdif card and dai link unless we rmmod snd-soc-imx-spdif and modprobe
it again due to the execution platform_driver_unregister() in probe() when
meeting -EPROBE_DEFER at Step 2.

2) After "imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok",
'rmmod snd-soc-imx-spdif' would cause kernel dump with warning:
WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90()
sysfs: can not remove 'dapm_widget', no directory
This should be caused by disordered resourse releasing of the whole link.
And trying to unregister the card and then CODEC dev can't fix this issue.

Thus this patch just provides a simple fix to these two bugs by using the
snd-soc-dummy in the core instead of seperate snd-soc-spdif-tx/rx so that
there's no need to handle the registering and unregistering of CODEC or
CODEC dai any more.
Signed-off-by: NNicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

From "ASoC: make snd_soc_dai_link more symmetrical", can we see that
the name of CPU DAI maybe omitted. If the DAI name is omitted, try to
use the component name instead.
Signed-off-by: NXiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

On the Dell machines with codec whose Subsystem Id is 0x10280640,
no external microphone can be detected when plugging a 3-ring headset.
Using ALC255_FIXUP_DELL1_MIC_NO_PRESENCE can fix this problem.

The codec (Vendor ID: 0x10ec0255) on the machine belongs to alc_269
family.

BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When wm8904 work in DSP mode B, we still need to configure it to
work in DSP mode. Or else, it will work in Right Justified mode.
Signed-off-by: NBo Shen <voice.shen@atmel.com>
Acked-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org

Some devices are getting very close to the limit whilst polling the RAM
start, this patch adds a small delay to this loop to give a longer
startup timeout.
Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org

When the process is sleeping at the SNDRV_PCM_STATE_PAUSED
state from the wait_for_avail function, the sleep process will be woken by
timeout(10 seconds). Even if the sleep process wake up by timeout, by this
patch, the process will continue with sleep and wait for the other state.
Signed-off-by: NJongHo Kim <furmuwon@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch fixes the rates declared in the CPU DAI parameters:
- SNDRV_PCM_RATE_KNOT and the discrete rates SNDRV_PCM_RATE_xxx should
  not be used with SNDRV_PCM_RATE_CONTINUOUS,
- SNDRV_PCM_RATE_CONTINUOUS asks for rate_min and rate_max,
- the device may do streaming down to 5512Hz.
Signed-off-by: NJean-Francois Moine <moinejf@free.fr>
Signed-off-by: NMark Brown <broonie@linaro.org>

Reported-by: NKyung-Kwee Ryu <kyung-kwee.ryu@wolfsonmicro.com>
Signed-off-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org

    DPCM can dynamically alter the FE to BE PCM links at runtime based
on mixer/mux setting updates. Add soc_dpcm_runtime_update() calling in
get/put function for mixer/mux to support this feature.
Signed-off-by: NNenghua Cao <nhcao@marvell.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

On the Dell machines with codec whose Subsystem Id is 0x10280610,
0x10280629 or 0x1028063e, no external microphone can be detected when
plugging a 3-ring headset. If we add "model=dell-headset-multi" for
the snd-hda-intel.ko, the problem will disappear.

The codecs on these machines belong to alc_269 family.

BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

While enabling these machines, we found we would sometimes lose an
interrupt if we change hardware volume during playback, and that
disabling msi fixed this issue. (Losing the interrupt caused underruns
and crackling audio, as the one second timeout is usually bigger than
the period size.)

The machines were all machines from HP, running AMD Hudson controller,
and Realtek ALC282 codec.

Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1260225Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

AD1986A codec is a pretty old codec and has really many hidden
restrictions.  One of such is that each DAC is dedicated to certain
pin although there are possible connections.  Currently, the generic
parser tries to assign individual DACs as much as possible, and this
lead to two bad situations: connections where the sound actually
doesn't work, and connections conflicting other channels.

We may fix this by trying to find the best connections more harder,
but as of now, it's easier to give some hints for paired DAC/pin
connections and honor them if available, since such a hint is needed
only for specific codecs (right now only AD1986A, and there will be
unlikely any others in future).

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

On the Dell machines with codec whose Subsystem Id is 0x10280624,
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.

BugLink: https://bugs.launchpad.net/bugs/1259790
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In case a single HDA card has both HDMI and S/PDIF outputs, the S/PDIF
outputs will have their IEC958 controls created starting from index 16
and the HDMI controls will be created starting from index 0.

However, HDMI simple_playback_build_controls() as used by old VIA and
NVIDIA codecs incorrectly requests the IEC958 controls to be created
with an S/PDIF type instead of HDMI.
In case the card has other codecs that have HDMI outputs, the controls
will be created with wrong index=16, causing them to e.g. be unreachable
by the ALSA "hdmi" alias.

Fix that by making simple_playback_build_controls() request controls
with HDMI indexes.

Not many cards have an affected configuration, but e.g. ASUS M3N78-VM
contains an integrated NVIDIA HDA "card" with:
- a VIA codec that has, among others, an S/PDIF pin incorrectly
  labelled as an HDMI pin, and
- an NVIDIA MCP7x HDMI codec.

Reported-by: MysterX on #openelec
Tested-by: MysterX on #openelec
Signed-off-by: NAnssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org> # 3.8+
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Not all channels have been initialized, so far, especially when aamix
NID itself doesn't have amps but its leaves have.  This patch fixes
these holes.  Otherwise you might get unexpected loopback inputs,
e.g. from surround channels.

Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

On the Dell Inspiron 3045 machine (codec Subsystem Id: 0x10280628),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.

BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259437
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

On the Dell Optiplex 3030 machine (codec Subsystem Id: 0x10280623),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.

BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259435
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If snd_dmaengine_pcm_register()'s call to snd_soc_add_platform() fails,
all objects allocated during registration are leaked. Fix this by adding
error-handling code.
Signed-off-by: NStephen Warren <swarren@nvidia.com>
Acked-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@linaro.org>

If we update it here, the set_bias_level() of Codec driver won't be normally
called and we will then miss some essential procedures in set_bias_level() of
the Codec driver. Thus drop it.
Signed-off-by: NNicolin Chen <b42378@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

In tegra*_i2s_set_fmt(), in the (fmt == SND_SOC_DAIFMT_CBM_CFM) case,
"val" is never assigned to, but left uninitialized. The other case does
initialized it. Fix this by initializing val at the start of the
function, and only ever ORing into it.

Update the handling of "mask" so it works the same way for consistency.

Update tegra20_spdif.c to use the same code-style for consistency, even
though it doesn't happen to suffer from the same problem at present.
Signed-off-by: NStephen Warren <swarren@nvidia.com>
Reviewed-by: NThierry Reding <treding@nvidia.com>
Signed-off-by: NMark Brown <broonie@linaro.org>
Fixes: 0f163546 ("ASoC: tegra: use regmap more directly")
Cc: <stable@vger.kernel.org>

I've tested the old Dell Vostro 131 with the latest generic parser
and it works just fine, and as a bonus we get better jack detection
features in userspace. Therefore this quirk can be removed.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix the following warning when optimizing for size with gcc-4.6.4:
sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: NMikulas Patocka <mpatocka@redhat.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

DSPCLK_DIV can be only generated correctly after enabling SYSCLK. But if the
current bias_level hasn't reached SND_SOC_BIAS_ON, DAPM won't enable SYSCLK,
which would cause the calculation result from DSPCLK_DIV invalid since bit
DSPCLK_DIV will be finally turned to its true value after DAPM enables SYSCLK
while the driver won't calculate it again for the current instance. In this
circumstance, a playback which needs non-zero DSPCLK_DIV would be distorted
due to unexpected clock frequency resulted from an invalid DSPCLK_DIV value.

So this patch provisionally enables the SYSCLK to get a valid DSPCLK_DIV for
calculation and then disables it afterward.
Signed-off-by: NNicolin Chen <b42378@freescale.com>
Acked-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

This patch add quirk for Acer Aspire E-572:
- fix external mic
- limit mic boost for internal mic with maximal noise level of -24dB
Signed-off-by: NOleksij Rempel <linux@rempel-privat.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

MacBook Air 2,1 has a fairly different pin assignment from its brother
MBA 1,1, and yet another quirks are needed for pin 0x18 and 0x19,
similarly like what iMac 9,1 requires, in order to make the sound
working on it.
Reported-and-tested-by: NBruno Prémont <bonbons@linux-vserver.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Change sam9x5 with wm8731 work in DSP A mode, this will fix the
left/right channel swap issue.
Signed-off-by: NBo Shen <voice.shen@atmel.com>
Tested-by: NRichard Genoud <richard.genoud@gmail.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

According to the SSC specifiation, it should be enabled after DMA is
enabled. So, add trigger operation to make sure the right sequence.
Signed-off-by: NBo Shen <voice.shen@atmel.com>
Tested-by: NRichard Genoud <richard.genoud@gmail.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

The snd_soc_dai_digital_mute() here will be never executed because we only
decrease codec->active in snd_soc_close(). Thus correct it.
Signed-off-by: NNicolin Chen <b42378@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

This patch removed the redundant snd_soc_dai_digital_mute() in close() since
it's better to mute in hw_free() which's slightly earlier and symmetrical for
the case of reconfiguration: 'aplay 44k1.wav 48k.wav', for example, will be
open()->hw_params()->prepare(unmute)->playi1ng->hw_free(mute)->hw_params()->
parepare(unmute)->playing->hw_free(mute)->close()
Signed-off-by: NNicolin Chen <b42378@freescale.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

In the case of using jackpoll_ms instead of unsol events, the jack
was correctly detected, but ELD info was not refreshed on plug-in.

And without ELD info, no proper restriction of pcm, which can in turn
break sound output on some devices.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

I forgot to remove the hp_automute_hook from alc283_fixup_chromebook.
It doesn't need this for other chrome os machine.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it
works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0.
So, fix LRP for DSP mode as the datesheet specification.
Signed-off-by: NBo Shen <voice.shen@atmel.com>
Acked-by: NCharles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org

Create single model for HP.
The headset jack module was difference between other chrome book.
It need to manual control Mic jack detect.

Chrome OS loaded driver by models. Remove old assigned fixup table from
ALC269 fixup list entry.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

By trial and error, I found this patch could work around an issue
where the headset mic would stop working if you switch between the
internal mic and the headset mic, and the internal mic was muted.

It still takes a second or two before the headset mic actually starts
working, but still better than nothing.

Information update from Kailang:
  The verb was ADC digital mute(bit 6 default 1).
  Switch internal mic and headset mic will run alc_headset_mode_default.
  The coef index 0x11 will set to 0x0041.
  Because headset mode was fixed type. It doesn't need to run
  alc_determine_headset_type.
  So, the value still keep 0x0041. ADC was muted.

BugLink: https://bugs.launchpad.net/bugs/1256840Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

It seems that AD1986A cannot manage the dynamic pin on/off for
auto-muting, but rather gets confused.  Since each output has own amp,
let's use it instead.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Cc: <stable@vger.kernel.org> [v3.11+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ad_vmaster_eapd_hook() needs to handle the inverted EAPD case
properly, too.  Otherwise the output gets broken on Lenovo N100 with
AD1986A codec.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ASUS Z35HL laptop also needs the very same fix as the previous one
that was applied to ASUS W7J.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66231
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The static checker found a possible array overflow in atmel/abdac.c:
  static checker warning: "sound/atmel/abdac.c:373 set_sample_rates()
        error: buffer overflow 'dac->rates' 6 <= 6"

This patch papers over the buggy point, by ensuring that dac->rates[]
update not overflowing the actual array size.
Reported-by: NDan Carpenter <dan.carpenter@oracle.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>