Turing on the headphone amp interferes with the impedance measurement
used to detect a TRRS style headset microphone.  Delay the HP turn on
until 500ms after the jack is detected, allowing the mic detection
state machine to run to completion.
Signed-off-by: NChih-Chung Chang <chihchung@chromium.org>
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Used kmemdup instead of replicating it's behaviour with kmalloc followed
by memcpy.
Patch found using coccinelle.
Signed-off-by: NAlexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Used kmemdup instead of replicating it's behaviour with kmalloc followed
by memcpy.
Patch found using coccinelle.
Signed-off-by: NAlexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The recent fix for the independent HP reduced the availability of the
side surround output, because there are only 4 DACs for 7.1 and a HP
outputs.  Adjust the badness tables for VIA so that 7.1 outputs are
activated for the cost of missing independent HP.

Once when we implement the dynamic DAC switching to multiple outputs,
this conflicts will be eased in future...
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The lack of independent HP mode shouldn't be too bad, but currently
its badness is set a bit too high.  Let's lower it.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The standard badness values don't seem to fit to all preferences.
Some configuration prefer the side output over the headphone, some
want the speaker over the surround, etc.

This patch moves the badness table pointers into hda_gen_spec, so that
the codec driver can override them.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Merge back for-linus branch for the badness table adjustment for VIA codecs

* for-linus:
  ALSA: hda - Fix DAC assignment for independent HP
  ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
  ALSA: hda - Fix typo in checking IEC958 emphasis bit
  ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
  ALSA: snd-usb: mixer: propagate errors up the call chain
  ALSA: usb: Parse UAC2 extension unit like for UAC1
  ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver

The generic parser should evaluate the availability of the independent
HP when specified.  Otherwise a DAC without the direct connection to
the corresponding pin may be assigned for the HP, but the driver
doesn't check it at all.  The problem was actually seen on some
machines with VT1708s or equivalent codec, where DAC0 is assigned to
HP although it can be connected only via aamix.

This patch adds the badness evaluation for the independent HP to make
it working properly.
Reported-by: NLydia Wang <LydiaWang@viatech.com.cn>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Now that we have a "Headset Mic" name, let's use it for some devices
we know for sure has a headset mic jack.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Headset mic jacks, i e TRRS style jacks with Headphone Left,
Headphone Right, Mic and GND signals, are becoming increasingly
common and are now being shipped by several manufacturers.

Unfortunately, the HDA specification does not give us any hint
of whether a Mic pin belongs to such a jack or not, but it would
still be helpful for the user to know (especially if there is one
TRS Mic jack and one TRRS headset jack).

This new fixup causes the first (non-dock, non-internal) mic to
be a headset mic jack. The algorithm can be later refined if needed.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The current DSP loader code abuses snd_hda_lock_devices() for ensuring
the DSP loader not conflicting with the other normal operations.  But
this trick obviously doesn't work for the PM resume since the streams
are kept opened there where snd_hda_lock_devices() returns -EBUSY.
That means we need another lock mechanism instead of abuse.

This patch provides the new lock state to azx_dev.  Theoretically it's
possible that the DSP loader conflicts with the stream that has been
already assigned for another PCM.  If it's running, the DSP loader
should simply fail.  If not -- it's the case for PM resume --, we
should assign this stream temporarily to the DSP loader, and take it
back to the PCM after finishing DSP loading.  If the PCM is operated
during the DSP loading, it should get an error, too.
Reported-and-tested-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

There is a typo in convert_to_spdif_status() about checking the
emphasis IEC958 status bit.  It should check the given value instead
of the resultant value.
Reported-by: NMartin Weishart <martin.weishart@telosalliance.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Creation of individual mixer controls may fail, but that shouldn't cause
the entire mixer creation to fail. Even worse, if the mixer creation
fails, that will error out the entire device probing.

All the functions called by parse_audio_unit() should return -EINVAL if
they find descriptors that are unsupported or believed to be malformed,
so we can safely handle this error code as a non-fatal condition in
snd_usb_mixer_controls().

That fixes a long standing bug which is commonly worked around by
adding quirks which make the driver ignore entire interfaces. Some of
them might now be unnecessary.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-and-tested-by: NRodolfo Thomazelli <pe.soberbo@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In check_input_term() and parse_audio_feature_unit(), propagate the
error value that has been returned by a failing function instead of
-EINVAL. That helps cleaning up the error pathes in the mixer.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in
the same way when parsing the unit. Otherwise parse_audio_unit() fails when it
sees an extension unit on a UAC2 device.

UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1.
Signed-off-by: NTorstein Hegge <hegge@resisty.net>
Acked-by: NDaniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

I forgot to update spec->gpio_data in the automute hook, so it will be
overridden at the init sequence, thus the machine is still silent when
no headphone jack is plugged at boot time.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The new HP desktop machines have Realtek codecs and their LEDs are
controlled via GPIO as for many laptop models.  Add similar hooks as
well as in patch_sigmatel.c for controlling LEDs.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The verb to set up the digital beep via AC_VERB_SET_DIGI_CONVERT_2
should be executed at resume as well.  Use the cached write for it
being performed automatically at resume.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

While playing the digital beep tone, the codec shouldn't be turned
off.  This patch adds proper snd_hda_power_up()/down() calls at each
time when the beep is played or off.

Also, this fixes automatically an unnecessary codec power-up at
detaching the beep device when the beep isn't being played.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Instead of calling snd_hda_attach_beep_device() and
snd_hda_detach_beep_device() in each codec driver, move them to the
generic parser.  The codec driver just needs to set spec->beep_nid for
activating the digital beep.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Back-merged for refactoring beep stuff.

The argument passed to snd_hda_attach_beep_device() is a widget NID
while spec->beep_amp holds the composed value for amp controls.

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

This leaks the beep input device after module unload, which leads to
Oops.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321Signed-off-by: NTakashi Iwai <tiwai@suse.de>

"Playback Design" products need a 50ms delay after setting the USB
interface.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-by: NAndreas Koch <andreas@akdesigninc.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

UAC2 compliant audio devices may announce the capability to transport
raw audio data on their endpoints. Catch this and handle it as
'special' stream on the ALSA side.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-by: NAndreas Koch <andreas@akdesigninc.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This field may use up to 32 bits, so it should be handled as unsigned
int.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-by: NAndreas Koch <andreas@akdesigninc.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The maxpacksize field is given in some quirks, but it gets ignored (in
favour of wMaxPacketSize from the first endpoint.) This patch favours
the one in the quirk.

Digidesign Mbox and Mbox 2 are the only affected quirks and the devices
are assumed to be working without this patch. So for safety against the
values in the quirk being incorrect, remove them.

The datainterval is also ignored but there are not currently any quirks
which choose to override this.

Cc: Damien Zammit <damien@zamaudio.com>
Cc: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: NMark Hills <mark@xwax.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The hardware also has a PCM capture device which is not implemented in
this patch.

It may be possible to generalise this to Saffire 6 USB support and some
of the other Focusrite interfaces, but as I don't have access to these
devices we should wait until capture support is working first.

Capture support is not implemented because the code assumes the endpoint
to have its own interface (instead, it shares the interface with playback)
and some thought will be needed to lift this limitation.
Signed-off-by: NMark Hills <mark@xwax.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Correct spelling typos in Documentation/sound/alsa
Signed-off-by: NMasanari Iida <standby24x7@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

During the transition to the generic parser, the hook to the codec
specific automute function was forgotten.  This resulted in the silent
output on some MacBooks.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

"chn" here is a number between 0 and 255, but ->chn_info[] only has
16 elements so there is a potential write beyond the end of the
array.

If the seq_mode isn't SEQ_2 then we let the individual drivers
(either opl3.c or midi_synth.c) handle it.  Those functions all
do a bounds check on "chn" so I haven't changed anything here.
The opl3.c driver has up to 18 channels and not 16.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset
and check it in ca0132_download_dsp().
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Instead of using the dspload_is_loaded() function, check the dsp_state
that is kept in the spec.  The dspload_is_loaded() function returns
true if the DSP transfer was never started.  This false-positive leads
to multiple second delays when ca0132_setup_efaults() times out on
each write.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If dspload_image() fails, it was ignored and dspload_wait_loaded() was
still called.  dsp_loaded should never be set to true in this case,
skip it.  The check in dspload_wait_loaded() return true if the DSP is
loaded or if it never started.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If there are no internal speakers, we should not turn the eapd switch
off, because it might be necessary to keep high for Headphone.

BugLink: https://bugs.launchpad.net/bugs/1155016Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

So far, the driver doesn't power down the widget at going down to D3
when the widget node has an EAPD capability and EAPD is actually set
on all codecs unless codec->power_filter is set explicitly.
This caused a problem on some Conexant codecs, leading to click
noises, and we set it as NULL there.  But it is very unlikely that the
problem hits only these codecs.

Looking back at the development history, this workaround for EAPD was
introduced just for some laptops with STAC9200 codec, then we applied
it blindly for all.  Now, since it's revealed to have an ill effect,
we should disable this workaround per default and apply only for the
known requiring systems.

The EAPD workaround is implemented now as snd_hda_codec_eapd_power_filter(),
and this has to be set explicitly by the codec driver when needed.
As of now, only patch_stac9200() sets this one.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Use the dynamic array allocations for pins, converters and PCM arrays
instead of the fixed size arrays.  The modern HDMI codecs get more and
more pins, and we don't know the sensitive limit.

Most of the patch are spent for the straight conversions from the
fixed array access to snd_array helpers.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Make sure that the allocated buffer for reading the proc file won't
expose the uncleared kernel memory.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>