Just for cleaning up, introduce a new function get_sample_rate_v2()
for replacing two identical calls in set_sample_rate_v2().

No functional change.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The C-Media CM6631 USB receiver doesn't respond to changes in sample rate
while the interface is active. The same behavior is observed in other UAC2
hardware like the VIA VT1731.

Reset the interface after setting the sampling frequency on sample rate
changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is
used. Otherwise, the device will try to use the sample rate of the previous
stream, causing distorted sound on sample rate changes.

The reset is performed for all UAC2 devices, as it should not affect a
standards compliant device, but it is only necessary for C-Media CM6631,
VIA VT1731 and possibly others.

Failure to read sample rate from the device is not handled as an error in
set_sample_rate_v2(), as (permanent or intermittent) failure to read sample
rate isn't essential for a successful sample rate set.
Signed-off-by: NTorstein Hegge <hegge@resisty.net>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch let ELD debug message show 'pin_eld->monitor_present' which reflects
the real pin response to verb GET_PIN_SENSE.

'eld->monitor_present' should not be used here because 'eld' is a temp
structure now and so its "monitor_present" is not set.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Acked-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0.
Otherwise it will be returned uninitialized as non-zero after ELD info is got
successfully. Thus hdmi_present_sense() will always assume ELD info is invalid
by mistake, and /proc file system cannot show the proper ELD info.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Cc: stable@vger.kernel.org
Acked-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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>

The dma-sh7760 currently fails with the following compile error:
	sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer
	sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type
	sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer
	sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast
	sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer
	sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type
	sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe':
	sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type
	include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *'

This is due the misnaming of the snd_soc_platform_driver type name and 'ops'
field. The issue was introduced in commit f0fba2ad("ASoC: multi-component - ASoC
Multi-Component Support").
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

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>

The objects allocated by devm_* APIs are managed by devres and are freed when
the device is detached. Hence there is no need to use kfree() explicitly.
Signed-off-by: NSilviu-Mihai Popescu <silviupopescu1990@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Commit 552d1ef6 ("ASoC: core - Optimise and refactor pcm_new() to pass only
rtd") updated the pcm_new() callback to take the rtd as the only parameter. The
spear PCM driver (which was merged much later) still uses the old API. This
patch updates the driver to the new API.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Acked-by: NRajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

Source files shouldn't have the executable bit set.
Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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>

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>

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>

*path is not yet initialized when we check if the widget is connected.

The compiler also warns about this:
sound/soc/soc-dapm.c: In function 'is_connected_output_ep':
sound/soc/soc-dapm.c:824:18: warning: 'path' may be used uninitialized in this function
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>