Now that we have a flag for headphone mics, we can use that flag
in the jack creation instead of creating the jack manually.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

I never liked that we move our speaker and hp pins to line out
if there are not any line outs; but now that we do,
add some convenience functions to find hp and speaker pins even
if they have been moved.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This allows a specific mic to get the "Headphone Mic" name, in addition
to the existing "Headset Mic" name.

Also, it allows for a special mark: if the sequence number is set
to 0xc, that's an indication to prefer it for headset mic, and if it's
set to 0xd, that's an indication to prefer it for headphone mic.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Use dev_pm_ops instead of the deprecated legacy suspend/resume callbacks.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Acked-by: NHans-Christian Egtvedt <egtvedt@samfundet.no>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

It turns out the devices from Playback Design need the delay quirk
after usb_set_interface from clocks.c as well. Make it a proper
quirks function and factor out the code to quirks.c.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

For playback add the codec-side delay to the timestamp, for capture
subtract it.  This brings the timestamps in line with the time that
was recently added to the delay reporting.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Correct pin configs for the Acer AC700.  Most importantly indicate
that SPDIF is connected, it routes to HDMI out.
Similar to Aspire models, chain in the DMIC fixup and allow it to be
applied to this codec (ALC269VB) as well.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The Stumpy ChromeBox needs its pin configs fixed up.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The DSP in the CA0132 codec adds a variable latency to audio depending
on what processing is being done.  Add a new patch op to return that
latency for capture and playback streams.  The latency is determined
by which blocks are enabled and knowing how much latency is added by
each block.
Signed-off-by: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a new codec PCM ops, get_delay(), to obtain the codec/stream-
specific PCM delay count.  When it's NULL, nothing changes.

This new feature was requested for CA0132, which has significant
delays in the path depending on the running DSP code.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Some clocks might be read-only, e.g., external clocks (see also
UAC2 4.7.2.1).

In this case, setting the sample frequency will always fail
(even if the rate is equal to the current clock rate),
therefore do not write, but read the value and compare to the
requested rate.
If the clock is read only, avoid reading it twice.

If it doesn't match, return -ENXIO since the clock is invalid for
this configuration.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Show the error code returned from the USB subsystem in
the debug messages.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a module param to disable auto clock selection.
This is provided for users that expect the audio stream to
fail when the clock source is invalid (e.g., the word clock
was unintentionally disconnected).
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If a selector is available on a device, it may be pointing to a
clock source which is currently invalid.
If there is a valid clock source which can be selected, switch
to it.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Move the check that parse_audio_format_rates_v2() do after
receiving the clock source entity ID directly into the find
function and add a validation flag to the function.

This patch does not introduce any logic flow change.

It is provided to allow introducing automatic clock switching
easier later. By moving this uac_clock_source_is_valid callsite,
2 additional callsites can be avoided.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Replace the endianness conversions with the kernel-wide swabbing macros
in get/set_sample_rate_v2.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Correct spelling of snd_usb_endpoint_implict_feedback_sink in all
occurances.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Put EXPORT_SYMBOLS directly under the exported function.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Minor style fix, following a general code style in the kernel.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Change occurances of list_for_each into list_for_each_entry where
applicable.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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>