On some machines, there is a headset jack that can support both
headphone, headsets (of both CTIA and OMTP type) and mic-in.

On other machines, the headset jack supports headphone, headsets
(both CTIA and OMTP), but not mic-in.

This patch implements that functionality as different capture sources.

Buglink: https://bugs.launchpad.net/bugs/1169143Tested-by: NDavid Chen <david.chen@canonical.com>
Co-authored-by: NKailang <kailang@realtek.com>
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.

Userspace expected:  L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1

Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.

Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.
Signed-off-by: NCalvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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>

Back-merge for cleaning up usb-audio code the recent commit modified,
and further UAC2 autoclock patches.

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>

Merge tag 'asoc-fix-v3.9-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.9

A few more fixes here and there, including quite a few nasty driver
specific ones, but nothing that has a major general impact.

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>