Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.

That quirks handler can augment the already parsed formats bit-field,
by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop
flag in the audio format, if the driver should take care for the DOP
byte stuffing.

The only devices that are known to work with this are the ones with
a 'Playback Designs' vendor id.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There is quite some confusion around the bit-ordering in DSD samples,
and no general agreement that defines whether hardware is supposed to
expect the oldest sample in the MSB or the LSB of a byte.

ALSA will hence set the rule that on the software API layer, bytes
always carry the oldest bit in the most significant bit of a byte, and
the driver has to translate that at runtime in order to match the
hardware layout.

This patch adds support for this by adding a boolean flag to the
audio format struct.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com

 for the official document.

The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.

To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.

The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

For normal PCM transfer, this change has no effect, as the endpoint's
stride is always frame_bits/8. For DSD DOP streams, however, which is
added later, the hardware stride differs from the software stride, and
the endpoint has the correct information in these cases.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital



DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.

The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).

DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:

                                                  configured hardware
        176.4KHz   352.8kHz   705.6KHz     <----       sample rate

8-bit                2.8MHz     5.6MHz
16-bit    2.8Mhz     5.6MHz    11.2MHz

         `-----------------------------'
             actual DSD sample rates

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When pin default configs are overridden via patch option, these are
evaluated before fixups are applied.  Since some fixups change the
whole codec trees and/or add pins dynamically, this sanity check might
not pass when pins aren't present at the time the function is called.

We may reorder the execution, but an easier fix is simply to disable
this sanity check.

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

Fix to return a negative error code from the error handling
case instead of 0, as returned elsewhere in this function.

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The flag bus->shutdown implies that the control elements might have
been already destroyed.  When a codec is resumed at this state and
tries to call vmaster hook (e.g. in snd_hda_gen_init()), it would
refer to a non-existing object, resulting in Oops in the end.

This patch just adds a check of the flag in the caller side for
avoiding such a crash.

Though, the best would be to clear hook->sw_kctl by the destructor of
the corresponding ctl element, but vmaster uses its own private_free,
it can't be done easily.  So let it be for a while.

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

When graphics initializes the HDMI chip, sometimes this leads to
pins going into D3 and right channel being muted. If the audio driver
finishes initialization before the graphic driver does, this situation
becomes permanent.

This is a workaround that checks for this situation and corrects it on
playback prepare. It has been verified working on at least one machine.

BugLink: https://bugs.launchpad.net/bugs/1167270


Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When setting up the aamix output paths, use the primary DAC instead of
the individual DAC for each output as default.  Otherwise multiple
DACs will be turned on for a single aamix widget, which results in
doubly or more volumes, because the duplicated signals will be sent
through all these DACs for a single stream.

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

When we have a loopback mixer control, this should manage the state
whether the output paths include the aamix or not.  But the current
code blindly initializes the output paths with aamix = true, thus the
aamix is enabled unless the loopback mixer control is changed.

Also, update_aamix_paths() called by the loopback mixer control put
callback invokes snd_hda_activate_path() with aamix = true even for
disabling the mixing.  This leaves the aamix path even though the
loopback control is turned off.

This patch fixes these issues:
- Introduced aamix_default() helper to indicate whether with_aamix is
  true or false as default
- Fix the argument in update_aamix_paths() for disabling loopback

Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>

For capture, the delay through the codec contributes to the time stamp
of the sample recorded at the A to D.  Rename the codec time stamp
function appropriately.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: Updates for v3.10

A bunch of changes here, the most interesting one subsystem wise being
Morimoto-san's work to create snd_soc_component which doesn't do much
for now but will be pretty important going forwards:

 - Add a new component object type which will form the basis of moving
   to a more generic handling of SoC and off-SoC components, contributed
   by Kuninori Morimoto.
 - A fairly large set of cleanups for the dmaengine integration from
   Lars-Peter Clausen, starting to move towards being able to have a
   generic driver based on the library.
 - Performance optimisations to DAPM from Ryo Tsutsui.
 - Support for mixer control sharing in DAPM from Stephen Warren.
 - Multiplatform ARM cleanups from Arnd Bergmann.
 - New CODEC drivers for AK5385 and TAS5086 from Daniel Mack.

Commit 88a8516a (ALSA: usbaudio: implement USB autosuspend)
introduced autopm for all USB audio/MIDI devices.  However, many MIDI
devices, such as synthesizers, do not merely transmit MIDI messages but
use their MIDI inputs to control other functions.  With autopm, these
devices would get powered down as soon as the last MIDI port device is
closed on the host.

Even some plain MIDI interfaces could get broken: they automatically
send Active Sensing messages while powered up, but as soon as these
messages cease, the receiving device would interpret this as an
accidental disconnection.

Commit f5f16541

 (ALSA: usb-audio: Fix missing autopm for MIDI input)
introduced another regression: some devices (e.g. the Roland GAIA SH-01)
are self-powered but do a reset whenever the USB interface's power state
changes.

To work around all this, just disable autopm for all USB MIDI devices.

Reported-by: Laurens Holst
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

With this patch, a TRRS headset mic cannot be successfully detected
on the Asus X101CH, and we can also distinguish between headphone
and headset automatically.

Buglink: https://bugs.launchpad.net/bugs/1169138


Co-authored-by: Kailang <kailang@realtek.com>
Tested-by: Luis Henriques <luis.henriques@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

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/1169143


Tested-by: David Chen <david.chen@canonical.com>
Co-authored-by: Kailang <kailang@realtek.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi 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: Calvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>