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: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi 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: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi 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: NTakashi 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: NWei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: NTakashi 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: NTakashi 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/1167270Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi 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: NTakashi 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: NLydia Wang <LydiaWang@viatech.com.cn>
Cc: <stable@vger.kernel.org> [v3.9+]
Signed-off-by: NTakashi 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: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi 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/1169138Co-authored-by: NKailang <kailang@realtek.com>
Tested-by: NLuis Henriques <luis.henriques@canonical.com>
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi 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/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>

Commit b2ca7871 (ARM: S3C24XX: make gta02.h local) already replaced
the GTA02_GPIO_* constants in neo1973-wm8753.c but forgot to remove the
inclusion of mach/gta02.h before moving the file out of mach/.
Signed-off-by: NHeiko Stuebner <heiko@sntech.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The plat/regs-iis.h and plat/regs-ac97.h files in the samsung platform
are only needed by the ASoC drivers, so they can be moved into the same
directory, as one more step towards a multiplatform build.
Signed-off-by: NArnd Bergmann <arnd@arndb.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Firmwares may provide some firmware wide configuration regions which can
be configured by the coefficient files using the firmware ID as the
algorithm ID, include these in the algorithm list.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The idma_reg_addr_init function is used by the samsung i2s driver,
which can be a loadable module, so we have to export this function.
Signed-off-by: NArnd Bergmann <arnd@arndb.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The second argument to the module_device_table macro must be the
name of the device id array. In the samsung i2s driver, there
was a small typo, resulting in a build error when building it
as a loadable module.
Signed-off-by: NArnd Bergmann <arnd@arndb.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

With multiplatform kernels, we cannot use hardwired IRQ
numbers in device drivers. This changes the idma driver
to use a proper resource, like all other drivers do.
Signed-off-by: NArnd Bergmann <arnd@arndb.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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>

Reported-by: NRyo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

The Charge Pump needs the DSP clock to work properly, without it the
bypass to HP/LINEOUT is not working properly. This requirement is not
mentioned in the datasheet but has been confirmed by Mark Brown from
Wolfson.
Signed-off-by: NAlban Bedel <alban.bedel@avionic-design.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org

There's already a device revision stored in the core data structure,
don't duplicate it in the CODEC driver.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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>

A test for CONFIG_SND_SOC_UX500_AB5500 was added in v3.5. But there
never was a corresponding Kconfig symbol so this test has always
evaluated to true. And since AB5500 support was removed in v3.5 it
appears safe to remove this test and a few lines of code.
Signed-off-by: NPaul Bolle <pebolle@tiscali.nl>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The snd_pcm_hardware structs for playback and capture in the ux500 PCM are
identical, so remove one of them and use the same snd_pcm_hardware struct for
both playback and capture. Also move the defines used to initialize the
snd_pcm_hardware fields from ux500_pcm.h to ux500_pcm.c since that's the only
place where they are used.

Also drop the assignment of the snd_pcm_hardware struct to runtime->hw since
that is what the call to snd_soc_set_runtime_hwparams() right above it already
does, so the second assignment is redundant.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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>

Channel size settings will be made at the end of
davinci_mcasp_hw_params() routine and thus overwrite frame
format settings made for DIT mode. This patch fixes this issue
by taking op_mode into account. Tested with official PSP 3.2
kernel and sii9022a HDMI transmitter.
Signed-off-by: NYegor Yefremov <yegorslists@googlemail.com>
Tested-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

AFSX won't be used in DIT mode. The related pins are AHCLKX and
the data pins.
Signed-off-by: NYegor Yefremov <yegorslists@googlemail.com>
Acked-by: NVaibhav Bedia <vaibhav.bedia@ti.com>
Tested-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Commit 453807f3 ("ASoC: ep93xx: Use ep93xx_dma_params instead of
ep93xx_pcm_dma_params") introduced a small compile error by not updating the
name of the 'dma_port' field to 'port'. This patch fixes it.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Use the common DAI DMA data struct for fsl/imx, this allows us to use the common
helper function to configure the DMA slave config based on the DAI DMA data.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Tested-by: NShawn Guo <shawn.guo@linaro.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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>

changed is not initialized in path_power_down_sync, but it is expected
to be false in case no change happened in the loop. So set it to
false.
Signed-off-by: NJiri Slaby <jslaby@suse.cz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This reverts commit 6ab31741.

The commit [6ab31741: ALSA: hda - Allow power_save_controller option
override DCAPS] changed the behavior of power_save_controller so that
it can override the driver capability.  This assumed that this option
is rarely changed dynamically unlike power_save option.  Too naive.

It turned out that the user-space power-management tool tries to set
power_save_controller option to 1 together with power_save option
without knowing what's actually doing.  This enabled forcibly the
runtime PM of the controller,  which is known to be broken om many
chips thus disabled as default.

So, the only sane fix is to revert this commit again.  It was intended
to ease debugging/testing for runtime PM enablement, but obviously we
need another way for it.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171Reported-and-tested-by: NNikita Tsukanov <keks9n@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>