For legacy reasons the ASoC framework assumes that a CODEC INPUT or OUTPUT
widget that is not explicitly connected to a external source or sink is
potentially connected to a source or a sink and hence the framework treats
the widget itself as source (for INPUT) or sink (for OUTPUT). For this
reason a INPUT or OUTPUT widget that is really not connected needs to be
explicitly marked as so.

Setting the card's fully_routed flag will cause the ASoC core, once that all
widgets and routes have been registered, to go through the list of all
widgets and mark all INPUT and OUTPUT that are not externally connected as
non-connected. This essentially negates the default behaviour of treating
INPUT or OUTPUT widgets without external routes as sources or sinks.

This patch takes a different approach while getting the same result. Instead
of first marking INPUT and OUTPUT widgets as sinks/sources and then later
marking them as non-connected, just never mark them as a sink or a source if
the fully_routed flag is set on a card.

This requires a lot less code and also results in a slightly faster card
initialization since there is no need to iterate over all widgets and check
whether the INPUT and OUTPUT widgets are connected or not.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@kernel.org>

The connected flag of a widget only affects widgets that are either a source
or a sink. The MICBIAS widget is a simple pass-through widget though and
hence its behavior is the same regardless of whether the connected flag is
set or not.

Hence there is not much point in trying to automatically mark MICBIAS
widgets as non-connected, so just remove it.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@kernel.org>

The sst-haswell-pcm driver is a new user of CONFIG_PM_RUNTIME.

However, after commit b2b49ccb (PM: Kconfig: Set PM_RUNTIME if
PM_SLEEP is selected) PM_RUNTIME is always set if PM is set, so
#ifdef blocks depending on CONFIG_PM_RUNTIME may now be changed to
depend on CONFIG_PM.

Replace CONFIG_PM_RUNTIME with CONFIG_PM in
sound/soc/intel/sst-haswell-pcm.c.
Signed-off-by: NRafael J. Wysocki <rafael.j.wysocki@intel.com>
Acked-by: NMark Brown <broonie@kernel.org>

The Arcam rPAC seems to have the same problem - whenever anything
(alsamixer, udevd, 3.9+ kernel from 60af3d03, ..) attempts to
access mixer / control interface of the card, the firmware "locks up"
the entire device, resulting in
  SNDRV_PCM_IOCTL_HW_PARAMS failed (-5): Input/output error
from alsa-lib.

Other operating systems can somehow read the mixer (there seems to be
playback volume/mute), but any manipulation is ignored by the device
(which has hardware volume controls).

Cc: <stable@vger.kernel.org>
Signed-off-by: NJiri Jaburek <jjaburek@redhat.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add new support for ALC298 codec.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This corresponds with updated asihpi firmware in alsa-firmware repo
Signed-off-by: NEliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Increase size allocated for PAD (programme associated data) control.
This is used by newer tuner products.
Signed-off-by: NEliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Some products firmware is no longer being updated
e.g. dsp5000, dsp8700  but it should continue to work
with updated HPI versions.
Avoid regression by allowing this firmware to be loaded as
long as major version is the same.
Warn about mismatching versions, as matching versions are
preferred.
Signed-off-by: NEliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The num_controls field was incorrectly set to 0 causing 6i6 to not be
initialized. Set this to 9.
Reported-and-tested-by: NMark Roberts <sunifiram@gmail.com>
Signed-off-by: NChris J Arges <chris.j.arges@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Packard Bell EasyNote MX65 with AD1986A codec needs a few fixups,
namely, the pin config overrides to set only the known I/O pins and
the EAPD has to be turned on.  In addition, add stereo mix input
forcibly for avoiding the weird KDE behavior by this update.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=88251Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds native DSD support for two XMOS based DACs from Matrix Audio:
- X-Sabre
- Mini-i Pro
Signed-off-by: NJurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add new support for ALC256 codec.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

New Dell desktop needs to support headset mode for ALC3234.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

A crash can occur on some platforms where adsp is enumerated but codec is not
matched. Define codec_id as a pointer intead of an array so that it gets
initialized to NULL for the terminating element of sst_acpi_bytcr[] and
sst_acpi_chv[].
Signed-off-by: NKevin Strasser <kevin.strasser@linux.intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Skylake and Haswell have the same behavior on display audio. So this patch
applys Haswell fix-ups to Skylake.
Signed-off-by: NLibin Yang <libin.yang@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In case of error, the function platform_device_register_data()
returns ERR_PTR() and never returns NULL. The NULL test in the
return value check should be replaced with IS_ERR().
Signed-off-by: NWei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: NMark Brown <broonie@kernel.org>

... for distinguishing whether it's explicitly enabled via a user hint
or enabled by a driver as a fallback.  Now the former case corresponds
to HDA_HINT_STEREO_MIX_ENABLE while the latter to
HDA_HINT_STEREO_MIX_AUTO.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When the stereo mix input is explicitly enabled via a user hint, the
driver should create always a capture source enum ctl and disable the
auto-mic switch.  Otherwise the behavior gets confused.  For doing it,
this patch just sets spec->suppress_auto_mic flag appropriately.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

This patch adds codec ID (0x80862809) and module alias for Skylake
display codec.
Signed-off-by: NLibin Yang <libin.yang@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This code tends to use unsigned variables by default and it causes
signedness bugs when we use negative variables for error handling.
The "i" and "j" variables are used to iterated over small positive
values and so they should be type "int".  The "len" variable doesn't
*need* to be signed but it should be signed to make the code easier to
read and audit.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This code causes a static checker warning:

	sound/firewire/oxfw/oxfw.c:46 detect_loud_models()
	warn: signedness bug returning '(-2)'

The detect_loud_models() function should return false on falure, so that
we don't try to set up the loud code for hardware that doesn't support
it.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

After commit b2b49ccb (PM: Kconfig: Set PM_RUNTIME if PM_SLEEP is
selected) PM_RUNTIME is always set if PM is set, so #ifdef blocks
depending on CONFIG_PM_RUNTIME may now be changed to depend on
CONFIG_PM.

Replace CONFIG_PM_RUNTIME with CONFIG_PM everywhere under sound/.
Signed-off-by: NRafael J. Wysocki <rafael.j.wysocki@intel.com>
Acked-by: NTakashi Iwai <tiwai@suse.de>
Acked-by: NNicolin Chen <nicoleotsuka@gmail.com>
Acked-by: NBrian Austin <brian.austin@cirrus.com>
Acked-by: NMark Brown <broonie@kernel.org>

The codec driver uses regmap to do i2c read/write.
The codec driver started to use REGMAP_IRQ since:

5e3363ad
ASoC: rt5677: add GPIO IRQ support
Signed-off-by: NBen Zhang <benzh@chromium.org>
Signed-off-by: NMark Brown <broonie@kernel.org>

"card" is NULL if snd_card_new() fails.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

"card" is NULL if snd_card_new() fails.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Also update the documentation to the latest state.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The same fixup to enable EAPD is needed for ASUS Z99He with AD1986A
codec like another ASUS machine.
Reported-and-tested-by: NDmitry V. Zimin <pfzim@mail.ru>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This interface is designed for mixer/control application. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This commit adds MIDI functionality with an assumption of 'if the device
has MIDI comformant data channels in its stream formation, the device has
one MIDI port'.

When no streams have already started, MIDI functionality starts stream
with current sampling rate.

When MIDI functionality has already starts some streams and PCM
functionality is going to start streams at different sampling rate,
this driver stops streams once and changes sampling rate, then restarts
streams for both PCM/MIDI substreams.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In previous commit, a support for transmitted packets is added. This commit
add a support for capturing PCM samples.

When any streams are already started, this driver should not change sampling
rate of the device, thus this commit also adds a restriction of sampling rate
in this situation.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Previous commit adds support for some devices which can capture PCM samples.
These devices transmit AMDTP stream in non-blocking mode. This commit adds
functionality to handle AMDTP incoming stream.

OXFW seems to have two quirks:
 - Transmits packets with non-zero dbc in its beginning
 - Transmits packets with wrong values in syt field

For the first quirk, this commit adds CIP_SKIP_INIT_DBC_CHECK flag for
incoming stream to skip first check of dbc.

For the second quirk, this commit doesn't add duplex stream which
Fireworks/BeBoB drivers use. So OXFW driver generates syt value for outgoing
stream.

Here are examples of a sequence of packets transmitted by Behringer F-Control
Audio 202. There are differences between sequences of syt value when OXFW
driver transfers outgoing stream or not.

When driver gives no outgoing stream:
Index   Payload CIP_Header_0    CIP_Header_1
38      14      00020092        900103D1
39      12      00020098        900102FF
40      12      0002009D        9001027F
41      14      000200A2        90010396
42      14      000200A8        900102E8
43      12      000200AE        90010219
44      14      000200B3        90010331
45      12      000200B9        9001025F
46      14      000200BE        90010376
47      12      000200C4        900102A1
00      12      000200C9        9001023E
01      14      000200CE        90010358
02      12      000200D4        90010289
03      16      000200D9        900103A3
04      12      000200E0        900102DD
05      14      000200E5        900103F1
06      12      000200EB        90010335
07      12      000200F0        90010263
08      14      000200F5        9001037C
09      12      000200FB        900102AE

When driver gives outgoing stream:
Index   Payload CIP_Header_0    CIP_Header_1
38      12      000200BD        900104A8
39      14      000200C2        900104A8
40      12      000200C8        900104AC
41      14      000200CD        900104A9
42      12      000200D3        900104B1
43      14      000200D8        900104A8
44      12      000200DE        900104AA
45      14      000200E3        900104A9
46      14      000200E9        900104AE
47      12      000200EF        900104A8
00      14      000200F4        900104AD
01      12      000200FA        900104A7
02      14      000200FF        900104A9
03      12      00020005        900104A9
04      14      0002000A        900104B1
05      12      00020010        900104AA
06      14      00020015        900104AD
07      12      0002001B        900104A7
08      14      00020020        900104AC
09      12      00020026        900104A7
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Some devices produced by Behringer/Mackie are based on OXFW970/971. This
commit adds support for them. Additionally, this commit changes the way to
name card with some information in config rom.

Ids of some Mackie(Loud) models are not identified, therefore this commit
applies name detection for these models.

The devices support capture/playback of PCM-samples and some of them
supports capture/playback of MIDI messages. These functionalities are
implemented by followed commits.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In past commit, this driver can keep stream formations for each sampling
rate. So its stream functionality can decide stream formations with given
some parameters.

This commit moves related codes from PCM functionality to stream
functionality. Furthermore, to set stream format correctly, this commit
uses AV/C Stream Format Information command instead of AV/C Input/Output
Plug Signal Format command.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This commit adds proc interface to get information about stream
formation. This commit also adds snd_oxfw_stream_get_current_formation()
to get current stream formation.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In previous commit, this driver can get to know stream formations at
each supported sampling rates. This commit uses it to make PCM
rules/constraints and obsoletes hard-coded rules/constraints.

For this purpose, this commit adds 'struct snd_oxfw_stream_formation' and
snd_oxfw_stream_parse_format() to parse data channel formation of data
block.

According to datasheet of OXFW970/971, they support 32.0kHz to 196.0kHz.

As long as developers investigate, some devices are confirmed to have
several formats for the same sampling rate.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

OXFW970/971 may supports AV/C Stream Format Information Specification 1.1
Working Draft (Apr 2005, 1394TA). By using this command, drivers can get to know
stream formations which device supports.

This commit adds 'EXTENDED STREAM FORMAT INFORMATION' command. This command
has two subfunctions, 'SINGLE' and 'LIST'. Drivers can use 'SINGLE' subfunction
to know/set current formation of AMDTP stream, Drivers can use 'LIST'
subfunction to know an available formation of AMDTP stream in a certain sampling
rate.

But some devices don't implement the 'LIST' subfunction. So this commit uses
an assumption that 'if they don't implement it, they don't change stream
formation depending on current each sampling rate'. With this assumption, this
driver generates formations for such devices by:
 1.getting current formation by SINGLE subfunction
 2.getting supported sampling rates
 3.applying current formation for all of supported sampling rates

Followed commit implements a parser of this format information.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This is a preparation for more models. In following commit, members
of 'struct snd_card' related to name becomes to consists of vendor and
model strings in device's config-rom.

Current supported devices also has strings in their config rom, but the
strings are too long to name sound card, thus this driver still keep
hard-coded vendor and model names for them.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This commit adds a support for MIDI capture/playback

When MIDI substrams already start streaming and PCM substreams are going to
join at different sampling rate, streams are stopped once. Then sampling rate
is changed and streams are restarted.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This commit adds a support for capturing PCM samples.

When opposite PCM substream is already running, available sampling rate is
limited at current one.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>