The "chip" can't be NULL in hda_tegra_runtime_resume() because code would
crash otherwise. Let's remove the unnecessary check in order to clean up
code a tad.

Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works
Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested
Suggested-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20210120003154.26749-4-digetx@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Reset hardware on RPM-resume in order to bring it into a predictable
state.

Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works
Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested
Tested-by: Nicolas Chauvet <kwizart@gmail.com> # TK1 boot-tested
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20210120003154.26749-3-digetx@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Use clk_bulk helpers to make code cleaner. Note that this patch changed
the order in which clocks are enabled to make code look nicer, but this
doesn't matter in terms of hardware.

Tested-by: Peter Geis <pgwipeout@gmail.com> # Ouya T30 audio works
Tested-by: Matt Merhar <mattmerhar@protonmail.com> # Ouya T30 boot-tested
Tested-by: Nicolas Chauvet <kwizart@gmail.com> # TK1 boot-tested
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20210120003154.26749-2-digetx@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Drivers in ALSA firewire stack supports eventing to userspace
applications via ALSA hwdep interface. All of the drivers supports stream
lock events. Some of them supports their unique events according to
specification of target device.

ALSA bebob driver supports the stream lock event only. In the case, it's
enough to check condition only in loop with process blocking. However,
current implementation check it again after breaking the loop.

This commit removes the redundant check.

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Reported-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210125140208.26318-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Smatch complains that "count" is not clamped when "ff->dev_lock_changed"
and it leads to an information leak.  Fortunately, that's not actually
possible and the condition can be deleted.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/YA6n6I8EcNAO5ZFs@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Smatch complains that "count" isn't clamped properly and
"oxfw->dev_lock_changed" is false then it leads to an information
leak.  But it turns out that "oxfw->dev_lock_changed" is always
set and the condition can be removed.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/YA6ntkBxT/4DJ4YK@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Fix the following coccicheck warnings:

./sound/pci/hda/hda_intel.c:2309:3-23: WARNING: Assignment of
0/1 to bool variable.

Reported-by: Abaci Robot <abaci@linux.alibaba.com>
Signed-off-by: Jiapeng Zhong <abaci-bugfix@linux.alibaba.com>
Link: https://lore.kernel.org/r/1611559047-106928-1-git-send-email-abaci-bugfix@linux.alibaba.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Fix the following coccicheck warnings:

./sound/pci/hda/patch_conexant.c:570:2-20: WARNING:
Assignment of 0/1 to bool variable.

Reported-by: Abaci Robot <abaci@linux.alibaba.com>
Signed-off-by: Jiapeng Zhong <abaci-bugfix@linux.alibaba.com>
Link: https://lore.kernel.org/r/1610958469-65856-1-git-send-email-abaci-bugfix@linux.alibaba.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

At probing a UAC2/UAC3 device like NUX MG-300 USB interface, we get
error messages "RANGE setting not yet supported".  It comes the place
where the driver tries to determine the resolution of mixer volumes
via SET_CUR_RES and GET_CUR_RES verbs.  Those verbs aren't supported
on UAC2 and UAC3, hence the driver warns like the above.  Although the
driver handles this error and works as expected, it's still ugly to
show such errors unnecessarily.

This patch papers over the errors by applying the resolution detection
only for UAC1 and skipping it for UAC2/UAC3.

Reported-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210120213932.1971-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The current USB-audio driver gets an error at probing NUX MG-300 about
parsing the clocks.  This is because the firmware doesn't return the
proper connection of the clock selector that is connected to a single
clock; it's likely that the firmware was lazy^w optimized and the
inquiry wasn't handled.  Actually it makes little sense to inquire and
set up the single connection explicitly.

This patch fixes the issue by simply skipping the clock selector
inquiry if it's a single connection.

Reported-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20210120213932.1971-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This adds the Pioneer DJ DJM-750 to the quirks table and ensures
skip_pioneer_sync_ep() is (also) called: this device uses the vendor
ID of 0x08e4 (I'm not sure why they use multiple vendor IDs but many
just like to be awkward it seems).

Playback on all 8 channels works. I'll likely keep this working in the
future and submit futher patches and improvements as necessary.

Signed-off-by: Olivia Mackintosh <livvy@base.nu>
Link: https://lore.kernel.org/r/20210118130621.77miiie47wp7mump@base.nu
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Back-merge of 5.11 devel branch for more works on USB-audio.

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

For addressing the regression on Pioneer devices, we recently
corrected the quirk code to enable the implicit feedback mode on those
devices properly.  However, the devices still showed problems with the
full duplex operations with JACK, and after debug sessions, we figured
out that the older kernels that had worked with JACK also didn't use
the implicit feedback mode at all although they had the quirk code to
enable it; instead, the old code worked just to skip the normal sync
endpoint setup that would have been detected without it.  IOW, what
broke without the implicit-fb quirk in the past was the application of
the normal sync endpoint that is actually the capture data endpoint on
these devices.

This patch covers the overseen piece: it modifies the quirk code again
not to enable the implicit feedback mode but just to make the driver
skipping the sync endpoint detection.  This made the driver working
with JACK full-duplex mode again.

Still it's not quite clear why the implicit feedback doesn't work on
those devices yet; maybe it's about some issues in the URB setup.  But
at least, with this patch, the driver should work in the level of the
older kernels again.

Fixes: 167c9dc8

 ("ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices")
Link: https://lore.kernel.org/r/20210118075816.25068-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The UAC2/3 sample rate setup is based on the clock node, which is
usually shared in the interface, and can't be re-setup without
deselecting the interface once, and that's how the current code
behaves.  OTOH, the sample rate setup of UAC1 is per endpoint, hence
we basically need to call for each endpoint usage even if those share
the same interface.

This patch fixes the behavior of UAC1 to call always
snd_usb_init_sample_rate() in snd_usb_endpoint_configure().

Fixes: bf6313a0

 ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210118075816.25068-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The current sample rate setup function for UAC1 assumes only the first
endpoint retrieved from the interface:altset pair, but the rate set up
may be needed also for the secondary endpoint.  Also, retrieving the
endpoint number from the interface descriptor is redundant; we have
already the target endpoint in the given audioformat object.

This patch simplifies the code and corrects the target endpoint as
described in the above.  It simply refers to fmt->endpoint directly.

Also, this patch drops the pioneer_djm_set_format_quirk() that is
caleld from snd_usb_set_format_quirk(); this function does the sample
rate setup but for the capture endpoint (0x82), and that's exactly
what the change above fixes.

Link: https://lore.kernel.org/r/20210118075816.25068-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

LPE driver still has explicit calls of dma_set_mask() and
dma_set_coherent_mask().

Let's simplify with dma_set_mask_and_coherent().

Link: https://lore.kernel.org/r/20210114125412.993-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Many PCI drivers still have two explicit calls of dma_set_mask() and
dma_set_coherent_mask().

Let's simplify with dma_set_mask_and_coherent().

Link: https://lore.kernel.org/r/20210114125412.993-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add HD Audio Device PCI ID for the Intel Cometlake-R platform

Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Kai-Chuan Hsieh <kaichuan.hsieh@canonical.com>
Link: https://lore.kernel.org/r/20210115031515.13100-1-kaichuan.hsieh@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

I-ONIX FW810s was shipped in Lexicon brand of HARMAN International
industries, Inc 2009. The model uses TCD2220 ASIC as its communication
engine. TCAT general protocol is supported, its extension isn't.

This patch adds support for the model with hard-coded stream formats.

$ python3 ~/git/linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom
               ROM header and bus information block
               -----------------------------------------------------------------
400  04042b91  bus_info_length 4, crc_length 4, crc 11153
404  31333934  bus_name "1394"
408  e0008102  irmc 1, cmc 1, isc 1, bmc 0, cyc_clk_acc 0, max_rec 8 (512)
40c  000fd720  company_id 000fd7     |
410  007d7ecf  device_id 20007d7ecf  | EUI-64 000fd720007d7ecf

               root directory
               -----------------------------------------------------------------
414  00064c2d  directory_length 6, crc 19501
418  03000fd7  vendor
41c  8100000a  --> descriptor leaf at 444
420  17000001  model
424  8100000d  --> descriptor leaf at 458
428  0c0087c0  node capabilities per IEEE 1394
42c  d1000001  --> unit directory at 430

               unit directory at 430
               -----------------------------------------------------------------
430  000438f2  directory_length 4, crc 14578
434  12000fd7  specifier id
438  13000001  version
43c  17000001  model
440  8100000d  --> descriptor leaf at 474

               descriptor leaf at 444
               -----------------------------------------------------------------
444  000489d5  leaf_length 4, crc 35285
448  00000000  textual descriptor
44c  00000000  minimal ASCII
450  4c657869  "Lexi"
454  636f6e00  "con"

               descriptor leaf at 458
               -----------------------------------------------------------------
458  0006594b  leaf_length 6, crc 22859
45c  00000000  textual descriptor
460  00000000  minimal ASCII
464  492d4f4e  "I-ON"
468  49585f46  "IX_F"
46c  57383130  "W810"
470  53000000  "S"

               descriptor leaf at 474
               -----------------------------------------------------------------
474  0006594b  leaf_length 6, crc 22859
478  00000000  textual descriptor
47c  00000000  minimal ASCII
480  492d4f4e  "I-ON"
484  49585f46  "IX_F"
488  57383130  "W810"
48c  53000000  "S"

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20210115035623.148580-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The last remaining usage of strlcpy() in USB-audio driver is the setup
of the card longname string.  Basically we need to know whether any
non-empty string is set or not, and no real length is needed.
Refactor the code and use strscpy() instead.  After this change,
strlcpy() is gone from all sound/* code.

Link: https://lore.kernel.org/r/20210115100437.20906-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

USB-audio driver still contains two calls of strlcpy() because the
return size is evaluated.  Basically it just checks whether the string
is copied or not, but since strcpy() may return a negative error code,
we should check the negative value and treat as filled.

Link: https://lore.kernel.org/r/20210115095758.19707-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

snd_seq_oss_synth_make_info() didn't check the error code from
snd_seq_oss_midi_make_info(), and this leads to the call of strlcpy()
with the uninitialized string as the source, which may lead to the
access over the limit.

Add the proper error check for avoiding the failure.

Reported-by:  <syzbot+e42504ff21cff05a595f@syzkaller.appspotmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210115093428.15882-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

It turned out that VIA codecs also mute the sound in the lowest mixer
level.  Turn on the dac_min_mute flag to indicate the mute-as-minimum
in TLV like already done in Conexant and IDT codecs.

BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210559
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210114072453.11379-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The Acer Apire E5-575T laptop with codec ALC255 has a terrible
background noise comes from internal mic capture. And the jack
sensing dose not work for headset like some other Acer laptops.

This patch limits the internal mic boost on top of the existing
ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk for Acer Aspire E5-575T.

Signed-off-by: Chris Chiu <chiu@endlessos.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210114082728.74729-1-chiu@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Since the commit 5a6c3e11 ("ALSA: usb-audio: Add hw constraint for
implicit fb sync"), we apply the hw constraints for the implicit
feedback sync to make the secondary open aligned with the already
opened stream setup.  This change assumed that the secondary open is
performed after the first stream has been already set up, and adds the
hw constraints to sync with the first stream's parameters only when
the EP setup for the first stream was confirmed at the open time.
However, most of applications handling the full-duplex operations do
open both playback and capture streams at first, then set up both
streams.  This results in skipping the additional hw constraints since
the counter-part stream hasn't been set up yet at the open of the
second stream, and it eventually leads to "incompatible EP" error in
the end.

This patch corrects the behavior by always applying the hw constraints
for the implicit fb sync.  The hw constraint rules are defined so that
they check the sync EP dynamically at each invocation, instead.  This
covers the concurrent stream setups better and lets the hw refine
calls resolving to the right configuration.

Also this patch corrects a minor error that has existed in the debug
print that isn't built as default.

Fixes: 5a6c3e11

 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Link: https://lore.kernel.org/r/20210111081611.12790-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add HD Audio PCI ID and HDMI codec vendor ID for Intel AlderLake-P.

Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210113155629.4097057-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

We are able to power down the GPU and audio via the GPU driver
so flag these asics as supporting runtime pm.

Reviewed-by: Evan Quan <evan.quan@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Link: https://lore.kernel.org/r/20210105175245.963451-1-alexander.deucher@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

As snd_fw_async_midi_port.consume_bytes is unsigned int, and
NSEC_PER_SEC is 1000000000L, the second multiplication in

    port->consume_bytes * 8 * NSEC_PER_SEC / 31250

always overflows on 32-bit platforms, truncating the result.  Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.

Note that this assumes port->consume_bytes <= 16777.

Fixes: 531f4718

 ("ALSA: firewire-lib/firewire-tascam: localize async midi port")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-3-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>

As snd_ff.rx_bytes[] is unsigned int, and NSEC_PER_SEC is 1000000000L,
the second multiplication in

    ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250

always overflows on 32-bit platforms, truncating the result.  Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.

Note that this assumes ff->rx_bytes[port] <= 16777.

Fixes: 19174295

 ("ALSA: fireface: add transaction support")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-2-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Currently hda on tegra30 fails to open a stream with an input/output error.

For example:
speaker-test -Dhw:0,3 -c 2

speaker-test 1.2.2

Playback device is hw:0,3
Stream parameters are 48000Hz, S16_LE, 2 channels
Using 16 octaves of pink noise
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 64 to 16384
Period size range from 32 to 8192
Using max buffer size 16384
Periods = 4
was set period_size = 4096
was set buffer_size = 16384
 0 - Front Left
Write error: -5,Input/output error
xrun_recovery failed: -5,Input/output error
Transfer failed: Input/output error

The tegra-hda device was introduced in tegra30 but only utilized in
tegra124 until recent chips. Tegra210/186 work only due to a hardware
change. For this reason it is unknown when this issue first manifested.
Discussions with the hardware team show this applies to all current tegra
chips. It has been resolved in the tegra234, which does not have hda
support at this time.

The explanation from the hardware team is this:
Below is the striping formula referenced from HD audio spec.
   { ((num_channels * bits_per_sample) / number of SDOs) >= 8 }

The current issue is seen because Tegra HW has a problem with boundary
condition (= 8) for striping. The reason why it is not seen on
Tegra210/Tegra186 is because it uses max 2SDO lines. Max SDO lines is
read from GCAP register.

For the given stream (channels = 2, bps = 16);
ratio = (channels * bps) / NSDO = 32 / NSDO;

On Tegra30,      ratio = 32/4 = 8  (FAIL)
On Tegra210/186, ratio = 32/2 = 16 (PASS)
On Tegra194,     ratio = 32/4 = 8  (FAIL) ==> Earlier workaround was
applied for it

If Tegra210/186 is forced to use 4SDO, it fails there as well. So the
behavior is consistent across all these chips.

Applying the fix in [1] universally resolves this issue on tegra30-hda.
Tested on the Ouya game console and the tf201 tablet.

[1] commit 60019d8c

 ("ALSA: hda/tegra: workaround playback failure on
Tegra194")

Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Tested-by: Ion Agorria <ion@agorria.com>
Reviewed-by: Sameer Pujar <spujar@nvidia.com>
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Peter Geis <pgwipeout@gmail.com>
Link: https://lore.kernel.org/r/20210108135913.2421585-3-pgwipeout@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Current implementation defaults the hda clocks to clk_m. This causes hda
to run too slow to operate correctly. Fix this by defaulting to pll_p and
setting the frequency to the correct rate.

This matches upstream t124 and downstream t30.

Acked-by: Jon Hunter <jonathanh@nvidia.com>
Tested-by: Ion Agorria <ion@agorria.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Peter Geis <pgwipeout@gmail.com>
Link: https://lore.kernel.org/r/20210108135913.2421585-2-pgwipeout@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

MIXART.txt has been converted to ReST and renamed. Fix the reference
in alsa-configuration.rst.

Fixes: 3d8e8186

 ("ALSA: doc: ReSTize MIXART.txt")
Signed-off-by: Jonathan Neuschäfer <j.neuschaefer@gmx.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210101221942.1068388-1-j.neuschaefer@gmx.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Pioneer devices have both playback and capture streams sharing the
same iface/altsetting, and those need to be paired as implicit
feedback.  Instead of a half-baked (and broken) static quirk entry,
set up more generically for those devices by checking the number of
endpoints and the attribute of the secondary EP.

Fixes: bf6313a0

 ("ALSA: usb-audio: Refactor endpoint management")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There are devices that have multiple endpoints sharing the same
iface/altset not only for sync but also for the actual streams, and
the audioformat for such an endpoint needs to be handled with the
proper endpoint index; otherwise it confuses the endpoint management.

This patch extends the audioformat to annotate the endpoint index, and
put the proper ep_idx=1 to Pioneer device quirk entries accordingly.

Fixes: bf6313a0

 ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The current endpoint handling assumed (more or less) a unique 1:1
relation between the endpoint and the iface/altset.  The exception was
the sync EP without the implicit feedback which has usually the
secondary EP of the same altset.  This works fine for most devices,
but it turned out that some unusual devices like Pinoeer's ones have
both playback and capture endpoints in the same iface/altsetting and
use both for the implicit feedback mode.  For handling such a case, we
need to extend the endpoint management to take the shared interface
into account.

This patch does that: it adds a new object snd_usb_iface_ref for
managing the reference counts of the each USB interface that is used
by each endpoint.  The interface setup is performed only once for the
(sharing) endpoints, and the doubly initialization is avoided.

Along with this, the resource release of endpoints and interface
refcounts are put into a single function, snd_usb_endpoint_free_all()
instead of looping in the caller side.

Fixes: bf6313a0

 ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The implicit feedback mode needs to handle two endpoints and the
choice of the audioformat object for the sync EP is important since
this determines the compatibility of the hw_params.  The current code
uses the same audioformat object if both the main EP and the sync EP
point to the same iface/altsetting.  This was done in consideration of
the non-implicit-fb sync EP handling, and it doesn't match well with
the cases where actually to endpoints are defined in the sameiface /
altsetting like a few Pioneer devices.

Modify snd_usb_find_implicit_fb_sync_format() to pick up the
audioformat that is assigned in the counter-part substreams primarily,
so that the actual capture stream can be opened properly.  We keep the
same audioformat object only as a fallback in case nothing found,
though.

Fixes: 9fddc15e

 ("ALSA: usb-audio: Factor out the implicit feedback quirk code")
Link: https://lore.kernel.org/r/20210108075219.21463-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The recent change in the endpoint management moved the endpoint object
creation from the stream open time to the parser of the audio
descriptor.  It works fine for the standard audio, but it overlooked
the other places that create audio streams via quirks
(QUIRK_AUDIO_FIXED_ENDPOINT) like the reported a few Pioneer devices;
those call snd_usb_add_audio_stream() manually, hence they miss the
endpoints, eventually resulting in the error at opening streams.
Moreover, now the sync EP setup was moved to the explicit call of
snd_usb_audioformat_set_sync_ep(), and this needs to be added for
those places, too.

This patch addresses those regressions for quirks.  It adds a local
helper function add_audio_stream_from_fixed_fmt(), which does the all
needed tasks, and replaces the calls of snd_usb_add_audio_stream()
with this new function.

Fixes: 54cb3190

 ("ALSA: usb-audio: Create endpoint objects at parsing phase")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>

strlcpy is deprecated.  see: Documentation/process/deprecated.rst

Change the calls that do not use the strlcpy return value to the
preferred strscpy.

Done with cocci script:

@@
expression e1, e2, e3;
@@

-	strlcpy(
+	strscpy(
	e1, e2, e3);

This cocci script leaves the instances where the return value is
used unchanged.

After this patch, sound/ has 3 uses of strlcpy() that need to be
manually inspected for conversion and changed one day.

$ git grep -w strlcpy sound/
sound/usb/card.c:               len = strlcpy(card->longname, s, sizeof(card->longname));
sound/usb/mixer.c:      return strlcpy(buf, p->name, buflen);
sound/usb/mixer.c:                      return strlcpy(buf, p->names[index], buflen);

Miscellenea:

o Remove trailing whitespace in conversion of sound/core/hwdep.c

Link: https://lore.kernel.org/lkml/CAHk-=wgfRnXz0W3D37d01q3JFkr_i_uTL=V6A6G1oUZcprmknw@mail.gmail.com/

Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/22b393d1790bb268769d0bab7bacf0866dcb0c14.camel@perches.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Back-merge of 5.11-devel branch for syncing the result changes.

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

Merge tag 'asoc-fix-v5.11-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v5.11

A collection of mostly driver specific fixes, plus a maintainership
update for TI and a fix for DAPM driver removal paths.