Commit Graph

37721 Commits

Author SHA1 Message Date
Takashi Iwai
85db1cfb13 ALSA: usb-audio: Fix "RANGE setting not yet supported" errors
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>
2021-01-21 12:47:19 +01:00
Takashi Iwai
086b957cc1 ALSA: usb-audio: Skip the clock selector inquiry for single connections
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>
2021-01-21 12:47:06 +01:00
Olivia Mackintosh
b952ac76a2 ALSA: usb-audio: Add support for Pioneer DJM-750
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>
2021-01-18 17:39:42 +01:00
Takashi Iwai
c09e28cd12 Merge branch 'for-linus' into for-next
Back-merge of 5.11 devel branch for more works on USB-audio.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2021-01-18 17:39:23 +01:00
Takashi Iwai
532a208ad6 ALSA: usb-audio: Avoid implicit feedback on Pioneer devices
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: 167c9dc84e ("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>
2021-01-18 09:38:13 +01:00
Takashi Iwai
3784d449d7 ALSA: usb-audio: Set sample rate for all sharing EPs on UAC1
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: bf6313a0ff ("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>
2021-01-18 09:38:03 +01:00
Takashi Iwai
87cb9af9f8 ALSA: usb-audio: Fix UAC1 rate setup for secondary endpoints
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>
2021-01-18 09:37:53 +01:00
Takashi Iwai
915183b622 ALSA: x86: Simplify with dma_set_mask_and_coherent()
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>
2021-01-15 11:17:19 +01:00
Takashi Iwai
669f65eaeb ALSA: pci: Simplify with dma_set_mask_and_coherent()
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>
2021-01-15 11:17:00 +01:00
Kai-Chuan Hsieh
f84d3a1ec3 ALSA: hda: Add Cometlake-R PCI ID
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>
2021-01-15 11:15:53 +01:00
Takashi Sakamoto
9a08676fc5 ALSA: dice: add support for Lexicon I-ONIX FW810s
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>
2021-01-15 11:13:33 +01:00
Takashi Iwai
42ef170d57 ALSA: usb-audio: Convert the last strlcpy() usage
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>
2021-01-15 11:05:26 +01:00
Takashi Iwai
e8afdfdc81 ALSA: usb-audio: Convert remaining strlcpy() to strscpy()
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>
2021-01-15 11:05:20 +01:00
Takashi Iwai
217bfbb8b0 ALSA: seq: oss: Fix missing error check in snd_seq_oss_synth_make_info()
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>
2021-01-15 10:35:17 +01:00
Takashi Iwai
67ea698c39 ALSA: hda/via: Add minimum mute flag
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>
2021-01-14 15:11:15 +01:00
Chris Chiu
495dc7637c ALSA: hda/realtek - Limit int mic boost on Acer Aspire E5-575T
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>
2021-01-14 12:58:02 +01:00
Takashi Iwai
e4ea77f8e5 ALSA: usb-audio: Always apply the hw constraints for implicit fb sync
Since the commit 5a6c3e11c9 ("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: 5a6c3e11c9 ("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>
2021-01-13 22:25:27 +01:00
Kai Vehmanen
5e941fc033 ALSA: hda: Add AlderLake-P PCI ID and HDMI codec vid
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>
2021-01-13 17:02:43 +01:00
Alex Deucher
20c7842ed8 ALSA: hda/hdmi - enable runtime pm for CI AMD display audio
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>
2021-01-12 16:06:01 +01:00
Geert Uytterhoeven
9f65df9c58 ALSA: firewire-tascam: Fix integer overflow in midi_port_work()
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: 531f471834 ("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>
2021-01-12 14:57:38 +01:00
Geert Uytterhoeven
e7c22eeaff ALSA: fireface: Fix integer overflow in transmit_midi_msg()
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: 1917429578 ("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>
2021-01-12 14:57:29 +01:00
Peter Geis
615d435400 ALSA: hda/tegra: fix tegra-hda on tegra30 soc
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 60019d8c65 ("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>
2021-01-12 14:44:47 +01:00
Takashi Iwai
167c9dc84e ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices
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: bf6313a0ff ("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>
2021-01-08 23:43:39 +01:00
Takashi Iwai
eae4d054f9 ALSA: usb-audio: Annotate the endpoint index in audioformat
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: bf6313a0ff ("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>
2021-01-08 23:43:38 +01:00
Takashi Iwai
00272c6182 ALSA: usb-audio: Avoid unnecessary interface re-setup
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: bf6313a0ff ("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>
2021-01-08 23:43:37 +01:00
Takashi Iwai
5d15f1eb45 ALSA: usb-audio: Choose audioformat of a counter-part substream
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: 9fddc15e80 ("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>
2021-01-08 23:43:36 +01:00
Takashi Iwai
b2345a8a43 ALSA: usb-audio: Fix the missing endpoints creations for quirks
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: 54cb31901b ("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>
2021-01-08 23:43:35 +01:00
Joe Perches
75b1a8f9d6 ALSA: Convert strlcpy to strscpy when return value is unused
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>
2021-01-08 09:30:05 +01:00
Takashi Iwai
6dcb8bf9a1 Merge branch 'for-linus' into for-next
Back-merge of 5.11-devel branch for syncing the result changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2021-01-07 18:52:58 +01:00
Takashi Iwai
7b62275507 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.
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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.
2021-01-07 18:51:39 +01:00
Jeremy Szu
91bc156817 ALSA: hda/realtek: fix right sounds and mute/micmute LEDs for HP machines
* The HP ZBook Fury 15/17 G7 Mobile Workstation are using ALC285 codec
   which is using 0x04 to control mute LED and 0x01 to control micmute LED.

 * The right channel speaker is no sound and it needs to expose GPIO1 for
   initialing AMP.

Add quirks to support them.

Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210106130549.100532-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2021-01-07 11:40:39 +01:00
Takashi Iwai
4bfd6247fa ALSA: hda/via: Fix runtime PM for Clevo W35xSS
Clevo W35xSS_370SS with VIA codec has had the runtime PM problem that
looses the power state of some nodes after the runtime resume.  This
was worked around by disabling the default runtime PM via a denylist
entry.  Since 5.10.x made the runtime PM applied (casually) even
though it's disabled in the denylist, this problem was revisited.  The
result was that disabling power_save_node feature suffices for the
runtime PM problem.

This patch implements the disablement of power_save_node feature in
VIA codec for the device.  It also drops the former denylist entry,
too, as the runtime PM should work in the codec side properly now.

Fixes: b529ef2464 ("ALSA: hda: Add Clevo W35xSS_370SS to the power_save blacklist")
Reported-by: Christian Labisch <clnetbox@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210104153046.19993-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2021-01-04 16:34:43 +01:00
Timon Reinold
b0e1306302 ALSA: usb-audio: Add quirk for RC-505
BOSS RC-505 (shown by lsusb as "Roland Corp. RC-505") does require the
same quirk as these other BOSS devices.

Without this quirk it is neither possible to capture audio from nor to
write audio to the RC-505. Both just result in an empty audio
stream. With these changes both capture and playback seem to work
quite fine. MIDI funtionality was not tested.

Tested-by: Harry Reinold <harry.reinold@posteo.de>
Signed-off-by: Timon Reinold <tirei@agon.one>
Link: https://lore.kernel.org/r/20210102210835.21268-1-tirei@agon.one
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2021-01-03 08:58:39 +01:00
Takashi Iwai
3d5c5fdcee ALSA: hda/hdmi: Fix incorrect mutex unlock in silent_stream_disable()
The silent_stream_disable() function introduced by the commit
b1a5039759 ("ALSA: hda/hdmi: fix silent stream for first playback to
DP") takes the per_pin->lock mutex, but it unlocks the wrong one,
spec->pcm_lock, which causes a deadlock.  This patch corrects it.

Fixes: b1a5039759 ("ALSA: hda/hdmi: fix silent stream for first playback to DP")
Reported-by: Jan Alexander Steffens (heftig) <heftig@archlinux.org>
Cc: <stable@vger.kernel.org>
Acked-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210101083852.12094-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2021-01-01 21:08:53 +01:00
Kai-Heng Feng
a598098cc9 ALSA: hda/realtek: Enable mute and micmute LED on HP EliteBook 850 G7
HP EliteBook 850 G7 uses the same GPIO pins as ALC285_FIXUP_HP_GPIO_LED
to enable mute and micmute LED. So apply the quirk to enable the LEDs.

Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201230125636.45028-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-31 11:59:17 +01:00
PeiSen Hou
ce2e79b223 ALSA: hda/realtek: Add two "Intel Reference board" SSID in the ALC256.
Add two "Intel Reference boad" SSID in the alc256.
Enable "power saving mode" and Enable "headset jack mode".

Signed-off-by: PeiSen Hou <pshou@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/5978d2267f034c28973d117925ec9c63@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-31 11:57:48 +01:00
Manuel Jiménez
484229585a ALSA: hda/realtek: Add mute LED quirk for more HP laptops
HP Pavilion 13-bb0000 (SSID 103c:87c8) needs the same
quirk as other models with ALC287.

Signed-off-by: Manuel Jiménez <mjbfm99@me.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/X+s/gKNydVrI6nLj@HP-Pavilion-13
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-31 11:55:34 +01:00
bo liu
744a11abc5 ALSA: hda/conexant: add a new hda codec CX11970
The current kernel does not support the cx11970 codec chip.
Add a codec configuration item to kernel.

[ Minor coding style fix by tiwai ]

Signed-off-by: bo liu <bo.liu@senarytech.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201229035226.62120-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-31 11:40:21 +01:00
Takashi Iwai
3deba4d8f0 ALSA: usb-audio: Add quirk for BOSS AD-10
BOSS AD-10 requires the very same quirk like other BOSS devices to
enable the special implicit feedback mode.

Reported-and-tested-by: Martin Passing <martin@passing.name>
Link: https://lore.kernel.org/r/20201229083428.20467-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-29 09:34:59 +01:00
Jaroslav Kysela
1f092d1c88
ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad X395
The ThinkPad X395 latop does not have the internal digital
microphone connected to the AMD's ACP bridge, but it's advertised
via BIOS. The internal microphone is connected to the HDA codec.

Use DMI to block the microphone PCM device for this platform.

BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20201227164109.269973-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
2020-12-28 14:22:44 +00:00
Ravulapati Vishnu vardhan rao
a523e1538f
ASoC: amd: Replacing MSI with Legacy IRQ model
When we try to play and capture simultaneously we see that
interrupts are genrated but our handler is not being acknowledged,
After investigating further more in detail on this issue we found
that IRQ delivery via MSI from the ACP IP is unreliable and so sometimes
interrupt generated will not be acknowledged so MSI model shouldn't be used
and using legacy IRQs will resolve interrupt handling issue.

This patch replaces MSI interrupt handling with legacy IRQ model.

Issue can be reproduced easily by running below python script:

import subprocess
import time
import threading

def do2():
  cmd = 'aplay -f dat -D hw:2,1 /dev/zero -d 1'
    subprocess.call(cmd, stdin=subprocess.PIPE,
			stderr=subprocess.PIPE, shell=True)
    print('Play Done')

def run():
	for i in range(1000):
		do2()

def do(i):
    cmd = 'arecord -f dat -D hw:2,2 /dev/null -d 1'
    subprocess.call(cmd, stdout=subprocess.PIPE,
			stderr=subprocess.PIPE, shell=True)
    print(datetime.datetime.now(), i)

t = threading.Thread(target=run)
t.start()
for i in range(1000):
	do(i)

t.join()

After applying this patch issue is resolved.

Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20201222115929.11222-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2020-12-28 14:22:27 +00:00
Jaroslav Kysela
275565997a
ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad E14 Gen 2
The ThinkPad E14 Gen 2 latop does not have the internal digital
microphone connected to the AMD's ACP bridge, but it's advertised
via BIOS. The internal microphone is connected to the HDA codec.

Use DMI to block the microphone PCM device for this platform.

Reported-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20201227164037.269893-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
2020-12-28 14:20:43 +00:00
Mark Brown
f81325a05e Linux 5.11-rc1
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Merge tag 'v5.11-rc1' into asoc-5.11

Linux 5.11-rc1
2020-12-28 14:16:53 +00:00
Lars-Peter Clausen
64062869f0 ALSA: maestro: Use DIV_ROUND_CLOSEST() instead of open-coding it
Use DIV_ROUND_CLOSEST() instead of open-coding it. This documents intent
and makes it more clear what is going on for the casual reviewer.

Generated using the following the Coccinelle semantic patch.

// <smpl>
@r1@
expression x;
constant C1;
constant C2;
@@
 ((x) + C1) / C2

@script:python@
C1 << r1.C1;
C2 << r1.C2;
@@
try:
	if int(C1) * 2 != int(C2):
		cocci.include_match(False)
except:
	cocci.include_match(False)

@@
expression r1.x;
constant r1.C1;
constant r1.C2;
@@
-(((x) + C1) / C2)
+DIV_ROUND_CLOSEST(x, C2)
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201223172229.781-18-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-25 09:11:57 +01:00
Lars-Peter Clausen
e08eaf4003 ALSA: sis7019: Use DIV_ROUND_CLOSEST() instead of open-coding it
Use DIV_ROUND_CLOSEST() instead of open-coding it. This documents intent
and makes it more clear what is going on for the casual reviewer.

Generated using the following the Coccinelle semantic patch.

// <smpl>
@r1@
expression x;
constant C1;
constant C2;
@@
 ((x) + C1) / C2

@script:python@
C1 << r1.C1;
C2 << r1.C2;
@@
try:
	if int(C1) * 2 != int(C2):
		cocci.include_match(False)
except:
	cocci.include_match(False)

@@
expression r1.x;
constant r1.C1;
constant r1.C2;
@@
-(((x) + C1) / C2)
+DIV_ROUND_CLOSEST(x, C2)
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201223172229.781-17-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-25 09:11:56 +01:00
Lars-Peter Clausen
db43394690 ALSA: ens1370: Use DIV_ROUND_CLOSEST() instead of open-coding it
Use DIV_ROUND_CLOSEST() instead of open-coding it. This documents intent
and makes it more clear what is going on for the casual reviewer.

Generated using the following the Coccinelle semantic patch.

// <smpl>
@r1@
expression x;
constant C1;
constant C2;
@@
 ((x) + C1) / C2

@script:python@
C1 << r1.C1;
C2 << r1.C2;
@@
try:
	if int(C1) * 2 != int(C2):
		cocci.include_match(False)
except:
	cocci.include_match(False)

@@
expression r1.x;
constant r1.C1;
constant r1.C2;
@@
-(((x) + C1) / C2)
+DIV_ROUND_CLOSEST(x, C2)
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201223172229.781-16-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-25 09:11:56 +01:00
Lars-Peter Clausen
a8667a3f15 ALSA: trident: Use DIV_ROUND_CLOSEST() instead of open-coding it
Use DIV_ROUND_CLOSEST() instead of open-coding it. This documents intent
and makes it more clear what is going on for the casual reviewer.

Generated using the following the Coccinelle semantic patch.

// <smpl>
@@
expression x, y;
@@
-((x) + ((y) / 2)) / (y)
+DIV_ROUND_CLOSEST(x, y)

@r1@
expression x;
constant C1;
constant C2;
@@
 ((x) + C1) / C2

@script:python@
C1 << r1.C1;
C2 << r1.C2;
@@
try:
	if int(C1) * 2 != int(C2):
		cocci.include_match(False)
except:
	cocci.include_match(False)

@@
expression r1.x;
constant r1.C1;
constant r1.C2;
@@
-(((x) + C1) / C2)
+DIV_ROUND_CLOSEST(x, C2)
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201223172229.781-15-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-25 09:11:55 +01:00
Lars-Peter Clausen
4743feb9f4 ALSA: sonicvibes: Use DIV_ROUND_CLOSEST() instead of open-coding it
Use DIV_ROUND_CLOSEST() instead of open-coding it. This documents intent
and makes it more clear what is going on for the casual reviewer.

Generated using the following the Coccinelle semantic patch.

// <smpl>
@@
expression x, y;
@@
-((x) + ((y) / 2)) / (y)
+DIV_ROUND_CLOSEST(x, y)
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201223172229.781-14-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-25 09:11:55 +01:00
Lars-Peter Clausen
6b5edf1dcc ALSA: oss: Use DIV_ROUND_CLOSEST() instead of open-coding it
Use DIV_ROUND_CLOSEST() instead of open-coding it. This documents intent
and makes it more clear what is going on for the casual reviewer.

Generated using the following the Coccinelle semantic patch.

// <smpl>
@@
expression x, y;
@@
-((x) + ((y) / 2)) / (y)
+DIV_ROUND_CLOSEST(x, y)
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201223172229.781-13-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-25 09:11:54 +01:00
Lars-Peter Clausen
af102a88ad ALSA: vx: Use roundup() instead of open-coding it
Use roundup() instead of open-coding it. This documents intent
and makes it more clear what is going on for the casual reviewer.

Generated using the following the Coccinelle semantic patch.

// <smpl>
@@
expression x, y;
@@
-((((x) + (y) - 1) / (y)) * y)
+roundup(x, y)
// </smpl>

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Link: https://lore.kernel.org/r/20201223172229.781-12-lars@metafoo.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2020-12-25 09:11:53 +01:00