If SND_HDA_CODEC_CA0132 is enabled, the DSP support should be enabled as
well. Disabled DSP support leads to a hanging alsa system and no sound
output on the card otherwise. Tested on:
Takashi Iwai [Wed, 25 Mar 2020 10:33:21 +0000 (11:33 +0100)]
ALSA: usb-audio: Inform devices that need delayed registration
The USB-audio driver may call snd_card_register() multiple times as
its probe function is per USB interface while some USB-audio devices
may provide multiple interfaces to assign different streams although
they belong to the same device. This works in most cases but the
registration is racy, hence it may miss the device recognition,
e.g. PA doesn't see certain devices when hotplugged.
The recent addition of the delayed registration quirk allows to sync
the registration at the last known interface, and the previous commit
added a new module option to allow the dynamic setup for that
purpose.
Now, this patch tries to find out and notifies for such devices that
require the delayed registration. It shows a message like:
Found post-registration device assignment: 1234abcd:02
If you hit this message, you can pass delayed_register module option
like:
by just copying the last shown entry. If this works, it can be added
statically in the quirk list, registration_quirks[] found at the end
of sound/usb/quirks.c.
Takashi Iwai [Wed, 25 Mar 2020 10:33:20 +0000 (11:33 +0100)]
ALSA: usb-audio: Add delayed_register option
Add a new option for specifying the quirk for delayed registration of
the certain device. A list of devices can be passed in a form
ID:IFACE,ID:IFACE,ID:IFACE,....
where ID is the 32bit hex number combo of vendor and device IDs and
IFACE is the interface number to trigger the register.
When a matching device is probed, the card registration is delayed
until the given interface is probed. It's needed for syncing the
registration until the last interface when multiple interfaces are
provided for the same card.
A slight refactoring of the registration quirk code. Now it uses the
table lookup for easy additions in future. Also the return type was
changed to bool, and got a few more comments.
Takashi Iwai [Mon, 23 Mar 2020 17:06:42 +0000 (18:06 +0100)]
ALSA: core: Add snd_device_get_state() helper
A new small helper to get the current state of the device registration
for the given object. It'll be used for USB-audio driver to check the
delayed device registrations.
Takashi Iwai [Fri, 13 Mar 2020 13:02:41 +0000 (14:02 +0100)]
ALSA: hda: Use scnprintf() for string truncation
snd_hdac_codec_modalias() truncates the string to the given size and
returns its size, but it returned a wrong size from snprintf().
snprintf() returns the would-be-output size, not the actual size.
Use scnprintf() instead to return the correct size.
Takashi Iwai [Fri, 13 Mar 2020 13:02:23 +0000 (14:02 +0100)]
ALSA: pcm: Fix superfluous snprintf() usage
show_pcm_class() returns obviously a short string that can't overflow
PAGE_SIZE. And even if it were to overflow, using snprintf() there is
just wrong, as it doesn't return the correct size.
So simplify with sprintf() instead.
Both snd_pcm_plug_client_size() and snd_pcm_plug_slave_size() do the
almost same calculations of calling src_frames() and dst_frames() in
the chain, but just to the different directions with each other.
This patch simplifies those functions. Now they return -EINVAL for
the invalid direction, but practically seen, there is no functional
changes at all.
Takashi Iwai [Mon, 9 Mar 2020 09:59:22 +0000 (10:59 +0100)]
ALSA: line6: Fix endless MIDI read loop
The MIDI input event parser of the LINE6 driver may enter into an
endless loop when the unexpected data sequence is given, as it tries
to continue the secondary bytes without termination. Also, when the
input data is too short, the parser returns a negative error, while
the caller doesn't handle it properly. This would lead to the
unexpected behavior as well.
This patch addresses those issues by checking the return value
correctly and handling the one-byte event in the parser properly.
Takashi Iwai [Mon, 9 Mar 2020 08:21:48 +0000 (09:21 +0100)]
ALSA: pcm: oss: Avoid plugin buffer overflow
Each OSS PCM plugins allocate its internal buffer per pre-calculation
of the max buffer size through the chain of plugins (calling
src_frames and dst_frames callbacks). This works for most plugins,
but the rate plugin might behave incorrectly. The calculation in the
rate plugin involves with the fractional position, i.e. it may vary
depending on the input position. Since the buffer size
pre-calculation is always done with the offset zero, it may return a
shorter size than it might be; this may result in the out-of-bound
access as spotted by fuzzer.
This patch addresses those possible buffer overflow accesses by simply
setting the upper limit per the given buffer size for each plugin
before src_frames() and after dst_frames() calls.
Takashi Iwai [Sat, 7 Mar 2020 06:24:36 +0000 (07:24 +0100)]
Merge tag 'asoc-fix-v5.6-rc4' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.6
More fixes that have arrived since the merge window, spread out all
over. There's a few things like the operation callback addition for
rt1015 and the meson reset addition which add small new bits of
functionality to fix non-working systems, they're all very small and for
parts of newly added functionality.
ALSA: firewire: use KBUILD_MODNAME for struct driver.name instead of string
KBUILD_MODNAME is available to name kernel modules according to its object
name. This commit uses the macro instead of string for name field of
struct driver since drivers in ALSA firewire stack have the same name of
each object name.
ALSA: sgio2audio: Remove usage of dropped hw_params/hw_free functions
Commit ee88f4ebe575 ("ALSA: mips: Use managed buffer allocation") removed
superfluous hw_params/hw_free callbacks, but forgot to remove them where
they were used.
Takashi Iwai [Fri, 6 Mar 2020 08:12:31 +0000 (09:12 +0100)]
ALSA: usb-audio: Fix missing braces in some struct inits
The struct s1810c_state_packet contains the array in the first field
hence zero-initialization requires a more couple of braces. Fix the
compile warning pointing it out:
sound/usb/mixer_s1810c.c: In function 'snd_sc1810c_get_status_field':
sound/usb/mixer_s1810c.c:178:9: warning: missing braces around initializer [-Wmissing-braces]
Alexander Tsoy [Sat, 29 Feb 2020 15:18:15 +0000 (18:18 +0300)]
ALSA: usb-audio: Add support for MOTU MicroBook IIc
MicroBook IIc operates in UAC2 mode by default. This patch addresses
several issues with it:
- MicroBook II and IIc shares the same USB ID. We can distinguish them
by interface class.
- MaxPacketsOnly attribute is erroneously set in endpoint descriptors.
As a result this card produces noise with all sample rates other than
96 KHz. This also causes issues like IOMMU page faults and other
problems with host controller.
- Sample rate changes takes more than 2 seconds for this device. Clock
validity request returns false during that period, so the clock validity
quirk is required.
Randy Dunlap [Wed, 26 Feb 2020 05:32:49 +0000 (21:32 -0800)]
ALSA: korg1212: fix if-statement empty body warnings
Fix gcc warnings when -Wextra is used by using an empty do-while
block instead of <nothing>. Fixes these build warnings:
../sound/pci/korg1212/korg1212.c:674:44: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:708:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:730:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:853:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:1013:44: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:1035:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:1052:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:1066:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:1087:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:1094:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:1208:43: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
../sound/pci/korg1212/korg1212.c:2360:102: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
The Gigabyte X570 Aorus Master motherboard with ALC1220 codec
requires a similar workaround for Clevo laptops to enforce the
DAC/mixer connection path. Set up a quirk entry for that.
Olivier Moysan [Wed, 4 Mar 2020 10:24:06 +0000 (11:24 +0100)]
ASoC: stm32: sai: manage rebind issue
The commit e894efef9ac7 ("ASoC: core: add support to card rebind")
allows to rebind the sound card after a rebind of one of its component.
With this commit, the sound card is actually rebound,
but may be no more functional. The following problems have been seen
with STM32 SAI driver.
1) DMA channel is not requested:
With the sound card rebind the simplified call sequence is:
stm32_sai_sub_probe
snd_soc_register_component
snd_soc_try_rebind_card
snd_soc_instantiate_card
devm_snd_dmaengine_pcm_register
The problem occurs because the pcm must be registered,
before snd_soc_instantiate_card() is called.
Modify SAI driver, to change the call sequence as follows:
stm32_sai_sub_probe
devm_snd_dmaengine_pcm_register
snd_soc_register_component
snd_soc_try_rebind_card
2) DMA channel is not released:
dma_release_channel() is not called when
devm_dmaengine_pcm_release() is executed.
This occurs because SND_DMAENGINE_PCM_DRV_NAME component,
has already been released through devm_component_release().
devm_dmaengine_pcm_release() should be called before
devm_component_release() to avoid this problem.
Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component()
explicitly from SAI driver, to have the right sequence.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Message-Id: <20200304102406.8093-1-olivier.moysan@st.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Dan Carpenter [Tue, 3 Mar 2020 10:18:58 +0000 (13:18 +0300)]
ASoC: SOF: Fix snd_sof_ipc_stream_posn()
We're passing "&posn" instead of "posn" so it ends up corrupting
memory instead of doing something useful.
Fixes: 53e0c72d98ba ("ASoC: SOF: Add support for IPC IO between DSP and Host") Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200303101858.ytehbrivocyp3cnf@kili.mountain Signed-off-by: Mark Brown <broonie@kernel.org>
commit 1e5ddb6ba73894 ("ASoC: component: Add sync_stop PCM ops")
added snd_soc_pcm_component_sync_stop(), but it is checking
ioctrl instead of sync_stop. This is bug.
This patch fixup it.
Charles Keepax [Fri, 28 Feb 2020 15:31:45 +0000 (15:31 +0000)]
ASoC: dapm: Correct DAPM handling of active widgets during shutdown
commit c2caa4da46a4 ("ASoC: Fix widget powerdown on shutdown") added a
set of the power state during snd_soc_dapm_shutdown to ensure the
widgets powered off. However, when commit 39eb5fd13dff
("ASoC: dapm: Delay w->power update until the changes are written")
added the new_power member of the widget structure, to differentiate
between the current power state and the target power state, it did not
update the shutdown to use the new_power member.
As new_power has not updated it will be left in the state set by the
last DAPM sequence, ie. 1 for active widgets. So as the DAPM sequence
for the shutdown proceeds it will turn the widgets on (despite them
already being on) rather than turning them off.
ASoC: soc-pcm/soc-compress: don't use snd_soc_dapm_stream_stop()
commit b0edff42360ab4 ("ASoC: soc-pcm/soc-compress: use
snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP")
uses snd_soc_dapm_stream_stop() for soc_compr_free_fe()
and dpcm_fe_dai_shutdown() because it didn't care about pmdown_time.
But, it didn't need to care.
This patch rollback to original code.
Some system will wait unneeded timed-out without this patch.
Special Thanks for reporting to Chris Gorman.
...
intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3
intel_sst_acpi 808622A8:00: fw returned err -16
sst-mfld-platform sst-mfld-platform: ASoC: PRE_PMD: pcm0_in event failed: -16
...
Fixes: commit b0edff42360ab4 ("ASoC: soc-pcm/soc-compress: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP") Reported-by: Chris Gorman <chrisjohgorman@gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Link: https://lore.kernel.org/r/87lfowspeb.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie@kernel.org>
Dragos Tarcatu [Fri, 7 Feb 2020 18:53:25 +0000 (20:53 +0200)]
ASoC: topology: Fix memleak in soc_tplg_manifest_load()
In case of ABI version mismatch, _manifest needs to be freed as
it is just a copy of the original topology manifest. However, if
a driver manifest handler is defined, that would get executed and
the cleanup is never reached. Fix that by getting the return status
of manifest() instead of returning directly.
Dragos Tarcatu [Fri, 7 Feb 2020 18:53:24 +0000 (20:53 +0200)]
ASoC: topology: Fix memleak in soc_tplg_link_elems_load()
If soc_tplg_link_config() fails, _link needs to be freed in case of
topology ABI version mismatch. However the current code is returning
directly and ends up leaking memory in this case.
This patch fixes that.
Takashi Iwai [Tue, 18 Feb 2020 11:17:37 +0000 (12:17 +0100)]
ASoC: pcm: Fix possible buffer overflow in dpcm state sysfs output
dpcm_show_state() invokes multiple snprintf() calls to concatenate
formatted strings on the fixed size buffer. The usage of snprintf()
is supposed for avoiding the buffer overflow, but it doesn't work as
expected because snprintf() doesn't return the actual output size but
the size to be written.
Fix this bug by replacing all snprintf() calls with scnprintf()
calls.
Fixes: f86dcef87b77 ("ASoC: dpcm: Add debugFS support for DPCM") Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20200218111737.14193-4-tiwai@suse.de Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Tue, 18 Feb 2020 11:17:36 +0000 (12:17 +0100)]
ASoC: intel: skl: Fix possible buffer overflow in debug outputs
The debugfs output of intel skl driver writes strings with multiple
snprintf() calls with the fixed size. This was supposed to avoid the
buffer overflow but actually it still would, because snprintf()
returns the expected size to be output, not the actual output size.
Fix it by replacing snprintf() calls with scnprintf().
Fixes: d14700a01f91 ("ASoC: Intel: Skylake: Debugfs facility to dump module config") Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20200218111737.14193-3-tiwai@suse.de Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Tue, 18 Feb 2020 11:17:35 +0000 (12:17 +0100)]
ASoC: intel: skl: Fix pin debug prints
skl_print_pins() loops over all given pins but it overwrites the text
at the very same position while increasing the returned length.
Fix this to show the all pin contents properly.
Fixes: d14700a01f91 ("ASoC: Intel: Skylake: Debugfs facility to dump module config") Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Cezary Rojewski <cezary.rojewski@intel.com> Link: https://lore.kernel.org/r/20200218111737.14193-2-tiwai@suse.de Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Tue, 18 Feb 2020 09:14:09 +0000 (10:14 +0100)]
ALSA: hda: Use scnprintf() for printing texts for sysfs/procfs
Some code in HD-audio driver calls snprintf() in a loop and still
expects that the return value were actually written size, while
snprintf() returns the expected would-be length instead. When the
given buffer limit were small, this leads to a buffer overflow.
Use scnprintf() for addressing those issues. It returns the actually
written size unlike snprintf().
Samuel Holland [Mon, 17 Feb 2020 06:42:22 +0000 (00:42 -0600)]
ASoC: sun8i-codec: Fix setting DAI data format
Use the correct mask for this two-bit field. This fixes setting the DAI
data format to RIGHT_J or DSP_A.
Fixes: 36c684936fae ("ASoC: Add sun8i digital audio codec") Signed-off-by: Samuel Holland <samuel@sholland.org> Acked-by: Chen-Yu Tsai <wens@csie.org> Cc: stable@kernel.org Link: https://lore.kernel.org/r/20200217064250.15516-7-samuel@sholland.org Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Thu, 13 Feb 2020 11:20:59 +0000 (12:20 +0100)]
ALSA: usb-audio: Parse source ID of UAC2 effect unit
During parsing the input source, we currently cut off at the Effect
Unit node without parsing further its source id. It's no big problem,
so far, but it should be more consistent to parse it properly.
This patch adds the recursive parsing in parse_term_effect_unit().
It doesn't add anything in the audio unit parser itself, and the
effect unit itself is still skipped, though.
The UAC2 Effect Unit Descriptor has a slightly different definition
from other similar ones like Processing Unit or Extension Unit.
Define it here so that it can be used in USB-audio driver in a later
patch.
Nick Kossifidis [Sat, 15 Feb 2020 01:23:35 +0000 (03:23 +0200)]
ALSA: usb-audio: Add support for Presonus Studio 1810c
This patch adds support for Presonus Studio 1810c, a usb interface
that's UAC2 compliant with a few quirks and a few extra hw-specific
controls. I've tested all 3 altsettings and the added switch
controls and they work as expected.
More infos on the card:
https://www.presonus.com/products/Studio-1810c
Note that this work is based on packet inspection with
usbmon. I just wanted to get this card to work for using
it on our open-source radio station:
https://github.com/UoC-Radio
v2 address issues reported by Takashi:
* Properly get/set enum type controls
* Prevent race condition on switch_get/set
* Various control naming changes
* Various coding style fixes
v3 improve readability of sample rate filtering
and some other minor changes.
Takashi Iwai [Fri, 14 Feb 2020 17:16:43 +0000 (18:16 +0100)]
ALSA: pcm: oss: Unlock mutex temporarily for sleeping at read/write
ALSA PCM OSS layer calls the generic __snd_pcm_lib_xfer() helper for
the actual transfer of the audio data. The xfer helper may sleep long
for waiting for the enough space becoming empty for read/write, and
it does unlock/relock for the substream lock. This works fine, so
far, but a slight problem specific to OSS layer is that OSS layer
wraps yet more mutex (runtime->oss.params_lock) over
__snd_pcm_lib_xfer() call; so this mutex is still locked during a
possible long sleep, and it prevents the whole ioctl and other actions
applied to the given stream.
This patch adds the temporarily unlock and relock of the mutex around
__snd_pcm_lib_xfer() call in the OSS layer to be more friendly to the
concurrent accesses. The long mutex protection itself shouldn't be a
real issue for the normal systems, and its influence appears only on
strange things like fuzzers.
Takashi Iwai [Fri, 14 Feb 2020 11:13:16 +0000 (12:13 +0100)]
ALSA: rawmidi: Avoid bit fields for state flags
The rawmidi state flags (opened, append, active_sensing) are stored in
bit fields that can be potentially racy when concurrently accessed
without any locks. Although the current code should be fine, there is
also no any real benefit by keeping the bitfields for this kind of
short number of members.
This patch changes those bit fields flags to the simple bool fields.
There should be no size increase of the snd_rawmidi_substream by this
change.
Takashi Iwai [Fri, 14 Feb 2020 11:13:15 +0000 (12:13 +0100)]
ALSA: seq: Fix concurrent access to queue current tick/time
snd_seq_check_queue() passes the current tick and time of the given
queue as a pointer to snd_seq_prioq_cell_out(), but those might be
updated concurrently by the seq timer update.
Fix it by retrieving the current tick and time via the proper helper
functions at first, and pass those values to snd_seq_prioq_cell_out()
later in the loops.
snd_seq_timer_get_cur_time() takes a new argument and adjusts with the
current system time only when it's requested so; this update isn't
needed for snd_seq_check_queue(), as it's called either from the
interrupt handler or right after queuing.
Also, snd_seq_timer_get_cur_tick() is changed to read the value in the
spinlock for the concurrency, too.
Takashi Iwai [Fri, 14 Feb 2020 11:13:14 +0000 (12:13 +0100)]
ALSA: seq: Avoid concurrent access to queue flags
The queue flags are represented in bit fields and the concurrent
access may result in unexpected results. Although the current code
should be mostly OK as it's only reading a field while writing other
fields as KCSAN reported, it's safer to cover both with a proper
spinlock protection.
This patch fixes the possible concurrent read by protecting with
q->owner_lock. Also the queue owner field is protected as well since
it's the field to be protected by the lock itself.
Takashi Iwai [Fri, 14 Feb 2020 14:49:28 +0000 (15:49 +0100)]
ALSA: usb-audio: Don't create a mixer element with bogus volume range
Some USB-audio descriptors provide a bogus volume range (e.g. volume
min and max are identical), which confuses user-space.
This patch makes the driver skipping such a control element.
Takashi Iwai [Thu, 13 Feb 2020 06:03:49 +0000 (07:03 +0100)]
ALSA: pcm: Fix double hw_free calls
The commit 66f2d19f8116 ("ALSA: pcm: Fix memory leak at closing a
stream without hw_free") tried to fix the regression wrt the missing
hw_free call at closing without SNDRV_PCM_IOCTL_HW_FREE ioctl.
However, the code change dropped mistakenly the state check, resulting
in calling hw_free twice when SNDRV_PCM_IOCTL_HW_FRE got called
beforehand. For most drivers, this is almost harmless, but the
drivers like SOF show another regression now.
This patch adds the state condition check before calling do_hw_free()
at releasing the stream for avoiding the double hw_free calls.
Samuel Holland [Thu, 13 Feb 2020 06:11:44 +0000 (00:11 -0600)]
ASoC: codec2codec: avoid invalid/double-free of pcm runtime
The PCM runtime was freed during PMU in the case that the event hook
encountered an error. However, it is also unconditionally freed during
PMD. Avoid a double-free by dropping the call to kfree in the PMU hook.
Alexander Tsoy [Wed, 12 Feb 2020 23:54:50 +0000 (02:54 +0300)]
ALSA: usb-audio: Add clock validity quirk for Denon MC7000/MCX8000
It should be safe to ignore clock validity check result if the following
conditions are met:
- only one single sample rate is supported;
- the terminal is directly connected to the clock source;
- the clock type is internal.
This is to deal with some Denon DJ controllers that always reports that
clock is invalid.
Because of MAX BUFFER size in register,when user/app give small
buffer size produces noise of old data in buffer.
This patch rectifies this noise when using different
buffer sizes less than MAX BUFFER.
Takashi Iwai [Wed, 12 Feb 2020 08:10:47 +0000 (09:10 +0100)]
ALSA: hda/realtek - Fix silent output on MSI-GL73
MSI-GL73 laptop with ALC1220 codec requires a similar workaround for
Clevo laptops to enforce the DAC/mixer connection path. Set up a
quirk entry for that.
ALSA: hda_codec: Replace zero-length array with flexible-array member
The current codebase makes use of the zero-length array language
extension to the C90 standard, but the preferred mechanism to declare
variable-length types such as these ones is a flexible array member[1][2],
introduced in C99:
struct foo {
int stuff;
struct boo array[];
};
By making use of the mechanism above, we will get a compiler warning
in case the flexible array does not occur last in the structure, which
will help us prevent some kind of undefined behavior bugs from being
inadvertenly introduced[3] to the codebase from now on.
ALSA: hda/ca0132 - Replace zero-length array with flexible-array member
The current codebase makes use of the zero-length array language
extension to the C90 standard, but the preferred mechanism to declare
variable-length types such as these ones is a flexible array member[1][2],
introduced in C99:
struct foo {
int stuff;
struct boo array[];
};
By making use of the mechanism above, we will get a compiler warning
in case the flexible array does not occur last in the structure, which
will help us prevent some kind of undefined behavior bugs from being
inadvertenly introduced[3] to the codebase from now on.
ALSA: usb-midi: Replace zero-length array with flexible-array member
The current codebase makes use of the zero-length array language
extension to the C90 standard, but the preferred mechanism to declare
variable-length types such as these ones is a flexible array member[1][2],
introduced in C99:
struct foo {
int stuff;
struct boo array[];
};
By making use of the mechanism above, we will get a compiler warning
in case the flexible array does not occur last in the structure, which
will help us prevent some kind of undefined behavior bugs from being
inadvertenly introduced[3] to the codebase from now on.
ALSA: core: Replace zero-length array with flexible-array member
The current codebase makes use of the zero-length array language
extension to the C90 standard, but the preferred mechanism to declare
variable-length types such as these ones is a flexible array member[1][2],
introduced in C99:
struct foo {
int stuff;
struct boo array[];
};
By making use of the mechanism above, we will get a compiler warning
in case the flexible array does not occur last in the structure, which
will help us prevent some kind of undefined behavior bugs from being
inadvertenly introduced[3] to the codebase from now on.
Arvind Sankar [Tue, 11 Feb 2020 16:22:35 +0000 (11:22 -0500)]
ALSA: usb-audio: Apply sample rate quirk for Audioengine D1
The Audioengine D1 (0x2912:0x30c8) does support reading the sample rate,
but it returns the rate in byte-reversed order.
When setting sampling rate, the driver produces these warning messages:
[168840.944226] usb 3-2.2: current rate 4500480 is different from the runtime rate 44100
[168854.930414] usb 3-2.2: current rate 8436480 is different from the runtime rate 48000
[168905.185825] usb 3-2.1.2: current rate 30465 is different from the runtime rate 96000
As can be seen from the hexadecimal conversion, the current rate read
back is byte-reversed from the rate that was set.
Takashi Iwai [Tue, 11 Feb 2020 16:05:21 +0000 (17:05 +0100)]
ALSA: usb-audio: Fix UAC2/3 effect unit parsing
We've got a regression report about M-Audio Fast Track C400 device,
and the git bisection resulted in the commit e0ccdef92653 ("ALSA:
usb-audio: Clean up check_input_term()"). This commit was about the
rewrite of the input terminal parser, and it's not too obvious from
the change what really broke. The answer is: it's the interpretation
of UAC2/3 effect units.
In the original code, UAC2 effect unit is as if through UAC1
processing unit because both UAC1 PU and UAC2/3 EU share the same
number (0x07). The old code went through a complex switch-case
fallthrough, finally bailing out in the middle:
if (protocol == UAC_VERSION_2 &&
hdr[2] == UAC2_EFFECT_UNIT) {
/* UAC2/UAC1 unit IDs overlap here in an
* uncompatible way. Ignore this unit for now.
*/
return 0;
}
... and this special handling was missing in the new code; the new
code treats UAC2/3 effect unit as if it were equivalent with the
processing unit.
Actually, the old code was too confusing. The effect unit has an
incompatible unit description with the processing unit, so we
shouldn't have dealt with EU in the same way.
This patch addresses the regression by changing the effect unit
handling to the own parser function. The own parser function makes
the clear distinct with PU, so it improves the readability, too.
The EU parser just sets the type and the id like the old kernels.
Once when the proper effect unit support is added, we can revisit this
parser function, but for now, let's keep this simple setup as is.
Jabra Evolve 65 headset appears as if supporting lower rates than
48kHz, but it actually doesn't work but with 48kHz for playback.
This patch applies a workaround to enforce the 48kHz like LINE6
devices already did. The workaround is put in a unified helper
function, set_fixed_rate(), to be called from both places now.
Kai Vehmanen [Thu, 6 Feb 2020 20:02:23 +0000 (22:02 +0200)]
ASoC: SOF: Intel: hda: move i915 init earlier
To be compliant with i915 display driver requirements, i915 power-up
must be done before any HDA communication takes place, including
parsing the bus capabilities. Otherwise the initial codec probe
may fail.
Move i915 initialization earlier in the SOF HDA sequence. This
sequence is now aligned with the snd-hda-intel driver where the
display_power() call is before snd_hdac_bus_parse_capabilities()
and rest of the capability parsing.
Also remove unnecessary ifdef around hda_codec_i915_init(). There's
a dummy implementation provided if CONFIG_SND_SOC_SOF_HDA is not
enabled.
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: Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20200206200223.7715-3-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Kai Vehmanen [Thu, 6 Feb 2020 20:02:22 +0000 (22:02 +0200)]
ASoC: SOF: Intel: hda: fix ordering bug in resume flow
When HDA controller is resumed from suspend, i915 HDMI/DP
codec requires that following order of actions is kept:
- i915 display power up and configuration of link params
- hda link reset and setup
Current SOF HDA code delegates display codec power control
to the codec driver. This works most of the time, but in
runtime PM sequences, the above constraint may be violated.
On platforms where BIOS values for HDA link parameters do
not match hardware reset defaults, this may lead to errors
in HDA verb transactions after resume.
Fix the issue by explicitly powering the display codec
in the HDA controller resume/suspend calls, thus ensuring
correct ordering. Special handling is needed for the D0i3
flow, where display power must be turned off even though
DSP is left powered.
Now that we have more invocations of the display power helper
functions, the conditional checks surrounding each call have
been moved inside hda_codec_i915_display_power(). The two
special cases of display powering at initial probe are handled
separately. The intent is to avoid powering the display whenever
no display codecs are used.
Note that early powering of display was removed in
commit 687ae9e287b3 ("ASoC: intel: skl: Fix display power regression").
This change was also copied to the SOF driver. No failures
have resulted as hardware default values for link parameters
have worked out of the box. However with recent i915 driver
changes like done in commit 87c1694533c9 ("drm/i915: save
AUD_FREQ_CNTRL state at audio domain suspend"), this does not
hold anymore and errors are hit.
Cc: Takashi Iwai <tiwai@suse.de> 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: Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20200206200223.7715-2-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Kai Vehmanen [Thu, 6 Feb 2020 20:02:21 +0000 (22:02 +0200)]
ALSA: hda: do not override bus codec_mask in link_get()
snd_hdac_ext_bus_link_get() does not work correctly in case
there are multiple codecs on the bus. It unconditionally
resets the bus->codec_mask value. As per documentation in
hdaudio.h and existing use in client code, this field should
be used to store bit flag of detected codecs on the bus.
By overwriting value of the codec_mask, information on all
detected codecs is lost. No current user of hdac is impacted,
but use of bus->codec_mask is planned in future patches
for SOF.
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: Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20200206200223.7715-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Arnd Bergmann [Thu, 30 Jan 2020 13:05:45 +0000 (15:05 +0200)]
ASoC: atmel: fix atmel_ssc_set_audio link failure
The ssc audio driver can call into both pdc and dma backends. With the
latest rework, the logic to do this in a safe way avoiding link errors
was removed, bringing back link errors that were fixed long ago in commit 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") such as
sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio':
atmel_ssc_dai.c:(.text+0xac): undefined reference to `atmel_pcm_pdc_platform_register'
Fix it this time using Makefile hacks and a comment to prevent this
from accidentally getting removed again rather than Kconfig hacks.
Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com> Link: https://lore.kernel.org/r/20200130130545.31148-1-codrin.ciubotariu@microchip.com Reviewed-by: Michał Mirosław <mirq-linux@rere.qmqm.pl> Signed-off-by: Mark Brown <broonie@kernel.org>
Colin Ian King [Sat, 8 Feb 2020 22:42:06 +0000 (22:42 +0000)]
ALSA: info: remove redundant assignment to variable c
Variable c is being assigned with the value -1 that is never read,
it is assigned a new value in a following while-loop. The assignment
is redundant and can be removed.
Colin Ian King [Sat, 8 Feb 2020 22:27:56 +0000 (22:27 +0000)]
ALSA: hda: remove redundant assignment to variable timeout
Variable timeout is being assigned with the value 200 that is never
read, it is assigned a new value in a following do-loop. The assignment
is redundant and can be removed.
Colin Ian King [Sat, 8 Feb 2020 22:20:06 +0000 (22:20 +0000)]
ALSA: hdsp: remove redundant assignment to variable err
Variable err is being assigned with a value that is never read, it is
assigned a new value in the next statement. The assignment is redundant
and can be removed.
Takashi Iwai [Thu, 6 Feb 2020 16:39:44 +0000 (17:39 +0100)]
ALSA: pcm_dmaengine: Use pcm_for_each_format() macro for PCM format iteration
The new macro can fix the sparse warnings gracefully:
sound/core/pcm_dmaengine.c:429:50: warning: restricted snd_pcm_format_t degrades to integer
sound/core/pcm_dmaengine.c:429:55: warning: restricted snd_pcm_format_t degrades to integer
sound/core/pcm_dmaengine.c:429:79: warning: restricted snd_pcm_format_t degrades to integer
Takashi Iwai [Thu, 6 Feb 2020 16:39:43 +0000 (17:39 +0100)]
ALSA: pcm: Use a macro for parameter masks to reduce the needed cast
The parameter bit mask needs often explicit cast with __force,
e.g. for the PCM subformat type. Instead of adding __force at each
place, which is error prone, this patch introduces a new macro and
replaces the all bit shift with it. This fixes the sparse warnings
like the following:
sound/core/pcm_native.c:2508:30: warning: restricted snd_pcm_access_t degrades to integer
Takashi Iwai [Thu, 6 Feb 2020 16:39:42 +0000 (17:39 +0100)]
ALSA: pcm: Use standard macros for fixing PCM format cast
Simplify the code with the new macros for PCM format type iterations.
This fixes the sparse warnings nicely:
sound/core/pcm_native.c:2302:26: warning: restricted snd_pcm_format_t degrades to integer
sound/core/pcm_native.c:2306:54: warning: incorrect type in argument 1 (different base types)
sound/core/pcm_native.c:2306:54: expected restricted snd_pcm_format_t [usertype] format
sound/core/pcm_native.c:2306:54: got unsigned int [assigned] k
....
Takashi Iwai [Thu, 6 Feb 2020 16:39:41 +0000 (17:39 +0100)]
ALSA: dummy: Use standard macros for fixing PCM format cast
Simplify the code with the new macros for PCM format type iterations.
This fixes the sparse warnings nicely:
sound/drivers/dummy.c:906:25: warning: restricted snd_pcm_format_t degrades to integer
sound/drivers/dummy.c:908:25: warning: incorrect type in argument 1 (different base types)
sound/drivers/dummy.c:908:25: expected restricted snd_pcm_format_t [usertype] format
sound/drivers/dummy.c:908:25: got int [assigned] i
Takashi Iwai [Thu, 6 Feb 2020 16:39:40 +0000 (17:39 +0100)]
ALSA: usb-audio: Use pcm_for_each_format() macro for PCM format iterations
The new macro can fix the sparse warnings gracefully:
sound/usb/proc.c:73:31: warning: restricted snd_pcm_format_t degrades to integer
sound/usb/proc.c:73:38: warning: restricted snd_pcm_format_t degrades to integer
sound/usb/proc.c:73:61: warning: restricted snd_pcm_format_t degrades to integer
Takashi Iwai [Thu, 6 Feb 2020 16:39:39 +0000 (17:39 +0100)]
ALSA: pcm: More helper macros for reducing snd_pcm_format_t cast
snd_pcm_format_t is a strong-typed integer and requires the explicit
cast with __force if converted or compared with a normal integer
value. Since most of use cases do iterate over all formats and test /
set the mask, provide a couple of new helper macros that do the
explicit cast.
Takashi Iwai [Thu, 6 Feb 2020 16:39:38 +0000 (17:39 +0100)]
ALSA: aloop: Fix PCM format assignment
Fix sparse warnings about PCM format assignment regarding the strong
typed snd_pcm_format_t:
sound/drivers/aloop.c:352:45: warning: restricted snd_pcm_format_t degrades to integer
sound/drivers/aloop.c:355:39: warning: incorrect type in assignment (different base types)
sound/drivers/aloop.c:355:39: expected unsigned int format
sound/drivers/aloop.c:355:39: got restricted snd_pcm_format_t [usertype] format
sound/drivers/aloop.c:1435:34: warning: incorrect type in assignment (different base types)
sound/drivers/aloop.c:1435:34: expected long max
sound/drivers/aloop.c:1435:34: got restricted snd_pcm_format_t [usertype]
sound/drivers/aloop.c:1565:39: warning: incorrect type in assignment (different base types)
sound/drivers/aloop.c:1565:39: expected unsigned int format
sound/drivers/aloop.c:1565:39: got restricted snd_pcm_format_t [usertype]
Some code in this driver assigns an integer value to snd_pcm_format_t
via control API, and they need to be with the explicit cast.
Takashi Iwai [Thu, 6 Feb 2020 16:31:52 +0000 (17:31 +0100)]
ALSA: emu8000: Fix the cast to __user pointer
Fixes the sparse warnings. The cast to __user pointer needs __force:
sound/isa/sb/emu8000_pcm.c:528:9: warning: cast removes address space '<asn:1>' of expression
Takashi Iwai [Thu, 6 Feb 2020 16:31:51 +0000 (17:31 +0100)]
ALSA: emu10k1: Fix endianness annotations
The internal page tables are little endian, hence they should be
__le32 type. This fixes the relevant sparse warning:
sound/pci/emu10k1/emu10k1_main.c:2013:51: warning: incorrect type in assignment (different base types)
sound/pci/emu10k1/emu10k1_main.c:2013:51: expected unsigned int [usertype]
sound/pci/emu10k1/emu10k1_main.c:2013:51: got restricted __le32 [usertype]
Takashi Iwai [Thu, 6 Feb 2020 16:31:50 +0000 (17:31 +0100)]
ALSA: via82xx: Fix endianness annotations
The internal page tables are in little endian, hence they should be
__le32 type. This fixes the relevant sparse warnings:
sound/pci/via82xx.c:454:60: warning: incorrect type in assignment (different base types)
sound/pci/via82xx.c:454:60: expected unsigned int [usertype]
sound/pci/via82xx.c:454:60: got restricted __le32 [usertype]
....
Takashi Iwai [Thu, 6 Feb 2020 16:28:04 +0000 (17:28 +0100)]
ALSA: hda/hdmi: Move ELD parse and jack reporting into update_eld()
This is a final step of the cleanup series: move the HDMI ELD parser
call into update_eld() function so that we can unify the calls.
The ELD validity check is unified in update_eld(), too.
Along with it, the repoll scheduling is moved to update_eld() as well,
where sync_eld_via_acomp() just passes 0 for skipping it.
Takashi Iwai [Thu, 6 Feb 2020 16:28:03 +0000 (17:28 +0100)]
ALSA: hda/hdmi: Move runtime PM resume into hdmi_present_sense_via_verbs()
For improving the readability, move the runtime PM handling code from
hdmi_present_sense() to hdmi_present_sense_via_verbs(). Now
hdmi_present_sense() became symmetric for both audio-component and
legacy cases.
Takashi Iwai [Thu, 6 Feb 2020 16:28:02 +0000 (17:28 +0100)]
ALSA: hda/hdmi: Don't use standard hda_jack for generic HDMI jacks
The current HDMI codec driver code manages the jack detection in two
different ways: for Intel codecs with audio component, the driver
creates snd_jack objects by itself while the standard hda_jack stuff
is used for the rest. This was basically because the audio component
doesn't need the pin sense reading and the unsol event handling, hence
it just needs to report the corresponding jacks directly.
It was a bit messy but not too messy until the driver got DP-MST
support for Nvidia that re-uses the part of dyn_pcm_assign feature
while keeping the pin sense and the unsol event handling. Now, for
DP-MST, we use hda_jack for pin sensing and unsol events but use the
own snd_jack objects. Meanwhile for non-DP-MST, hda_jack is used for
pin sense and unsol events, and the jacks are bound on hda_jack.
Moreover, there is a polling mode support where the unsol event isn't
used. For those, we also have special handling.
For simplifying those messes, this patch unifies the snd_jack handling
over all generic HDMI codes. The driver creates snd_jack objects just
like Intel codecs did in the past but now for all devices. For the
system without audio component binding, we still need the pin sense
and the unsol event handling, and those are still done with the
hda_jack table as before. But hda_jack is no longer used for the
actual snd_jack handling.
Since the hda_jack is no longer used for jack reporting, we removed
snd_hda_jack_report_sync() calls, which also allowed to simplify the
return type of hda_present_sense() and co. pin_idx_to_pcm_jack() was
simplified as well because it behaves same for all cases now.
Note that the hda_jack is still used for the simple HDMI codecs; they
are really simple enough, so no big reason to change intrusively.
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