snd_pcm_lib_mmap_vmalloc() was supposed to be implemented with
somewhat special for vmalloc handling, but in the end, this turned to
just the default handler, i.e. NULL. As the situation has never
changed over decades, let's rip it off.
The size of in-kernel rawmidi buffers may be big up to 1MB, and it can
be specified freely by user-space; which implies that user-space may
trigger kmalloc() errors frequently.
This patch replaces the buffer allocation via kvmalloc() for dealing
with bigger buffers gracefully.
Unify a few open codes with helper functions to improve the
readability. Minor behavior changes (rather fixes) are:
- runtime->drain clearance is done within lock
- active_sensing is updated before resizing buffer in
SNDRV_RAWMIDI_IOCTL_PARAMS ioctl.
Other than that, simply code cleanups.
The SNDRV_RAWMIDI_IOCTL_PARAMS ioctl may resize the buffers and the
current code is racy. For example, the sequencer client may write to
buffer while it being resized.
As a simple workaround, let's switch to the resized buffer inside the
stream runtime lock.
ALSA: usb-audio: Tidy up logic for Processing Unit min/max values
This patch refactors the processing units min/max calculation logic
for the mixer controls and fixes an issue where the Mode Select
checking of the Up/Down mixers doesn't differentiate between the
UAC1 and UAC2 Control Selector (0x02) and the UAC3 one which is
different (0x01).
ALSA: usb-audio: Unify virtual type units type to UAC3 values
The Audio Control interface descriptor subtypes do not match
across all the UAC versions. That makes reusability of the
"virtual type" (Mixer, Processors, Selectors, etc) terminals
difficult. It also makes the mixer get the default names for
the virtual terminals wrong due to the overlap.
This patch proposes an unified approach by always using the most
comprehensive spec version to define them all (in this case UAC3).
ALSA: usb-audio: Add support for Processing Units in UAC3
This patch adds support for the Processig Units defined in
the UAC3 spec. The main difference with the previous specs
is the lack of on/off switches in the controls for these
units and the addiction of the new Multi Function Processing
Unit.
The current version of the UAC3 spec doesn't define any
useful controls for the new Multi Function Processing Unit
so no control will get created once this unit is parsed.
ALSA: usb-audio: Processing Unit controls parsing in UAC2
Current support for UAC2 Processing Units does the parsing
as one control per bit in the bitmap. However, the UAC2 spec
defines the controls as bit pairs where b01 means read-only
and b11 means read/write control.
This patch fixes that and uses the helper functions for checking
controls readability/writability when the control is defined as
bit pairs (UAC2 and UAC3).
ALSA: usb-audio: Add support for Selector Units in UAC3
This patch add support for Selector Units and Clock Selector Units
defined in the new UAC3 spec.
Selector Units play a really important role in the new UAC3 spec as
Processing Units do not define an on/off switch control anymore.
This forces topology designers to add bypass paths in the topology
to enable/dissable the Processing Units.
Timo Wischer [Tue, 10 Jul 2018 15:28:45 +0000 (17:28 +0200)]
ALSA: pcm: Fix snd_interval_refine first/last with open min/max
Without this commit the following intervals [x y), (x y) were be
replaced to (y-1 y) by snd_interval_refine_last(). This was also done
if y-1 is part of the previous interval.
With this changes it will be replaced with [y-1 y) in case of y-1 is
part of the previous interval. A similar behavior will be used for
snd_interval_refine_first().
This commit adapts the changes for alsa-lib of commit 9bb985c ("pcm: snd_interval_refine_first/last: exclude value only if
also excluded before")
Signed-off-by: Timo Wischer <twischer@de.adit-jv.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO.
This needs to be configurable for modern hardware like DSPs where no
pointer update in milliseconds can indicate terminal DSP errors.
Add a substream variable to set the wait time in ms. This allows userspace
and drivers to recover more quickly from terminal DSP errors.
Hui Wang [Fri, 6 Jul 2018 07:14:11 +0000 (15:14 +0800)]
ALSA: hda/realtek - two more lenovo models need fixup of MIC_LOCATION
We have two new lenovo desktop models which need to apply the fixup of
ALC294_FIXUP_LENOVO_MIC_LOCATION, and they have the same pin cfg as
the machine with subsystem id:0x17aa3136, now use the pincfg table
to apply the fixup for them.
Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: pcm: Use snd_pcm_stop_xrun() for xrun injection
Basically the xrun injection routine can simply call the standard
helper snd_pcm_stop_xrun(), but with one exception: it may be called
even when the stream is closed.
Make snd_pcm_stop_xrun() more robust and check the NULL runtime state,
and simplify xrun injection code by calling it.
The PCM xrun injection triggers directly snd_pcm_stop() without the
standard xrun handler, hence it's not recorded on the event buffer.
Ditto for snd_pcm_stop_xrun() call and SNDRV_PCM_IOCTL_XRUN ioctl.
They are inconvenient from the debugging POV.
Let's make them to trigger XRUN via the standard helper more
consistently.
Chris Wilson [Wed, 27 Jun 2018 06:25:32 +0000 (07:25 +0100)]
ALSA: hda - Handle pm failure during hotplug
Obtaining the runtime pm wakeref can fail, especially in a hotplug
scenario where i915.ko has been unloaded. If we do not catch the
failure, we end up with an unbalanced pm.
v2 additions by tiwai:
hdmi_present_sense() checks the return value and handle only a
negative error case and bails out only if it's really still suspended.
Also, snd_hda_power_down() is called at the error path so that the
refcount is balanced.
Along with it, the spec->pcm_lock is taken outside
hdmi_present_sense() in the caller side, so that it won't cause
deadlock at reentrace via runtime resume.
v3 fix by tiwai:
Missing linux/pm_runtime.h is included.
References: 222bde03881c ("ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug") Signed-off-by: Chris Wilson <chris@chris-wilson.co.uk> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
John Ogness [Sun, 1 Jul 2018 15:28:08 +0000 (17:28 +0200)]
ALSA: usb-midi: use irqsave() in USB's complete callback
The USB completion callback does not disable interrupts while acquiring
the lock. We want to remove the local_irq_disable() invocation from
__usb_hcd_giveback_urb() and therefore it is required for the callback
handler to disable the interrupts while acquiring the lock.
The callback may be invoked either in IRQ or BH context depending on the
USB host controller.
Use the _irqsave() variant of the locking primitives.
Signed-off-by: John Ogness <john.ogness@linutronix.de> Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
John Ogness [Sun, 1 Jul 2018 15:28:07 +0000 (17:28 +0200)]
ALSA: usb: caiaq: audio: use irqsave() in USB's complete callback
The USB completion callback does not disable interrupts while acquiring
the lock. We want to remove the local_irq_disable() invocation from
__usb_hcd_giveback_urb() and therefore it is required for the callback
handler to disable the interrupts while acquiring the lock.
The callback may be invoked either in IRQ or BH context depending on the
USB host controller.
Use the _irqsave() variant of the locking primitives.
Signed-off-by: John Ogness <john.ogness@linutronix.de> Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de> Acked-by: Daniel Mack <daniel@zonque.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both calls to of_find_node_by_name() and of_get_next_child() return a
node pointer with refcount incremented thus it must be explicidly
decremented here after the last usage. As we are assured to have a
refcounted np either from the initial
of_find_node_by_name(NULL, name); or from the of_get_next_child(gpio, np)
in the while loop if we reached the error code path below, an
x of_node_put(np) is needed.
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org> Fixes: commit f3d9478b2ce4 ("[ALSA] snd-aoa: add snd-aoa") Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 27 Jun 2018 07:54:46 +0000 (09:54 +0200)]
ALSA: hda/hdmi - Don't fall back to generic when i915 binding fails
When i915 component binding fails, it means that HDMI isn't applicable
anyway. Although the probe with the generic HDMI parser would still
work, it's essentially useless, hence better to be left unbound.
This patch mimics the probe_id field at failing the i915 component
binding so that the generic HDMI won't be bound after that.
Rakesh Ughreja [Sat, 2 Jun 2018 03:53:58 +0000 (22:53 -0500)]
ALSA: hdac: add extended ops in the hdac_bus
Add extended ops in the hdac_bus to allow calling the ASoC HDAC library
ops to reuse the legacy HDA codec drivers with ASoC framework.
Extended ops are used by the legacy codec drivers to call into
hdac_hda library, in the subsequent patches..
Rakesh Ughreja [Sat, 2 Jun 2018 03:53:57 +0000 (22:53 -0500)]
ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init
Remove memory allocation within snd_hdac_ext_bus_device_init, to make
its behaviour identical to snd_hdac_bus_device_init. So that caller
can allocate the parent data structure containing hdac_device.
This API change helps in reusing the legacy HDA codec drivers with
ASoC platform drivers.
Rakesh Ughreja [Sat, 2 Jun 2018 03:53:56 +0000 (22:53 -0500)]
ALSA: hda: split snd_hda_codec_new function
Split snd_hda_codec_new into two separate functions.
snd_hda_codec_device_init allocates memory and registers with bus.
snd_hda_codec_device_new initialializes the fields and performs
snd_device_new. This enables reuse of legacy HDA codec drivers as ASoC
codec drivers.
In addition mark some functions with EXPORT_SYMBOL_GPL so that it can
be called by ASoC wrapper around the legacy HDA driver (hdac_hda).
Rakesh Ughreja [Sat, 2 Jun 2018 03:54:00 +0000 (22:54 -0500)]
ALSA: hdac: ext: add wait for codec to respond after link reset
As per HDA spec section 4.3 - Codec Discovery, the software shall wait
for atleast 521usec for codec to respond after link reset.
With the multi-link capability each link is turned ON/OFF individually.
Link controller drives reset signal when it is turned ON.
Rakesh Ughreja [Sat, 2 Jun 2018 03:53:51 +0000 (22:53 -0500)]
ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead
This patch removes the hdac_ext_driver structure. The legacy and
enhanced HDaudio capabilities can be handled in a backward-compatible
way without separate definitions.
Rakesh Ughreja [Sat, 2 Jun 2018 03:53:50 +0000 (22:53 -0500)]
ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead
This patch removes the hdac_ext_bus structure. The legacy and
enhanced HDaudio capabilities can be handled in a backward-compatible
way without separate definitions.
Follow-up patches in this series handle the driver definition.
Rakesh Ughreja [Sat, 2 Jun 2018 03:53:49 +0000 (22:53 -0500)]
ALSA: hdac: Remove usage of struct hdac_ext_device and use hdac_device instead
This patch removes the hdac_ext_device structure. The legacy and
enhanced HDaudio capabilities can be handled in a backward-compatible
way without separate definitions.
Follow-up patches in this series handle the bus and driver definitions.
Takashi Iwai [Wed, 27 Jun 2018 07:03:51 +0000 (09:03 +0200)]
ALSA: hda - Move in_pm accessors to HDA core
The in_pm atomic in hdac_device is an important field used as a flag
as well as a refcount for PM. The existing snd_hdac_power_up/down
helpers already refer to it in the HD-audio core code, while the code
to actually setting the value (atomic_inc() / _dec()) is open-coded in
HDA legacy side, which is hard to find.
This patch adds the helper functions to set/reset the in_pm counter to
HDA core and use them in HDA legacy side, for making it clearer who /
where the PM is managed.
There is no functional changes, just code refactoring.
Reviewed-by: Chris Wilson <chris@chris-wilson.co.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 27 Jun 2018 12:59:00 +0000 (14:59 +0200)]
ALSA: intel_hdmi: Use strlcpy() instead of strncpy()
hdmi_lpe_audio_probe() copies the pcm name string via strncpy(), but
as a gcc8 warning suggests, it misses a NUL terminator, and unlikely
the expected result.
Takashi Iwai [Tue, 26 Jun 2018 05:42:40 +0000 (07:42 +0200)]
ALSA: hda/realtek - Comprehensive model list for ALC662 & co
ALC662 and others have far more fixup entries than the model table.
Let's add more model string entries so that user can test / debug
without compiling kernels at each time.
Takashi Iwai [Mon, 25 Jun 2018 09:13:59 +0000 (11:13 +0200)]
ALSA: seq: Fix UBSAN warning at SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT ioctl
The kernel may spew a WARNING with UBSAN undefined behavior at
handling ALSA sequencer ioctl SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT:
UBSAN: Undefined behaviour in sound/core/seq/seq_clientmgr.c:2007:14
signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Call Trace:
__dump_stack lib/dump_stack.c:77 [inline]
dump_stack+0x122/0x1c8 lib/dump_stack.c:113
ubsan_epilogue+0x12/0x86 lib/ubsan.c:159
handle_overflow+0x1c2/0x21f lib/ubsan.c:190
__ubsan_handle_add_overflow+0x2a/0x31 lib/ubsan.c:198
snd_seq_ioctl_query_next_client+0x1ac/0x1d0 sound/core/seq/seq_clientmgr.c:2007
snd_seq_ioctl+0x264/0x3d0 sound/core/seq/seq_clientmgr.c:2144
....
It happens only when INT_MAX is passed there, as we're incrementing it
unconditionally. So the fix is trivial, check the value with
INT_MAX. Although the bug itself is fairly harmless, it's better to
fix it so that fuzzers won't hit this again later.
Takashi Iwai [Mon, 25 Jun 2018 09:09:11 +0000 (11:09 +0200)]
ALSA: timer: Fix UBSAN warning at SNDRV_TIMER_IOCTL_NEXT_DEVICE ioctl
The kernel may spew a WARNING about UBSAN undefined behavior at
handling ALSA timer ioctl SNDRV_TIMER_IOCTL_NEXT_DEVICE:
UBSAN: Undefined behaviour in sound/core/timer.c:1524:19
signed integer overflow: 2147483647 + 1 cannot be represented in type 'int'
Call Trace:
__dump_stack lib/dump_stack.c:77 [inline]
dump_stack+0x122/0x1c8 lib/dump_stack.c:113
ubsan_epilogue+0x12/0x86 lib/ubsan.c:159
handle_overflow+0x1c2/0x21f lib/ubsan.c:190
__ubsan_handle_add_overflow+0x2a/0x31 lib/ubsan.c:198
snd_timer_user_next_device sound/core/timer.c:1524 [inline]
__snd_timer_user_ioctl+0x204d/0x2520 sound/core/timer.c:1939
snd_timer_user_ioctl+0x67/0x95 sound/core/timer.c:1994
....
It happens only when a value with INT_MAX is passed, as we're
incrementing it unconditionally. So the fix is trivial, check the
value with INT_MAX. Although the bug itself is fairly harmless, it's
better to fix it so that fuzzers won't hit this again later.
Currently patch_alc269() calls the fixup with HDA_FIXUP_ACT_PRE_PROBE
before setting up the codec model-specific setups (e.g. setting
codec_variant or mixer_nid setup). This is rather confusing as others
do call the *_PRE_PROBE fixup after such a setup. Due to this
disorder, we have to override spec->shutup not at the usual
HDA_FIXUP_ACT_PRE_PROBE but the unusual HDA_FIXUP_ACT_PROBE time.
This patch corrects the fixup call orders in patch_alc269(), and also
corrects the action to set up spec->shutup accordingly.
Takashi Iwai [Wed, 20 Jun 2018 15:27:50 +0000 (17:27 +0200)]
ALSA: hda/via - Use standard verb containers
In this patch, the remaining static init verbs in VIA codec driver are
converted to the standard snd_hda_add_verbs() calls. The conversion
is straightforward, but one change to be noted is the place of calls:
since these verbs are supposed to be executed at the beginning of the
init / resume procedure, we need to add snd_hda_add_verbs() calls
before calling the other parsers.
Takashi Iwai [Wed, 20 Jun 2018 15:20:42 +0000 (17:20 +0200)]
ALSA: hda/via - Simplify control management
This patch replaces the control element creations in VIA codec driver
with the standard snd_hda_gen_add_kctl() calls as a cleanup. There
are two major fields targeted by this patch: the beep controls and
static init controls.
The former is converted just like other codec drivers do. The
spec->beep_amp field can be eliminated by this change as well.
The latter, static init controls, are replaced simply with explicit
snd_hda_gen_add_kctl() calls.
After these conversions, via_build_controls() becomes superfluous and
replaced with snd_hda_gen_build_controls(), too.
Takashi Iwai [Wed, 20 Jun 2018 20:36:17 +0000 (22:36 +0200)]
ALSA: hda/via - Rewrite with error goto
Currently VIA codec driver invokes via_free() at each place of the
error path. Move the error handling to the end of each function
commonly and do goto-error as a standard idiom.
This is a preliminary patch for the further cleanups, and no
functional changes.
Takashi Iwai [Wed, 20 Jun 2018 15:06:58 +0000 (17:06 +0200)]
ALSA: hda/conexant - Clean up beep code
Like the previous commit for Realtek codec, the similar cleanup work
can be applied to Conexant codec, too. A slight difference is that
the call of cx_auto_parse_beep() is moved after
snd_hda_gen_parse_auto_config(). It's not strictly needed, but it'd
be good to make the creation of such beep mixers at the end, which
matches with the former situation.
Along with this conversion, cx_auto_build_controls() becomes just
calling snd_hda_gen_build_controls(), so it's simply replaced with
snd_hda_gen_build_controls.
Takashi Iwai [Wed, 20 Jun 2018 10:52:46 +0000 (12:52 +0200)]
ALSA: hda/realtek - Use common helper for creating beep controls
In the Realtek codec driver, we used to build kctl elements for beep
mixer in the own build_controls callback. This is an open-code and
can be covered by the standard feature of the generic parser with
snd_hda_gen_add_kctl() instead.
Also, after the conversion, spec->beep_amp becomes superfluous; hence
it's removed along with the conversion.
Takashi Iwai [Wed, 20 Jun 2018 10:50:11 +0000 (12:50 +0200)]
ALSA: hda/realtek - Use common helper for creating ALC268 beep controls
The beep mixer controls are the only remaining stuff that uses
spec->mixers[] array, and they can be well converted to the standard
helper in the generic parser, snd_hda_gen_add_kctl().
This simplifies the code, especially the superfluous mixers and
num_mixers fields can be now removed from alc_spec.
Takashi Iwai [Tue, 19 Jun 2018 21:05:47 +0000 (23:05 +0200)]
ALSA: hda/realtek - Use common GPIO mask for ALC660VD ASUS fixup
The ALC660VD_FIX_ASUS_GPIO1 quirk requires to set up GPIO bit0 ON
while bit 1 OFF. Implement the fixup function and convert from the
static init verbs.
Dell XPS13 has multi-step fixups, and one of them
(ALC288_FIXUP_DELL_XPS_13_GPIO6) corresponds to the management of GPIO
bit6 (0x40). It used to be a static init verbs (to turn *off* the
bit6).
In this patch, we convert it as the gpio_mask and gpio_dir
initializations folded in the existing fixup function. With this
change, ALC288_FIXUP_DELL_XPS_13_GPIO6 becomes superfluous, thus it's
removed.
Takashi Iwai [Tue, 19 Jun 2018 21:00:03 +0000 (23:00 +0200)]
ALSA: hda/realtek - Convert some manual GPIO setups
This patch converts the remaining static init verbs for GPIO bits with
the common gpio_* fields management. Only the verbs setting the GPIO
data bits are targeted in this patch. The rest will be changed in
later patches.
Takashi Iwai [Tue, 19 Jun 2018 20:47:30 +0000 (22:47 +0200)]
ALSA: hda/realtek - Simplify mute LED GPIO handling
Now we can simplify the mute LED GPIO handling as well. Each fixup
dealing with GPIO for the mute LED controls defined the static init
verbs, and they are converted to the common GPIO bit fields with the
new helper, alc_fixup_hp_gpio_led().
Takashi Iwai [Tue, 19 Jun 2018 20:28:22 +0000 (22:28 +0200)]
ALSA: hda/realtek - Add GPIO data update helper
For updating GPIO bits dynamically, provide a new helper, and use it
from the alc260 automute hook. This helper will be used by other
places in future, too.
Takashi Iwai [Tue, 19 Jun 2018 20:22:41 +0000 (22:22 +0200)]
ALSA: hda/realtek - Manage GPIO bits commonly
Currently the GPIO bits are managed by individual verbs in some cases
while toggled dynamically in other cases. For simplifying the GPIO
management, define the GPIO mask, dir and data bits in alc_spec
fields, and refer to / set them consistently from all places.
As a first step, along with the definition of the new gpio_* fields,
this patch replaces the static verbs that are used at initialization
and fixups with the common helper functions.
Some devices have the overrides of spec->init_amp at
HDA_FIXUP_ACT_PROBE just because alc_ssid_check() gives the
false-positive values from the SSID.
For more consistent behavior, define the logic in the following way:
- Define ALC_INIT_UNDEFINED as the default value before calling
alc_ssid_check()
- Each fixup may set up spec->init_amp with another value at
HDA_FIXUP_ACT_PRE_PROBE
- At detection, check whether spec->init_amp is ALC_INIT_UNDEFINED or
not; if it's different, we skip the detection
Also, it turned out that ASUS TX300 requires the spec->init_amp
override, too; currently it ignores the GPIO bits implicitly by its
static init verb, but this will be changed in the later patchset.
Robert Jarzmik [Fri, 22 Jun 2018 19:28:34 +0000 (21:28 +0200)]
ALSA: ac97: add codecs devicetree binding
Add a devicetree binding for codecs. This is especially useful if the
AC97 bitclk clock is provided by the codec, as it has to be described in
the devicetree description for the ac97 bus code to aquire it.
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hui Wang [Mon, 25 Jun 2018 06:40:56 +0000 (14:40 +0800)]
ALSA: hda/realtek - Fix the problem of two front mics on more machines
We have 3 more Lenovo machines, they all have 2 front mics on them,
so they need the fixup to change the location for one of two mics.
Among these 3 Lenovo machines, one of them has the same pin cfg as the
machine with subid 0x17aa3138, so use the pin cfg table to apply fixup
for them. The rest machines don't share the same pin cfg, so far use
the subid to apply fixup for them.
Fixes: a3dafb2200bf ("ALSA: hda/realtek - adjust the location of one mic") Cc: <stable@vger.kernel.org> Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 19 Jun 2018 10:45:53 +0000 (12:45 +0200)]
ALSA: hda - Sanity check of unexpected cap_sync_hook override
There are a couple of places setting cap_sync_hook in the codec
drivers, and they just overwrite the value. Add a sanity check via
WARN_ON() in case if an old non-NULL value is overridden and
forgotten.
Takashi Iwai [Tue, 19 Jun 2018 10:42:03 +0000 (12:42 +0200)]
ALSA: hda - Allow multiple ADCs for mic mute LED controls
Instead of refusing, allow the configuration with the multiple ADCs
(thus multiple capture switches) for enabling the mic mute LED.
This has been done for Sigmatel/IDT codecs, and we treat the OR-ed
values from all capture switches as the boolean condition.
Takashi Iwai [Tue, 19 Jun 2018 10:36:36 +0000 (12:36 +0200)]
ALSA: hda/conexant - Use the mic-mute LED helper
Convert to use the common helper for controlling the mic mute LED for
HP laptops, just as we've done for Realtek codecs. This will give the
mic mute LED enum as gratis.
Takashi Iwai [Tue, 19 Jun 2018 10:29:13 +0000 (12:29 +0200)]
ALSA: hda/realtek - Use the mic-mute LED helper for HP and others
Similar as the previous commit, convert to use the common helper for
controlling the mic mute LED for HP and other machines in the Realtek
codec driver, too. This will give the mic mute LED enum as gratis.
Takashi Iwai [Mon, 18 Jun 2018 15:28:45 +0000 (17:28 +0200)]
ALSA: hda - Use the common helper for thinkpad_acpi mic mute LED handling
Use the new common helper for setting up and controlling the mic mute
LED over thinkpad_acpi. This also provides a new mixer enum "Mic
Mute-LED Mode" (that was present only for Dell models), which allows
user to choose the mic mute LED behavior. For example, if you want
the mic mute LED turned on only while mic is on, choose "Follow
Capture" there.
Takashi Iwai [Mon, 18 Jun 2018 15:26:12 +0000 (17:26 +0200)]
ALSA: hda - Move mic mute LED helper to the generic parser
Move the code for setting up and controlling the mic mute LED hook
from dell-wmi helper to the generic parser, so that it can be referred
from the multiple driver codes.
Takashi Iwai [Thu, 21 Jun 2018 11:33:53 +0000 (13:33 +0200)]
ALSA: hda - Force to link down at runtime suspend on ATI/AMD HDMI
Henning Kühn reported that the discrete AMD GPU on his hybrid graphics
laptop no longer runtime-suspends due to the recent commit 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller").
The root cause is that the HDMI codec on AMD GPU doesn't support
CLKSTOP and EPSS, which are currently mandatory for powering down the
HD-audio link at runtime suspend. Because the HD-audio link is still
up, HD-audio controller driver blocks the transition to D3.
For addressing the regression, this patch adds a new flag to indicate
the forced link-down, and sets it for AMD HDMI codecs appropriately
in the codec driver.
Fixes: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller")
Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=106957 Reported-by: Lukas Wunner <lukas@wunner.de> Reported-and-tested-by: Henning Kühn <prg@cooco.de> Cc: <stable@vger.kernel.org> # v4.17+ Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Mon, 18 Jun 2018 12:07:55 +0000 (21:07 +0900)]
ALSA: firewire-motu: add support for Motu Traveler
This commit adds support for MOTU Traveler, launched in 2005, discontinued
quite before. As a result, transmission of PCM frame and MIDI messages is
available via ALSA PCM and RawMIDI/Sequencer interfaces.
This model supports sampling transmission frequency up to 192.0 kHz, and
AES/EBU on XLR interface and ADAT on optical interface. Unlike
Motu 828MkII, Windows driver can switch fetching mode for DSP, like
mute/unmute feature.
Although this commit enables high sampling transmission frequency, actual
sound from this model is not good. As long as I tested, it's silence at
176.4 kHz, and it includes hissing noise at 192.0 kHz. In my opinion, as I
reported at 3526ce7f9ba7 ('ALSA: firewire-motu: add MOTU specific protocol
layer'), timestamping on source packet header (SPH) may not still be good
for this model as well.
root directory
-----------------------------------------------------------------
414 0004c65c directory_length 4, crc 50780
418 030001f2 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 8d000006 --> eui-64 leaf at 438
424 d1000001 --> unit directory at 428
unit directory at 428
-----------------------------------------------------------------
428 00035955 directory_length 3, crc 22869
42c 120001f2 specifier id
430 13000009 version
434 17107800 model
Takashi Sakamoto [Mon, 18 Jun 2018 12:07:54 +0000 (21:07 +0900)]
ALSA: firewire-motu: cancel chunk alignment for protocol version 2
For MOTU protocol version 2, this driver arranges the number of data
chunks to align chunks to quadlet data channel. However, MOTU Traveler
has padding bytes in the end of data block at high clock mode.
This commit removes the arrangement. Fortunately, at low and middle clock
mode, supported model for v2 protocol (828mkII) gets no influence from this
change because all of combination for data chunks are just aligned to
quadlet data channel.
Takashi Sakamoto [Mon, 18 Jun 2018 12:07:53 +0000 (21:07 +0900)]
ALSA: firewire-motu: add a flag for AES/EBU on XLR interface
MOTU Traveler supports AES/EBU on XLR interface and data block of rx/tx
packet includes two chunk for the interface. This commit adds a flag
for this purpose.
Takashi Sakamoto [Mon, 18 Jun 2018 12:07:52 +0000 (21:07 +0900)]
ALSA: firewire-motu: add a flag for chunks for main 1/2 out
This driver explicitly assumes that all of supported models have main data
chunk separated from chunk for analog ports. However, MOTU Traveler doesn't
support the separated main data chunk.
This commit adds a flag for the separated main data chunk.
Takashi Sakamoto [Mon, 18 Jun 2018 12:07:51 +0000 (21:07 +0900)]
ALSA: firewire-motu: suppless consumption for unused element of array in stack
In MOTU firewire protocol, data block consists of 24 bit data chunks except
for one quadlet for source packet header (SPH). The number of data chunk in
a data block is different between three clock modes; low, middle and high.
When unit supports ADAT on optical interface, the data block includes some
chunks for ADAT channels. These ADAT chunks are unavailable at high mode.
This driver has local functions to calculate the number of ADAT chunks. But
They uses stack for three clock modes. This is useless for higher mode.
Arnd Bergmann [Mon, 18 Jun 2018 15:41:01 +0000 (17:41 +0200)]
ALSA: line6: stop using get_seconds()
The get_seconds() function is deprecated because it truncates the
timestamp to 32 bits, so all users should change to ktime_get_seconds()
or ktime_get_real_seconds().
The firmware interface for passing the timestamp is also limited to
32 bits, so this patch only has the cosmetic effect of avoiding the
old interface.
Stefan Agner [Sun, 17 Jun 2018 13:46:49 +0000 (15:46 +0200)]
ALSA: ctxfi: use enum type CT_SUM_CTL where appropriate
Currently a variable of type enum CT_AMIXER_CTL is used for enum
CT_SUM_CTL values. This leads to warnings when using clang:
sound/pci/ctxfi/ctmixer.c:945:32: warning: implicit conversion from enumeration type
'enum CT_SUM_CTL' to different enumeration type 'enum CT_AMIXER_CTL'
[-Wenum-conversion]
for (i = AMIXER_MASTER_F, j = SUM_IN_F;
~ ^~~~~~~~
sound/pci/ctxfi/ctmixer.c:975:29: warning: implicit conversion from enumeration type
'enum CT_SUM_CTL' to different enumeration type 'enum CT_AMIXER_CTL'
[-Wenum-conversion]
for (i = AMIXER_PCM_F, j = SUM_IN_F; i <= AMIXER_PCM_S; i++, j++) {
~ ^~~~~~~~
Introduce enum CT_SUM_CTL k and it instead.
Signed-off-by: Stefan Agner <stefan@agner.ch> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stefan Agner [Sun, 17 Jun 2018 13:46:29 +0000 (15:46 +0200)]
ALSA: ice1724: remove unused array
This fixes a warning seen with clang:
sound/pci/ice1712/prodigy_hifi.c:321:28: warning: variable 'wm_vol' is not
needed and will not be emitted [-Wunneeded-internal-declaration]
static const unsigned char wm_vol[256] = {
^
Signed-off-by: Stefan Agner <stefan@agner.ch> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: hda/ca0132: Fix DMic data rate for Alienware M17x R4
The commentary says to use various parameters, and lays out what
the mapping is... The code used a 32KHz rate when the comment
says that it needs to use a 48KHz rate. And this has been the
case since day one.
On the Alienware M17x R4, the DMic used to have exceptionally quiet
pickup and a lot of noise. Changing the data rate fixes both of
these issues.
Searching the kernel bug tracker for ca0132-related issues shows no
mention of this being an issue for other hardware, and I have no
other hardware to test with, so a quirk is used to limit the effect
to just the M17x R4.