Takashi Iwai [Thu, 24 Jan 2019 08:45:36 +0000 (09:45 +0100)]
ALSA: hda - Use standard device registration for beep
Currently the registration and free of beep input device was done
manually from the register and the disconnect callbacks of the
assigned codec object. This seems working in most cases, but this may
be a cause of some races at probe. Moreover, due to these manual
calls, the total code became unnecessarily lengthy.
This patch rewrites the beep registration code to follow the standard
sound device object style. This allows us reducing the code, in
addition to avoiding the nested device registration calls.
Takashi Iwai [Tue, 29 Jan 2019 13:14:51 +0000 (14:14 +0100)]
ALSA: hda/realtek - Apply ALC294 hp init also for S4 resume
The init sequence for ALC294 headphone stuff is needed not only for
the boot up time but also for the resume from hibernation, where the
device is switched from the boot kernel without sound driver to the
suspended image. Since we record the PM event in the device
power_state field, we can now recognize the call pattern and apply the
sequence conditionally.
Takashi Iwai [Tue, 29 Jan 2019 13:03:33 +0000 (14:03 +0100)]
ALSA: hda - Record the current power state before suspend/resume calls
Currently we deal with single codec and suspend codec callbacks for
all S3, S4 and runtime PM handling. But it turned out that we want
distinguish the call patterns sometimes, e.g. for applying some init
sequence only at probing and restoring from hibernate.
This patch slightly modifies the common PM callbacks for HD-audio
codec and stores the currently processed PM event in power_state of
the codec's device.power field, which is currently unused. The codec
callback can take a look at this event value and judges which purpose
it's being called.
Kailang Yang [Tue, 29 Jan 2019 07:38:21 +0000 (15:38 +0800)]
ALSA: hda/realtek - Fixed hp_pin no value
Fix hp_pin always no value.
[More notes on the changes:
The hp_pin value that is referred in alc294_hp_init() is always zero
at the moment the function gets called, hence this is actually
useless as in the current code.
And, this kind of init sequence should be called from the codec init
callback, instead of the parser function. So, the first fix in this
patch to move the call call into its own init_hook.
OTOH, this function is needed to be called only once after the boot,
and it'd take too long for invoking at each resume (where the init
callback gets called). So we add a new flag and invoke this only
once as an additional fix.
The one case is still not covered, though: S4 resume. But this
change itself won't lead to any regression in that regard, so we
leave S4 issue as is for now and fix it later. -- tiwai ]
Fixes: bde1a7459623 ("ALSA: hda/realtek - Fixed headphone issue for ALC700") Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jussi Laako [Mon, 28 Jan 2019 22:47:01 +0000 (00:47 +0200)]
ALSA: usb-audio: Cleanup DSD whitelist
XMOS/Thesycon family of USB Audio Class firmware flags DSD altsetting
separate from the PCM ones. Thus the DSD altsetting can be auto-detected
based on the flag and doesn't need maintaining specific altsetting
whitelist.
In addition, static VID:PID-to-altsetting whitelisting causes problems
when firmware update changes the altsetting, or same VID:PID is reused
for another device that has different kind of firmware.
This patch removes existing explicit whitelist mappings for XMOS VID
(0x20b1) and Thesycon VID (0x152a).
Also corrects placement of Hegel HD12 and NuPrime DAC-10 to keep list
sorted based on VID.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Mon, 28 Jan 2019 11:40:58 +0000 (20:40 +0900)]
ALSA: dice: add support for Solid State Logic Duende Classic/Mini
Duende Classic was produced by Solid State Logic in 2006, as a
first model of Duende DSP series. The following model, Duende Mini
was produced in 2008. They are designed to receive isochronous
packets for PCM frames via IEEE 1394 bus, perform signal processing by
downloaded program, then transfer isochronous packets for converted
PCM frames.
These two models includes the same embedded board, consists of several
ICs below:
- Texus Instruments Inc, TSB41AB3 for physical layer of IEEE 1394 bus
- WaveFront semiconductor, DICE II STD ASIC for link/protocol layer
- Altera MAX 3000A CPLD for programs
- Analog devices, SHARC ADSP-21363 for signal processing (4 chips)
This commit adds support for the two models to ALSA dice driver. Like
support for the other devices, packet streaming is just available.
Userspace applications should be developed if full features became
available; e.g. program uploader and parameter controller.
Takashi Iwai [Fri, 25 Jan 2019 16:11:32 +0000 (17:11 +0100)]
ALSA: pcm: Fix tight loop of OSS capture stream
When the trigger=off is passed for a PCM OSS stream, it sets the
start_threshold of the given substream to the boundary size, so that
it won't be automatically started. This can be problematic for a
capture stream, unfortunately, as detected by syzkaller. The scenario
is like the following:
- In __snd_pcm_lib_xfer() that is invoked from snd_pcm_oss_read()
loop, we have a check whether the stream was already started or the
stream can be auto-started.
- The function at this check returns 0 with trigger=off since we
explicitly disable the auto-start.
- The loop continues and repeats calling __snd_pcm_lib_xfer() tightly,
which may lead to an RCU stall.
This patch fixes the bug by simply allowing the wait for non-started
stream in the case of OSS capture. For native usages, it's supposed
to be done by the caller side (which is user-space), hence it returns
zero like before.
(In theory, __snd_pcm_lib_xfer() could wait even for the native API
usage cases, too; but I'd like to stay in a safer side for not
breaking the existing stuff for now.)
Olek Poplavsky [Fri, 25 Jan 2019 04:30:03 +0000 (23:30 -0500)]
ALSA: usb-audio: Add Opus #3 to quirks for native DSD support
This patch adds quirk VID/PID IDs for the Opus #3 DAP (made by 'The Bit')
in order to enable Native DSD support.
[ NOTE: this could be handled in the generic way with fp->dvd_raw if
we add 0x10cb to the vendor whitelist, but since 0x10cb shows a
different vendor name (Erantech), put to the individual entry at
this time -- tiwai ]
Takashi Iwai [Fri, 25 Jan 2019 16:31:59 +0000 (17:31 +0100)]
ALSA: pcm: Use the common error path in __snd_pcm_lib_xfer()
An open-coded error path in __snd_pcm_lib_xfer() can be replaced with
the simple goto to the common error path. This also makes the error
handling more consistent, i.e. when some samples have been already
processed, return that size instead of the error code.
Takashi Iwai [Wed, 23 Jan 2019 16:59:40 +0000 (17:59 +0100)]
ALSA: proc: Avoid possible leaks of snd_info_entry objects
This patch changes the parent pointer assignment of snd_info_entry
object to be always non-NULL. More specifically,check the parent
argument in snd_info_create_module_entry() & co, and assign
snd_proc_root if NULL is passed there.
This assures that the proc object is always freed when the root is
freed, so avoid possible memory leaks. For example, some error paths
(e.g. snd_info_register() error at snd_minor_info_init()) may leave
snd_info_entry object although the proc file itself is freed.
Takashi Iwai [Wed, 23 Jan 2019 15:44:38 +0000 (16:44 +0100)]
ALSA: pcm: Simplify proc file destruction
The proc files are recursively freed by calling with the root
snd_info_entry object, so we don't have to keep each object for
releasing one by one. Move the release of the PCM stream proc root at
the beginning, so that we can remove the redundant code and resource.
Takashi Sakamoto [Thu, 24 Jan 2019 09:32:03 +0000 (18:32 +0900)]
ALSA: fireface: comment cleanup about destination address of async transactions for MIDI messages
In Fireface series, registration of higher 4 bytes of destination
address for asynchronous transaction of MIDI messages is done by
a write transaction to model-specific register.
On the other hand, registration of lower 4 bytes of the address is
selectable from 4 options. A register for this registration includes
the other purpose options such as input attenuation. Thus this
driver expects userspace applications to configure the register.
Actual behaviour for the asynchronous transaction is different
depending on protocols. In former protocol, destination offset
of each transaction is the same as the registered address even if
it is block request. In latter models, destination offset of each
transaction is the offset of previous transaction plus 4 byte
and the transaction is quadlet request.
This commit cleanups comments about the above mechanism.
Takashi Iwai [Sun, 13 Jan 2019 09:15:03 +0000 (10:15 +0100)]
ALSA: pcm: Cleanup snd_pcm_stream_lock() & co
After the previous code refactoring, the PCM stream locking code
became nothing but the PCM group lock with self_group object. Use the
existing helper function for simplifying the code.
Takashi Iwai [Tue, 22 Jan 2019 13:29:51 +0000 (14:29 +0100)]
ALSA: pcm: Remove down_write() hack for snd_pcm_link_rwsem
Remove the hackish down_write_nonfifo() that was introduced as a
workaround of rwsem deadlock.
It used to be a problem for non-atomic PCM streams that take the rwsem
for the locking and hit the high lock contention. Since the current
PCM locking refactoring, we'll no longer hit it as the hot code-paths
don't take global locks.
Takashi Iwai [Sun, 13 Jan 2019 08:50:33 +0000 (09:50 +0100)]
ALSA: pcm: More fine-grained PCM link locking
We have currently two global locks, a rwlock and a rwsem, that are
used for managing linking the PCM streams. Due to these global locks,
once when a linked stream is used, the lock granularity suffers a
lot.
This patch attempts to eliminate the former global lock for atomic
ops. The latter rwsem needs remaining because of the loosy way of the
loop calls in snd_pcm_action_nonatomic(), as well as for avoiding the
deadlock at linking. However, these are used far rarely, actually
only by two actions (prepare and reset), where both are no timing
critical ones. So this can be still seen as a good improvement.
The basic strategy to eliminate the rwlock is to assure group->lock at
adding or removing a stream to / from the group. Since we already
takes the group lock whenever taking the all substream locks under the
group, this shouldn't be a big problem. The reference to group
pointer in snd_pcm_substream object is protected by the stream lock
itself.
However, there are still pitfalls: a race window at re-locking and the
lifecycle of group object. The former is a small race window for
dereferencing the substream group object opened while snd_pcm_action()
performs re-locking to avoid ABBA deadlocks. This includes the unlink
of group during that window, too. And the latter is the kfree
performed after all streams are removed from the group while it's
still dereferenced.
For addressing these corner cases, two new tricks are introduced:
- After re-locking, the group assigned to the stream is checked again;
if the group is changed, we retry the whole procedure.
- Introduce a refcount to snd_pcm_group object, so that it's freed
only when it's empty and really no one refers to it.
(Some readers might wonder why not RCU for the latter. RCU in this
case would cost more than refcounting, unfortunately. We take the
group lock sooner or later, hence the performance improvement by RCU
would be negligible. Meanwhile, because we need to deal with
schedulable context depending on the pcm->nonatomic flag, it'll become
dynamic RCU/SRCU switch, and the grace period may become too long.)
Along with these changes, there are a significant amount of code
refactoring. The complex group re-lock & ref code is factored out to
snd_pcm_stream_group_ref() function, for example.
Takashi Sakamoto [Tue, 22 Jan 2019 13:17:05 +0000 (22:17 +0900)]
ALSA: fireface: support rx MIDI functionality for Fireface UCX
In latter model of Fireface series, asynchronous transaction includes
a prefix byte to indicate the way to decode included MIDI bytes.
Upper 4 bits of the prefix byte indicates port number, and the rest 4
bits indicate the way to decode rest of bytes for MIDI messages.
Basically the rest bits indicates the number of bytes for MIDI message.
However, if the last byte of each MIDi message is included, the rest
bits are 0xf. For example:
This commit supports encoding scheme for the above and allows
applications to transfer MIDI messages via ALSA rawmidi interface.
An unused member (running_status) is reused to keep state of
transmission of system exclusive messages.
For your information, this is a dump of config rom.
Takashi Sakamoto [Tue, 22 Jan 2019 13:17:02 +0000 (22:17 +0900)]
ALSA: fireface: support tx MIDI functionality of Fireface UCX
Fireface UCX transfers asynchronous transactions for MIDI messages.
One transaction includes quadlet data therefore it can transfer 3
message bytes as maximum. Base address of the destination is
configured by two settings; a register for higher 8 byte of the
address, and a bitflag to option register indicates lower 8byte.
The register for higher address is 0x'ffff'0000'0034. Unfortunately,
firmware v24 includes a bug to ignore registered value for the
destination address and transfers to 0x0001xxxxxxxx always. This
driver doesn't work well if the bug exists, therefore users should
install the latest firmware (v27).
The bitflag is a part of value to be written to option register
(0x'ffff'0000'0014).
This register includes more options but they are not relevant to
packet streaming or MIDI functionality. This driver don't touch it.
Furthermore, the transaction is sent to address offset incremented
by 4 byte to the offset in previous time. When it reaches base address
plus 0x7c, next offset is the base address.
Content of the transaction includes a prefix byte. Upper 4 bits of
the byte indicates port number, and the rest 4 bits indicate the way
to decode rest of bytes for MIDI message.
Except for system exclusive messages, the rest bits are the same as
status bits of the message without channel bits. For system exclusive
messages, the rest bits are encoded according to included message bytes.
For example:
message: f0 7e 7f 09 01 f7
offset: content (little endian, port 0)
'0000: 0x04f07e7f
'0004: 0x070901f7
This commit supports decoding scheme for the above and allows
applications to receive MIDI messages via ALSA rawmidi interface.
The lower 8 bytes of destination address is fixed to 0x'0000'0000,
thus this driver expects userspace applications to configure option
register with bitflag 0x00002000 in advance.
Takashi Sakamoto [Tue, 22 Jan 2019 13:17:01 +0000 (22:17 +0900)]
ALSA: fireface: add model-dependent parameter for address range to receive async transaction
In Fireface series, drivers can register destination address for
asynchronous transaction which transfers MIDI messages from device.
In former models, all of the transactions arrive at the registered
address without any offset. In latter models, each of the transaction
arrives at the registered address with sequential offset within 0x00
to 0x7f. This seems to be for discontinuity detection.
This commit adds model-dependent member for the address range.
Takashi Sakamoto [Tue, 22 Jan 2019 13:17:00 +0000 (22:17 +0900)]
ALSA: fireface: change prototype of handler for async transaction with MIDI messages
In a series of Fireface, devices transfer asynchronous transaction with
MIDI messages. In the transaction, content is different depending on
models. ALSA fireface driver has protocol-dependent handler to pick up
MIDI messages from the content.
In latter models of the series, the transaction is transferred to range
of address sequentially. This seems to check continuity of transferred
messages.
This commit changes prototype of the handler to receive offset of
address for received transactions.
Sameer Pujar [Tue, 22 Jan 2019 07:33:21 +0000 (13:03 +0530)]
ALSA: hda/tegra: add driver flag for runtime PM
AZX_DCAPS_PM_RUNTIME flag is added to indicate support for runtime PM.
azx_has_pm_runtime() is used to check if above is enabled and thus
forbid runtime PM calls if needed.
Sameer Pujar [Tue, 22 Jan 2019 07:33:20 +0000 (13:03 +0530)]
ALSA: hda/tegra: implement runtime suspend/resume
This patch moves clock enable/disable from system resume/suspend to
runtime resume/suspend respectively. Along with this hda controller
chip init or stop is also moved. System resume/suspend can invoke
runtime callbacks and do necessary setup.
chip->running can be used to check for probe completion and device
access during runtime_resume or runtime_suspend can be avoided if
probe is not yet finished. This helps to avoid kernel panic during
boot where runtime PM callbacks can happen from system PM.
Sameer Pujar [Tue, 22 Jan 2019 07:33:19 +0000 (13:03 +0530)]
ALSA: hda/tegra: remove redundant clock enable API
Explicit clock enable is not required during probe, as this would be
managed by runtime PM calls. Clock can be enabled/disabled in runtime
resume/suspend. This way it is easier to balance clock enable/disable
counts.
Sameer Pujar [Tue, 22 Jan 2019 07:33:17 +0000 (13:03 +0530)]
ALSA: hda/tegra: get clock handles early in probe
Moved devm_clk_get() API calls to a separate function and the same
can be called early in the probe. This is done before runtime PM
for the device is enabled. The runtime resume/suspend callbacks can
later enable/disable clocks respectively(the support would be added
in subsequent patches). Clock handles should be available by the
time runtime suspend/resume calls can happen.
Sameer Pujar [Tue, 22 Jan 2019 07:33:16 +0000 (13:03 +0530)]
ALSA: hda/tegra: runtime power management support
This patch enables runtime power management(runtime PM) support for
hda. pm_runtime_enable() and pm_runtime_disable() are added during
device probe and remove respectively. The runtime PM callbacks will
be forbidden if hda controller does not have support for runtime PM.
pm_runtime_get_sync() and pm_runtime_put() are added for hda register
access. The callbacks for above will be added in subsequent patches.
Takashi Iwai [Sun, 13 Jan 2019 08:40:21 +0000 (09:40 +0100)]
ALSA: pcm: Make PCM linked list consistent while re-grouping
Make a common helper to re-assign the PCM link using list_move() instead
of open code with manual list_del() and list_add_tail(). This assures
the consistency and we can get rid of snd_pcm_group.count field -- its
purpose is only to check whether the list is singular, and we can know
it by list_is_singular() call now.
The snd_card_unref() call in snd_pcm_link() looks suspicious through a
quick glance, but it's a correct usage; this is needed just because
the file descriptor check in is_pcm_file() calls the helper
snd_lookup_minor_data() that keeps the card refcount.
Despite of the correctness, the code still looks confusing.
Basically, keeping the card ref for the whole code isn't needed
as fdget() blocks the release of the opened file. Hence it's more
understandable if snd_card_unref() is moved into is_pcm_file(), then
the caller doesn't have to take care after the call.
Takashi Sakamoto [Sun, 20 Jan 2019 08:25:53 +0000 (17:25 +0900)]
ALSA: fireface: add support for Fireface UCX
Fireface UFX was shipped by RME GmbH in 2012. This model supports later
protocol for management of isochronous communication and synchronization
of sampling transmission frequency.
This commit adds support for the model. At present, it's not clear how
to encode MIDI messages and decide destination address for asynchronous
transaction, thus this commit adds support for isochronous communication
for PCM frames only.
Takashi Sakamoto [Sun, 20 Jan 2019 08:25:47 +0000 (17:25 +0900)]
ALSA: fireface: obsolete proc node to leave one node
In a series of Fireface, latter protocol has no way for drivers to
retrieve current clock configuration. On the other hand, this driver
has proc node for it.
This commit removes a proc node to dump both clock configuration
and synchronization status in one proc node.
Takashi Iwai [Mon, 21 Jan 2019 08:10:00 +0000 (09:10 +0100)]
ALSA: hda - Fix unused variable warning
The unused variable was forgotten to be removed and now we get a
compiler warning:
sound/pci/hda/hda_codec.c: In function 'hda_codec_runtime_suspend':
sound/pci/hda/hda_codec.c:2926:18: warning: unused variable 'pcm'
Takashi Iwai [Fri, 18 Jan 2019 16:37:14 +0000 (17:37 +0100)]
Merge branch 'topic/pcm-device-suspend' into for-next
Pull the PCM suspend improvement / cleanup.
This moves the most of snd_pcm_suspend*() calls into PCM's own device
PM ops. There should be no change from the functionality POV.
Takashi Iwai [Fri, 18 Jan 2019 14:17:17 +0000 (15:17 +0100)]
Merge tag 'asoc-fix-v5.0-rc2' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
Quite a big batch of fixes here. There's a couple of things going on,
the main one is that we found some issues with not deferring probe when
we should, causing us to skip some driver initialization. The fixes for
this then in turn exposed some issues with how we were searching for
components which had previously gone unnoticed due to the original
issue.
There's also been the normal driver specific stuff and there's been what
looks like several batches of automated scanning for issues which have
generated quite a large set of smaller fixes for potential crashes and
missed error handling.
Check return value from call to devm_kzalloc() in order to prevent a
potential NULL pointer dereference.
Also, notice that it makes no sense to allocate any resources if
res = platform_get_resource(pdev, IORESOURCE_MEM, 0); fails,
so move the call to devm_kzalloc() below the mentioned code.
Lastly, improve the use of sizeof in the call to devm_kzalloc() by
changing it from sizeof(struct i2s_dev_data) to sizeof(*adata)
This issue was detected with the help of Coccinelle.
Fixes: ac289c7ec0bc ("ASoC: amd: add ACP3x PCM platform driver") Cc: stable@vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Silvio Cesare [Tue, 15 Jan 2019 03:27:27 +0000 (04:27 +0100)]
ASoC: imx-audmux: change snprintf to scnprintf for possible overflow
Change snprintf to scnprintf. There are generally two cases where using
snprintf causes problems.
1) Uses of size += snprintf(buf, SIZE - size, fmt, ...)
In this case, if snprintf would have written more characters than what the
buffer size (SIZE) is, then size will end up larger than SIZE. In later
uses of snprintf, SIZE - size will result in a negative number, leading
to problems. Note that size might already be too large by using
size = snprintf before the code reaches a case of size += snprintf.
2) If size is ultimately used as a length parameter for a copy back to user
space, then it will potentially allow for a buffer overflow and information
disclosure when size is greater than SIZE. When the size is used to index
the buffer directly, we can have memory corruption. This also means when
size = snprintf... is used, it may also cause problems since size may become
large. Copying to userspace is mitigated by the HARDENED_USERCOPY kernel
configuration.
The solution to these issues is to use scnprintf which returns the number of
characters actually written to the buffer, so the size variable will never
exceed SIZE.
Signed-off-by: Silvio Cesare <silvio.cesare@gmail.com> Cc: Timur Tabi <timur@kernel.org> Cc: Nicolin Chen <nicoleotsuka@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Xiubo Li <Xiubo.Lee@gmail.com> Cc: Fabio Estevam <fabio.estevam@nxp.com> Cc: Dan Carpenter <dan.carpenter@oracle.com> Cc: Kees Cook <keescook@chromium.org> Cc: Will Deacon <will.deacon@arm.com> Cc: Greg KH <greg@kroah.com> Signed-off-by: Willy Tarreau <w@1wt.eu> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Reviewed-by: Kees Cook <keescook@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
There is a potential NULL pointer dereference in case devm_kzalloc()
fails and returns NULL.
Fix this by adding a NULL check on rt5514_dsp.
This issue was detected with the help of Coccinelle.
Fixes: 6eebf35b0e4a ("ASoC: rt5514: add rt5514 SPI driver") Cc: stable@vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Silvio Cesare [Sat, 12 Jan 2019 15:28:43 +0000 (16:28 +0100)]
ASoC: dapm: change snprintf to scnprintf for possible overflow
Change snprintf to scnprintf. There are generally two cases where using
snprintf causes problems.
1) Uses of size += snprintf(buf, SIZE - size, fmt, ...)
In this case, if snprintf would have written more characters than what the
buffer size (SIZE) is, then size will end up larger than SIZE. In later
uses of snprintf, SIZE - size will result in a negative number, leading
to problems. Note that size might already be too large by using
size = snprintf before the code reaches a case of size += snprintf.
2) If size is ultimately used as a length parameter for a copy back to user
space, then it will potentially allow for a buffer overflow and information
disclosure when size is greater than SIZE. When the size is used to index
the buffer directly, we can have memory corruption. This also means when
size = snprintf... is used, it may also cause problems since size may become
large. Copying to userspace is mitigated by the HARDENED_USERCOPY kernel
configuration.
The solution to these issues is to use scnprintf which returns the number of
characters actually written to the buffer, so the size variable will never
exceed SIZE.
Signed-off-by: Silvio Cesare <silvio.cesare@gmail.com> Cc: Liam Girdwood <lgirdwood@gmail.com> Cc: Mark Brown <broonie@kernel.org> Cc: Dan Carpenter <dan.carpenter@oracle.com> Cc: Kees Cook <keescook@chromium.org> Cc: Will Deacon <will.deacon@arm.com> Cc: Greg KH <greg@kroah.com> Signed-off-by: Willy Tarreau <w@1wt.eu> Signed-off-by: Mark Brown <broonie@kernel.org>
Matthias Reichl [Tue, 15 Jan 2019 16:51:07 +0000 (17:51 +0100)]
ASoC: core: Don't defer probe on optional, NULL components
cpu and platform are optional components in DAI links. For example
codec-codec links usually have no platform set.
Call snd_soc_find_component only if the name or of_node of
a cpu or platform is set. Otherwise it will return NULL and
soc_init_dai_link bails out immediately with -EPROBE_DEFER,
meaning registering a card with NULL cpu or platform in DAI links
can never succeed.
Fixes: 8780cf1142a5 ("ASoC: soc-core: defer card probe until all component is added to list") Signed-off-by: Matthias Reichl <hias@horus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Tue, 15 Jan 2019 09:49:47 +0000 (10:49 +0100)]
ALSA: doc: Update the description about PCM suspend procedure
The PCM suspend procedure was changed for drivers, so that they don't
have to call snd_pcm_suspend*() in each callback any longer. Update
the documentation to adapt the changes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 11 Jan 2019 16:57:29 +0000 (17:57 +0100)]
ALSA: atiixp: Move PCM suspend/resume code into trigger callback
ATIIXP driver supports the full PCM resume and saves/restores the
running PCM pointer. This used to be done in the suspend and resume
callbacks together with snd_pcm_suspend() call. But since we moved
the snd_pcm_supsend*() call in PCM device PM ops, this should be moved
to a more appropriate place, i.e. the trigger callback.
Along with the movement of the PCM suspend/resume code, remove the
superfluous snd_pcm_suspend_all() call, too.
Reviewed-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 11 Jan 2019 14:58:39 +0000 (15:58 +0100)]
ALSA: pcm: Suspend streams globally via device type PM ops
Until now we rely on each driver calling snd_pcm_suspend*() explicitly
at its own PM handling. However, this can be done far more easily by
setting the PM ops to each actual snd_pcm device object.
This patch adds the device_type object for PCM stream and assigns to
each PCM stream object. The type contains only the PM ops for system
suspend; we don't need to deal with the resume in general.
The suspend hook simply calls snd_pcm_suspend_all() for the given PCM
streams. This implies that the PM order is correctly put, i.e. PCM is
suspended before the main (or codec) driver, which should be true in
general. If a special ordering is needed, you'd need to adjust the
device PM order manually later.
This patch introduces a new flag, snd_pcm.no_device_suspend, too.
With this flag set, the PCM device object won't invoke
snd_pcm_suspend_all() by itself. This is needed for ASoC who wants to
manage the PM call orders in its serialized way, and the flag is set
in soc_new_pcm() as default.
For the non-ASoC world, we can get rid of the manual snd_pcm_suspend
calls. This will be done in the later patches.
Reviewed-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Mon, 14 Jan 2019 23:29:36 +0000 (23:29 +0000)]
ASoC: core: Make snd_soc_find_component() more robust
There are some use cases where you're checking for a lot of things on a
card and it makes sense that you might end up trying to call
snd_soc_find_component() without either a name or an of_node. Currently
in that case we try to dereference the name and crash but it's more
useful to allow the caller to just treat that as a case where we don't
find anything, that error handling will already exist.
Inspired by a patch from Ajit Pandey fixing some callers.
Fixes: 8780cf1142a5 ("ASoC: soc-core: defer card probe until all component is added to list") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Curtis Malainey [Fri, 11 Jan 2019 00:21:04 +0000 (16:21 -0800)]
ASoC: soc-core: fix init platform memory handling
snd_soc_init_platform initializes pointers to snd_soc_dai_link which is
statically allocated and it does this by devm_kzalloc. In the event of
an EPROBE_DEFER the memory will be freed and the pointers are left
dangling. snd_soc_init_platform sees the dangling pointers and assumes
they are pointing to initialized memory and does not reallocate them on
the second probe attempt which results in a use after free bug since
devm has freed the memory from the first probe attempt.
Since the intention for snd_soc_dai_link->platform is that it can be set
statically by the machine driver we need to respect the pointer in the
event we did not set it but still catch dangling pointers. The solution
is to add a flag to track whether the pointer was dynamically allocated
or not.
Signed-off-by: Curtis Malainey <cujomalainey@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Sameer Pujar [Mon, 14 Jan 2019 18:21:12 +0000 (23:51 +0530)]
ALSA: hda: program stripe control for codec
Program codec stripe through AC_VERB_SET_STRIPE_CONTROL to use multiple
sdo lines if supported. Audio needs to be striped across number of sdo
lines for simultaneous playbacks of higher resolutions to work.
This needs to be implemented only for an Audio Output Converter and only
if the stripe bit(AC_WCAP_STRIPE) of Audio Widget Capabilities parameter
is 1.
Sameer Pujar [Mon, 14 Jan 2019 18:21:11 +0000 (23:51 +0530)]
ALSA: hda: program stripe bits for controller
Platforms having multiple SORs and hdmi/dp sinks require higher
bandwidth to support simultaneous playbacks of higher resolution.
If hda controller supports multiple SDO lines, STRIPE can be used
to indicate how many of the SDO lines the stream should be striped
across.
During stream start stripe control bits are programmed to use given
number of sdo lines and the same is cleared during stream stop.
Sameer Pujar [Mon, 14 Jan 2019 18:21:10 +0000 (23:51 +0530)]
ALSA: hda: add register offset for stripe control
bits 16:17 in SD_CTL register refer to stripe control. Added an
offset register(AZX_REG_SD_CTL_3B) to have exclusive read/write
of corresponding register byte. This helps to avoid unnecessary
32-bit read/write of SD_CTL whenever only stripe or other bits of
corresponding byte need to be updated. Also HD audio spec defines
SD_CTL as 3 byte register.
SD_CTL_STRIPE_MASK(0x3) can be used for stripe control programming
and when updating AZX_REG_SD_CTL_3B.
Sameer Pujar [Mon, 14 Jan 2019 18:21:09 +0000 (23:51 +0530)]
ALSA: hda: Add api to program stripe control bits
Controllers and codecs can support striping of audio out across
multiple SDO lines. The number of supported SDO lines can be
specific to chip. GCAP register can be read to know the maximum
supported SDO lines.
snd_hdac_get_stream_stripe_ctl() is exposed to program stripe bits
on controller and codec side.
stripe value: 0 for 1SDO, 1 for 2SDO, 2 for 4SDO lines, etc.,
Sameer Pujar [Mon, 14 Jan 2019 18:21:08 +0000 (23:51 +0530)]
ALSA: hda: add verbs for stripe control
Controllers can support multiple Serial Data Out(SDO) lines, for
extended outbound bandwidth, to pump data to all codecs on the link.
Codecs can sample data present on SDO.
Add verbs AC_VERB_GET_STRIPE_CONTROL and AC_VERB_SET_STRIPE_CONTROL
These can be used to program usage of SDO lines for codec.
Takashi Iwai [Tue, 8 Jan 2019 10:37:31 +0000 (11:37 +0100)]
ASoC: intel: skl: Fix display power regression
Since the refactoring of HD-audio display power management, the
display power status is managed per domain. Meanwhile the ASoC
hdac_hdmi driver still keeps and relies (incorrectly) on the
refcounting together with ASoC skl driver, and this leads to the
display state always on.
This patch is an attempt to address the regression by simplifying the
PM code of ASoC skl and hdac_hdmi drivers. Basically, since the
refactoring, we don't have to manage the display power at HD-audio
controller suspend / resume but only at HD-audio HDMI codec suspend /
resume. So the patch drops the superfluous snd_hdac_display_power()
calls in skl driver.
Meanwhile, in hdac_hdmi side, we rewrite the PM call just to re-use
the runtime PM callbacks like other drivers do. Now the logic is
simple: turn off at suspend and turn on at resume.
The patch also fixes the possibly missing display-power off at skl
driver removal as well as some error paths at probe.
Fixes: 029d92c289bd ("ALSA: hda: Refactor display power management") Reported-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rohit kumar [Thu, 10 Jan 2019 09:02:41 +0000 (14:32 +0530)]
ASoC: soc-core: Hold client_mutex around soc_init_dai_link()
soc_init_dai_link() calls soc_find_component() which needs
to be within client_mutex lock. Add client_mutex lock around
soc_init_dai_link() in snd_soc_register_card() to avoid
lockdep warning.
Fixes: 8780cf1142a5 ("ASoC: soc-core: defer card probe until all component is added to list") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Rohit kumar <rohitkr@codeaurora.org> Signed-off-by: Ajit Pandey <ajitp@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
Mac Chiang [Wed, 5 Dec 2018 10:11:19 +0000 (18:11 +0800)]
ASoC: Intel: Boards: move the codec PLL configuration to _init
move the codec PLL to rt5682_codec_init, because codec only need to config the clock source/PLL once.
As the result, remove the platform_clock_controls since no need to control clock anymore.
Signed-off-by: Shuming Fan <shumingf@realtek.com> Signed-off-by: Mac Chiang <mac.chiang@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
Ajit Pandey [Wed, 9 Jan 2019 08:47:07 +0000 (14:17 +0530)]
ASoC: soc-core: defer card probe until all component is added to list
DAI component probe is not called if it is not present
in component list during sound card registration.
Check if component is available in component list for
platform and cpu dai before soundcard registration.
Signed-off-by: Ajit Pandey <ajitp@codeaurora.org> Signed-off-by: Rohit kumar <rohitkr@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
YueHaibing [Wed, 9 Jan 2019 02:20:16 +0000 (02:20 +0000)]
ALSA: usb-audio: Remove set but not used variable 'first_ch_bits'
Fixes gcc '-Wunused-but-set-variable' warning:
sound/usb/mixer.c: In function 'parse_audio_feature_unit':
sound/usb/mixer.c:1838:28: warning:
variable 'first_ch_bits' set but not used [-Wunused-but-set-variable]
Keyon Jie [Wed, 9 Jan 2019 08:20:51 +0000 (16:20 +0800)]
ALSA: hda: Fix a mask wrong issue in snd_hdac_stream_start()
To enable SIE(Stream Interrupt Enable) in snd_hdac_stream_start(), we
should set both mask and value to be "1 << azx_dev->index" for register
update, the mask was 0, here fix it.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keyon Jie [Wed, 9 Jan 2019 08:20:50 +0000 (16:20 +0800)]
ALSA: hda: Fix mismatches for register mask and value in hdac controller
E.g. for azx_int_enable(), we should set both mask and value to be
"AZX_INT_CTRL_EN | AZX_INT_GLOBAL_EN"(the mask was 0) to enable
controller CIE and GIE.
We have similar issues on setting AZX_GCTL_RESET and AZX_GCTL_UNSOL,
here try to correct all of them.
Signed-off-by: Keyon Jie <yang.jie@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kangjie Lu [Wed, 26 Dec 2018 02:29:48 +0000 (20:29 -0600)]
ASoC: atom: fix a missing check of snd_pcm_lib_malloc_pages
snd_pcm_lib_malloc_pages() may fail, so let's check its status and
return its error code upstream.
Signed-off-by: Kangjie Lu <kjlu@umn.edu> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
b-ak [Mon, 7 Jan 2019 17:00:22 +0000 (22:30 +0530)]
ASoC: tlv320aic32x4: Kernel OOPS while entering DAPM standby mode
During the bootup of the kernel, the DAPM bias level is in the OFF
state. As soon as the DAPM framework kicks in it pushes the codec
into STANDBY state.
The probe function doesn't prepare the clock, and STANDBY state
does a clk_disable_unprepare() without checking the previous state.
This leads to an OOPS.
Not transitioning from an OFF state to the STANDBY state fixes the
problem.
Signed-off-by: b-ak <anur.bhargav@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
Hui Peng [Tue, 25 Dec 2018 23:11:52 +0000 (18:11 -0500)]
ALSA: usb-audio: Fix an out-of-bound read in create_composite_quirks
In `create_composite_quirk`, the terminating condition of for loops is
`quirk->ifnum < 0`. So any composite quirks should end with `struct
snd_usb_audio_quirk` object with ifnum < 0.
for (quirk = quirk_comp->data; quirk->ifnum >= 0; ++quirk) {
.....
}
the data field of Bower's & Wilkins PX headphones usb device device quirks
do not end with {.ifnum = -1}, wihch may result in out-of-bound read.
This Patch fix the bug by adding an ending quirk object.
Takashi Iwai [Wed, 2 Jan 2019 16:12:21 +0000 (17:12 +0100)]
ALSA: usb-audio: Always check descriptor sizes in parser code
There are a few places where we access the data without checking the
actual object size from the USB audio descriptor. This may result in
OOB access, as recently reported.
This patch addresses these missing checks. Most of added codes are
simple bLength checks in the caller side. For the input and output
terminal parsers, we put the length check in the parser functions.
For the input terminal, a new argument is added to distinguish between
UAC1 and the rest, as they treat different objects.