More updates for v3.20:
- Lots of refactoring from Lars-Peter Clausen, moving drivers to more
data driven initialization and rationalizing a lot of DAPM usage.
- Much improved handling of CDCLK clocks on Samsung I2S controllers.
- Lots of driver specific cleanups and feature improvements.
- CODEC support for TI PCM514x and TLV320AIC3104 devices.
- Board support for Tegra systems with Realtek RT5677.
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Merge tag 'asoc-v3.20-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.20
More updates for v3.20:
- Lots of refactoring from Lars-Peter Clausen, moving drivers to more
data driven initialization and rationalizing a lot of DAPM usage.
- Much improved handling of CDCLK clocks on Samsung I2S controllers.
- Lots of driver specific cleanups and feature improvements.
- CODEC support for TI PCM514x and TLV320AIC3104 devices.
- Board support for Tegra systems with Realtek RT5677.
Conflicts:
sound/soc/intel/sst-mfld-platform-pcm.c
Since the device is no longer hidden but embedded into each component,
we no longer need snd_get_device(). Let's drop it and relevant codes.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that all callers have been replaced with
snd_device_register_for_dev(), let's drop the obsolete device
registration code and concentrate only on the code handling struct
device directly. That said,
- remove the old snd_device_register(),
- rename snd_device_register_for_dev() with snd_device_register(),
- drop superfluous arguments from snd_device_register(),
- change snd_unregister_device() to pass the device pointer directly
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like previous patches, this one embeds the struct device into struct
snd_compr. As the dev field wasn't used beforehand, it's reused as
the new device struct.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous change for the timer device, this patch changes the
device management for the ALSA sequencer device using the struct
device directly.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a relatively straightforward change, using the struct device
directly for managing the ALSA timer device.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... instead of card's device. This will be helpful to distinguish
errors from multiple rawmidi devices on a single card.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like previous patches, this changes the device management for rawmidi,
embedding the struct device into struct snd_rawmidi. The required
change is more or less same as hwdep device.
The currently unused dev field is reused as the new embedded struct
field now.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like previous patches, at this time we embed the struct device into
PCM object. However, this needs a bit more caution: struct snd_pcm
doesn't own one device but two, for both playback and capture! Thus
not struct snd_pcm but struct snd_pcm_str object contains the device.
Along with this change, pcm->dev field is dropped for avoiding
confusion. It was meant to point to a non-standard parent. But,
since now we can touch each struct device directly, we can manipulate
the parent field easily there, too.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous patch, this one embeds the device object into hwdep
object. For a proper object lifecycle, it's freed in the release
callback.
This also allows us to create sysfs entries via passing to the groups
field of the device without explicit function calls. Since each
driver can see the device and touch its groups field directly, we
don't need to delegate in hwdep core any longer. So, remove the
groups field from snd_hwdep, and let the user (in this case only
hda_hwdep.c) modify the device groups.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch embeds a struct device for the control device into the card
object and avoid the device creation at registration time.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new helper function snd_device_initialize() to initialize
the device object for sound devices.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of open-coding the search over the control file loop, provide
a helper function for the preferred subdevice assigned to the current
process.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for the further work on embedding struct
device into each sound device instance. It changes
snd_register_device*() helpers to receive the device object directly
for skipping creating a device there.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add helper functions to allow drivers to specify several disjoint
ranges for a variable. In particular, there is a codec (PCM512x) that
has a hole in its supported range of rates, due to PLL and divider
restrictions.
This is like snd_pcm_hw_constraint_list(), but for ranges instead of
points.
Signed-off-by: Peter Rosin <peda@axentia.se>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Due to SNDRV_SEQ_ADDRESS_BROADCAST, not all 256 port number values can
be used.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the last subscriber to a "Through" port has been removed, the
subscribed destination ports might still be active, so it would be
wrong to send "all sounds off" and "reset controller" events to them.
The proper place for such a shutdown would be the closing of the actual
MIDI port (and close_substream() in rawmidi.c already can do this).
This also fixes a deadlock when dummy_unuse() tries to send events to
its own port that is already locked because it is being freed.
Reported-by: Peter Billam <peter@www.pjb.com.au>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_midi_event_free() function tests whether its argument is NULL and then
returns immediately. Thus the test around the call is not needed.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain.
Some audio devices require notification of drain events
in order to properly drain and shutdown an audio stream.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of opencoding them use the standard roundup_pow_of_two() and
rounddown_pow_of_two() helper functions. This gets rids one of the few users
of the custom ld2() function and also makes it a bit more obvious what the
code does.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the msbits constraints requires to specify a specific sample
format width for which the constraint should be applied. But often the
number of most significant bits is not sample format specific, but rather a
absolute limit. E.g. the PCM interface might accept 32-bit and 24-bit
samples, but the DAC has a 16-bit resolution and throws away the LSBs. In
this case for both 32-bit and 24-bit format msbits should be set to 16. This
patch extends snd_pcm_hw_constraint_msbits() so that a wildcard constraint
can be setup that is applied for all formats with a sample width larger than
the specified msbits. Choosing the wildcard constraint is done by setting
the sample width parameter of the function to 0.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the sound card is made up of discrete components, each with their own
driver (e.g. like in the ASoC case), we might end up with multiple msbits
constraint rules installed. Currently this will result in msbits being set
to whatever the last rule set it to.
This patch updates the behavior of the rule to choose the minimum (other
than zero) of all the installed rules.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit [7a2e9ddc: ALSA: usb-audio: Add native DSD support for
Denon/Marantz DACs] requires the new format definition that has
landed only in for-next branch.
The functions snd_seq_oss_timer_delete() and vunmap() perform also input
parameter validation. Thus the test around the call is not needed.
This issue was detected by using the Coccinelle software.
Signed-off-by: Markus Elfring <elfring@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds
DSD_U16_BE and DSD_U32_BE sample formats.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Acked-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a copy & paste error:
Warning(sound/core/pcm_native.c:1112): Excess function parameter 'state' description in 'snd_pcm_stop_xrun'
The state argument was dropped from snd_pcm_stop_xrun().
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper function snd_pcm_stop_xrun() to the standard sequnce
lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the
existing open codes with this helper.
The function checks the PCM running state to prevent setting the wrong
state, too, for more safety.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to know the offset for the id that was passed to the function, not
the offset of the first id of the control (which is always 0).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some state changes (e.g. snd_pcm_stop()) sets the runtime state after
calling snd_timer_notify(). This is basically racy, since the
notification may wakes up the user even before the state change.
Although the possibility is low, we should set the state before the
notifications.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This merges the USB-audio disconnect fix and resolves the conflicts
so that we can continue working on development of usb-audio stuff.
Conflicts:
sound/usb/card.c
This patch adds a new proc entry for PCM substreams to inject an
XRUN. When a PCM substream is running and any value is written to its
xrun_injection proc file, the driver triggers XRUN. This is a useful
feature for debugging XRUN and error handling code paths.
Note that this entry is enabled only when CONFIG_SND_PCM_XRUN_DEBUG is
set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA PCM core has a mechanism tracking the PCM hwptr updates for
analyzing XRUNs. But its log is limited (up to 10) and its log output
is a kernel message, which is hard to handle.
In this patch, the hwptr logging is moved to the tracing
infrastructure instead of its own. Not only the hwptr updates but
also XRUN and hwptr errors are recorded on the trace log, so that user
can see such events at the exact timing.
The new "snd_pcm" entry will appear in the tracing events:
# ls -F /sys/kernel/debug/tracing/events/snd_pcm
enable filter hw_ptr_error/ hwptr/ xrun/
The hwptr is for the regular hwptr update events. An event trace
looks like:
aplay-26187 [004] d..3 4012.834761: hwptr: pcmC0D0p/sub0: POS: pos=488, old=0, base=0, period=1024, buf=16384
"POS" shows the hwptr update by the explicit position update call and
"IRQ" means the hwptr update by the interrupt,
i.e. snd_pcm_period_elapsed() call. The "pos" is the passed
ring-buffer offset by the caller, "old" is the previous hwptr, "base"
is the hwptr base position, "period" and "buf" are period- and
buffer-size of the target PCM substream.
(Note that the hwptr position displayed here isn't the ring-buffer
offset. It increments up to the PCM position boundary.)
The XRUN event appears similarly, but without "pos" field.
The hwptr error events appear with the PCM identifier and its reason
string, such as "Lost interrupt?".
The XRUN and hwptr error reports on kernel message are still left, can
be turned on/off via xrun_debug proc like before. But the bit 3, 4, 5
and 6 bits of xrun_debug proc are dropped by this patch. Also, along
with the change, the message strings have been reformatted to be a bit
more consistent.
Last but not least, the hwptr reporting is enabled only when
CONFIG_SND_PCM_XRUN_DEBUG is set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While converting to dev_*(), the message showing the invalid PCM
position was wrongly tagged as if an XRUN although it's actually a
BUG. This patch corrects the message again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function snd_pcm_action_lock_irq() can be much simplified by
simply wrapping snd_pcm_action() with the stream lock. This was
rather the original idea, but later it was open coded for
optimization. However, looking at the optimization part closely, one
notices that the probability of the optimized path is quite low; in
normal situations, the linked stream action happens only for the
triggered substream, thus the operation becomes identical. So the
code simplification has a clear win, especially because we have now
doubly codes for both atomic and non-atomic locks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few functions have no proper documentation yet, so let's add them.
Along with it, remove superfluous blank line between the closing brace
and EXPORT_SYMBOL() line.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In compat mode, we copy each field of snd_pcm_status struct but don't
touch the reserved fields, and this leaves uninitialized values
there. Meanwhile the native ioctl does zero-clear the whole
structure, so we should follow the same rule in compat mode, too.
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On x86, using dma_mmap_coherent() for the pages allocated via
dma_alloc_coherent() results in a warning like:
aplay:32536 map pfn RAM range req uncached-minus for [mem 0x21d500000-0x21d51ffff], got write-back
Until the issue is addressed in the core side, take back to the old
good way in PCM code only for x86.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
this is a series of patches to just convert the plain info callback
for enum ctl elements to snd_ctl_elem_info(). Also, it includes the
extension of snd_ctl_elem_info(), for catching the unexpected string
cut-off and handling the zero items.
Some architectures like PARISC is known not to support mmap properly
with the DMA buffer, where dma_mmap_coherent() returns -EINVAL
unconditionally. From the API POV, we should rather drop the mmap
support there and expose it before the user-space tries to call mmap.
The patch contains again ugly ifdef's, unfortunately, as there is no
global flag indicating this. Once when such macro is defined, we can
get rid of this instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we have consistently dma_mmap_coherent() for all architectures,
the current ifdef and arch-specific codes in pcm core can be cleaned
up gracefully.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As PCM core handles the multiple linked streams in parallel, lockdep
gets confused (partly because of weak annotations) and spews the
false-positive warnings. This hasn't been a problem for long time but
the latest PCM lock path update seems to have woken up a sleeping
dog.
Here is an attempt to paper over this issue: pass the lock subclass
just calculated from the depth in snd_pcm_action_group(). Also, a
(possibly) wrong lock subclass set in snd_pcm_action_lock_mutex() is
dropped, too.
Reported-and-tested-by: Arthur Marsh <arthur.marsh@internode.on.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although this is weird, some drivers want to allow empty control
elements intentionally, e.g. the number of items may change depending
on the firmware status. Let the function simply returning in such a
case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we're calling request_module() asynchronously now, we can get
rid of the autoload lock in snd_seq_device_register_driver(), as well
as in the snd-seq driver registration itself. This enables the
automatic loading of dependent sequencer modules, such as
snd-seq-virmidi from snd-emu10k1-synth.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the sequencer module binding is performed independently from
the card module itself. The reason behind it is to keep the sequencer
stuff optional and allow the system running without it (e.g. for using
PCM or rawmidi only). This works in most cases, but a remaining
problem is that the binding isn't done automatically when a new driver
module is probed. Typically this becomes visible when a hotplug
driver like usb audio is used.
This patch tries to address this and other potential issues. First,
the seq-binder (seq_device.c) tries to load a missing driver module at
creating a new device object. This is done asynchronously in a workq
for avoiding the deadlock (modprobe call in module init path).
This action, however, should be enabled only when the sequencer stuff
was already initialized, i.e. snd-seq module was already loaded. For
that, a new function, snd_seq_autoload_init() is introduced here; this
clears the blocking of autoloading, and also tries to load all pending
driver modules.
Reported-by: Adam Goode <agoode@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the unlock loop of snd_pcm_action_group(), the object "s" is used
as the check of nonatomic PCM, but it should be rather "s1", which is
the iterator of the loop. This supposedly causes a kernel panic when
the substreams in operatino are linked.
Fixes: 257f8cce5d ('ALSA: pcm: Allow nonatomic trigger operations')
Reported-and-tested-by: Arthur Marsh <arthur.marsh@internode.on.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a NULL check in snd_pci_quirk_lookup() so that NULL can be passed
as a pci_dev pointer. This fixes the possible NULL dereferences in
HD-audio drivers.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- More componentisation work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes and
enhancements for the associated CODEC drivers, this is going to need
a lot quirks over time due to the lack of any firmware description of
the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest
Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX
processors.
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Merge tag 'asoc-v3.18' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.18
- More componentisation work from Lars-Peter, this time mainly
cleaning up the suspend and bias level transition callbacks.
- Real system support for the Intel drivers and a bunch of fixes and
enhancements for the associated CODEC drivers, this is going to need
a lot quirks over time due to the lack of any firmware description of
the boards.
- Jack detect support for simple card from Dylan Reid.
- A bunch of small fixes and enhancements for the Freescale drivers.
- New drivers for Analog Devices SSM4567, Cirrus Logic CS35L32, Everest
Semiconductor ES8328 and Freescale cards using the ASRC in newer i.MX
processors.
The calculated frame size was wrong because snd_pcm_format_physical_width()
actually returns the number of bits, not bytes.
Use snd_pcm_format_size() instead, which not only returns bytes, but also
simplifies the calculation.
Fixes: 8bea869c5e ("ALSA: PCM midlevel: improve fifo_size handling")
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
XMOS based USB DACs with native DSD support expose this feature via a USB
alternate setting. The audio format is either 32-bit raw or a 32-bit PCM format.
To utilize this feature on linux this patch introduces a new 32-bit DSD
sampleformat DSD_U32_LE.
A follow up patch will add a quirk for XMOS based devices to utilize the new format.
Further patches will add support to alsa-lib.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changing an interval boundary to a multiple of the step size makes that
boundary exact.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The min parameter was not used by any caller. And if it were used,
underflows in the calculations could lead to incorrect results.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit for the non-atomic PCM ops added more codes to
snd_pcm_stream_lock() and its variants. Since they are inlined
functions, it resulted in a significant code size bloat. For reducing
the size bloat, this patch changes the inline functions to the normal
function calls. The export of rwlock and rwsem are removed as well,
since they are referred only in pcm_native.c now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, many PCM operations are performed in a critical section
protected by spinlock, typically the trigger and pointer callbacks are
assumed to be atomic. This is basically because some trigger action
(e.g. PCM stop after drain or xrun) is done in the interrupt handler.
If a driver runs in a threaded irq, however, this doesn't have to be
atomic. And many devices want to handle trigger in a non-atomic
context due to lengthy communications.
This patch tries all PCM calls operational in non-atomic context.
What it does is very simple: replaces the substream spinlock with the
corresponding substream mutex when pcm->nonatomic flag is set. The
driver that wants to use the non-atomic PCM ops just needs to set the
flag and keep the rest as is. (Of course, it must not handle any PCM
ops in irq context.)
Note that the code doesn't check whether it's atomic-safe or not, but
trust in 100% that the driver sets pcm->nonatomic correctly.
One possible problem is the case where linked PCM substreams have
inconsistent nonatomic states. For avoiding this, snd_pcm_link()
returns an error if one tries to link an inconsistent PCM substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Right now we set 0 as the silence data for DSD_U8 and DSD_U16 formats,
but this is actually wrong. 0 is rather the most negative value.
Alternatively, we may take the repeating 0x69 pattern like ffmpeg
deploys.
Reference: https://ffmpeg.org/pipermail/ffmpeg-cvslog/2014-April/076427.html
Suggested-by: Alexander E. Patrakov <patrakov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_info_get_line() documents that its last parameter must be one
less than the buffer size, but this API design guarantees that
(literally) every caller gets it wrong.
Just change this parameter to have its obvious meaning.
Reported-by: Tommi Rantala <tt.rantala@gmail.com>
Cc: <stable@vger.kernel.org> # v2.2.26+
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This has been a pretty exciting release in terms of the framework, we've
finally got support for multiple CODECs attached to a single DAI link
which has been something there's been interest in as long as I've been
working on ASoC. A big thanks to Benoit and Misael for their work on
this.
Otherwise it's been a fairly standard release for development, including
more componentisation work from Lars-Peter and a good selection of both
CODEC and CPU drivers.
- Support for multiple CODECs attached to a single DAI, enabling
systems with for example multiple DAC/speaker drivers on a single
link, contributed by Benoit Cousson based on work from Misael Lopez
Cruz.
- Support for byte controls larger than 256 bytes based on the use of
TLVs contributed by Omair Mohammed Abdullah.
- More componentisation work from Lars-Peter Clausen.
- The remainder of the conversions of CODEC drivers to params_width()
- Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
TAS2552.
- Lots of updates and fixes, especially to the DaVinci, Intel,
Freescale, Realtek, and rcar drivers.
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Merge tag 'asoc-v3.17' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.17
This has been a pretty exciting release in terms of the framework, we've
finally got support for multiple CODECs attached to a single DAI link
which has been something there's been interest in as long as I've been
working on ASoC. A big thanks to Benoit and Misael for their work on
this.
Otherwise it's been a fairly standard release for development, including
more componentisation work from Lars-Peter and a good selection of both
CODEC and CPU drivers.
- Support for multiple CODECs attached to a single DAI, enabling
systems with for example multiple DAC/speaker drivers on a single
link, contributed by Benoit Cousson based on work from Misael Lopez
Cruz.
- Support for byte controls larger than 256 bytes based on the use of
TLVs contributed by Omair Mohammed Abdullah.
- More componentisation work from Lars-Peter Clausen.
- The remainder of the conversions of CODEC drivers to params_width()
- Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
TAS2552.
- Lots of updates and fixes, especially to the DaVinci, Intel,
Freescale, Realtek, and rcar drivers.
For controlling the new fields more strictly, add sw_params.proto
field indicating the protocol version of the user-space. User-space
should fill the SNDRV_PCM_VERSION value it's built with, then kernel
can know whether the new fields should be evaluated or not.
And now tstamp_type field is evaluated only when the valid value is
set there. This avoids the wrong override of tstamp_type to zero,
which is SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I previously added an integer overflow check here but looking at it now,
it's still buggy.
The bug happens in snd_compr_allocate_buffer(). We multiply
".fragments" and ".fragment_size" and that doesn't overflow but then we
save it in an unsigned int so it truncates the high bits away and we
allocate a smaller than expected size.
Fixes: b35cc82258 ('ALSA: compress_core: integer overflow in snd_compr_allocate_buffer()')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For allowing adjusting the timestamp type on the fly, add it to
sw_params. The existing ioctl is still kept for compatibility.
Along with this, increment the PCM protocol version.
The extension was suggested by Clemens Ladisch.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No functional change.
Acked-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case of _3LE/_3BE formats the samples are stored in 3 consecutive bytes
without padding it to 4 bytes. This means that the DMA needs to be able to
support 3 bytes word length in order to read/write the samples from memory
correctly. Originally the code treated 24 bits physical length samples as
they were 32 bits which leads to corruption when playing or recording audio.
The hw.formats field has already been prepared to exclude formats not
supported by the DMA engine in use, which means that only on platforms where
3 bytes is supported by the DMA will be able to use this format.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
params_physical_width() is available via pcm_params.h
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a newline and, while at it, remove a space and redundant braces.
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ALSA control code expects that the range of assigned indices to a control is
continuous and does not overflow. Currently there are no checks to enforce this.
If a control with a overflowing index range is created that control becomes
effectively inaccessible and unremovable since snd_ctl_find_id() will not be
able to find it. This patch adds a check that makes sure that controls with a
overflowing index range can not be created.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Each control gets automatically assigned its numids when the control is created.
The allocation is done by incrementing the numid by the amount of allocated
numids per allocation. This means that excessive creation and destruction of
controls (e.g. via SNDRV_CTL_IOCTL_ELEM_ADD/REMOVE) can cause the id to
eventually overflow. Currently when this happens for the control that caused the
overflow kctl->id.numid + kctl->count will also over flow causing it to be
smaller than kctl->id.numid. Most of the code assumes that this is something
that can not happen, so we need to make sure that it won't happen
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A control that is visible on the card->controls list can be freed at any time.
This means we must not access any of its memory while not holding the
controls_rw_lock. Otherwise we risk a use after free access.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two issues with the current implementation for replacing user
controls. The first is that the code does not check if the control is actually a
user control and neither does it check if the control is owned by the process
that tries to remove it. That allows userspace applications to remove arbitrary
controls, which can cause a user after free if a for example a driver does not
expect a control to be removed from under its feed.
The second issue is that on one hand when a control is replaced the
user_ctl_count limit is not checked and on the other hand the user_ctl_count is
increased (even though the number of user controls does not change). This allows
userspace, once the user_ctl_count limit as been reached, to repeatedly replace
a control until user_ctl_count overflows. Once that happens new controls can be
added effectively bypassing the user_ctl_count limit.
Both issues can be fixed by instead of open-coding the removal of the control
that is to be replaced to use snd_ctl_remove_user_ctl(). This function does
proper permission checks as well as decrements user_ctl_count after the control
has been removed.
Note that by using snd_ctl_remove_user_ctl() the check which returns -EBUSY at
beginning of the function if the control already exists is removed. This is not
a problem though since the check is quite useless, because the lock that is
protecting the control list is released between the check and before adding the
new control to the list, which means that it is possible that a different
control with the same settings is added to the list after the check. Luckily
there is another check that is done while holding the lock in snd_ctl_add(), so
we'll rely on that to make sure that the same control is not added twice.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The user-control put and get handlers as well as the tlv do not protect against
concurrent access from multiple threads. Since the state of the control is not
updated atomically it is possible that either two write operations or a write
and a read operation race against each other. Both can lead to arbitrary memory
disclosure. This patch introduces a new lock that protects user-controls from
concurrent access. Since applications typically access controls sequentially
than in parallel a single lock per card should be fine.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
do_posix_clock_monotonic_gettime() is a leftover from the initial
posix timer implementation which maps to ktime_get_ts().
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes PORT_EXIT messages are lost when a process is exiting.
This happens if you subscribe to the announce port with client A,
then subscribe to the announce port with client B, then kill client A.
Client B will not see the PORT_EXIT message because client A's port is
closing and is earlier in the announce port subscription list. The
for each loop will try to send the announcement to client A and fail,
then will stop trying to broadcast to other ports. Killing B works fine
since the announcement will already have gone to A. The CLIENT_EXIT
message does not get lost.
How to reproduce problem:
*** termA
$ aseqdump -p 0:1
0:1 Port subscribed 0:1 -> 128:0
*** termB
$ aseqdump -p 0:1
*** termA
0:1 Client start client 129
0:1 Port start 129:0
0:1 Port subscribed 0:1 -> 129:0
*** termB
0:1 Port subscribed 0:1 -> 129:0
*** termA
^C
*** termB
0:1 Client exit client 128
<--- expected Port exit as well (before client exit)
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_seq_event_dup returns -ENOMEM in some buffer-full conditions,
but usually returns -EAGAIN. Make -EAGAIN trigger the overflow
condition in snd_seq_fifo_event_in so that the fifo is cleared
and -ENOSPC is returned to userspace as stated in the alsa-lib docs.
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently snd_dmaengine_pcm_trigger() calls dmaengine_pause()
unconditinally during device suspend. In case where DMA controller
doesn't support PAUSE/RESUME functionality, this call is not able
to stop the DMA controller. In this scenario, audio playback doesn't
resume after device resume.
Calling dmaengine_pause/dmaengine_terminate_all conditionally fixes
the issue.
It has been tested with audio playback on Samsung platform having
PL330 DMA controller which doesn't support PAUSE/RESUME.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix format string mismatch in snd_seq_midisynth_register_port().
Argument type of p is unsigned int.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/1305480
The kerneloops-daemon scans dmesg for common crash signatures, among
which is 'BUG:'. The message emitted by the PCM library is really a
warning, so the most expedient thing to do seems to be to change the
string.
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_compr_open function would always return 0 even if the compressed
ops open function failed, obviously this is incorrect. Looks like this
was introduced by a small typo in:
commit a0830dbd4e
ALSA: Add a reference counter to card instance
This patch returns the value from the compressed op as it should.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
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Merge tag 'asoc-v3.15' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.15
Quite a busy release for ASoC this time, more on janitorial work than
exciting new features but welcome nontheless:
- Lots of cleanups from Takashi for enumerations; the original API for
these was error prone so he's refactored lots of code to use more
modern APIs which avoid issues.
- Elimination of the ASoC level wrappers for I2C and SPI moving us
closer to converting to regmap completely and avoiding some
randconfig hassle.
- Provide both manually and transparently locked DAPM APIs rather than
a mix of the two fixing some concurrency issues.
- Start converting CODEC drivers to use separate bus interface drivers
rather than having them all in one file helping avoid dependency
issues.
- DPCM support for Intel Haswell and Bay Trail platforms.
- Lots of work on improvements for simple-card, DaVinci and the Renesas
rcar drivers.
- New drivers for Analog Devices ADAU1977, TI PCM512x and parts of the
CSR SiRF SoC.
A few code cleanups and optimizations. In addition, drop
snd_device_disconnect() that isn't used at all, and drop the return
values from snd_device_free*().
Another slight difference by this change is that now the device state
will become always SNDRV_DEV_REGISTERED no matter whether dev_register
ops is present or not. It's for better consistency. There should be
no impact for the current tree, as the state isn't checked.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Basically, the device type specifies the priority of the device to be
registered / freed, too. However, the priority value isn't well
utilized but only it's checked as a group. This results in
inconsistent register and free order (where each of them should be in
reversed direction).
This patch simplifies the device list management code by simply
inserting a list entry at creation time in an incremental order for
the priority value. Since we can just follow the link for register,
disconnect and free calls, we don't have to specify the group; so the
whole enum definitions are also simplified as well.
The visible change to outside is that the priorities of some object
types are revisited. For example, now the SNDRV_DEV_LOWLEVEL object
is registered before others (control, PCM, etc) and, in return,
released after others. Similarly, SNDRV_DEV_CODEC is in a lower
priority than SNDRV_DEV_BUS for ensuring the dependency.
Also, the unused SNDRV_DEV_TOPLEVEL, SNDRV_DEV_LOWLEVEL_PRE and
SNDRV_DEV_LOWLEVEL_NORMAL are removed as a cleanup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like PCM, allow hwdep to be assigned to a different parent device
than the card. It'll be used for the HD-audio codec device in the
later patches.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For referring to a different object from sysfs ops, take hwdep
private_data as stored via dev_set_drvdata() at creating the device
object. In that way, the same sysfs ops can be used by different
device types.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of calling each time device_create_file(), create the groups
of sysfs attribute files at once in a normal way. Add a new helper
function, snd_get_device(), to return the associated device object,
so that we can handle the sysfs addition locally.
Since the sysfs file addition is done differently now,
snd_add_device_sysfs_file() helper function is removed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow modules to use it, fixing a build failure when the newly added
ADAU1977 driver is built as a module.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
While moving the card device into struct snd_card, the reference to
the assigned card in sysfs show/store callbacks were forgotten to be
refreshed, still accessing to the no longer used drvdata. Fix these
places to refer correctly via container_of().
Also, remove the superfluous NULL checks since it's guaranteed to be
non-NULL now.
Fixes: 8bfb181c17 ('ALSA: Embed card device into struct snd_card')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using __bitwise and typedefs for the attributes of snd_device struct
isn't so useful, and rather it worsens the readability. Let's drop
them and use the straightforward enum.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() & co as much as possible. If not available (no device
assigned at the calling point), use pr_xxx() helpers instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() & co as much as possible. If not available (no device
assigned at the calling point), use pr_xxx() helpers instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() & co as much as possible. If not available (no device
assigned at the calling point), use pr_xxx() helpers instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() & co as much as possible. If not available (no device
assigned at the calling point), use pr_xxx() helpers instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() & co as much as possible. If not available (no device
assigned at the calling point), use pr_xxx() helpers instead.
For simplicity, introduce new helpers for pcm stream, pcm_err(), etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use dev_err() & co as much as possible. If not available (no device
assigned at the calling point), use pr_xxx() helpers instead.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop the own refcount but use the standard device refcounting via
get_device() and put_device(). Introduce a new completion to snd_card
instead of the wait queue for syncing the last release, which is used
in snd_card_free().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As prepared in the previous patch, we are ready to create a device
struct for the card object in snd_card_create() now. This patch
changes the scheme from the old style to:
- embed a device struct for the card object into snd_card struct,
- initialize the card device in snd_card_create() (but not register),
- registration is done in snd_card_register() via device_add()
The actual card device is stored in card->card_dev. The card->dev
pointer is kept unchanged and pointing to the parent device as before
for compatibility reason.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a part of preliminary works for modernizing the ALSA device
structure. So far, we set card->dev at later point after the object
creation. Because of this, the core layer doesn't always know which
device is being handled before it's actually registered, and it makes
impossible to show the device in error messages, for example. The
first goal is to achieve a proper struct device initialization at the
very beginning of probing.
As a first step, this patch introduces snd_card_new() function (yes
there was the same named function in the very past), in order to
receive the parent device pointer from the very beginning.
snd_card_create() is marked as deprecated.
At this point, there is no functional change other than that. The
actual change of the device creation scheme will follow later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last argument, name, of snd_oss_register_device() is nowhere
referred in the function in the current code. Let's drop it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the lengthy #if defined(XXX) || defined(XXX_MODULE) with the
new IS_ENABLED() macro.
The patch still doesn't cover all ifdefs. For example, the dependency
on CONFIG_GAMEPORT is still open-coded because this also has an extra
dependency on MODULE. Similarly, an open-coded ifdef in pcm_oss.c and
some sequencer-related stuff are left untouched.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixed 2 typos in DocBook/alsa-driver-api.xml.
It is because this file is generated by make xmldocs,
I have to fix typos within source files.
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two loops that are almost identical but only with different
checks. Refactor them with a simple helper, and give a bit more
comments what's doing there.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The test here is intended intended to prevent shift wrapping bugs when
we do "1U << idx2". We should consider the number of bits in a u32
instead of the number of bytes.
[fix another chunk similarly by tiwai]
Fixes: 7bb2491b35 ('ALSA: Add kconfig to specify the max card numbers')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A bit of special care is necessary when creating the intersection of two rate
masks. This comes from the special meaning of the SNDRV_PCM_RATE_CONTINUOUS and
SNDRV_PCM_RATE_KNOT bits, which needs special handling when intersecting two
rate masks. SNDRV_PCM_RATE_CONTINUOUS means the hardware supports all rates in a
specific interval. SNDRV_PCM_RATE_KNOT means the hardware supports a set of
discrete rates specified by a list constraint. For all other cases the supported
rates are specified directly in the rate mask.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The failures of buffer preallocations at driver initializations aren't
critical but it's still helpful to inform, so that user can know that
something doesn't work as expected.
For example, the recent page allocator change triggered regressions,
but developers didn't notice until recently because the driver didn't
complain.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of keeping a separate snd-page-alloc module, merge into the
core snd-pcm module, as we don't need to keep it as an individual
module due to the drop of page reservation.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nowadays we have CMA for obtaining the contiguous memory pages
efficiently. Let's kill the old kludge for reserving the memory pages
for large buffers. It was rarely useful (only for preserving pages
among module reloading or a little help by an early boot scripting),
used only by a couple of drivers, and yet it gives too much ugliness
than its benefit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
commit f0e9c080 - "ALSA: compress: change the way sample rates are sent to
kernel" changed the way sample rates are sent. So now we don't need to check for
PCM_RATE_xxx in kernel
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the process is sleeping at the SNDRV_PCM_STATE_PAUSED
state from the wait_for_avail function, the sleep process will be woken by
timeout(10 seconds). Even if the sleep process wake up by timeout, by this
patch, the process will continue with sleep and wait for the other state.
Signed-off-by: JongHo Kim <furmuwon@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two peaks in diffstat are for the audio EQ init of IDT codecs and the
EMU2004 usb mixer addition, both of which are pretty device-specific,
so safe to apply. The rest are a bunch of small fixes, most of them
are regression fixes.
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Merge tag 'sound-fix-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Two peaks in diffstat are for the audio EQ init of IDT codecs and the
EMU2004 usb mixer addition, both of which are pretty device-specific,
so safe to apply. The rest are a bunch of small fixes, most of them
are regression fixes"
* tag 'sound-fix-3.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (26 commits)
ALSA: hda - load EQ params into IDT codec on HP bNB13 systems
ASoC: cs42l52: Correct MIC CTL mask
ASoC: wm8962: Turn on regcache_cache_only before disabling regulator
ALSA: jack: Unregister input device at disconnection
ALSA: pcsp: Fix the order of input device unregistration
ASoC: fsl: imx-pcm-fiq: omit fiq counter to avoid harm in unbalanced situations
ASoC: blackfin: Fix missing break
ALSA: usb-audio: add front jack channel selector for EMU0204
ALSA: hda - Don't clear the power state at snd_hda_codec_reset()
ASoC: arizona: Fix typo in name of EQ coefficient controls
ALSA: hda - Control EAPD for Master volume on Lenovo N100
ALSA: hda - Don't turn off EAPD for headphone on Lenovo N100
ALSA: isa: not allocating enough space
ALSA: snd-aoa: two copy and paste bugs
ASoC: wm8997: Correct typo in ISRC mux routes
ALSA: hda - Check keep_eapd_on before inv_eapd
ALSA: hda - Fix Line Out automute on Realtek multifunction jacks
ALSA: msnd: Avoid duplicated driver name
ALSA: compress_core: don't return -EBADFD from poll if paused
ALSA: hda - hdmi: Fix wrong baseline length in ATI/AMD generated ELD
...
Since gen_pool_dma_alloc() is introduced, we implement it to simplify code.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
The recent change in sysfs triggered a kernel WARNING at unloading a
sound driver like
WARNING: CPU: 3 PID: 2247 at fs/sysfs/group.c:214 sysfs_remove_group+0xe8/0xf0()
sysfs group ffffffff81ab7b20 not found for kobject 'event14'
for each jack instance. It's because the unregistration of jack input
device is done in dev_free callback, which is called after
snd_card_disconnect(). Since device_unregister(card->card_dev) is
called in snd_card_disconnect(), the whole sysfs entries belonging to
card->card_dev have been already removed recursively. Thus this
results in a warning as input_unregister_device() yet tries to
unregister the already removed sysfs entry.
For fixing this mess, we need to unregister the jack input device at
dev_disconnect callback so that it's called before unregistering the
card->card_dev.
Reviwed-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull vfs updates from Al Viro:
"All kinds of stuff this time around; some more notable parts:
- RCU'd vfsmounts handling
- new primitives for coredump handling
- files_lock is gone
- Bruce's delegations handling series
- exportfs fixes
plus misc stuff all over the place"
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs: (101 commits)
ecryptfs: ->f_op is never NULL
locks: break delegations on any attribute modification
locks: break delegations on link
locks: break delegations on rename
locks: helper functions for delegation breaking
locks: break delegations on unlink
namei: minor vfs_unlink cleanup
locks: implement delegations
locks: introduce new FL_DELEG lock flag
vfs: take i_mutex on renamed file
vfs: rename I_MUTEX_QUOTA now that it's not used for quotas
vfs: don't use PARENT/CHILD lock classes for non-directories
vfs: pull ext4's double-i_mutex-locking into common code
exportfs: fix quadratic behavior in filehandle lookup
exportfs: better variable name
exportfs: move most of reconnect_path to helper function
exportfs: eliminate unused "noprogress" counter
exportfs: stop retrying once we race with rename/remove
exportfs: clear DISCONNECTED on all parents sooner
exportfs: more detailed comment for path_reconnect
...
Pausing audio playback is not an illegal state so it doesn't
seem sensible for poll() to return -EBADFD on a paused stream.
There's also no reason to assume that we can't write more data
to the DSP while playback is paused. Remove the -EBADFD so that
a stream in paused state will still report the buffer
availability from poll(). It is up to the user process to
manage its state so that it knows whether it is paused or not.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by VInod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The drain and drain_notify callback were blocked by low level driver
until the draining was complete. Due to this being invoked with big
fat mutex held, others ops like reading timestamp, calling pause, drop
were blocked.
So to fix this we add a new snd_compr_drain_notify() API. This would
be required to be invoked by low level driver when drain or partial
drain has been completed by the DSP. Thus we make the drain and
partial_drain callback as non blocking and driver returns immediately
after notifying DSP. The waiting is done while releasing the lock so
that other ops can go ahead.
[ The commit 917f4b5cba was wrongly applied from the preliminary
patch. This commit corrects to the final version.
Sorry for inconvenience! -- tiwai ]
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
CC: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unable to handle kernel NULL pointer dereference at virtual address 00000008
pgd = d5300000
[00000008] *pgd=0d265831, *pte=00000000, *ppte=00000000
Internal error: Oops: 17 [#1] PREEMPT ARM
CPU: 0 PID: 2295 Comm: vlc Not tainted 3.11.0+ #755
task: dee74800 ti: e213c000 task.ti: e213c000
PC is at snd_pcm_info+0xc8/0xd8
LR is at 0x30232065
pc : [<c031b52c>] lr : [<30232065>] psr: a0070013
sp : e213dea8 ip : d81cb0d0 fp : c05f7678
r10: c05f7770 r9 : fffffdfd r8 : 00000000
r7 : d8a968a8 r6 : d8a96800 r5 : d8a96200 r4 : d81cb000
r3 : 00000000 r2 : d81cb000 r1 : 00000001 r0 : d8a96200
Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user
Control: 10c5387d Table: 15300019 DAC: 00000015
Process vlc (pid: 2295, stack limit = 0xe213c248)
[<c031b52c>] (snd_pcm_info) from [<c031b570>] (snd_pcm_info_user+0x34/0x9c)
[<c031b570>] (snd_pcm_info_user) from [<c03164a4>] (snd_pcm_control_ioctl+0x274/0x280)
[<c03164a4>] (snd_pcm_control_ioctl) from [<c0311458>] (snd_ctl_ioctl+0xc0/0x55c)
[<c0311458>] (snd_ctl_ioctl) from [<c00eca84>] (do_vfs_ioctl+0x80/0x31c)
[<c00eca84>] (do_vfs_ioctl) from [<c00ecd5c>] (SyS_ioctl+0x3c/0x60)
[<c00ecd5c>] (SyS_ioctl) from [<c000e500>] (ret_fast_syscall+0x0/0x48)
Code: e1a00005 e59530dc e3a01001 e1a02004 (e5933008)
---[ end trace cb3d9bdb8dfefb3c ]---
This is provoked when the ASoC front end is open along with its backend,
(which causes the backend to have a runtime assigned to it) and then the
SNDRV_CTL_IOCTL_PCM_INFO is requested for the (visible) backend device.
Resolve this by ensuring that ASoC internal backend devices are not
visible to userspace, just as the commentry for snd_pcm_new_internal()
says it should be.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Acked-by: Mark Brown <broonie@linaro.org>
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When no proper id string is given, the driver tries to fall back to
copy the proc_root name string via strcpy(), but this might overflow
the fixed string size. Let's use strlcpy().
Spotted by coverity CID 139008.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
dmab->area and addr fields should be cleared at the head of
snd_malloc_dev_iram() as especially dmab->area is used to indicate the
allocation failure / fallback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
module->name is a fixed array, so we can check the empty contents
straightforwardly in module_slot_match().
Spotted by coverity CID 1056786.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the standard PCM helper function to figure out the sample bytes
instead of hardcodec PCM format checks in
snd_hwparams_to_dma_slave_config().
The patch also extends the format check for 8 bytes formats although
no one should match so far.
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The drain and drain_notify callback were blocked by low level driver untill the
draining was complete. Due to this being invoked with big fat mutex held, others
ops like reading timestamp, calling pause, drop were blocked.
So to fix this we add a new snd_compr_drain_notify() API. This would be required
to be invoked by low level driver when drain or partial drain has been completed
by the DSP. Thus we make the drain and partial_drain callback as non blocking
and driver returns immediately after notifying DSP.
The waiting is done while relasing the lock so that other ops can go ahead.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
CC: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that we can't use gen_pool_*() functions on archs
without CONFIG_GENERIC_ALLOCATOR (resulting in missing symbols), since
linux/genalloc.h doesn't provide dummy functions for all. We'd be
able to fix linux/genalloc.h size, but I take an easier path for
now...
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now it's quite common that an SoC contains its on-chip internal RAM.
By using this RAM space for DMA buffer during audio playback/record,
we can shutdown the voltage for external RAM to save power.
So add new DEV type with iram malloc()/free() and accordingly modify
current default mmap() for the iram circumstance.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently we assume that userspace will shut down the compressed stream
correctly. However, if userspcae dies (e.g. cplay & ctrl-C) we dont
stop the stream before freeing it.
This now checks that the stream is stopped before freeing.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
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Merge tag 'asoc-v3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.12
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
regressions in the special cases for non-DAPM CODECs and make it
easier to integrate with other components on boards. All existing
drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
compile test.
When users of SND_DMAENGINE_PCM are built as module, the config symbol
SND_DMAENGINE_PCM must be tristate, otherwise the linker will fail.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For the PXA DMA rework, we need the generic dmaengine implementation
that currently lives in sound/soc for standalone (non-ASoC) AC'97
support.
Move it to sound/core, and rename the Kconfig symbol.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
the return value of SNDRV_COMPRESS_VERSION always return default -ENOTTY as the
return value was never updated for this call
assign return value from put_user()
Reported-by: Haynes <hgeorge@codeaurora.org>
CC: stable@vger.kernel.org
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We've got bug reports that the module loading stuck on Debian system
with 3.10 kernel. The debugging session revealed that the initial
registration of OSS sequencer clients stuck at module loading time,
which involves again with request_module() at the init phase. This is
triggered only by special --install stuff Debian is using, but it's
still not good to have such loops.
As a workaround, call the registration part asynchronously. This is a
better approach irrespective of the hang fix, in anyway.
Reported-and-tested-by: Philipp Matthias Hahn <pmhahn@pmhahn.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A relative calm release at this time with a flat diffstat.
The only significant change in the ALSA core side is the support for
more than 32 card instances, configurable via kconfig.
Other than that, in both ASoC and other parts, mostly some
improvements and fixes on the driver side.
- hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes
- hda: Haswell HDMI audio fixes, runtime PM improvements
- hda: Intel BayTrail support, ALC5505 DSP support
- es1968: MediaForte M56VAP support
- usb-audio: Improved support for Yamaha/Roland devices
- usb-audio: M2Tech hiFace, Audio Advantage Micro II support
- hdspm: wordclock fixes
- ASoC: Pending fixes for WM8962
- ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500
- ASoC: Generalisation of the Bluetooth and HDMI stub drivers
- ASoC: SSM2518 and RT5640 codec drivers.
- ASoC: Tegra CPUs with RT5640 machine driver
- ASoC: AC'97 refactoring bug fixes
- ASoC: ADAU1701 driver fixes
- Clean up of *_set_drvdata() in a wide range of drivers
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Merge tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"A relative calm release at this time with a flat diffstat. The only
significant change in the ALSA core side is the support for more than
32 card instances, configurable via kconfig. Other than that, in both
ASoC and other parts, mostly some improvements and fixes on the driver
side.
- hda: More quirks for ALC269-variants on Dell & co, VIA codec fixes
- hda: Haswell HDMI audio fixes, runtime PM improvements
- hda: Intel BayTrail support, ALC5505 DSP support
- es1968: MediaForte M56VAP support
- usb-audio: Improved support for Yamaha/Roland devices
- usb-audio: M2Tech hiFace, Audio Advantage Micro II support
- hdspm: wordclock fixes
- ASoC: Pending fixes for WM8962
- ASoC: Cleanups and fixes for Blackfin, SGTL5000 and UX500
- ASoC: Generalisation of the Bluetooth and HDMI stub drivers
- ASoC: SSM2518 and RT5640 codec drivers.
- ASoC: Tegra CPUs with RT5640 machine driver
- ASoC: AC'97 refactoring bug fixes
- ASoC: ADAU1701 driver fixes
- Clean up of *_set_drvdata() in a wide range of drivers"
* tag 'sound-3.11' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (284 commits)
ALSA: vmaster: Fix the regression of missing vmaster hook call
ALSA: hda - Add Dell SSID to support Headset Mic recording
ASoC: adau1701: remove control_data assignment
ASoC: adau1701: more direct regmap usage
ASoC: ac97: fixup multi-platform AC'97 module build failure
ASoC: pxa2xx: fixup multi-platform AC'97 build failures
ASoC: tegra20-ac97: Remove unused variable
ASoC: tegra20-ac97: Remove duplicate error message
ALSA: usb-audio: Add Audio Advantage Micro II
ASoC: tas5086: fix Mid-Z implementation
ASoC: tas5086: fix TAS5086_CLOCK_CONTROL register size
ALSA: Replace the magic number 44 with const
ALSA: hda - Fix the max length of control name in generic parser
ALSA: hda - Guess what, it's two more Dell headset mic quirks
ALSA: hda - Yet another Dell headset mic quirk
ALSA: hda - Add support for ALC5505 DSP power-save mode
ASoC: mfld: Remove unused variable
ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
ALSA: usb-audio: claim autodetected PCM interfaces all at once
ALSA: usb-audio: remove superfluous Roland quirks
...
The commit [1ca2f2ec: ALSA: vmaster: Add snd_ctl_sync_vmaster() helper
function] changed master_put() function and the check for the required
vmaster hook call is wrongly performed now, which results in the
missing hook call upon "Master Playback Switch" value changes.
This patch corrects the check logic.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new helper function, snd_ctl_sync_vmaster(), which updates
the slave put callbacks forcibly as well as calling the hook. This
will be used in the upcoming patch in HD-audio codec driver for
toggling the mute in vmaster slaves.
Along with the new function, the old snd_ctl_sync_vmaster_hook() is
replaced as a macro calling with the argument hook_only=true.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* for-linus: (635 commits)
ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310
ALSA: hda - Fix pin configurations for MacBook Air 4,2
ALSA: usb-audio: work around Android accessory firmware bug
ALSA: hda - Headset mic support for three more machines
Linux 3.10-rc6
smp.h: Use local_irq_{save,restore}() in !SMP version of on_each_cpu().
powerpc: Fix missing/delayed calls to irq_work
powerpc: Fix emulation of illegal instructions on PowerNV platform
powerpc: Fix stack overflow crash in resume_kernel when ftracing
snd_pcm_link(): fix a leak...
use can_lookup() instead of direct checks of ->i_op->lookup
move exit_task_namespaces() outside of exit_notify()
fput: task_work_add() can fail if the caller has passed exit_task_work()
xfs: don't shutdown log recovery on validation errors
xfs: ensure btree root split sets blkno correctly
xfs: fix implicit padding in directory and attr CRC formats
xfs: don't emit v5 superblock warnings on write
mei: me: clear interrupts on the resume path
mei: nfc: fix nfc device freeing
mei: init: Flush scheduled work before resetting the device
...
Currently ALSA supports up to 32 card instances when the dynamic minor
is used. While 32 cards are usually big enough for normal use cases,
there are sometimes weird requirements with more card support.
Actually, this limitation, 32, comes from the index option, where you
can pass the bit mask to assign the card. Other than that, we can
actually give more cards up to the minor number limits (currently 256,
which can be extended more, too).
This patch adds a new Kconfig to specify the max card numbers, and
changes a few places to accept more than 32 cards.
The only incompatibility with high card numbers would be the handling
of index option. The index option can be still used to pass the
bitmask for card assignments, but this works only up to 32 slots.
More than 32, no bitmask style option is available but only a single
slot can be specified via index option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA core tries to add a suffix as "_1" automatically when the given
id string conflicts. The current code assumes implicitly that the max
card number is 16 so that the single hex "_X" suffix can be put.
However, with the dynamic device management, the card can be at most
32, so it can put even a non-hex character there. Also, when the max
card number is increased in future, this would result in worse.
This patch rewrites the code to add the suffix string in a simpler
(thus cleaner) way. It can support up to three digits, so it should
suffice for most requirements.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ops field of the snd_pcm_substream struct is never modified inside the ALSA
core. Making it const allows drivers to declare their snd_pcm_ops struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone mic
and headset mic support, jack_modes hint consolidation, proper beep
attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack
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Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"Mostly many small changes spread as seen in diffstat in sound/*
directory by this update. A significant change in the subsystem level
is the introduction of snd_soc_component, which will help more generic
handling of SoC and off-SoC components.
Also, snd_BUG_ON() macro is enabled unconditionally now due to its
misuses, so people might hit kernel warnings (it's a good thing for
us).
- compress-offload: support for capture by Charles Keepax
- HD-audio: codec delay support by Dylan Reid
- HD-audio: improvements/fixes in generic parser: better headphone
mic and headset mic support, jack_modes hint consolidation, proper
beep attach/detachment, generalized power filter controls by David
Henningsson, et al
- HD-audio: Improved management of HDMI codec pins/converters
- HD-audio: Better pin/DAC assignment for VIA codecs
- HD-audio: Haswell HDMI workarounds
- HD-audio: ALC268 codec support, a few new quirks for Chromebooks
- USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
fix by Clemens Ladisch
- USB: support for DSD formats by Daniel Mack
- USB: A few UAC2 device endian/cock fixes by Eldad Zack
- USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
Yamaha THRxx devices
- HDSPM: updates for TCO controls by Adrian Knoth
- ASoC: Add a snd_soc_component object type for generic handling of
SoC and off-SoC components by Kuninori Morimoto,
- dmaengine: a large set of cleanups and conversions by Lars-Peter
Clausen
- ASoC DAPM: performance optimizations from Ryo Tsutsui
- ASoC DAPM: support for mixer control sharing by Stephen Warren
- ASoC: multiplatform ARM cleanups from Arnd Bergmann
- ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"
* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
ALSA: pcm_format_to_bits strong-typed conversion
ALSA: compress: fix the states to check for allowing read
ALSA: hda - Move Thinkpad X220 to use auto parser
ALSA: USB: adjust for changed 3.8 USB API
ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
sound: oss/dmabuf: use dma_map_single
ALSA: ali5451: use mdelay instead of large udelay constants
ALSA: hda - Add the support for ALC286 codec
ALSA: usb-audio: USB quirk for Yamaha THR10C
ALSA: usb-audio: USB quirk for Yamaha THR5A
ALSA: usb-audio: USB quirk for Yamaha THR10
ALSA: usb-audio: Fix autopm error during probing
ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
ALSA: sound kconfig typo
ALSA: emu10k1: Fix dock firmware loading
ASoC: ux500: forward declare msp_i2s_platform_data
ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
...
Pull VFS updates from Al Viro,
Misc cleanups all over the place, mainly wrt /proc interfaces (switch
create_proc_entry to proc_create(), get rid of the deprecated
create_proc_read_entry() in favor of using proc_create_data() and
seq_file etc).
7kloc removed.
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs: (204 commits)
don't bother with deferred freeing of fdtables
proc: Move non-public stuff from linux/proc_fs.h to fs/proc/internal.h
proc: Make the PROC_I() and PDE() macros internal to procfs
proc: Supply a function to remove a proc entry by PDE
take cgroup_open() and cpuset_open() to fs/proc/base.c
ppc: Clean up scanlog
ppc: Clean up rtas_flash driver somewhat
hostap: proc: Use remove_proc_subtree()
drm: proc: Use remove_proc_subtree()
drm: proc: Use minor->index to label things, not PDE->name
drm: Constify drm_proc_list[]
zoran: Don't print proc_dir_entry data in debug
reiserfs: Don't access the proc_dir_entry in r_open(), r_start() r_show()
proc: Supply an accessor for getting the data from a PDE's parent
airo: Use remove_proc_subtree()
rtl8192u: Don't need to save device proc dir PDE
rtl8187se: Use a dir under /proc/net/r8180/
proc: Add proc_mkdir_data()
proc: Move some bits from linux/proc_fs.h to linux/{of.h,signal.h,tty.h}
proc: Move PDE_NET() to fs/proc/proc_net.c
...
for reading compressed data, we need to allow when we are paused, draining or
stopped.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Cc: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Cc: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few more fixes, nothing too major though the DMA changes fix modular
builds.
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Merge tag 'asoc-v3.10-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.10
A few more fixes, nothing too major though the DMA changes fix modular
builds.
Like the previous patch by Dan, we should clear the data to be
returned from certain compress ioctls, namely,
snd_compr_get_codec_caps() and snd_compr_get_params().
This time, we can simply replace kmalloc() with kzalloc().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the ->get_caps() function doesn't clear the buffer then there would
stack information leaked to userspace. For example,
soc_compr_get_caps() can return success without clearing the buffer.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch reworks the writes to use cumulative values thus making the
app_pointer unecessary and removing it.
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only tested as far as build.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The buffer passed to the copy callback should not be const because the
copy callback can be used for capture and playback.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The app_pointer is managed locally by the compress core for memory
mapped DSPs but for DSPs that are not memory mapped this would have to
be manually updated from within the DSP driver itself, which is hardly
very idiomatic.
This patch switches to using the cumulative values to calculate the
available buffer space because these are already gracefully passed out
of the DSP driver to the compress core and otherwise should be
functionally equivalent.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is my example conversion of a few existing mmap users. The pcm
mmap case is one of the more straightforward ones.
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital
DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.
The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).
DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:
configured hardware
176.4KHz 352.8kHz 705.6KHz <---- sample rate
8-bit 2.8MHz 5.6MHz
16-bit 2.8Mhz 5.6MHz 11.2MHz
`-----------------------------'
actual DSD sample rates
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only part of proc_dir_entry the code outside of fs/proc
really cares about is PDE(inode)->data. Provide a helper
for that; static inline for now, eventually will be moved
to fs/proc, along with the knowledge of struct proc_dir_entry
layout.
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
script/kernel-doc reports the following type of warnings (when run in verbose
mode):
Warning(sound/core/init.c:152): No description found for return value of
'snd_card_create'
To fix that:
- add missing descriptions of function return values
- use "Return:" sections to describe those return values
Along the way:
- complete some descriptions
- fix some typos
Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a value of a vmaster slave control is changed, the ctl change
notification is sometimes ignored. This happens when the master
control overrides, e.g. when the corresponding master control is
muted. The reason is that slave_put() returns the value of the actual
slave put callback, and it doesn't reflect the virtual slave value
change.
This patch fixes the function just to return 1 whenever a slave value
is changed.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "dev" variable could be out of bounds. Calling
snd_seq_oss_synth_is_valid() checks that it is is a valid device
which has been opened. We check this inside set_note_event() so
this function can't succeed without a valid "dev". But we need to
do the check earlier to prevent invalid dereferences and memory
corruption.
One call tree where "dev" could be out of bounds is:
-> snd_seq_oss_oob_user()
-> snd_seq_oss_process_event()
-> extended_event()
-> note_on_event()
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull vfs pile (part one) from Al Viro:
"Assorted stuff - cleaning namei.c up a bit, fixing ->d_name/->d_parent
locking violations, etc.
The most visible changes here are death of FS_REVAL_DOT (replaced with
"has ->d_weak_revalidate()") and a new helper getting from struct file
to inode. Some bits of preparation to xattr method interface changes.
Misc patches by various people sent this cycle *and* ocfs2 fixes from
several cycles ago that should've been upstream right then.
PS: the next vfs pile will be xattr stuff."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs: (46 commits)
saner proc_get_inode() calling conventions
proc: avoid extra pde_put() in proc_fill_super()
fs: change return values from -EACCES to -EPERM
fs/exec.c: make bprm_mm_init() static
ocfs2/dlm: use GFP_ATOMIC inside a spin_lock
ocfs2: fix possible use-after-free with AIO
ocfs2: Fix oops in ocfs2_fast_symlink_readpage() code path
get_empty_filp()/alloc_file() leave both ->f_pos and ->f_version zero
target: writev() on single-element vector is pointless
export kernel_write(), convert open-coded instances
fs: encode_fh: return FILEID_INVALID if invalid fid_type
kill f_vfsmnt
vfs: kill FS_REVAL_DOT by adding a d_weak_revalidate dentry op
nfsd: handle vfs_getattr errors in acl protocol
switch vfs_getattr() to struct path
default SET_PERSONALITY() in linux/elf.h
ceph: prepopulate inodes only when request is aborted
d_hash_and_lookup(): export, switch open-coded instances
9p: switch v9fs_set_create_acl() to inode+fid, do it before d_instantiate()
9p: split dropping the acls from v9fs_set_create_acl()
...
this add new API for sound compress to support gapless playback.
As noted in Documentation change, we add API to send metadata of encoder and
padding delay to DSP. Also add API for indicating EOF and switching to
subsequent track
Also bump the compress API version
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_compr_update_tstamp() can only fill in the snd_compr_tstamp
if the codec implements the pointer() function. If that happened
the code was previously returning uninitialized garbage in the
tstamp because it wasn't initialized anywhere.
This change zero-fills the tstamp in the two places it is used
before calling snd_compr_update_tstamp(), and also has
snd_compr_update_tstamp() return an error indication if it
can't provide a tstamp. For the case of snd_compr_calc_avail()
it ignores this error because we still need to return info on
the available buffer space even if we can't provide tstamp
info - when the tstamp is not valid all fields are now
guaranteed to be zero.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pull trivial branch from Jiri Kosina:
"Usual stuff -- comment/printk typo fixes, documentation updates, dead
code elimination."
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (39 commits)
HOWTO: fix double words typo
x86 mtrr: fix comment typo in mtrr_bp_init
propagate name change to comments in kernel source
doc: Update the name of profiling based on sysfs
treewide: Fix typos in various drivers
treewide: Fix typos in various Kconfig
wireless: mwifiex: Fix typo in wireless/mwifiex driver
messages: i2o: Fix typo in messages/i2o
scripts/kernel-doc: check that non-void fcts describe their return value
Kernel-doc: Convention: Use a "Return" section to describe return values
radeon: Fix typo and copy/paste error in comments
doc: Remove unnecessary declarations from Documentation/accounting/getdelays.c
various: Fix spelling of "asynchronous" in comments.
Fix misspellings of "whether" in comments.
eisa: Fix spelling of "asynchronous".
various: Fix spelling of "registered" in comments.
doc: fix quite a few typos within Documentation
target: iscsi: fix comment typos in target/iscsi drivers
treewide: fix typo of "suport" in various comments and Kconfig
treewide: fix typo of "suppport" in various comments
...
Return the value obtained from snd_pcm_hw_constraint_minmax() instead
of -EINVAL. Silences the following smatch warning:
sound/core/pcm_native.c:2003 snd_pcm_hw_constraints_complete() info:
why not propagate 'err' from snd_pcm_hw_constraint_minmax() instead of -22?
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some comments misspell "registered"; this fixes them. No code changes.
Signed-off-by: Adam Buchbinder <adam.buchbinder@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The dereference snd_pcm_plug_stream(plug) should come after the NULL
check snd_BUG_ON(!plug).
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dereference snd_pcm_plug_stream(plug) should come after the NULL
check snd_BUG_ON(!plug).
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are uncovered cases whether the card refcount introduced by the
commit a0830dbd isn't properly increased or decreased:
- OSS PCM and mixer success paths
- When lookup function gets NULL
This patch fixes these places.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=50251
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When disconnect callback is called, each component should wake up
sleepers and check card->shutdown flag for avoiding the endless sleep
blocking the proper resource release.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.
The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix races at PCM disconnection:
- while a PCM device is being opened or closed
- while the PCM state is being changed without lock in prepare,
hw_params, hw_free ops
Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 4eeaaeaea (ALSA: core: add hooks for audio timestamps) added the
new audio_tstamp field to struct snd_pcm_status. However, struct
timespec requires 64-bit alignment, so the 64-bit compiler would insert
32 bits of padding before this field, which broke SNDRV_PCM_IOCTL_STATUS
with error messages like this:
kernel: unknown ioctl = 0x80984120
To solve this, insert the padding explicitly so that it can be taken
into account when calculating the ABI structure size.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA did not provide any direct means to infer the audio time for A/V
sync and system/audio time correlations (eg. PulseAudio).
Applications had to track the number of samples read/written and
add/subtract the number of samples queued in the ring buffer. This
accounting led to small errors, typically several samples, due to the
two-step process. Computing the audio time in the kernel is more
direct, as all the information is available in the same routines.
Also add new .audio_wallclock routine to enable fine-grain synchronization
between monotonic system time and audio hardware time.
Using the wallclock, if supported in hardware, allows for a
much better sub-microsecond precision and a common drift tracking for
all devices sharing the same wall clock (master clock).
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keep track of boundary crossing when hw_ptr
exceeds boundary limit and wraps-around. This
will help keep track of total number
of frames played/received at the kernel level
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A long time ago, in v2.4, VM_RESERVED kept swapout process off VMA,
currently it lost original meaning but still has some effects:
| effect | alternative flags
-+------------------------+---------------------------------------------
1| account as reserved_vm | VM_IO
2| skip in core dump | VM_IO, VM_DONTDUMP
3| do not merge or expand | VM_IO, VM_DONTEXPAND, VM_HUGETLB, VM_PFNMAP
4| do not mlock | VM_IO, VM_DONTEXPAND, VM_HUGETLB, VM_PFNMAP
This patch removes reserved_vm counter from mm_struct. Seems like nobody
cares about it, it does not exported into userspace directly, it only
reduces total_vm showed in proc.
Thus VM_RESERVED can be replaced with VM_IO or pair VM_DONTEXPAND | VM_DONTDUMP.
remap_pfn_range() and io_remap_pfn_range() set VM_IO|VM_DONTEXPAND|VM_DONTDUMP.
remap_vmalloc_range() set VM_DONTEXPAND | VM_DONTDUMP.
[akpm@linux-foundation.org: drivers/vfio/pci/vfio_pci.c fixup]
Signed-off-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Cc: Alexander Viro <viro@zeniv.linux.org.uk>
Cc: Carsten Otte <cotte@de.ibm.com>
Cc: Chris Metcalf <cmetcalf@tilera.com>
Cc: Cyrill Gorcunov <gorcunov@openvz.org>
Cc: Eric Paris <eparis@redhat.com>
Cc: H. Peter Anvin <hpa@zytor.com>
Cc: Hugh Dickins <hughd@google.com>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: James Morris <james.l.morris@oracle.com>
Cc: Jason Baron <jbaron@redhat.com>
Cc: Kentaro Takeda <takedakn@nttdata.co.jp>
Cc: Matt Helsley <matthltc@us.ibm.com>
Cc: Nick Piggin <npiggin@kernel.dk>
Cc: Oleg Nesterov <oleg@redhat.com>
Cc: Peter Zijlstra <a.p.zijlstra@chello.nl>
Cc: Robert Richter <robert.richter@amd.com>
Cc: Suresh Siddha <suresh.b.siddha@intel.com>
Cc: Tetsuo Handa <penguin-kernel@I-love.SAKURA.ne.jp>
Cc: Venkatesh Pallipadi <venki@google.com>
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support of
D3 clock-stop. Also changing the power_save option in sysfs kicks
off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in HD-audio
are continued cleanups and standardization for the generic auto
parser and bug fixes (HBR, device-specific fixups), in addition to
the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode.
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support
of D3 clock-stop. Also changing the power_save option in sysfs
kicks off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in
HD-audio are continued cleanups and standardization for the generic
auto parser and bug fixes (HBR, device-specific fixups), in
addition to the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode."
Fix up various arm soc header file reorg conflicts.
* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
ALSA: hda - Add new codec ALC283 ALC290 support
ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
ALSA: hda - fix indices on boost volume on Conexant
ALSA: aloop - add locking to timer access
ALSA: hda - Fix hang caused by race during suspend.
sound: Remove unnecessary semicolon
ALSA: hda/realtek - Fix detection of ALC271X codec
ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
ALSA: hda - make a generic unsol event handler
ASoC: codecs: Add DA9055 codec driver
ASoC: eukrea-tlv320: Convert it to platform driver
ALSA: ASoC: add DT bindings for CS4271
ASoC: wm_hubs: Ensure volume updates are handled during class W startup
ASoC: wm5110: Adding missing volume update bits
ASoC: wm5110: Add OUT3R support
ASoC: wm5110: Add AEC loopback support
ASoC: wm5110: Rename EPOUT to HPOUT3
ASoC: arizona: Add more clock rates
ASoC: arizona: Add more DSP options for mixer input muxes
...
Passing struct snd_dma_buffer pointer instead, so that they work no
matter whether real SG buffer is used or not.
This is a preliminary work for the HD-audio DSP loader code.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit ALSA: compress_core: integer overflow in snd_compr_allocate_buffer()
added a new error check for input params.
this add new routine for input checks and moves buffer overflow check to this
new routine. This allows the error value to be propogated to user space
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These are 32 bit values that come from the user, we need to check for
integer overflows or we could end up allocating a smaller buffer than
expected.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always
false and it will never do compress capture. The test for O_WRONLY is
also slightly off. The original test would consider "->flags =
(O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid.
I've also removed the pr_err() because that could flood dmesg.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In general, mono streams have no dedicated speaker assignment, thus
they should be rather marked as UNKNOWN position.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The VOLATILE flag was added to control elements by
snd_pcm_add_chmap_ctls() just because I didn't want to have a
side-effect of "alsactl restore". But now the set operation doesn't
allow to change the value unless the PCM stream is in PREAPRED state,
there is no reason to keep this flag. Let's rip it off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SNDRV_CTL_ELEM_ACCESS_VOLATILE bit flag wasn't properly inherited
at creating control elements via snd_ctl_new1().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements the basic data types for the standard channel
mapping API handling.
- The definitions of the channel positions and the new TLV types are
added in sound/asound.h and sound/tlv.h, so that they can be
referred from user-space.
- Introduced a new helper function snd_pcm_add_chmap_ctls() to create
control elements representing the channel maps for each PCM
(sub)stream.
- Some standard pre-defined channel maps are provided for
convenience.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the main ALSA version number from the kernel ALSA driver.
The ALSA driver package release diverges from the upstream. This may
confuse users to see the same ALSA version for many kernel releases
and this version lost it's original purpose and connection.
The "ioctl" APIs have own version numbers, so the user space may check
for specific API changes only.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
uinfo has been allocated in this function and should be
freed before leaving from the error handling cases.
spatch with a semantic match is used to found this problem.
(http://coccinelle.lip6.fr/)
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context. The biggest batch are for the newly
added Arizona drivers.
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Merge tag 'asoc-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Additional updates for 3.6
A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context. The biggest batch are for the newly
added Arizona drivers.
The new LTO EXPORT_SYMBOL references symbols even without CONFIG_MODULES.
Since these functions are macros in this case this doesn't work.
Add a ifdef to fix the build.
Signed-off-by: Andi Kleen <ak@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Containing only a few really small/trivial fixes.
The only urgent fix is a regression fix of HDMI codec probing,
introduced in 3.6-rc1. The rest are HD-audio specific fixes and
a copule of minor bug fixes in PCM core and the old emu10k1.
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Merge tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Containing only a few really small/trivial fixes. The only urgent fix
is a regression fix of HDMI codec probing, introduced in 3.6-rc1. The
rest are HD-audio specific fixes and a copule of minor bug fixes in
PCM core and the old emu10k1."
* tag 'sound-3.6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Fix double quirk for Quanta FL1 / Lenovo Ideapad
ALSA: hda - Fix ugly debug prints with CONFIG_SND_VERBOSE_PRINTK=y
ALSA: hda - remove redundant auto quirks for conexant 506x
ALSA: hda - remove quirk for Dell Vostro 1015
ALSA: hda - add dock support for Thinkpad X230
ALSA: hda - Fix regression of HDMI codec probing
ALSA: hda - add dock support for Thinkpad T430s
ALSA: emu10k1: Avoid access to invalid pages when period=1
ALSA: PCM: Fix possible memory leaks in the error path
Make the output logging routine independent of the KERN_<LEVEL> style.
Signed-off-by: Joe Perches <joe@perches.com>
Cc: Kay Sievers <kay.sievers@vrfy.org>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
They aren't modified by the core so the drivers can declare them const.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is essentially the reverse of snd_pcm_rate_to_rate_bit().
This is generally useful as the Compress API uses the rate bit
directly and it helps to be able to map back to the actual sample
rate.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai