Commit Graph

714 Commits

Author SHA1 Message Date
Linus Torvalds a2013a13e6 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
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
  ...
2012-12-13 12:00:02 -08:00
Denis Washington 1d31affbef ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3
The only required change is to extend the existing Xonar U1
mixer quirks to the U3, which seems to be controlled the same
way.

Signed-off-by: Denis Washington <denisw@online.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-12 11:32:54 +01:00
Jurgen Kramer 9621055fbb ALSA: usb6fire: prevent driver panic state when stopping
The patch below prevents the 6fire usb driver going into panic state
when stopping playing. On some systems the urb in handler
(usb6fire_pcm_in_urb_handler) is being called while urbs are being
killed off, this causes the driver to set panic state and can result in
the kernel warning 'URB %p submitted while active'.

Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-07 15:03:34 +01:00
Bill Pemberton 14c56706f9 ALSA: snd-usb-caiaq: remove __dev* attributes
CONFIG_HOTPLUG is going away as an option.  As result the __dev*
markings will be going away.

Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.

Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-07 07:35:11 +01:00
Bill Pemberton 87f9796a03 ALSA: snd-usb-6fire: remove __dev* attributes
CONFIG_HOTPLUG is going away as an option.  As result the __dev*
markings will be going away.

Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.

Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-07 07:34:46 +01:00
Eldad Zack 0d9741c0e0 ALSA: usb-audio: sync ep init fix for audioformat mismatch
Commit 947d299686 , "ALSA: snd-usb:
properly initialize the sync endpoint", while correcting the
initialization of the sync endpoint when opening just the data
endpoint, prevents devices that has a sync endpoint, with a channel
number different than that of the data endpoint, from functioning.
Due to a different channel and period bytes count, attempting to
initialize the sync endpoint will fail at the usb host driver.
For example, when using xhci:

 cannot submit urb 0, error -90: internal error

With this patch, if a sync endpoint has multiple audioformats, a
matching audioformat is preferred. An audioformat must be found
with at least one channel and support the requested sample rate
and PCM format, otherwise the stream will not be opened.

If the number of channels differ between the selected audioformat
and the requested format, adjust the period bytes count accordingly.
It is safe to perform the calculation on the basis of the channel
count, since the requested PCM audio format and the rate must be
supported by the selected audioformat.

Cc: Jeffrey Barish <jeff_barish@earthlink.net>
Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-04 08:14:31 +01:00
Takashi Iwai f5f165418c ALSA: usb-audio: Fix missing autopm for MIDI input
The commit [88a8516a: ALSA: usbaudio: implement USB autosuspend] added
the support of autopm for USB MIDI output, but it didn't take the MIDI
input into account.

This patch adds the following for fixing the autopm:
- Manage the URB start at the first MIDI input stream open, instead of
  the time of instance creation
- Move autopm code to the common substream_open()
- Make snd_usbmidi_input_start/_stop() more robust and add the running
  state check

Reviewd-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-04 07:27:44 +01:00
Takashi Iwai 59866da9e4 ALSA: usb-audio: Avoid autopm calls after disconnection
Add a similar protection against the disconnection race and the
invalid use of usb instance after disconnection, as well as we've done
for the USB audio PCM.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=51201

Reviewd-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-04 07:27:27 +01:00
David Henningsson 9b4ef97757 ALSA: usb - Don't create "Speaker" mixer controls on headphones and headsets
A lot of headsets/headphones have a "Speaker" mixer control. This confuses
PulseAudio to think it is a speaker instead of a headphone/headset.
Therfore, we rename it to "Headphone".

We determine if something is a headphone similar to how udev determines
form factor (see 78-sound-card.rules).

BugLink: https://bugs.launchpad.net/bugs/1082357
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 13:59:47 +01:00
Eldad Zack ca10a7ebdf ALSA: usb-audio: FT C400 sync playback EP to capture EP
The playback endpoint uses implicit feedback mode, similar
to the M-Audio FTU. Like with the FTU, we need to associate
the sync pipe ourselves.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:45:18 +01:00
Eldad Zack 09d8e3a71d ALSA: usb-audio: Fast Track C400 mixer controls
Add a mixer quirks for the M-Audio Fast Track C400
and create the following:

* Volume controls
* Effect Type (reusing FTU controls)
* Effect Volume
* Effect Send/Return
* Effect Program
* Effect Feedback

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:44:55 +01:00
Eldad Zack d50ed624e4 ALSA: usb-audio: Fast Track C400 mixer ranges
Add ranges for various Fast Track C400 controls, as observed
while using the vendor's mixer control software (res values
are an estimation).

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:44:42 +01:00
Eldad Zack 76f74bca73 ALSA: usb-audio: M-Audio Fast Track C400 quirks table
Adds a quirks table for the M-Audio Fast Track C400.
Thanks to Clemens Ladisch <clemens@ladisch.de> for pointing out that
the table must be sorted.

Based on the following patch from the alsa-devel list:
http://mailman.alsa-project.org/pipermail/alsa-devel/2012-May/051676.html

See also:
http://mailman.alsa-project.org/pipermail/alsa-devel/2012-April/051219.html

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:44:17 +01:00
Eldad Zack d847ce0e9a ALSA: usb-audio: parameterize FTU effect unit control
Adds the unit ID and the control as parameters to the creation of the
effect unit control for the M-Audio Fast Track Ultra. This allows the
code to be shared with other devices that use different unit ID and
control, such as the M-Audio Fast Track C400.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:43:55 +01:00
Eldad Zack 5dae5fd240 ALSA: usb-audio: skip UAC2 EFFECT_UNIT
Current code mishandles the case where the device is a UAC2
and the bDescriptorSubtype is a UAC2 Effect Unit (0x07).
It tries to parse it as a Processing Unit (which is similar to two
other UAC1 units with overlapping subtypes), but since the structure
is different (See: 4.7.2.10, 4.7.2.11 in UAC2 standard), the parsing
is done incorrectly and prevents the device from initializing.
For now, just ignore the unit.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:43:31 +01:00
Eldad Zack 9f81410592 ALSA: usb-audio: add control index offset
Currently, channel IDs exceeding 31 (0x1f) cannot be used.
The channel ID is derived from the cmask. Extending cmask
to a 64-bit type would only allow it to go up to 63 (0x3f).
Some devices have channel IDs exceeding that as well.
To address that, add an offset to the mixer element which
is then accounted for in the UAC set/get functions.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:43:12 +01:00
Eldad Zack 28acb12014 ALSA: usb-audio: use sender stride for implicit feedback
For implicit feedback endpoints, the number of bytes for each packet
is matched by the corresponding synchronizing endpoint.
The size is calculated by taking the actual size and dividing it by
the stride - currently by the endpoint's stride, but we should use the
synchronization source's stride.
This is evident when the number of channels differ between the
synchronization source and the implicitly fed-back endpoint, as with
M-Audio Fast Track C400 - the synchronization source (capture)
has 4 channels, while the implicit feedback mode endpoint has 6.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:42:54 +01:00
Eldad Zack fde854bdaf ALSA: usb-audio: replace hardcoded value with const
In this context, 0x01 is USB_ENDPOINT_XFER_ISOC.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:42:33 +01:00
Takashi Iwai 04324ccc75 ALSA: usb-audio: add channel map support
Add the support for channel maps of the PCM streams on USB audio
devices.  The channel map information is already found in
ChannelConfig descriptor entries, which haven't been referred until
now.

Each chmap entry is added to audioformat list entry and copied to TLV
dynamically instead of creating a whole chmap array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-26 16:24:02 +01:00
Takashi Iwai 48779a0b8f ALSA: usb-audio: fix delay account during pause
When a playback stream is paused, the stream isn't actually stopped,
thus we still need to take care of the in-flight data amount for the
delay calculation.  Otherwise the value of subs->last_delay is no
longer reliable and can give a bogus value after resuming from pause.
This will result in "delay: estimated XX, actual YY" error messages.

Also, during pause after all in flight data are processed
(i.e. last_delay = 0), we don't have to calculate the actual delay
from the current frame.  Give a short path in such a case.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23 16:07:11 +01:00
Takashi Iwai 3f94fad095 ALSA: usb-audio: ignore delay calculation for capture stream
It doesn't make sense to calculate the delay for capture streams in
the current implementation.  It's always zero, so we should skip the
computation in snd_usb_pcm_pointer() in the case of capture.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23 15:37:32 +01:00
Takashi Iwai 2ba509a6ba Merge branch 'for-linus' into for-next 2012-11-22 21:22:39 +01:00
Daniel Mack 947d299686 ALSA: snd-usb: properly initialize the sync endpoint
Jeffrey Barish reported an obvious bug in the pcm part of the usb-audio
driver which causes the code to not initialize the sync endpoint from
configure_endpoint().

Reported-by: Jeffrey Barish <jeff_barish@earthlink.net>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22 21:22:33 +01:00
Takashi Iwai b0db6063db ALSA: usb-audio: process pending stop at PCM hw_free and close
PCM hw_free and close should wait until all the pending stop
operations have been finished.  Basically only PCM trigger callback
should use non-wait calls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:43:58 +01:00
Takashi Iwai b2eb950de2 ALSA: usb-audio: stop both data and sync endpoints asynchronously
As we are stopping the endpoints asynchronously now, it's better to
trigger the stop of both data and sync endpoints and wait for pending
stopping operations, instead of the sequential trigger-and-wait
procedure.

So the wait argument in snd_usb_endpoint_stop() is dropped, and it's
expected that the caller synchronizes explicitly by calling
snd_usb_endpoint_sync_pending_stop().  (Actually there is only one
place calling this, so it was safe to change.)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:43:56 +01:00
Takashi Iwai ccc1696d52 ALSA: usb-audio: simplify endpoint deactivation code
For further code simplification, drop the conditional call for
usb_kill_urb() with can_wait argument in deactivate_urbs(), and use
only usb_unlink_urb() and wait_clear_urbs() pairs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:43:54 +01:00
Takashi Iwai a9bb36261e ALSA: usb-audio: simplify snd_usb_endpoint_start/stop arguments
Reduce the redundant arguments for snd_usb_endpoint_start() and
snd_usb_endpoint_stop().  Also replaced from int to bool.

No functional changes by this commit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:43:40 +01:00
Takashi Iwai 20d32022a8 ALSA: usb-audio: Deprecate async_unlink option
The async unlink behavior has been working over years.  The option was
provided only as a workaround for 2.4.x kernel.  Let's get rid of it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:37:40 +01:00
Sachin Kamat 8ad10dc6d3 ALSA: usb-audio: Return meaningful error codes instead of -1 in format.c
Also, silences the following smatch warning:
sound/usb/format.c:170 parse_audio_format_rates_v1() warn:
returning -1 instead of -ENOMEM is sloppy

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:31:52 +01:00
Sachin Kamat 27b2a22c71 ALSA: usb/6fire: Fix potential NULL pointer dereference in comm.c
'rt' was dereferenced before the NULL check.
Moved the code after the check.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 10:43:52 +01:00
Takashi Iwai 87af0b80c9 Merge branch 'for-linus' into for-next
Merge the recent HD-audio codec change for fixing recursive suspend
calls.

Conflicts:
	sound/pci/hda/hda_codec.c
2012-11-19 21:25:27 +01:00
Adam Buchbinder 48fc7f7e78 Fix misspellings of "whether" in comments.
"Whether" is misspelled in various comments across the tree; this
fixes them. No code changes.

Signed-off-by: Adam Buchbinder <adam.buchbinder@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2012-11-19 14:31:35 +01:00
Takashi Iwai 0ced14fbda Merge branch 'usb-midi-fix-3.7' of git://git.alsa-project.org/alsa-kprivate into for-linus
Merge a regression fix for USB MIDI on non-standard usb-audio drivers
by Clemens.
2012-11-19 09:55:06 +01:00
Clemens Ladisch e99ddfde6a ALSA: ua101, usx2y: fix broken MIDI output
Commit 88a8516a21 (ALSA: usbaudio: implement USB autosuspend) added
autosuspend code to all files making up the snd-usb-audio driver.
However, midi.c is part of snd-usb-lib and is also used by other
drivers, not all of which support autosuspend.  Thus, calls to
usb_autopm_get_interface() could fail, and this unexpected error would
result in the MIDI output being completely unusable.

Make it work by ignoring the error that is expected with drivers that do
not support autosuspend.

Reported-by: Colin Fletcher <colin.m.fletcher@googlemail.com>
Reported-by: Devin Venable <venable.devin@gmail.com>
Reported-by: Dr Nick Bailey <nicholas.bailey@glasgow.ac.uk>
Reported-by: Jannis Achstetter <jannis_achstetter@web.de>
Reported-by: Rui Nuno Capela <rncbc@rncbc.org>
Cc: Oliver Neukum <oliver@neukum.org>
Cc: 2.6.39+ <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2012-11-18 17:15:24 +01:00
Joe Perches 190006f9d6 ALSA: usb-audio: use bitmap_weight
Use bitmap_weight to count the total number of bits set in bitmap.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-17 11:35:07 +01:00
Takashi Iwai 10e44239f6 ALSA: usb-audio: Fix mutex deadlock at disconnection
The recent change for USB-audio disconnection race fixes introduced a
mutex deadlock again.  There is a circular dependency between
chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a
device is opened during the disconnection operation:

A. snd_usb_audio_disconnect() ->
     card.c::register_mutex ->
       chip->shutdown_rwsem (write) ->
         snd_card_disconnect() ->
           pcm.c::register_mutex ->
             pcm->open_mutex

B. snd_pcm_open() ->
     pcm->open_mutex ->
       snd_usb_pcm_open() ->
         chip->shutdown_rwsem (read)

Since the chip->shutdown_rwsem protection in the case A is required
only for turning on the chip->shutdown flag and it doesn't have to be
taken for the whole operation, we can reduce its window in
snd_usb_audio_disconnect().

Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-14 15:29:09 +01:00
Martin Schwenke 1762a59d8e ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i6
Probing this device currently fails in snd_usb_audio_probe() because
the call to snd_usb_create_mixer() fails.  This is due to unknown or
non-standard interface descriptor subtypes in parse_audio_unit():

  usbaudio: unit 51: unexpected type 0x09
  snd-usb-audio: probe of 1-8:1.0 failed with error -5

Some people are working around this by recompiling usb-audio with the
call to snd_usb_create_mixer() commented out.  It would be nice to
avoid that.

While the best idea would be to look into the mixer creation failure,
a reasonable short-term solution is to use quirks to only probe the
trouble-free interfaces.  This allows audio and MIDI interfaces to be
used without any obvious issues.

Interface 0 is the main one to ignore.  It contains lots of
control-fu, including the unexpected interface descriptor subtypes.
Interface 5 is for firmware updates and I'm not sure how to get
support for this.  Interface 3 is some sort of control interface that
I don't understand:

    Interface Descriptor:
      bLength                 9
      bDescriptorType         4
      bInterfaceNumber        3
      bAlternateSetting       0
      bNumEndpoints           0
      bInterfaceClass         1 Audio
      bInterfaceSubClass      1 Control Device
      bInterfaceProtocol      0
      iInterface              0
      AudioControl Interface Descriptor:
        bLength                 9
        bDescriptorType        36
        bDescriptorSubtype      1 (HEADER)
        bcdADC               1.00
        wTotalLength            9
        bInCollection           1
        baInterfaceNr( 0)       1

Signed-off-by: Martin Schwenke <martin@meltin.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-13 09:47:13 +01:00
Takashi Iwai 17a4adbe68 Merge branch 'for-linus' into for-next 2012-11-08 15:58:25 +01:00
Takashi Iwai f58161ba1b ALSA: usb-audio: Fix crash at re-preparing the PCM stream
There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback.  It turned out that the problem is that we don't
wait until all URBs are killed.

This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181

Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-08 08:56:44 +01:00
Takashi Iwai a5d00dc3a4 Merge branch 'for-linus' into for-next
... for migrating the core changes for USB-audio disconnection fixes
2012-10-30 11:08:25 +01:00
Takashi Iwai 888ea7d5ac ALSA: usb-audio: Fix races at disconnection in mixer_quirks.c
Similar like the previous commit, cover with chip->shutdown_rwsem
and chip->shutdown checks.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:07:05 +01:00
Takashi Iwai 34f3c89fda ALSA: usb-audio: Use rwsem for disconnect protection
Replace mutex with rwsem for codec->shutdown protection so that
concurrent accesses are allowed.

Also add the protection to snd_usb_autosuspend() and
snd_usb_autoresume(), too.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:07:00 +01:00
Takashi Iwai 978520b75f ALSA: usb-audio: Fix races at disconnection
Close some races at disconnection of a USB audio device by adding the
chip->shutdown_mutex and chip->shutdown check at appropriate places.

The spots to put bandaids are:
- PCM prepare, hw_params and hw_free
- where the usb device is accessed for communication or get speed, in
 mixer.c and others; the device speed is now cached in subs->speed
 instead of accessing to chip->dev

The accesses in PCM open and close don't need the mutex protection
because these are already handled in the core PCM disconnection code.

The autosuspend/autoresume codes are still uncovered by this patch
because of possible mutex deadlocks.  They'll be covered by the
upcoming change to rwsem.

Also the mixer codes are untouched, too.  These will be fixed in
another patch, too.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:06:54 +01:00
Kees Cook f598158aa4 ALSA: sound/usb: remove CONFIG_EXPERIMENTAL
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.

Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-25 00:17:47 +02:00
Didier Villevalois c902466800 ALSA: usb-audio: Add quirk for Reloop Play
The Reloop Audio needs a fixed endpoint quirk with S24_3LE format and
UAC_EP_CS_ATTR_SAMPLE_RATE attribute.

Signed-off-by: Didier Villevalois <ptitjes@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-23 16:38:14 +02:00
Pete Leigh 7a75e742fa ALSA: usb-audio: USB audio quirk for Roland VG-99 advanced mode
Without this quirk the VG-99 will work in standard mode (set under
USB on System menu page 2) giving 16 bits at 44.1 Khz audio in/out
but no midi, and is not recognised when set to advanced mode.

After applying this, I can also use the VG-99 in advanced mode: 24
24 bits audio in/out at 44.1 Khz, and midi in/out. Sysex is so far
untested.

In standard mode, the device appears with ID 0x00b3, so the
behaviour isn't affected by this quirk.

Thanks to Clemens Ladisch for simplifying and correcting my initial
attempt!

Signed-off-by: Pete Leigh <pete.leigh@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-21 12:05:03 +02:00
Wei Yongjun 950f40fdd4 ALSA: snd-usb: remove unused variable in init_pitch_v2()
The variable ep is initialized but never used
otherwise, so remove the unused variable.

dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-21 10:43:27 +02:00
Linus Torvalds 2fc07efa22 Sound updates #2 for 3.7-rc1
This update contains a few cleanup works, regression/stable fixes
 gathered since the last pull request.
 
 - Clean up with generic hd-audio jack handling code by David
   Henningsson
 - A few regression fixes for standardized HD-audio auto-parser
 - Misc clean-up and small fixes
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates #2 from Takashi Iwai:
 "This update contains a few cleanup works, regression/stable fixes
  gathered since the last pull request.

   - Clean up with generic hd-audio jack handling code by David
     Henningsson
   - A few regression fixes for standardized HD-audio auto-parser
   - Misc clean-up and small fixes"

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - do not detect jack on internal speakers for Realtek
  ALSA: hda - Fix missing beep on ASUS X43U notebook
  ALSA: hda - Remove AZX_DCAPS_POSFIX_COMBO
  ALSA: hda - Warn an allocation for an uninitialized array
  ALSA: hda/cirrus - Add missing init/free of hda_gen_spec
  ALSA: hda - Fix memory leaks at error path in patch_cirrus.c
  ALSA: hda - Add missing hda_gen_spec to struct via_spec
  ALSA: hda - remove "Mic Jack Mode" for headset jacks (Latitude Exx30)
  ALSA: hda - make Cirrus codec use generic unsol event handler
  ALSA: hda - make VIA codec use generic unsol event handler
  ALSA: hda - Remove dead GPIO code for VIA codec
  ALSA: usb-audio: Add TASCAM US122 MKII playback
2012-10-12 12:31:28 +09:00
Konstantin Khlebnikov 314e51b985 mm: kill vma flag VM_RESERVED and mm->reserved_vm counter
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>
2012-10-09 16:22:19 +09:00
Linus Torvalds f5a246eab9 Sound updates for 3.7-rc1
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
  ...
2012-10-09 07:07:14 +09:00