Commit Graph

737 Commits

Author SHA1 Message Date
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
Oto Petřík
613769fcab ALSA: usb-audio: Add TASCAM US122 MKII playback
Added quirk to provide at least playback-only support.

Signed-off-by: Oto Petrik <oto.petrik@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-08 15:16:33 +02:00
Daniel Mack
8dce30c891 ALSA: snd-usb: fix next_packet_size calls for pause case
Also fix the calls to next_packet_size() for the pause case. This was
missed in 245baf983 ("ALSA: snd-usb: fix calls to next_packet_size").

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reported-and-tested-by: Christian Tefzer <ctrefzer@gmx.de>
Cc: stable@kernel.org
[ Taking directly because Takashi is on vacation  - Linus ]
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2012-09-27 16:46:15 -07:00
David Henningsson
c10514394e ALSA: usb - disable broken hw volume for Tenx TP6911
While going through Ubuntu bugs, I discovered this patch being
posted and a confirmation that the patch works as expected.

Finding out how the hw volume really works would be preferrable
to just disabling the broken one, but this would be better than
nothing.

Credit: sndfnsdfin (qawsnews)
BugLink: https://bugs.launchpad.net/bugs/559939
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-20 10:48:47 +02:00
Takashi Iwai
384dc085c3 ALSA: usb-audio: Avoid unnecessary EP setups in prepare
The recent fix for USB suspend breakage moved the code to set up EP
from hw_params to prepare, but it means also the EP setup might be
called multiple times unnecessarily because the prepare callback can
be called multiple times without starting the stream (e.g. OSS
emulation).

This patch adds a new flag to struct snd_usb_substream indicating
whether the setup of EP is required, and do it only when necessary,
i.e. right after hw_params or suspend.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19 08:08:16 +02:00
Dylan Reid
61a709504b ALSA: usb-audio: Move configuration to prepare.
Move interface and endpoint configuration from hw_params to prepare
callback.  During system suspend/resume when the USB device power isn't
cycled the interface and endpoint configuration need to be set before
audio playback can continue.  Resume involves another call to prepare
but not to hw_params, moving it here allows a playing stream to continue
after resume.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19 08:08:11 +02:00
Dylan Reid
35ec7aa298 ALSA: usb-audio: Don't require hw_params in endpoint.
Change the interface to configure an endpoint so that it doesn't require
a hw_params struct.  This will allow it to be called from prepare
instead of hw_params, configuring it after system resume.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19 08:07:52 +02:00
Dylan Reid
715a170563 ALSA: usb-audio: set period_bytes in substream.
Set the peiod_bytes member of snd_usb_substream.  It was no longer being
set, but will be needed to resume properly in a future commit.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19 08:07:34 +02:00
Takashi Iwai
0528842690 Merge branch 'for-linus' into for-next
To merge HD-audio fixes back to 3.7 development line
2012-09-11 16:46:36 +02:00
Takashi Iwai
1213a205f9 ALSA: usb-audio: Fix bogus error messages for delay accounting
The recent fix for the missing fine delayed time adjustment gives
strange error messages at each start of the playback stream, such as
  delay: estimated 0, actual 352
  delay: estimated 353, actual 705

These come from the sanity check in retire_playback_urb().  Before the
stream is activated via start_endpoints(), a few silent packets have
been already sent.  And at this point the delay account is still in
the state as if the new packets are just queued, so the driver gets
confused and spews the bogus error messages.

For fixing the issue, we just need to check whether the received
packet is valid, whether it's zero sized or not.

Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-06 15:00:15 +02:00
Daniel Mack
2b58fd5b31 ALSA: snd-usb: Add quirks for Playback Designs devices
Playback Designs' USB devices have some hardware limitations on their
USB interface. In particular:

 - They need a 20ms delay after each class compliant request as the
   hardware ACKs the USB packets before the device is actually ready
   for the next command. Sending data immediately will result in buffer
   overflows in the hardware.
 - The devices send bogus feedback data at the start of each stream
   which confuse the feedback format auto-detection.

This patch introduces a new quirks hook that is called after each
control packet and which adds a delay for all devices that match
Playback Designs' USB VID for now.

In addition, it adds a counter to snd_usb_endpoint to drop received
packets on the floor. Another new quirks function that is called once
an endpoint is started initializes that counter for these devices on
their sync endpoint.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com>
Supported-by: Demian Martin <demianm_1@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-04 11:31:14 +02:00
Marko Friedemann
c05fce586d ALSA: USB: Support for (original) Xbox Communicator
Added support for Xbox Communicator to USB quirks.

Signed-off-by: Marko Friedemann <mfr@bmx-chemnitz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-03 10:14:25 +02:00
Daniel Mack
2e4a263ca8 ALSA: snd-usb: fix cross-interface streaming devices
Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface")
saved us some unnecessary calls to snd_usb_set_interface() but ignored
the fact that there is at least one device out there which operates on
two endpoint in different interfaces simultaniously.

Take care for this by catching the case where data and sync endpoints
are located on different interfaces and calling snd_usb_set_interface()
between the start of the two endpoints.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Robert M. Albrecht <linux@romal.de>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 21:04:53 +02:00
Daniel Mack
245baf983c ALSA: snd-usb: fix calls to next_packet_size
In order to support devices with implicit feedback streaming models,
packet sizes are now stored with each individual urb, and the PCM
handling code which fills the buffers purely relies on the size fields
now.

However, calling snd_usb_audio_next_packet_size() for all possible
packets in an URB at once, prior to letting the PCM code do its job
does in fact not lead to the same behaviour than what the old code did:
The PCM code will break its loop once a period boundary is reached,
consequently using up less packets that it really could.

As snd_usb_audio_next_packet_size() implements a feedback mechanism to
the endpoints phase accumulator, the number of calls to that function
matters, and when called too often, the data rate runs out of bounds.

Fix this by making the next_packet function public, and call it from the
PCM code as before if the packet data sizes are not defined.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 21:03:48 +02:00
Daniel Mack
fbcfbf5f67 ALSA: snd-usb: restore delay information
Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB
frame counter") were unfortunately lost during the refactoring of the
snd-usb driver in 3.5.

This patch adds them back, restoring the correct delay information
behaviour.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 21:03:08 +02:00
Pavel Roskin
03d2f44e96 ALSA: snd-usb: use list_for_each_safe for endpoint resources
snd_usb_endpoint_free() frees the structure that contains its argument.

Signed-off-by: Pavel Roskin <proski@gnu.org>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 18:17:45 +02:00
Daniel Mack
015618b902 ALSA: snd-usb: Fix URB cancellation at stream start
Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in
PCM capture stream") fixed a scheduling-while-atomic bug that happened
when snd_usb_endpoint_start was called from the trigger callback, which
is an atmic context. However, the patch breaks the idea of the endpoints
reference counting, which is the reason why the driver has been
refactored lately.

Revert that commit and let snd_usb_endpoint_start() take care of the URB
cancellation again. As this function is called from both atomic and
non-atomic context, add a flag to denote whether the function may sleep.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30 07:46:27 +02:00
Takashi Iwai
48ee7cb8b4 ALSA: usb-audio: Remove obsoleted fields in struct snd_usb_substream
The two entries are duplicated in struct snd_usb_endpoint.
Seems forgotten in the last clean-up.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-28 16:30:02 -07:00
Takashi Iwai
ddf83485d7 Merge branch 'for-linus' into for-next
Conflicts:
	sound/pci/hda/hda_codec.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 22:14:26 +02:00
Takashi Iwai
e9ba389c5f ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream
A PCM capture stream on usb-audio causes a scheduling-while-atomic
BUG, as reported in the bugzilla entry below.  It's because
snd_usb_endpoint_start() is called at first at trigger START for a
capture stream, and this function contains the left-over EP
deactivation codes.  The problem doesn't happen for a playback stream
because the function is called at PCM prepare time, which can sleep.

This patch fixes the BUG by moving the EP deactivation code into the
PCM prepare callback.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-16 08:04:07 +02:00
Andy Shevchenko
793ea49c47 ALSA: print small buffers via %*ph[C]
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-06 11:09:50 +02:00
Daniel Mack
aff252a848 ALSA: snd-usb: fix clock source validity index
uac_clock_source_is_valid() uses the control selector value to access
the bmControls bitmap of the clock source unit. This is wrong, as
control selector values start from 1, while the bitmap uses all
available bits.

In other words, "Clock Validity Control" is stored in D3..2, not D5..4
of the clock selector unit's bmControls.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-01 10:24:16 +02:00
Takashi Iwai
f0913cd16e Merge branch 'topic/misc' into for-next
Generic updates for sound 3.6
2012-07-18 13:53:29 +02:00
Daniel Mack
68e67f40b7 ALSA: snd-usb: move calls to usb_set_interface
The rework of the snd-usb endpoint logic moved the calls to
snd_usb_set_interface() into the snd_usb_endpoint implemenation. This
changed the order in which these calls are issued to the device, and
thereby caused regressions for some webcams.

Fix this by moving the calls back to pcm.c for now to make it work again
and use snd_usb_endpoint_activate() to really tear down all remaining
URBs in the flight, consequently fixing another regression caused by USB
packets on the wire after altsetting 0 has been selected.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Philipp Dreimann <philipp@dreimann.net>
Reported-by: Joseph Salisbury <joseph.salisbury@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-13 09:31:42 +02:00
Takashi Iwai
9e9b594661 ALSA: usb-audio: Fix the first PCM interface assignment
In the new PCM streaming logic, the interface number is assigned to
usb stream instance (subs->interface) after the format and rate setups
are succeeded, but some codes are still passing subs->interface as the
reference to helper functions.  This leads to initializing with an
invalid iface number (-1).

This patch replaces the wrong references with the ones from the target
fmt correctly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-06 08:11:43 +02:00
Daniel Mack
da185443c1 ALSA: snd-usb-caiaq: initialize card pointer
Fixes the following warning:

  CC [M]  sound/usb/caiaq/device.o
sound/usb/caiaq/device.c: In function ‘snd_probe’:
sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used
uninitialized in this function [-Wmaybe-uninitialized]

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-27 12:26:19 +02:00
Clemens Ladisch
74953e2010 ALSA: usb-audio: add BOSS GT-100 support
Reported-by: John McFarland <mcfarljm@gmail.com>
Tested-by: John McFarland <mcfarljm@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-25 11:11:24 +02:00
Oleksij Rempel
b64a1ba9d3 ALSA: snd_usb_audio: ignore ctrl errors on QuickCam Pro for Notebooks
This webcam works mostly ok, exept with skype.
Skype sends lots of ctrl messages to dynamically ajust
record level. If for some reasons it pokes some error
every thing goes broken:
- first pulseaudio blocks sound for all apps
- then video is reseted
- then skype freez

dmesg has lots of messages like:
cannot set freq 16000 to ep 0x86"

Setting ignore_ctl_error=1 fixes this problem.

Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-20 08:29:08 +02:00
Oleksij Rempel
05b9afd5b7 ALSA: snd_usb_audio: ignore ctrl errors on QuickCam E3500
if this cam is pluged in, pulse audio can't initiate capture
device.
dmesg has lots of messages like:
"cannot set freq 16000 to ep 0x86"

Setting ignore_ctl_error=1 fixes this problem.

Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-20 08:28:57 +02:00
Daniel Mack
0b1d8e0908 ALSA: 6fire: use NULL instead of 0 for pointer assignment
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-18 09:36:38 +02:00
Daniel Mack
afe25967ec ALSA: snd-usb: make snd_usb_substream_capture_trigger static
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-18 09:32:53 +02:00
Daniel Mack
7fb75db139 ALSA: snd-usb: fix sync pipe check
Fix a bogus sanity check for sync pipe in pcm.c. This flaw was
introduced during the streaming logic refactorization.

While at it, improve the error messages that are generated in such cases.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: <ben@b1c1l1.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-18 08:36:36 +02:00
Mark Hills
989b01385f ALSA: usb-audio: Convert table to preferred C99 format
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-11 12:49:56 +02:00
Mark Hills
b71dad181a ALSA: usb-audio: Use a table of mixer controls
Allow mixer controls to be provided clearly in a table, to avoid
quantity of error checking at each use.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-11 12:49:43 +02:00
Takashi Iwai
8260ef075b ALSA: usb-audio: Fix substream assignments
In 3.5 kernel, the endpoint is assigned dynamically for the
substreams, but the PCM assignment still checks the presence of the
endpoint pointer.  This ended up in duplicated PCM substream creations
at probing time, resulting in kernel warnings like:

WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6()
Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2
Call Trace:
 [<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c
 [<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48
 [<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b
 [<ffffffff811292f0>] proc_register+0x169/0x1a6
 [<ffffffff8112962e>] create_proc_entry+0x74/0x8c
 [<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd]
 [<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm]
 [<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio]
 [<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio]
 [<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb
 [<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio]
 [<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio]
 [<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio]
 .....

This patch fixes the regression by checking the fixed endpoint number
for each substream instead of the endpoint pointer.

Reported-and-tested-by: Jamie Heilman <jamie@audible.transient.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-08 09:01:37 +02:00
Clemens Ladisch
5cd5d7c449 ALSA: usb-audio: fix rate_list memory leak
The array of sample rates is reallocated every time when opening
the PCM device, but was freed only once when unplugging the device.

Reported-by: "Alexander E. Patrakov" <patrakov@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-31 10:25:44 +02:00
Daniel Mack
97f8d3b650 ALSA: snd-usb: fix stream info output in /proc
Set some substream struct members to make the proc interface code work
again.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-21 12:51:08 +02:00
Takashi Iwai
e182534d4b ALSA: usb-audio - Call get_min_max_*() after determining the name string
get_min_max_with_quirks() must be called after the control id name
string is determined, but the current code changes the id name string
after calling the function.

Reported-by: Christian Melki <christian.melki@ericsson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-15 08:35:00 +02:00
Mark Hills
7df4a691fb ALSA: usb-audio: Fix comment
Explained by Takashi in <s5hfwbtmz0q.wl%tiwai@suse.de>

> The reason is because get_min_max*() isn't called in the place you
> created these controls, and get_min_max() would be called only for
> integer volumes later even if uninitialized.  A short cut for booleans.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-11 21:27:36 +02:00
Daniel Mack
07a5e9d4fd ALSA: snd-usb: fix some typos in endpoint.c documentation
Also be more specific about some details while at it.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 20:16:18 +02:00
Andrew Morton
68853fa30c ALSA: usb-audio: sound/usb/endpoint.c: suppress warning
sound/usb/endpoint.c: In function 'queue_pending_output_urbs':
sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function

Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:10:10 +02:00
Takashi Iwai
baba2e0d2b ALSA: usb-audio: Add missing error checks in snd_ebox44_create_mixer()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:07:38 +02:00
Felix Homann
d34bf14851 ALSA: usb-audio: M-Audio Fast Track Ultra: Add effect controls
This adds controls for the effects section on the FTU devices.
Some of these controls need volume quirks. They are added to
mixer.c.

[fixed missing break by tiwai]

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:06:06 +02:00
Felix Homann
cfe8f97c82 ALSA: usb-audio: Rename Fast Track Ultra mixer quirk functions
This is in preparation for more FTU controls to come.
Should help keeping names a bit shorter.

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:02:11 +02:00
Felix Homann
25ee7ef8fa ALSA: usb-audio: Add TLV to M-Audio Fast Track Ultra controls
This adds db gain information to M-Audio Fast Track Ultra (8R) devices.

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:01:46 +02:00
Felix Homann
285de9c08b ALSA: usb-audio: Rename and export mixer_vol_tlv
Rename mixer_vol_tlv to snd_usb_mixer_vol_tlv and export it to make
it reuseable in mixer_quirks.c.

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:01:27 +02:00
Felix Homann
8a4d1d397b ALSA: usb-audio: Unify M-Audio Fast Track Ultra and Ebox-44 mixer quirks.
Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().

[fixed minor checkpatch.pl warnings by tiwai]

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:00:45 +02:00
Daniel Mack
c89a5d9cac ALSA: snd-usb: remove refactorization left-overs
Drop some struct members and definitions that became obsolete during
the refactorization of the driver.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-21 17:40:28 +02:00
Takashi Iwai
56599bb020 Merge branch 'topic/usb-endpoint' into topic/misc 2012-04-18 07:57:32 +02:00
Mark Hills
7536c301f8 ALSA: snd-usb-audio: Replace mixer for Electrix Ebox-44
The mixer units from the firmware are corrupt, and even where they
are valid they presents mono controls as L and R channels of
stereo.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-15 15:40:08 +02:00
Mark Hills
284a8dd6f0 ALSA: snd-usb-audio: Skip un-parseable mixer units instead of erroring
Some interfaces reference endpoints which do not exists. To
accomodate these, do not fail completely, but skip over them.

This allows the Electrix Ebox-44 with earlier firmware to be
detected and used for audio.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-15 15:39:55 +02:00
Takashi Iwai
22026c1a7b ALSA: usb: Remove obsoleted fields from struct snd_usb_substream
Many fields have been moved to struct snd_usb_endpoint.
Also fix the proc output to correspond to the new structure.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 12:57:39 +02:00
Takashi Iwai
85f71932e5 ALSA: usb: Fix fill_max flag set
ep->fill_max is a 1 bit flag, thus it has to be boolean.
  sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params':
  sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 12:41:54 +02:00
Takashi Iwai
c5ee4ec828 ALSA: usb: Remove unused variable
sound/usb/endpoint.c: In function ‘deactivate_urbs’:
sound/usb/endpoint.c:520:16: warning: unused variable ‘flags’ [-Wunused-variable]

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:27:28 +02:00
Daniel Mack
94c27215bc ALSA: snd-usb: add some documentation
Document the new streaming code and some of the functions so that
contributers can catch up easier.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:25:24 +02:00
Daniel Mack
c75a8a7ae5 ALSA: snd-usb: add support for implicit feedback
Implicit feedback is a streaming mode that does not rely on dedicated
sync endpoints but uses the information provided by record streams to
clock output streams. Now that the streaming logic is decoupled from the
PCM streams, this is easy to implement.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:32 +02:00
Daniel Mack
d399ff9593 ALSA: snd-usb: remove old streaming logic
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:23 +02:00
Daniel Mack
edcd3633e7 ALSA: snd-usb: switch over to new endpoint streaming logic
With the previous commit that added the new streaming model, all
endpoint and streaming related code is now in endpoint.c, and pcm.c
only acts as a wrapper for handling the packet's payload.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:08 +02:00
Daniel Mack
8fdff6a319 ALSA: snd-usb: implement new endpoint streaming model
This patch adds a new generic streaming logic for audio over USB.

It defines a model (snd_usb_endpoint) that handles everything that
is related to an USB endpoint and its streaming. There are functions to
activate and deactivate an endpoint (which call usb_set_interface()),
and to start and stop its URBs. It also has function pointers to be
called when data was received or is about to be sent, and pointer to
a sync slave (another snd_usb_endpoint) that is informed when data has
been received.

A snd_usb_endpoint knows about its state and implements a refcounting,
so only the first user will actually start the URBs and only the last
one to stop it will tear them down again.

With this sort of abstraction, the actual streaming is decoupled from
the pcm handling, which makes the "implicit feedback" mechanisms easy to
implement.

In order to split changes properly, this patch only adds the new
implementation but leaves the old one around, so the the driver doesn't
change its behaviour. The switch to actually use the new code is
submitted separately.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:23:42 +02:00
Daniel Mack
596580d0ee ALSA: snd-usb: add snd_usb_audio-wide mutex
This is needed for new card-wide list operations.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:21:55 +02:00
Takashi Iwai
44c76a960a Merge branch 'topic/misc' into for-linus 2012-03-18 18:22:33 +01:00
Takashi Iwai
0717d0f5d2 ALSA: usb-audio - Fix build error by consitification of rate list
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-03-15 16:14:38 +01:00
Torsten Schenk
adef39c0ea ALSA: snd-usb-6fire: Select missing SND_VMASTER option in Kconfig
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-25 11:07:19 +01:00
Torsten Schenk
06bb4e7435 ALSA: snd-usb-6fire: add analog input volume control
Add a stereo volume control for analog input channel pair 1/2.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:51:26 +01:00
Torsten Schenk
d97c735a10 ALSA: snd-usb-6fire: add mute control for analog outputs
Add a mute control for every analog output channel.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:51:16 +01:00
Torsten Schenk
f90ffbf3c6 ALSA: snd-usb-6fire: add individual volume control for analog channels
Add a stereo volume control for every analog output pair 1/2, 3/4, 5/6.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:51:06 +01:00
Torsten Schenk
8e247a9c90 ALSA: snd-usb-6fire: add tlv to controls
Remove the soft log-conversion and add a dB scale according to
the DAC documentation instead.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:50:56 +01:00
Torsten Schenk
c596758f57 ALSA: snd-usb-6fire: remove driver version information
Remove unused driver version information from the individual files.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:50:45 +01:00
Mark Hills
cb74eb15ac ALSA: snd-usb-caiaq: Fix the return of XRUN
Commit 3702b08 added a lock, but did not account for the case of
SNDRV_PCM_POS_XRUN, which would get immediately overwritten.

This could be bundled into one if-else-if statement, but the goto
helps to clarify the 'exceptional' case.

Thanks to Andreas Pape for spotting this.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 08:34:58 +01:00
Xi Wang
8866f405ef ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk()
A malicious USB device could feed in a large nr_rates value.  This would
cause the subsequent call to kmemdup() to allocate a smaller buffer than
expected, leading to out-of-bounds access.

This patch validates the nr_rates value and reuses the limit introduced
in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow
in parse_uac2_sample_rate_range()").

Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-15 14:58:15 +01:00
Masanari Iida
6e8d5d2f17 ALSA: usx2y: Fix typo in usbusx2yaudio.c and usx2yhwdeppcm.c
Correct spelling "propably" to "probably" and "activ" to "active"
in sound/usb/usx2y/usbusx2yaudio.c and usx2yhwdeppcm.c

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-15 14:56:11 +01:00
Clemens Ladisch
927c9423dd ALSA: usb-audio: add Edirol UM-3G support
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-08 09:46:34 +01:00
Linus Torvalds
a429638cac Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
  ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
  ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
  ALSA: usb-audio: add Yamaha MOX6/MOX8 support
  ALSA: virtuoso: add S/PDIF input support for all Xonars
  ALSA: ice1724 - Support for ooAoo SQ210a
  ALSA: ice1724 - Allow card info based on model only
  ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
  ALSA: hdspm - Provide unique driver id based on card serial
  ASoC: Dynamically allocate the rtd device for a non-empty release()
  ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
  ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
  ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
  ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
  ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
  ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
  ALSA: hda/cirrus - support for iMac12,2 model
  ASoC: cx20442: add bias control over a platform provided regulator
  ALSA: usb-audio - Avoid flood of frame-active debug messages
  ALSA: snd-usb-us122l: Delete calls to preempt_disable
  mfd: Put WM8994 into cache only mode when suspending
  ...

Fix up trivial conflicts in:
 - arch/arm/mach-s3c64xx/mach-crag6410.c:
	renamed speyside_wm8962 to tobermory, added littlemill right
	next to it
 - drivers/base/regmap/{regcache.c,regmap.c}:
	duplicate diff that had already come in with other changes in
	the regmap tree
2012-01-12 08:00:30 -08:00
Clemens Ladisch
8c3f5d8a9b ALSA: usb-audio: add Yamaha MOX6/MOX8 support
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11 09:24:53 +01:00
Takashi Iwai
80c8a2a372 ALSA: usb-audio - Avoid flood of frame-active debug messages
With some buggy devices, the usb-audio driver may give "frame xxx active"
kernel messages too often.  Better to keep it as debug-only using
snd_printdd(), and also add the rate-limit for avoiding floods.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738681

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09 11:40:46 +01:00
Karsten Wiese
d0f3a2eb90 ALSA: snd-usb-us122l: Delete calls to preempt_disable
They are not needed here.

Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09 11:31:30 +01:00
Xi Wang
4fa0e81b83 ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range()
A malicious USB device may feed in carefully crafted min/max/res values,
so that the inner loop in parse_uac2_sample_rate_range() could run for
a long time or even never terminate, e.g., given max = INT_MAX.

Also nr_rates could be a large integer, which causes an integer overflow
in the subsequent call to kmalloc() in parse_audio_format_rates_v2().
Thus, kmalloc() would allocate a smaller buffer than expected, leading
to a memory corruption.

To exploit the two vulnerabilities, an attacker needs physical access
to the machine to plug in a malicious USB device.

This patch makes two changes.

1) The type of "rate" is changed to unsigned int, so that the loop could
   stop once "rate" is larger than INT_MAX.

2) Limit nr_rates to 1024.

Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 16:03:12 +01:00
Greg Kroah-Hartman
ff4b8a57f0 Merge branch 'driver-core-next' into Linux 3.2
This resolves the conflict in the arch/arm/mach-s3c64xx/s3c6400.c file,
and it fixes the build error in the arch/x86/kernel/microcode_core.c
file, that the merge did not catch.

The microcode_core.c patch was provided by Stephen Rothwell
<sfr@canb.auug.org.au> who was invaluable in the merge issues involved
with the large sysdev removal process in the driver-core tree.

Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2012-01-06 11:42:52 -08:00