Commit Graph

782 Commits

Author SHA1 Message Date
Clemens Ladisch
7da5804648 ALSA: usb-audio: fix Roland A-PRO support
The quirk for the Roland/Cakewalk A-PRO keyboards accidentally used the
wrong interface number, which prevented the driver from attaching to the
device.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.37+ <stable@vger.kernel.org>
2013-01-31 21:21:59 +01:00
Antonio Ospite
aa53f98674 ALSA: usb: cosmetics, remove a leading space
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-29 15:11:13 +01:00
Antonio Ospite
febd1cc438 ALSA: caiaq: fix use of MODULE_SUPPORTED_DEVICES()
It looks like MODULE_SUPPORTED_DEVICES() is not implemented yet, but
still, having the entries in the list consistently separated by commas
and with balanced parenthesis won't hurt.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-29 15:10:57 +01:00
Clemens Ladisch
d56268fb10 ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices
Commit 23caaf19b1 (ALSA: usb-mixer: Add support for Audio Class v2.0)
forgot to adjust the length check for UAC 2.0 feature unit descriptors.
This would make the code abort on encountering a feature unit without
per-channel controls, and thus prevented the driver to work with any
device having such a unit, such as the RME Babyface or Fireface UCX.

Reported-by: Florian Hanisch <fhanisch@uni-potsdam.de>
Tested-by: Matthew Robbetts <wingfeathera@gmail.com>
Tested-by: Michael Beer <beerml@sigma6audio.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: 2.6.35+ <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-27 10:22:56 +01:00
Takashi Iwai
86b2723725 ALSA: Make snd_printd() and snd_printdd() inline
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
  sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
  sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]

For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros.  This should have the same effect but shut up warnings like
above.

But since we had already put ifdefs, changing to inline functions
would trigger compile errors.  So, such ifdefs is removed in this
patch.

In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too.  For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.

Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-25 18:32:14 +01:00
Takashi Iwai
e152f18027 Merge branch 'for-linus' into for-next
This is a preliminary merge before the upcoming merge of generic parser
branch.
2013-01-23 08:31:34 +01:00
Eldad Zack
39e95156b9 ALSA: usb-audio: selector map for M-Audio FT C400
Add names of the clock sources for the M-Audio Fast Track
C400.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:06:11 +01:00
Eldad Zack
83e3acd494 ALSA: usb-audio: M-Audio FT C400 skip packet quirk
Attain constant real-world latency by skipping 16 data packets.
The number of packets to be skipped was found by trial and error.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:06:03 +01:00
Eldad Zack
2aad272b3f ALSA: usb-audio: correct M-Audio C400 clock source quirk
Taking another look at the C400 descriptors, I see now that there is
a clock selector (0x80) for this device.
Right now, the clock source points to the internal clock (0x81), which
is also valid. When the external clock source (0x82) is selected in the
mixer, and the rates mismatch (if it's free-running it is fixed to
48KHz), xruns will occur.

Set the clock ID to the clock selector unit (0x81), which then
allows the validation code to function correctly.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:05:57 +01:00
David Henningsson
b98ae2729d ALSA: usb - fix race in creation of M-Audio Fast track pro driver
A patch in the 3.2 kernel caused regression with hotplugging the
M-Audio Fast track pro, or sound after suspend. I don't have the
device so I haven't done a full analysis, but it seems userspace
(both udev and pulseaudio) got confused when a card was created,
immediately destroyed, and then created again.

However, at least one person in the bug report (martin djfun)
reports that this patch resolves the issue for him. It also leaves
a message in the log:
"snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is
a bit misleading. It is better than non-working audio, but maybe
there's a more elegant solution?

BugLink: https://bugs.launchpad.net/bugs/1095315
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:03:03 +01:00
Takashi Iwai
31be5425d7 ALSA: usb-audio: Fix NULL dereference by access to non-existing substream
The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for
audioformat mismatch] introduced the correction of parameters to be
set for sync EP.  But since the new code assumes that the sync EP is
always paired with the data EP of another direction, it triggers Oops
when a device only with a single direction is used.

This patch adds a proper check of sync EP type and the presence of the
paired substream for avoiding the crash.

Reported-and-tested-by: Jens Axboe <axboe@kernel.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-11 11:12:17 +01:00
Sachin Kamat
e8e7da23c9 ALSA: usb-audio: Make ebox44_table static
Fixes the following sparse warning:
sound/usb/mixer_quirks.c:1209:23: warning:
symbol 'ebox44_table' was not declared. Should it be static?

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:22:25 +01:00
Damien Zammit
b7b435e81b ALSA: usb-audio: Fix kernel panic of Digidesign Mbox2 quirk
This patch is based on 3.8-rc1. It fixes two things:
1) A kernel panic caused by incorrect allocation of a u8 variable
   "bootresponse".
2) A noisy dmesg (urb status -32) caused by broken pipe to an
   invalid midi endpoint.

It is also a little cleaner because there is no need for a new
QUIRK_MIDI type as suggested by kernel developers, since the device
follows exactly the MIDIMAN protocol.

Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-04 09:53:17 +01:00
Alexander Schremmer
8f7f3ab15e ALSA: usb-audio: Add support for Creative BT-D1 via usb sound quirks
Support the Creative BT-D1 Bluetooth USB audio device. Before this
patch, Linux had trouble finding the correct USB descriptors and bailed
out with these messages:

 no or invalid class specific endpoint descriptor

Now it still prints these messages on hotplug:

 snd-usb-audio: probe of ...:1.0 failed with error -5
 snd-usb-audio: probe of ...:1.2 failed with error -5
 snd-usb-audio: probe of ...:1.3 failed with error -5

But the device works correctly, including the HID support.

The patch is diff'ed against 3.8-rc1 but should apply to older kernels
as well.

Signed-off-by: Alexander Schremmer <alex@alexanderweb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-03 14:26:48 +01:00
Pierre-Louis Bossart
e4cc615340 ALSA: usb-audio: support delay calculation on capture streams
Enable delay report on capture path. The delay is reset when an
URB is retired and increment at each call to .pointer based
on frame counter changes. The precision of the delay
information is limited to 1ms as in the playback case.

This reverts commit 3f94fad095.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-24 10:53:57 +01:00
Linus Torvalds
03c850ec32 Sound fixes for 3.8-rc1
This update contains overall only driver-specific fixes.
 Slightly large LOC are seen in usb-audio driver for a couple of new
 device quirks and cs42l71 ASoC driver for enhanced features.
 The others are a few small (regression) fixes HD-audio, and yet other
 small / trival ASoC fixes.
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Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This update contains overall only driver-specific fixes.  Slightly
  large LOC are seen in usb-audio driver for a couple of new device
  quirks and cs42l71 ASoC driver for enhanced features.  The others are
  a few small (regression) fixes HD-audio, and yet other small / trival
  ASoC fixes."

* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
  ALSA: HDA: Fix sound resume hang
  ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
  ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup
  ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs
  ASoC: atmel-ssc: change disable to disable in dts node
  ASoC: Prevent pop_wait overwrite
  ALSA: usb-audio: ignore-quirk for HP Wireless Audio
  ALSA: hda - Always turn on pins for HDMI/DP
  ALSA: hda - Fix pin configuration of HP Pavilion dv7
  ASoC: core: Fix splitting of log messages
  ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT
  ASoC: cs42l73: Add DAPM events for power down.
  ASoC: cs42l73: Add DMIC's as DAPM inputs.
  ASoC: sigmadsp: Fix endianness conversion issue
  ASoC: tpa6130a2: Use devm_* APIs
2012-12-20 07:52:13 -08:00
Damien Zammit
cb99864d40 ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
This patch is the result of a lot of trial and error, since there are no specs
available for the device.

Full duplex support is provided, i.e. playback and recording in stereo.
The format is hardcoded at 48000Hz @ 24 bit, which is the maximum that the
device supports.  Also, MIDI in and MIDI out both work.

Users will notice that the S/PDIF light also flashes when playback or recording
is active.  I believe this means that S/PDIF input/output is simultaneously
activated with the analogue i/o during use.
But this particular functionality remains untested.

Note that this particular version of the patch is so far untested on the
physical hardware because I have not compiled a full kernel with the changes.
However, extensive testing has been done by many users of the hardware
who believe other versions of my patch have worked since circa 2009.

[Modified to make a function static by tiwai]

Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-19 11:27:22 +01:00
Eldad Zack
df68f10643 ALSA: usb-audio: ignore-quirk for HP Wireless Audio
As Joe Cooper <swelljoe@gmail.com> reported, "On most HP Envy laptops
the snd-usb-audio module causes the system to become unresponsive and
Gnome Shell 3 to crash.".
See also:
 http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/057729.html

Add a quirk to ignore this device (for now) to solve the instability
issue and allow other USB audio devices to be used.

Reported-by: Joe Cooper <swelljoe@gmail.com>
Tested-by: Isaac Smith <hunternet93@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-15 11:13:10 +01:00
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
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