ALSA: sound/core: merge list_del()/list_add_tail() to list_move_tail()

Merge list_del() + list_add_tail() to list_move_tail().

Signed-off-by: Nicolas Kaiser <nikai@nikai.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: ctxfi - use list_move() instead of list_del()/list_add() combination

Signed-off-by: Kirill A. Shutemov <kirill@shutemov.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: ctxfi - Clear input settings before initialization

Clear input settings before initialization.

Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: ctxfi - Fix SPDIF status retrieval

SDPIF status retrieval always returned the default settings instead of
the actual ones.

Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: ctxfi - Fix incorrect SPDIF status bit mask

SPDIF status mask creation was incorrect.

Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: ctxfi - Fix microphone boost codes/comments

microphone boost was set at +12dB, not +20dB (like in Windows driver
and in adc_conf structure declaration), some comments added.

Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: atiixp - Fix wrong time-out checks during ac-link reset

The time-out in snd_atiixp_aclink_reset() is wrongly checked, and
it resulted in exiting from the loop at the first iteration.

Reported-by: Amir Shamsuddin <AmirS2+alsa@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: intel8x0m: append 'm' to "r_intel8x0"

Appending an 'm' will distinguish it from a similar struct in intel8x0.c

Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: intel8x0m: add 'm' as "suffix" to static functions

Adding an 'm' will distinguish them from identical names in intel8x0.c.

Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: intel8x0m: wait a bit before warm reset check

At every resume a laptop I use prints this message (at KERN_ERR level):
    ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2]

The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what
seems to be happening is that the register involved indicated a warm
reset for some time (as the ICH_AC97WARM bit was set) but by the time
the warning is printed, and that same register is checked again, that
bit is already cleared and only the ICH_AC97COLD bit is still set.

It turns out a warm reset needs some time to settle, but it is currently
checked right away. The test therefore fails the first time it is done
and schedule_timeout_uninterruptible() will be called. Once we return
from that jiffies is already (far) past end_time on this laptop, so we
exit the loop, print a warning, and exit the function while the warm
reset actually succeeded.

A way to fix this is to call usleep_range() after writing to the
register involved. A handful of tests suggest 500 usecs is a safe value.
(This might punish the "finish cold reset" case, but on this laptop such
a cold reset apparently never happens, so I can't say for sure.)

While we're at it drop the extra single tick from end_time, as it looks
rather silly.

Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usbaudio: implement USB autosuspend

Devices are autosuspended if no pcm nor midi channel is open
Mixer devices may be opened. This way they are active when
in use to play or record sound, but can be suspended while
users have a mixer application running.

[Small clean-ups using static inline by tiwai]

Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usbaudio: fix suspend/resume

- ESHUTDOWN must be correctly handled
- the optional interrupt endpoint's URB must be stopped and restarted

Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'fix/misc' into topic/misc

ALSA: usb-audio - Add "cval->res = 384" quirk for Logitech Webcam C600

One more affected devices: Logitech Webcam C600 (046d:0808)

Volume range before quirk is 6400, after (also real) is 16.

Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio - Add volume range check and warn if it too big

Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

sound: Use sound_register_*() for additional OSS minor devices

Since OSS driver creates the device entries for /dev/audio* and
/dev/dspW* by itself without coping with sound_core, it leads to
conflicts with others and let sysfs spewing warnings.

This patch rewrites the registration part of OSS driver to use
the standard method also for additional minor devices.

Reported-by: Steven Rostedt <rostedt@goodmis.org> (with ktest.pl)
Tested-by: Steven Rostedt <rostedt@goodmis.org> (with ktest.pl)
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Use %zd for size_t argument in error message

Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: control: clean up snd_ctl_hole_check()

The return value of snd_ctl_hole_check() is used only to detect whether
to continue the loop in snd_ctl_find_hole() or not, so we can simplify
the code by changing this return type to a boolean.  Also rename this
function to better show what it actually does.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: control: fix numid conflict check for new controls

The purpose of the snd_ctl_hole_check() function is to find conflicts
between the numerical IDs of the new control and those of any existing
controls.  However, it would fail to detect an existing control whose
count is smaller than the new control's count and whose interval of IDs
is entirely contained in the interval of the new control's IDs.

To fix this, use the correct formula to detect overlapping intervals,
which happens to simplify the condition.

This problem was not encountered so far because ALSA does not yet allow
drivers to allocate specific control IDs.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - Add firmware ID for older AES32 cards.

The current AES32 firmware revision ID is 234, however, a user confirmed
that everything works fine with the previous revision, too.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - Fix midi port initialization

Depending on the model and the presence of a TCO module, the number of
midi ports varies. Some have 1 port (MADIface), some have 2 (default),
with TCO, there are 3.

Don't hardcode the number of midi ports to initialize.

This patch also fixes a boot lockup on MADIface.

[Coding-style fixes by tiwai]

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - Abort on unknown firmware revision

Don't continue if we cannot detect the real card type, otherwise, all
subsequent functions, especially strcpy(), would fail, leaving the whole
driver in an unusable state.

Without such a protection, dmesg would look like this:

Pid: 1525, comm: modprobe Not tainted 2.6.38-rc6 #1 MSI MS-7250/MS-7250
EIP: 0060:[<c114e700>] EFLAGS: 00010296 CPU: 1
EIP is at strcpy+0x10/0x30
EAX: f4d33e58 EBX: f5990800 ECX: f4d33e58 EDX: 00000000
ESI: 00000000 EDI: f4d33e58 EBP: f5990930 ESP: f3dd3e0c
DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068
f53d4204 00000000 f90bfc03 00000001 00000001 f3dd3e64 c10fedff f3dd3e7c
f3d854e4 f53d4000 c10307b8 00000001 c10307b8 f5990860 c10307b8 00000001
c10332e5 f59908bc c12f4339 f59908bc f5990860 c11ef207 f4d33e00 f53d4000
[<f90bfc03>] ? snd_hdspm_probe+0x7e7/0x1166 [snd_hdspm]
[<c10fedff>] ? __sysfs_add_one+0x1f/0xf0
[<c10307b8>] ? get_parent_ip+0x8/0x20
[<c10307b8>] ? get_parent_ip+0x8/0x20
[<c10307b8>] ? get_parent_ip+0x8/0x20
[<c10332e5>] ? add_preempt_count+0xa5/0xd0
[<c12f4339>] ? _raw_spin_lock_irqsave+0x19/0x40
[<c11ef207>] ? pm_runtime_enable+0x17/0x80
[<c1160172>] ? local_pci_probe+0x42/0xb0
[<c1161091>] ? pci_device_probe+0x61/0x80
[<c11e9587>] ? driver_probe_device+0x77/0x180
[<c11600f0>] ? pci_match_device+0xa0/0xc0
[<c11e9709>] ? __driver_attach+0x79/0x80
[<c11e9690>] ? __driver_attach+0x0/0x80
[<c11e8e32>] ? bus_for_each_dev+0x52/0x80
[<c11e9416>] ? driver_attach+0x16/0x20
[<c11e9690>] ? __driver_attach+0x0/0x80
[<c11e8771>] ? bus_add_driver+0xa1/0x220
[<c1160f20>] ? pci_device_remove+0x0/0xf0
[<c11e99a3>] ? driver_register+0x63/0x120
[<c11612bd>] ? __pci_register_driver+0x3d/0xb0
[<c1001132>] ? do_one_initcall+0x32/0x160
[<f90d6000>] ? alsa_card_hdspm_init+0x0/0x14 [snd_hdspm]
[<c10676a9>] ? sys_init_module+0x99/0x1e0
[<c10ade3d>] ? sys_close+0x6d/0xc0
[<c1002c90>] ? sysenter_do_call+0x12/0x26
---[ end trace 239f0661c558378b ]---

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - Add updated firmware revision for AES32

In contrast to the PCIe version (RME AES), the PCI version (RME AES32)
has a different firmware revision.

This patch adds the missing PCI revision.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - AES32: Use define instead of hardcoded channel count

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: powermac - Allow input from microphone on Tumbler

Allow input from microphone on remaining PowerBooks and iBooks with Tumbler.
Tested only with PowerBook G4 Gigabit Ethernet (PowerBook3,3).

[Fixed to use of_machine_is_compatible by tiwai]

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

sound:core:seq:seq_ports.c Remove one to many n's in a word.

The Patch below removes one to many "n's" in a word..

Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: emu10k1 - emu10k1_main.c remove one to many l's in the word.

The patch below removes an extra "l" in the word.

Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - patch_realtek.c remove one to many l's in the word.

The patch below removes an extra "l" in the word.

Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - Fix lock/sync reporting on MADI and AES32

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - prevent reading unitialized stack memory

Original patch by Dan Rosenberg <drosenberg@vsecurity.com> under commit
e68d3b316ab7b02a074edc4f770e6a746390cb7d. I'm copying his text here:

The SNDRV_HDSPM_IOCTL_GET_CONFIG_INFO ioctl in hdspm.c allow unprivileged
users to read uninitialized kernel stack memory, because several fields
of the hdspm_config struct declared on the stack are not altered
or zeroed before being copied back to the user.  This patch takes care
of it.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - fix sync check on AES32

Fredrik Lingvall <fredrik.lingvall@gmail.com> has discovered wrong
frequency and sync detection on AES32. According to him, the provided
patch fixes these issues.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - Remove input selector on MADIface

In contrast to the RME MADI card, coax/optical selection on the MADIface
is done via a physical switch located at the breakout box. Obviously,
the driver cannot switch ports in software.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - Fix DS/QS output channel mappings on RME MADI/MADIface

Caused by two typos, no output channel mappings were assigned for
MADI/MADIface at double/quad speed.

The channel mapping is indeed identical to the single speed mapping, the
cards will simply use the first N channels.

Signed-off-by: Florian Faber <faber@faberman.de>
Signed-off-by: Fredrik Lingvall <fredrik.lingvall@gmail.com>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - Restrict channel count on RME AES/AES32

Without calling an appropriate rule, AES/AES32 cards would announce a
theoretical channel count of 64 (HDSPM_MAX_CHANNELS), leading to the
already known bug:

[37422.640481] ------------[ cut here ]------------
[37422.640487] WARNING: at sound/pci/rme9652/hdspm.c:5449
snd_hdspm_ioctl+0x18f/0x202 [snd_hdspm]()
[37422.640489] Hardware name: PRIMERGY RX100 S6
[37422.640490] BUG? (info->channel >= hdspm->max_channels_in)
[37422.640492] Modules linked in: snd_hdspm snd_seq_midi ipmi_watchdog
ipmi_poweroff ipmi_si ipmi_devintf ipmi_msghandler i2c_i801 e1000e
snd_rawmidi power_meter [last unloaded: snd_hdspm]
[37422.640501] Pid: 22231, comm: jackd Tainted: G      D W
2.6.36-gentoo-r5 #5
[37422.640502] Call Trace:
[37422.640508]  [<ffffffff8103db3a>] warn_slowpath_common+0x80/0x98
[37422.640511]  [<ffffffff8103dbe6>] warn_slowpath_fmt+0x41/0x43
[37422.640514]  [<ffffffff81034306>] ? get_parent_ip+0x11/0x42
[37422.640518]  [<ffffffffa0055763>] snd_hdspm_ioctl+0x18f/0x202
[snd_hdspm]
[37422.640522]  [<ffffffff813fd626>] snd_pcm_channel_info+0x73/0x7c
[37422.640525]  [<ffffffff814001e9>] snd_pcm_common_ioctl1+0x326/0xb01
[37422.640527]  [<ffffffff81034306>] ? get_parent_ip+0x11/0x42
[37422.640531]  [<ffffffff8105be6c>] ? __srcu_read_unlock+0x3b/0x59
[37422.640533]  [<ffffffff81400bce>] snd_pcm_capture_ioctl1+0x20a/0x227
[37422.640537]  [<ffffffff811e599c>] ? file_has_perm+0x90/0x9e
[37422.640540]  [<ffffffff81400c15>] snd_pcm_capture_ioctl+0x2a/0x2e
[37422.640543]  [<ffffffff810f2c69>] do_vfs_ioctl+0x404/0x453
[37422.640546]  [<ffffffff810f2d09>] sys_ioctl+0x51/0x74
[37422.640549]  [<ffffffff81002aab>] system_call_fastpath+0x16/0x1b
[37422.640552] ---[ end trace 0cd919cd68118082 ]---

We already have all the right values in place, we simply have to inform
the upper layers about this restriction.

Note that snd_hdspm_hw_rule_rate_out_channels and
snd_hdspm_hw_rule_rate_in_channels must not be called on AES32, because
the channel count is always 16, no matter of the samplerate in use.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hdspm - Fix buffer handling on RME MADI/MADIface/AES(32)

Only RayDAT and AIO provide sane buffer pointers that can be used with
HDSPM_BufferPositionMask, on all other cards, this would result in a
wrong HW pointer leading to xruns and these messages:

[260808.916788] BUG: pcmC0D0p:0, pos = 2976, buffer size = 1024, period size = 512
[260808.961124] BUG: pcmC0D0c:0, pos = 4944, buffer size = 1024, period size = 512

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hpdsm - RME AES(32): Fix missing channel mappings

On RME AES and AES(32), none of the required information
(max_channels_in, max_channels_out, channel mappings, port names) was
set, leading to the BUG below.

This patch adds the missing bits, thus fixing the bug.

125.058768] ------------[ cut here ]------------
[  125.058773] WARNING: at sound/pci/rme9652/hdspm.c:5389
snd_hdspm_ioctl+0x10c/0x1d8 [snd_hdspm]()
[  125.058775] Hardware name: PRIMERGY RX100 S6
[  125.058777] BUG? (info->channel >= hdspm->max_channels_out)
[  125.058778] Modules linked in: ipmi_watchdog ipmi_poweroff ipmi_si
ipmi_devintf ipmi_msghandler snd_hdspm power_meter e1000e snd_rawmidi
i2c_i801
[  125.058787] Pid: 3652, comm: audacity Tainted: G        W
2.6.36-gentoo-r5 #5
[  125.058788] Call Trace:
[  125.058792]  [<ffffffff8103db3a>] warn_slowpath_common+0x80/0x98
[  125.058796]  [<ffffffff8103dbe6>] warn_slowpath_fmt+0x41/0x43
[  125.058800]  [<ffffffffa006761a>] snd_hdspm_ioctl+0x10c/0x1d8
[snd_hdspm]
[  125.058803]  [<ffffffff813fd626>] snd_pcm_channel_info+0x73/0x7c
[  125.058806]  [<ffffffff814001e9>] snd_pcm_common_ioctl1+0x326/0xb01
[  125.058809]  [<ffffffff810c604c>] ? __do_fault+0x361/0x3a6
[  125.058812]  [<ffffffff81400e23>] snd_pcm_playback_ioctl1+0x20a/0x227
[  125.058815]  [<ffffffff811e599c>] ? file_has_perm+0x90/0x9e
[  125.058818]  [<ffffffff81400e6a>] snd_pcm_playback_ioctl+0x2a/0x2e
[  125.058821]  [<ffffffff810f2c69>] do_vfs_ioctl+0x404/0x453
[  125.058824]  [<ffffffff810f2d09>] sys_ioctl+0x51/0x74
[  125.058827]  [<ffffffff81002aab>] system_call_fastpath+0x16/0x1b
[  125.058830] ---[ end trace 5bddb08e5d4cbeb1 ]---

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Florian Faber <faber@faberman.de>
Signed-off-by: Fredrik Lingvall <fredrik.lingvall@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: fix oops due to cleanup race when disconnecting

When a USB audio device is disconnected, snd_usb_audio_disconnect()
kills all audio URBs.  At the same time, the application, after being
notified of the disconnection, might close the device, in which case
ALSA calls the .hw_free callback, which should free the URBs too.

Commit de1b8b93a0ba "[ALSA] Fix hang-up at disconnection of usb-audio"
prevented snd_usb_hw_free() from freeing the URBs to avoid a hang that
resulted from this race, but this introduced another race because the
URB callbacks could now be executed after snd_usb_hw_free() has
returned, and try to access already freed data.

Fix the first race by introducing a mutex to serialize the disconnect
callback and all PCM callbacks that manage URBs (hw_free and hw_params).

Reported-and-tested-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Cc: <stable@kernel.org>
[CL: also serialize hw_params callback]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA: Fix mic initialization in VIA auto parser

This typo caused some microphone inputs not to be correctly
initialized on VIA codecs.

Reported-By: Mark Goldstein <goldstein.mark@gmail.com>
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: fix one memory leak in sound jack

Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Reviewed-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA - au88x0 - add Playback Volume to 10 bands Equalizer Controls

Add " Playback Volume" to 10 bands Equalizer Controls of au88x0 so that
alsa-lib won't regard them as "Capture Volume".

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA: Do not announce false surround in Conexant auto

Without this patch, one line-out and one speaker and
Conexant's auto parser would announce (non-working) surround
capabilities.

BugLink: http://bugs.launchpad.net/bugs/721126
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA: Conexant auto: Handle multiple connections to ADC node

Conexant 20641 has several inputs to its ADC node, with one selector
and individual amps for all inputs. This patch adds support in the
Conexant auto parser to handle that case.

It also means that the pin node's volume is being renamed to "Boost"
to avoid name clash with the new volume controls on the ADC node.

BugLink: http://bugs.launchpad.net/bugs/719524
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: azt3328: hook up new emulated AC97 on AC97 patch side

Make newly created AC97 emulation of azt3328 known to the AC97 layer
side.
- relocate common functions to the top (due to definition after use)
- rename control names
- adjust 3D settings to the card's custom layout of this register

Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: azt3328: add custom AC97 semi-emulation use standard ALSA AC97 layer

Make use of the very flexible ALSA ac97 layer (hooks for custom I/O!)
on this weird AC97 copycat hardware,
via semi-extended I/O translation/emulation.

Some 5kB binary/loaded size saved (well... additional huge AC97 module
penalty not factored in, of course ;-P).
Given that the driver previously had 20kB that's not bad,
but the much more important thing is to have AC97 layer stress-tested
with a thoroughly weird AC97 copycat (or, simply put, if it were not for
this AC97 test aspect, this effort would merely have been a nut job ;).

Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: ac97: replace open-coded, error-prone stuff with AC97 bit defines

Use AC97 macros (sometimes already existing, or newly added)
instead of error-prone repetition of open-coded values.

Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA: Add position_fix quirk for an Asus device

The bug reporter claims that position_fix=1 is needed for his
microphone to work. The controller PCI vendor-id is [1002:4383] (rev 40).

Reported-by: Kjell L.
BugLink: http://bugs.launchpad.net/bugs/718402
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: caiaq - Fix possible string-buffer overflow

Use strlcpy() to assure not to overflow the string array sizes by
too long USB device name string.

Reported-by: Rafa <rafa@mwrinfosecurity.com>
Cc: stable <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: au88x0 - Modify pointer callback to give accurate playback position

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: reconstruct some dispatcher functions to use switch-case

The number of cases has increased so use switch-case rather than
if-statements.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: add support for Native Instruments MK2 devices

The MK2 generation of Native Instruments' sound cards are in fact
compliant to the USB audio standard of version 2 and other approved USB
standards. However, they come up as vendor-specific device when first
connected but can be told to come up with a new set of descriptors
upon their next enumeration. The interfaces announced by the new
descriptors will be handled by the kernel's class drivers. This is done
by issuing a vendor specific device request and sending the device to
reset.

There are also some vendor-specific USB requests for some mixer elements
that can't be exported in a standard compliant way. The driver now
supports them with quirks handling mechanisms.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: snd-usb-caiaq: Add support for Traktor Audio 2

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: core: sparse cleanups

Change the core code where sparse complains.  In most cases, this means
just adding annotations to confirm that we indeed want to do the dirty
things we're doing.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'fix/asoc' into for-linus

ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662

This netbook has a only one jack output and an internal mic.

By default, mic and jack sense aren't working. Using lenovo-101e
parameters makes both work.

The device seems based on a Sharetronic Q70, so this should fix audio for
this model too.

Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hrtimer: remove superfluous tasklet invocation

Commit bb758e9637e5ddc removed snd_hrtimer_callback() from the hardware
interrupt handler, thus moving it into a tasklet, but did not tell the
ALSA timer framework about this, so the timer handling would now be done
in the ALSA timer tasklet scheduled from another tasklet.

To fix this, add the flag to tell the ALSA timer framework that the
timer handler is already being invoked in a tasklet.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hrtimer: handle delayed timer interrupts

If a timer interrupt was delayed too much, hrtimer_forward_now() will
forward the timer expiry more than once.  When this happens, the
additional number of elapsed ALSA timer ticks must be passed to
snd_timer_interrupt() to prevent the ALSA timer from falling behind.

This mostly fixes MIDI slowdown problems on highly-loaded systems with
badly behaved interrupt handlers.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Arthur Marsh <arthur.marsh@internode.on.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - HPI v4.06

Firmware version check depends on hpi version. Update so correct firmware
is accepted.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Fix outstream start trigger for non-mmap adapters.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Tighten firmware version requirements.

Difference in major.minor between driver and firmware is an error now.
Release version mismatch give a warning.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Ensure all adapter data is cleared on device removal.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Minor define updates

HPI version 4.05.32
Tweak HPI error code for backward compatibility.
Add BUILD to build-related defines.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - New functions prep for interrupt driven streams.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Use consistent err return variable, change some bad variable names.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Remove unused code and data.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Clarify firmware id selection.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Allow adapters with duplicate index jumpers to be discovered.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Add volume mute control.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Add snd_card_set_dev to init.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Replace adapter list with single item in subsys response.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Cosmetic + a minor comments.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Remove int flag polling code preparing for stream interrupts.

Interrupt flag used for message handshake will be required for
stream interrupts, so conditionally compiled code without
HPI6205_NO_HSR_POLL defined can never be used;  removing it.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Code cleanup.

Remove unused function.
Simplify hpi_alloc_control_cache.
Remove useless assignment to struct subsequently freed.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Reduce number of error codes returned to upper layers.

Create and use HPI_ERROR_DSP_COMMUNICATION _DSP_BOOTLOAD, rather than
backend-specific error codes (now returned as data with the error).

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Remove unused subsys pointer from all HPI functions.

asihpi.c don't link playback and capture streams, there is too much
offset between them.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Update error codes.

Some error codes had duplicate meanings. Just use one.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Checkpatch line lengths etc.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Add include guard.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Add adapter index to cache info for debug.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Rewrite PCM timer function. Update control names.

Reported samples_played from card may be inaccurate, so don't use it.
Update control names to be closer to alsa standard practice.
Also fixed some accidentally lowercased strings.

[Removed adriver.h inclusion for external module builds by tiwai]

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Interrelated HPI tidy up.

Remove many unused functions.
Update some message and cache structs.
Use pci info directly from pci_dev.
Allow control cache elements with variable size, and handle
large message/response from dsp.
hpi6000 and hpi6205: fix error path when adapter bootload fails.
hpimsgx.c get rid of code duplicated in hpicmn.c

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - HPI 4.05.14

All enum values numeric for easier finding, particularly error codes.
Remove many unused declarations.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Simplify debug logging.

Log HPI messages and responses in consistent numeric format,
which can be post-processed to get strings.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Poison adapter_index in message. Remove unused function.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: asihpi - Switch to dev_printk.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: HDA: Add subwoofer quirk for Acer Aspire 8942G

According to the reporter, node 0x15 needs to be muted for subwoofer
to stop sounding. This pin is marked as unused by BIOS, so fix that.

BugLink: http://bugs.launchpad.net/bugs/715877
Cc: stable@kernel.org (2.6.37+)
Reported-by: Hans Peter
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - Don't handle empty patch files

When an empty string is passed to patch option, the driver should
ignore it.  Otherwise it gets an error by trying to load it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - Fix missing CA initialization for HDMI/DP

The commit 53d7d69d8ffdfa60c5b66cc2e9ee0774aaaef5c0
    ALSA: hdmi - support infoframe for DisplayPort
dropped the initialization of CA field accidentally.
This resulted in only two-channel LPCM mode on Nvidia machines.

Reference: kernel bug 28592
https://bugzilla.kernel.org/show_bug.cgi?id=28592

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>

ALSA: usbaudio - Enable the E-MU 0204 USB

Signed-off-by: Joseph Teichman <josteich@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: USB: 6fire: signedness bug in usb6fire_pcm_prepare()

rt->rate is an unsigned char so it's never equal to -1.  It's not a huge
problem because the invalid rate is caught inside the call to
usb6fire_pcm_set_rate() which returns -EINVAL.  But if we fix the test
then it prints out the correct error message so that's good.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - switch lfe with side in mixer for 4930g

Built-in sub-woofer can now be controlled by lfe slider instead of
side slider on Acer Aspire 5930g

Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge branch 'fix/asoc' into for-linus

ASoC: Improve WM8994 digital power sequencing

On WM8994 revision D and earlier ensure optimal sequencing with
simultaneous usage of AIF1 and AIF2 by tying the signals together
so if paths through both are connected the streams are started
simultaneously.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org

ASoC: Create an AIF1ADCDAT signal widget to match AIF2

Due to the different routing for AIF1 and AIF2 we weren't using a
single widget to represent the ADCDAT signal. For consistency add
one.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org

asoc: davinci: da830/omap-l137: correct cpu_dai_name

McASP1 is used on the DA830/OMAP-L137 platform for the codec.
This is different from the DA850/OMAP-L138 platform which uses McASP0.

This is fixed by adding a new snd_soc_dai_link struct.

Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init()

The .card member of the snd_soc_pcm_runtime structure pointed to by the
snd_soc_dai_link.init() argument used to be initialized before the
function being called. This has changed, probably unintentionally,
after recent refactorings. Since the function implementations are free
to make use of this pointer, move its assignment back before the
function is called to avoid NULL pointer dereferences.

Created and tested on Amstrad Delta againts linux-2.6.38-rc2

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ALSA: use linux/io.h to fix compile warnings

For helping to reduce Greert's regression list...
  src/sound/drivers/mtpav.c: error: implicit declaration of function 'inb'
  src/sound/drivers/mtpav.c: error: implicit declaration of function 'outb'
  ...

Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: hda - Fix memory leaks in conexant jack arrays

The Conexant codec driver adds the jack arrays in init callback which
may be called also in each PM resume.  This results in the addition of
new jack element at each time.

The fix is to check whether the requested jack is already present in
the array.

Reference: Novell bug 668929
https://bugzilla.novell.com/show_bug.cgi?id=668929

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ASoC: CX20442: fix NULL pointer dereference

The CX20442 codec driver never provided the snd_soc_codec_driver's
.reg_cache_default member. With the latest ASoC framework changes, it
seems to be referred unconditionally, resulting in a NULL pointer
dereference if missing. Provide it.

Created and tested on Amstrad Delta against linux-2.6.38-rc2

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

ASoC: Amstrad Delta: fix const related build error

The Amstrad Delta ASoC driver used to override the digital_mute()
callback, expected to be not provided by the on-board CX20442 CODEC
driver, with its own implementation. While this is still posssible when
substituting the whole empty snd_soc_dai_driver.ops member (the CX20442
case), replacing snd_soc_dai_ops.digital_mute only is no longer correct
after the snd_soc_dai_driver.ops member has been constified, and results
in build error.

Drop this actually not used code path in hope the CX20442 driver never
provides its own snd_soc_dai_ops structure.

Created and tested against linux-2.6.38-rc2

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'topic/hda' into fix/hda