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* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (257 commits)
[ARM] Update mach-types
ARM: 5636/1: Move vendor enum to AMBA include
ARM: Fix pfn_valid() for sparse memory
[ARM] orion5x: Add LaCie NAS 2Big Network support
[ARM] pxa/sharpsl_pm: zaurus c3000 aka spitz: fix resume
ARM: 5686/1: at91: Correct AC97 reset line in at91sam9263ek board
ARM: 5640/1: This patch modifies the support of AC97 on the at91sam9263 ek board
ARM: 5689/1: Update default config of HP Jornada 700-series machines
ARM: 5691/1: fix cache aliasing issues between kmap() and kmap_atomic() with highmem
ARM: 5688/1: ks8695_serial: disable_irq() lockup
ARM: 5687/1: fix an oops with highmem
ARM: 5684/1: Add nuc960 platform to w90x900
ARM: 5683/1: Add nuc950 platform to w90x900
ARM: 5682/1: Add cpu.c and dev.c and modify some files of w90p910 platform
ARM: 5626/1: add suspend/resume functions to amba-pl011 serial driver
ARM: 5625/1: fix hard coded 4K resource size in amba bus detection
MMC: MMCI: convert realview MMC to use gpiolib
ARM: 5685/1: Make MMCI driver compile without gpiolib
ARM: implement highpte
ARM: Show FIQ in /proc/interrupts on CONFIG_FIQ
...
Fix up trivial conflict in arch/arm/kernel/signal.c.
It was due to the TIF_NOTIFY_RESUME addition in commit d0420c83f ("KEYS:
Extend TIF_NOTIFY_RESUME to (almost) all architectures") and follow-ups.
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Conflicts:
MAINTAINERS
arch/arm/mm/fault.c
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devel-stable
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Move the definition of the "generic" IRQ in the process.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
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'w90x900' into devel
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The WM8731 driver has been updated to allow registration via normal
device model methods rather than from within the ASoC driver probe
so update the AT91SAM9G20-EK to make use of this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Andrew Victor <linux@maxim.org.za>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
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* topic/ymfpci:
sound: ymfpci: increase timer resolution to 96 kHz
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Allow the interval timer to be programmed with its full 96 kHz
precision.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/usb-audio:
ALSA: usb-audio - Fix types taken in min()
sound: usb-audio: do not make URBs longer than sync packet interval
sound: usb-audio: add MIDI drain callback
sound: usb-audio: use multiple output URBs
sound: usb-audio: use multiple input URBs
sound: usb-audio: Xonar U1 digital output support
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Fix the compile warning due to different integer types used in min():
sound/usb/usbaudio.c: In function 'init_substream_urbs':
sound/usb/usbaudio.c:1087: warning: comparison of distinct pointer types lacks a cast
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Using more packets in one URB do avoid interrupts does not make sense
when we have a sync pipe whose packets generate interrupts more often.
Therefore, limit the URB size to the synchronization packet interval.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When draining, instead of waiting for fifty milliseconds, just wait for
the currently active URBs to complete. This cuts the usual waiting time
down to one USB frame, or zero in the common case when there is no URB.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some newer USB MIDI interfaces use rather small packet sizes, so to get
enough bandwidth, we have to be able to send multiple packets in one USB
frame, so we have to use multiple URBs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some newer USB MIDI interfaces use rather small packet sizes, so to get
enough bandwidth, we have to be able to receive multiple packets in one
USB frame, so we have to use multiple URBs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add support for the Asus Xonar U1. This device is mostly class compliant, but
the digital output requires a vendor-specific request.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/tlv-minmax:
ALSA: usb-audio - Correct bogus volume dB information
ALSA: usb-audio - Use the new TLV_DB_MINMAX type
ALSA: Add new TLV types for dBwith min/max
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Some USB devices give bogus dB information and it screws up PA.
It's better to detect a broken value and correct it in the driver
before exposing the value to the outside.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use the new TLV_DB_MINMAX type instead of TLV_DB_SCALE.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add new types for TLV dB scale specified with min/max values instead
of min/step since the resolution can't match always with the one
a device provides. For example, usb audio devices give 1/256 dB
resolution while ALSA TLV is based on 1/100 dB resolution.
The new min/max types have less problems because the possible
rounding error happens only at min/max.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/soundcore-preclaim:
sound: make OSS device number claiming optional and schedule its removal
sound: request char-major-* module aliases for missing OSS devices
chrdev: implement __[un]register_chrdev()
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If any OSS support is enabled, regardless of built-in or module,
sound_core claims full OSS major number (that is, the old 0-255
region) to trap open attempts and request sound modules using custom
module aliases. This feature is redundant as chrdev already has such
mechanism. This preemptive claiming prevents alternative OSS
implementation.
The custom module aliases are scheduled to be removed and the previous
patch made soundcore emit the standard chrdev aliases too to help
transition.
This patch schedule the feature for removal in a year and makes it
optional so that developers and distros can try new things in the
meantime without rebuilding the kernel. The pre-claiming can be
turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel parameter
soundcore.preclaim_oss.
As this allows sound minors to be individually grabbed by other users,
this patch updates sound_insert_unit() such that if registering
individual device region fails, it tries the next available slot.
For details on removal plan, please read the entry added by this patch
in feature-removal-schedule.txt .
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Till now missing OSS devices emitted sound-slot/service-* module
alises instead of the standard char-major-* if a missing device number
is opened if soundcore is loaded. The custom module aliases don't
have any inherent benefit than backward compatibility.
sound-slot/service-* module aliases is scheduled to be removed and to
help the transition this patch makes soundcore emit the standard
module alises along with the custom ones.
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/snd-printk:
ALSA: Fixed a typo of printk()
ALSA: Add debug module option
ALSA: core - strip too long file names in snd_print*()
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Add debug module option to snd core.
This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value. debug=0 means no debug messsages.
As default, it's set to the verbose level 2.
Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When modules are built with M= option, they pass long file paths to
__FILE__. This results in ugly outputs of snd_print*() when
CONFIG_SND_VERBOSE_PRINTK is set.
This patch adds a check of the path and strips the leading path dirs
if the file name is an absolute path to improve the readability of logs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/pcm-estrpipe-in-pm:
ALSA: pcm - Tell user that stream to be rewound is suspended
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Return STRPIPE instead of EBADF when userspace attempts to rewind
of forward a stream that was suspended in meanwhile, so that it
can be recovered by snd_pcm_recover().
This was causing Pulseaudio to unload the ALSA sink module under a race
condition when it attempted to rewind the stream right after resume from
suspend, before writing to the stream which would cause it to revive the
stream otherwise. Tested to work with Pulseaudio patched to attempt to
snd_pcm_recover() upon receiving an error from snd_pcm_rewind().
Signed-off-by: Lubomir Rintel <lkundrak@v3.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/pcm-drain-nonblock:
ALSA: pcm - Increase protocol version
ALSA: pcm - Fix drain behavior in non-blocking mode
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The current PCM core has the following problems regarding PCM draining
in non-blocking mode:
- the current f_flags isn't checked in snd_pcm_drain(), thus changing
the mode dynamically via snd_pcm_nonblock() after open doesn't work.
- calling drain in non-blocking mode just return -EAGAIN error, but
doesn't provide any way to sync with draining.
This patch fixes these issues.
- check file->f_flags in snd_pcm_drain() properly
- when O_NONBLOCK is set, PCM core sets the stream(s) to DRAIN state
but quits ioctl immediately without waiting the whole drain; the
caller can sync the drain manually via poll()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/oxygen:
sound: oxygen: work around MCE when changing volume
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When the volume is changed continuously (e.g., when the user drags a
volume slider with the mouse), the driver does lots of I2C writes.
Apparently, the sound chip can get confused when we poll the I2C status
register too much, and fails to complete a read from it. On the PCI-E
models, the PCI-E/PCI bridge gets upset by this and generates a machine
check exception.
To avoid this, this patch replaces the polling with an unconditional
wait that is guaranteed to be long enough.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Johann Messner <johann.messner at jku.at>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/oss:
ALSA: allocation may fail in snd_pcm_oss_change_params()
sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
sound: fix OSS MIDI output data loss
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Allocation may fail, show if it did.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
[Additional fix for invalid runtime->oss.prepare flag set by tiwai]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since !LI_CCFG_* evaluates to 0, this did not change anything to
cfgval and ctlval.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In the 2.1.6 kernel, the output loop in midi_poll() was changed to
enable interrupts during the outputc() call. Unfortunately, the check
whether the device has accepted the current byte ("ok") was moved behind
the code that removes the byte from the output queue, so one byte would
be lost every time the hardware FIFO is full.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/misc:
ALSA: Remove unneeded ifdef from sound/core.h
ALSA: Remove struct snd_monitor_file from public sound/core.h
ALSA: Release v1.0.21
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The struct snd_monitor_file is used locally only in sound/core/init.c,
thus it should be moved there from the public sound/core.h.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/midi:
sound: rawmidi: disable active-sensing-on-close by default
sound: seq_oss_midi: remove magic numbers
sound: seq_midi: do not send MIDI reset when closing
seq-midi: always log message on output overrun
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Sending an Active Sensing message when closing a port can interfere with
the following data if the port is reopened and a note-on is sent before
the device's timeout has elapsed. Therefore, it is better to disable
this setting by default.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Instead of using magic numbers for the controlles sent when resetting
a port, use the symbols from asoundef.h.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Sending a MIDI reset message when closing a port is wrong because we
only want to shut the device up, not to reset all settings.
Furthermore, many devices ignore this message.
Fortunately, the RawMIDI layer already shuts the device up, so we can
ignore this matter here.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It turns out that the main cause of output buffer overruns is not slow
drivers but applications that generate too many messages. Therefore, it
makes more sense to make that error message always visible, and to
rate-limit it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/ice1724-pm:
ALSA: ice1724 - Fix section mismatch
ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD2
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Now snd_vt1724_chip_reset() is used in the resume callback, thus
it cannot be __devinit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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I've built a small HTPC and had to add suspend/resume support in ice1724
driver. There seem to be 3 existing bugs related to that:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3748
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2314
Due to hardware (un)availability, I only enabled the fix for Audiotrak
Prodigy HD2 card, which is installed in my HTPC. However, most of my code
should be reusable in the future on other ice1724-based cards as well (as
long as people add card-specific peices of code). The fix is currently based
on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel).
Signed-off-by: Igor Chernyshev <igor.ch75+alsa at gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/hdsp:
ALSA: hdsp - allow proc reporting with disconnected io box
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the hdsp driver refuses to report any information via the proc
interface, if the io box is not connected. with this patch, the
content of the control and status registers is printed before the
iobox check.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* topic/hda: (92 commits)
ALSA: hda - Use auto model for HP laptops with ALC268 codec
ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
ALSA: hda - Add support of Alienware M17x laptop
ALSA: hda - Remove dead codes from patch_sigmatel.c
ALSA: hda - Fix input source selection of IDT92HD73xx
ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT
ALSA: hda - Unmute docking line-out as default with AD1984A codec
ALSA: hda - Add another entry for Nvidia HDMI device
ALSA: hda - Add missing GPIO initialization for AD1984A laptop model
ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop model
ALSA: hda - Fix ALC268/ALC269 headphone pin routing
ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT codecs
ALSA: hda - Add more quirk for HP laptops with AD1984A
ALSA: hda - Add / fix model entries for HD-audio driver
ALSA: hda - Add full audio support on Acer Aspire 7730G notebook
ALSA: hda - Improve auto-cfg mixer name for ALC662
ALSA: hda - Improve auto-cfg mixer name for ALC861-VD
ALSA: hda - Improve auto-cfg mixer name for ALC262
ALSA: hda - Improve auto-cfg mixer name for ALC260
ALSA: hda - Improve auto-cfg mixer name for ALC880
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The HP laptops with ALC268 codec seem working better with model=auto
than model=toshiba; e.g. the auto model fixes missing digital outputs.
Let's fix quirk entry to choose auto model explicitly.
Tested-by: Jens Jorgensen <jbj1@ultraemail.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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