Frank Mandarino [Fri, 24 Nov 2006 14:49:39 +0000 (15:49 +0100)]
[ALSA] Update AT91 ASoC driver for 2.6.19 kernel.
Changes were required to support latest AT91 header files.
Also updated to remove AT91RM9200-specific code in the ASoC
platform drivers to support the AT91SAM9260 and AT91SAM9261
chips, but no testing was performed on these chips.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Fri, 24 Nov 2006 14:42:07 +0000 (15:42 +0100)]
[ALSA] intel8x0 - Add spdif_aclink option
Added spdif_aclink module option to specify whether the board
has SPDIF over AC-link or a direct connection from the controller
chip.
NForce and ICH4 (or newer) boards may be equipped with SPDIF
through AC97 codec. In such a case, SPDIF should be handled
as if the old ICH style (the same slot for analog and digital).
A quirk list is added to detect this automatically for known
hardwares.
Corresponds to ALSA bug#2637.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Giuliano Pochini [Fri, 24 Nov 2006 12:03:58 +0000 (13:03 +0100)]
[ALSA] echoaudio, add TLV support
This patch adds TLV support to the echoaudio driver.
All gains are in the range -127dB to +6dB with steps of 1dB, and -128 is
mute. VU-meters levels go from -128 to 0dB. The input gain of the Layla20
ranges from -25dB to +25dB in steps of 0.5dB.
Takashi Iwai [Wed, 22 Nov 2006 10:52:52 +0000 (11:52 +0100)]
[ALSA] hda-codec - Add asus-laptop model for ALC861 (ALC660)
Added a new model 'asus-laptop' for ASUS F2*/F3* laptops
with ALC861 (equivalent with ALC660) codec chip.
Also fixed the model for PCI SSID 1043:1338.
Corresponding to ALSA bug#2480.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Tue, 14 Nov 2006 11:30:52 +0000 (12:30 +0100)]
[ALSA] hda-codec - Add support for Sony UX-90s
Added the model entry (model=hippo) for Sony UX-90s with ALC262 codec.
Although the device has no SPDIF output, the hippo model adds a
PCM output, but it must be harmless.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Tobin Davis [Tue, 14 Nov 2006 11:13:39 +0000 (12:13 +0100)]
[ALSA] Add Conexant audio support to the HD Audio driver
This driver adds limited support for the Conexant 5045 and 5047 HD Audio
codecs. Some issues still need to be resolved. The code is based
primarily on code from the Analog Devices AD1981 support and the Realtek
ALC260 support. Some code came from the original code developed by Alex
Pototskiy (see alsa bugtracker 2485).
Signed-off-by: Tobin Davis <tdavis@dsl-only.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Thu, 9 Nov 2006 15:47:26 +0000 (16:47 +0100)]
[ALSA] ice1724 - Add support of M-Audio Audiophile 192
Added the (experimental) support of M-Audio Audiophile 192 board.
Currently, the analog and the digital playbacks seem working fine.
The inputs seem not working as far as I've tested yet.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Thu, 9 Nov 2006 15:35:01 +0000 (16:35 +0100)]
[ALSA] ASoC - mixer name changes for older OSS app support
This patch suggested by Richard Purdie changes the names of some WM8731
and WM8750 mixers so that they will be recognised by some older OSS
mixer apps.
Changes:-
o WM8731 Playback changed to Master Playback
o WM8750 Out1 changed to Headphone
o WM8750 Out2 changed to Speaker
Takashi Iwai [Mon, 6 Nov 2006 14:38:55 +0000 (15:38 +0100)]
[ALSA] hdspm - Fix printk warnings
sound/pci/rme9652/hdspm.c: In function 'snd_hdspm_hw_params':
sound/pci/rme9652/hdspm.c:3681: warning: format '%08X' expects type 'unsigned int', but argument 4 has type 'unsigned char *'
sound/pci/rme9652/hdspm.c:3692: warning: format '%08X' expects type 'unsigned int', but argument 4 has type 'unsigned char *'
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Instead of using a somewhat algorithmic approach of initializing the
YSS225's registers, just use a simple series of port/value pairs.
This makes it easier to later replace or entirely remove the register
data blob.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Hubert Kahlert [Tue, 31 Oct 2006 14:31:27 +0000 (15:31 +0100)]
[ALSA] Fix mask to stop AT91 SSC clock on shutdown
This patch by Frank Mandarino and Hubert Kahlert fixes a bug in the AT91
SSC (i2s) shutdown code that would erroneously disable other AT91
peripheral clocks.
Signed-off-by: Hubert Kahlert <hkahlert@hk-datentechnik.de> Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Tue, 24 Oct 2006 16:25:29 +0000 (18:25 +0200)]
[ALSA] ac97 - Suppress power-saving mode on non-supporting drivers
Don't enable power-saving mode on drivers that don't support
it. The supporting drivers set AC97_SCAP_POWER_SAVE to scaps
at creation of ac97 instance.
Currently enable on the following drivers: intel8x0, intel8x0m,
atiixp, atiixp-modem, via82xx and via82xx-modem.
Also, a bit clean up of power-saving stuff:
- Don't create an own workq
- Remove superfluous ifdefs
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Thu, 19 Oct 2006 18:35:56 +0000 (20:35 +0200)]
[ALSA] ASoC: Add support for BCLK based on (Rate * Chn * Word Size)
This patch adds support for the DAI BCLK to be generated by multiplying
Rate * Channels * Word Size (RCW).
This now gives 3 options for BCLK clocking and synchronisation :-
1. BCLK = Rate * x
2. BCLK = MCLK / x
3. BCLK = Rate * Chn * Word Size. (New)
Changes:-
o Add support for RCW generation of BCLK
o Update Documentation to include RCW.
o Update DAI documentation for label = value DAI modes.
o Add RCW support to wm8731, wm8750 and pxa2xx-i2s drivers.
Frank Mandarino [Thu, 19 Oct 2006 16:22:53 +0000 (18:22 +0200)]
[ALSA] ASoC AT91 DAI modes update
This patch by Frank Mandarino updates the AT91RM9200 I2S DAI audio modes
as follows:-
o fixes a typo in the 16k mode
o removes experimental 24k mode
o adds a 32k mode.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Remy Bruno [Tue, 17 Oct 2006 10:41:56 +0000 (12:41 +0200)]
[ALSA] hdsp - Add DDS register support for RME9632 rev >= 152
Add DDS register support for RME9632 rev >= 152.
This register sets the sample rate for these cards and is required
in addition to the standard control register. It corresponds to a
quartz divisor.
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Kailang Yang [Tue, 17 Oct 2006 10:32:26 +0000 (12:32 +0200)]
[ALSA] hda-codec - Add new modesl for Realtek codecs
Changes from Realtek driver:
- New models hippo and hippo_1 for ALC262
- New models tagra-dig and tagra-2ch-dig for ALC883
- New id for ALC660 codec chip
Signed-off-by: Kailang Yang <kailang@realtek.com.tw> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Mon, 16 Oct 2006 19:19:48 +0000 (21:19 +0200)]
[ALSA] ASoC - Fix build warnings in soc-core.c
This patch fixes some build warnings in soc-core.c
Changes:-
o Check the return value of soc_ac97_dev_register()
o Check return value of calls to device_create_file()
Remy Bruno [Mon, 16 Oct 2006 10:46:32 +0000 (12:46 +0200)]
[ALSA] hdspm: Add support for AES32
Add support for AES32. Difference between MADI and AES32 is done
through revision. Master support is not finished for now (RME so-called DDS
feature is not supported yet)
Signed-off-by: Remy Bruno <remy.bruno@trinnov.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Fri, 13 Oct 2006 10:33:56 +0000 (12:33 +0200)]
[ALSA] ASoC DAI capabilities labelling
This patch suggested by Takashi changes the DAI capabilities definitions
in pxa-i2s.c, at91rm9200-i2s.c, wm8731.c, wm8750.c and wm9712.c to use a
label = value style.
Liam Girdwood [Thu, 12 Oct 2006 12:29:03 +0000 (14:29 +0200)]
[ALSA] ASoC pxa2xx AC97 support
This patch adds pxa2xx AC97 ASoC audio support. It's based on
sound/arm/pxa-ac97 by Nicolas Pitre with the following differences.
o Modified driver structure to use ASoC core PCM callbacks.
o Removed AC97 configuration function (all handled in ASoC core)
o Added and exported ASoC DAI configuration table.
o Added DMA support for AUX DAC and Mic ADC
o Separated out AC97 reset into cold and warm reset functions.
From: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Nicolas Pitre <nico@cam.org> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Thu, 12 Oct 2006 12:26:55 +0000 (14:26 +0200)]
[ALSA] ASoC pxa2xx DMA support
This patch adds pxa2xx ASoC DMA audio support. It's based on
sound/arm/pxa-pcm.c by Nicolas Pitre with the following differences.
o Modified driver structure to use ASoC core PCM callbacks and data
structures.
o Registration with ASoC core.
From: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Nicolas Pitre <nico@cam.org> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Clemens Ladisch [Mon, 9 Oct 2006 06:17:48 +0000 (08:17 +0200)]
[ALSA] pci: select FW_LOADER instead of depending on it
Let the AudioScience, Echoaudio and Riptide drivers select FW_LOADER
instead of depending on it so that they can be configured without having
to enable FW_LOADER manually.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Frank Mandarino [Fri, 6 Oct 2006 16:40:25 +0000 (18:40 +0200)]
[ALSA] ASoC AT91RM92000 I2S support
This patch adds I2S support to the Atmel AT91RM9200 CPU.
Features:-
o Playback/Capture supported.
o 16 Bit data size.
o 8k - 48k sample rates.
o ssc0, ssc1 and ssc2 supported as I2S ports.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Richard Purdie [Fri, 6 Oct 2006 16:37:32 +0000 (18:37 +0200)]
[ALSA] ASoC codecs: WM9712 support
This patch adds ASoC support for the WM9712 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o Aux DAC.
o 8k - 48k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Richard Purdie [Fri, 6 Oct 2006 16:36:39 +0000 (18:36 +0200)]
[ALSA] ASoC codecs: WM8750 support
This patch adds ASoC support for the WM8750 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o 16 & 24 bit audio.
o 8k - 96k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Richard Purdie [Fri, 6 Oct 2006 16:36:07 +0000 (18:36 +0200)]
[ALSA] ASoC codecs: WM8731 support
This patch adds ASoC support for the WM8731 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o 16 & 24 bit audio.
o 8k - 96k sample rates.
o DAPM.
Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Liam Girdwood [Fri, 6 Oct 2006 16:34:51 +0000 (18:34 +0200)]
[ALSA] ASoC: documentation & maintainer
This patch adds documentation describing the ASoC architecture and a
maintainer entry for ASoC.
The documentation includes the following files:-
codec.txt: Codec driver internals.
DAI.txt: Description of Digital Audio Interface standards and how to
configure a DAI within your codec and CPU DAI drivers.
dapm.txt: Dynamic Audio Power Management.
platform.txt: Platform audio DMA and DAI.
machine.txt: Machine driver internals.
pop_clicks.txt: How to minimise audio artifacts.
clocking.txt: ASoC clocking for best power performance.
Richard Purdie [Fri, 6 Oct 2006 16:32:18 +0000 (18:32 +0200)]
[ALSA] ASoC: dynamic audio power management (DAPM)
This patch adds Dynamic Audio Power Management (DAPM) to ASoC.
Dynamic Audio Power Management (DAPM) is designed to allow portable and
handheld Linux devices to use the minimum amount of power within the
audio subsystem at all times. It is independent of other kernel PM and
as such, can easily co-exist with the other PM systems.
DAPM is also completely transparent to all user space applications as
all power switching is done within the ASoC core. No code changes or
recompiling are required for user space applications. DAPM makes power
switching decisions based upon any audio stream (capture/playback)
activity and audio mixer settings within the device.
DAPM spans the whole machine. It covers power control within the entire
audio subsystem, this includes internal codec power blocks and machine
level power systems.
There are 4 power domains within DAPM:-
1. Codec domain - VREF, VMID (core codec and audio power)
Usually controlled at codec probe/remove and suspend/resume, although
can be set at stream time if power is not needed for sidetone, etc.
2. Platform/Machine domain - physically connected inputs and outputs
Is platform/machine and user action specific, is configured by the
machine driver and responds to asynchronous events e.g when HP are
inserted
3. Path domain - audio subsystem signal paths
Automatically set when mixer and mux settings are changed by the user.
e.g. alsamixer, amixer.
4. Stream domain - DAC's and ADC's.
Enabled and disabled when stream playback/capture is started and stopped
respectively. e.g. aplay, arecord.
All DAPM power switching decisions are made automatically by consulting
an audio routing map of the whole machine. This map is specific to each
machine and consists of the interconnections between every audio
component (including internal codec components).
Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Frank Mandarino [Fri, 6 Oct 2006 16:31:09 +0000 (18:31 +0200)]
[ALSA] ASoC: core code
This patch is the core of ASoC functionality.
The ASoC core is designed to provide the following features :-
o Codec independence. Allows reuse of codec drivers on other platforms
and machines.
o Platform driver code reuse. Reuse of platform specific audio DMA and
DAI drivers on different machines.
o Easy I2S/PCM digital audio interface configuration between codec and
SoC. Each SoC interface and codec registers their audio interface
capabilities with the core at initialisation. The capabilities are
subsequently matched and configured at run time for best power and
performance when the application hw params are known.
o Machine specific controls/operations: Allow machines to add controls
and operations to the audio subsystem. e.g. volume control for speaker
amp.
To achieve all this, ASoC splits an embedded audio system into 3
components :-
1. Codec driver: The codec driver is platform independent and contains
audio controls, audio interface capabilities, codec dapm and codec IO
functions.
2. Platform driver: The platform driver contains the audio dma engine
and audio interface drivers (e.g. I2S, AC97, PCM) for that platform.
3. Machine driver: The machine driver handles any machine specific
controls and audio events. i.e. turning on an amp at start of playback.
Signed-off-by: Frank Mandarino <fmandarino@endrelia.com> Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Richard Purdie [Fri, 6 Oct 2006 16:20:14 +0000 (18:20 +0200)]
[ALSA] ASoC: core and dapm headers
This patch adds the ASoC and DAPM headers.
Features:-
o Defines Digital Audio Interface (DAI) API
o Defines Codec, Platform and Machine API
o Defines Dynamic Audio Power Management API
Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Takashi Iwai [Fri, 6 Oct 2006 15:06:39 +0000 (17:06 +0200)]
[ALSA] intel8x0 - Use pci_iomap
Use pci_iomap and ioread*/iowrite*() functions for accessing
hardwares. pci_iomap is suitable for hardwares like ICH and
compatible that have both PIO and MMIO.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Johannes Berg [Thu, 5 Oct 2006 14:02:22 +0000 (16:02 +0200)]
[ALSA] alsa core: convert to list_for_each_entry*
This patch converts most uses of list_for_each to list_for_each_entry all
across alsa. In some place apparently an item can be on a list with
different pointers so of course that isn't compatible with list_for_each, I
therefore didn't touch those places.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Johannes Berg [Thu, 5 Oct 2006 13:08:23 +0000 (15:08 +0200)]
[ALSA] aoa: fix up i2sbus_attach_codec
This patch changes i2sbus_attach_codec to implement a proper error handling
strategy using labels to jump to the right part. Since it has an elaborate
set-up sequence it also needs that tear-down, which I had hard-coded
inbetween all the checks. This increases readability and should reduce .text
size as well.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Johannes Berg [Thu, 5 Oct 2006 13:07:23 +0000 (15:07 +0200)]
[ALSA] aoa: set device pointer in pcms
This patch makes a few whitespace cleanups and makes i2sbus assign the new
struct device pointer in struct snd_pcm so that the proper device symlink
shows up in sysfs.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Johannes Berg [Thu, 5 Oct 2006 13:06:34 +0000 (15:06 +0200)]
[ALSA] alsa core: add struct device pointer to struct snd_pcm
This patch adds a struct device pointer to struct snd_pcm in order to be
able to give it a different device than the card. It defaults to the card's
device, however, so it should behave identically for drivers not touching
the field.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Johannes Berg [Thu, 5 Oct 2006 13:05:34 +0000 (15:05 +0200)]
[ALSA] allow registering an alsa device with struct device pointer
This patch adds snd_register_device_for_dev taking a struct device
pointer to link the new device to and makes snd_register_device a simple
static inline wrapper around it.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
Jochen Voss [Wed, 4 Oct 2006 16:08:43 +0000 (18:08 +0200)]
[ALSA] Enable the analog loopback of the Revolution 5.1
Enable the analog loopback of the Revolution 5.1 card.
This patch adds support for the PT2258 volume controller and modifies
the Revolution 5.1 driver to make use of this facility. This allows
to control the analog loopback of the card.
Jochen Voss [Wed, 4 Oct 2006 16:04:10 +0000 (18:04 +0200)]
[ALSA] Enable capture from line-in and CD on Revolution 5.1
Enable capture from line-in and CD on the Revolution 5.1 card.
This patch adds support for switching between the 5 input channels of
the AK5365 ADC and modifies the Revolution 5.1 driver to make use of
this facility. Previously the capture channel was fixed to channel 0
(microphone on the Revolution 5.1 card).
Linus Torvalds [Thu, 8 Feb 2007 18:37:22 +0000 (10:37 -0800)]
Merge branch 'upstream-linus' of master.kernel.org:/pub/scm/linux/kernel/git/mfasheh/ocfs2
* 'upstream-linus' of master.kernel.org:/pub/scm/linux/kernel/git/mfasheh/ocfs2: (22 commits)
configfs: Zero terminate data in configfs attribute writes.
[PATCH] ocfs2 heartbeat: clean up bio submission code
ocfs2: introduce sc->sc_send_lock to protect outbound outbound messages
[PATCH] ocfs2: drop INET from Kconfig, not needed
ocfs2_dlm: Add timeout to dlm join domain
ocfs2_dlm: Silence some messages during join domain
ocfs2_dlm: disallow a domain join if node maps mismatch
ocfs2_dlm: Ensure correct ordering of set/clear refmap bit on lockres
ocfs2: Binds listener to the configured ip address
ocfs2_dlm: Calling post handler function in assert master handler
ocfs2: Added post handler callable function in o2net message handler
ocfs2_dlm: Cookies in locks not being printed correctly in error messages
ocfs2_dlm: Silence a failed convert
ocfs2_dlm: wake up sleepers on the lockres waitqueue
ocfs2_dlm: Dlm dispatch was stopping too early
ocfs2_dlm: Drop inflight refmap even if no locks found on the lockres
ocfs2_dlm: Flush dlm workqueue before starting to migrate
ocfs2_dlm: Fix migrate lockres handler queue scanning
ocfs2_dlm: Make dlmunlock() wait for migration to complete
ocfs2_dlm: Fixes race between migrate and dirty
...
projects / linux-2.6 / log