Daniel T Chen [Sun, 21 Nov 2010 19:01:14 +0000 (14:01 -0500)]
ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J
BugLink: https://launchpad.net/bugs/677652
The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers. Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality. Testing was done
with an alsa-driver build from 2010-11-21.
Reported-and-tested-by: Joan Creus Cc: <stable@kernel.org> [2.6.35+] Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Andreas Mohr [Sun, 21 Nov 2010 11:09:32 +0000 (12:09 +0100)]
ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer
. Fix PulseAudio "ALSA driver bug" issue
(if we have two alternated areas within a 64k DMA buffer, then max
period size should obviously be 32k only).
Back references:
http://pulseaudio.org/wiki/AlsaIssues
http://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction
When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.
PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.
Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: Andreas Mohr <andi@lisas.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Daniel T Chen [Sat, 20 Nov 2010 15:20:35 +0000 (10:20 -0500)]
ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup
BugLink: https://launchpad.net/bugs/677830
The original reporter states that the subwoofer does not mute when
inserting headphones. We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).
Reported-and-tested-by: i-NoD Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Thu, 18 Nov 2010 08:53:54 +0000 (09:53 +0100)]
ALSA: timer: automatically load the high-resolution timer
Increase the default timer limit so that snd-hrtimer.ko can be
automatically loaded when needed, e.g., when used as the default
sequencer timer. This replaces the check for the obsolete
CONFIG_SND_HPET.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Thu, 18 Nov 2010 08:53:07 +0000 (09:53 +0100)]
ALSA: pcm: optimize xrun detection in no-period-wakeup mode
Add a lightweight condition on top of the xrun checking so that we can
avoid the division when the application is calling the update function
often enough.
Suggested-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Thu, 18 Nov 2010 08:43:52 +0000 (09:43 +0100)]
ALSA: pcm: detect xruns in no-period-wakeup mode
When period wakeups are disabled, successive calls to the pointer update
function do not have a maximum allowed distance, so xruns cannot be
detected with the pointer value only.
To detect xruns, compare the actually elapsed time with the time that
should have theoretically elapsed since the last update. When the
hardware pointer has wrapped around due to an xrun, the actually elapsed
time will be too big by about hw_ptr_buffer_jiffies.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Mon, 15 Nov 2010 09:46:23 +0000 (10:46 +0100)]
ALSA: pcm: support for period wakeup disabling
This patch allows to disable period interrupts which are
not needed when the application relies on a system timer
to wake-up and refill the ring buffer. The behavior of
the driver is left unchanged, and interrupts are only
disabled if the application requests this configuration.
The behavior in case of underruns is slightly different,
instead of being detected during the period interrupts the
underruns are detected when the application calls
snd_pcm_update_avail, which in turns forces a refresh of the
hw pointer and shows the buffer is empty.
More specifically this patch makes a lot of sense when
PulseAudio relies on timer-based scheduling to access audio
devices such as HDAudio or Intel SST. Disabling interrupts
removes two unwanted wake-ups due to period elapsed events
in low-power playback modes. It also simplifies PulseAudio
voice modules used for speech calls.
To quote Lennart "This patch looks very interesting and
desirable. This is something have long been waiting for."
Daniel T Chen [Sun, 21 Nov 2010 19:01:14 +0000 (14:01 -0500)]
ALSA: hda: Use hp-laptop quirk to enable headphones automute for Asus A52J
BugLink: https://launchpad.net/bugs/677652
The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers. Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality. Testing was done
with an alsa-driver build from 2010-11-21.
Reported-and-tested-by: Joan Creus Cc: <stable@kernel.org> [2.6.35+] Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Andreas Mohr [Sun, 21 Nov 2010 11:09:32 +0000 (12:09 +0100)]
ALSA: azt3328: period bug fix (for PA), add missing ACK on stop timer
. Fix PulseAudio "ALSA driver bug" issue
(if we have two alternated areas within a 64k DMA buffer, then max
period size should obviously be 32k only).
Back references:
http://pulseaudio.org/wiki/AlsaIssues
http://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction
When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.
PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.
Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: Andreas Mohr <andi@lisas.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Daniel T Chen [Sat, 20 Nov 2010 15:20:35 +0000 (10:20 -0500)]
ALSA: hda: Add Samsung R720 SSID for subwoofer pin fixup
BugLink: https://launchpad.net/bugs/677830
The original reporter states that the subwoofer does not mute when
inserting headphones. We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).
Reported-and-tested-by: i-NoD Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/669092
ALC887 does not have any volume control ability on the mixer NIDs,
so put the volume controls on the dac NIDs instead. Without this
patch, ALC887 users cannot use alsamixer at all.
Cc: stable@kernel.org Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Daniel T Chen [Mon, 1 Nov 2010 05:14:51 +0000 (01:14 -0400)]
ALSA: ac97: Apply quirk for Dell Latitude D610 binding Master and Headphone controls
BugLink: https://launchpad.net/bugs/669279
The original reporter states: "The Master mixer does not change the
volume from the headphone output (which is affected by the headphone
mixer). Instead it only seems to control the on-board speaker volume.
This confuses PulseAudio greatly as the Master channel is merged into
the volume mix."
Fix this symptom by applying the hp_only quirk for the reporter's SSID.
The fix is applicable to all stable kernels.
Reported-and-tested-by: Ben Gamari <bgamari@gmail.com> Cc: <stable@kernel.org> [2.6.32+] Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Axel Lin [Fri, 19 Nov 2010 07:48:06 +0000 (15:48 +0800)]
ASoC: uda134x - set reg_cache_default to uda134x_reg
After checking the code in 2.6.36,
I found this is missing during multi-component conversion.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Multi-component commit f0fba2ad broke a few things which this patch should
fix. Tested on the DM355 EVM. I've been as careful as I can, but it would be
good if those with access to other Davinci boards could test.
--
The multi-component commit put the initialisation of
snd_soc_dai.[capture|playback]_dma_data into snd_soc_dai_ops.hw_params of the
McBSP, McASP & VCIF drivers (davinci-i2s.c, davinci-mcasp.c & davinci-vcif.c).
The initialisation had to be moved from the probe function in these drivers
because davinci_*_dai changed from snd_soc_dai to snd_soc_dai_driver.
Unfortunately, the DMA params pointer is needed by davinci_pcm_open (in
davinci-pcm.c) before hw_params is called. I have moved the initialisation to
a new snd_soc_dai_ops.startup function in each of these drivers. This fix
indicates that all platforms that use davinci-pcm must have been broken and
need to test with this fix.
--
The multi-component commit also changed the McBSP driver name from
"davinci-asp" to "davinci-i2s" in davinci-i2s.c without updating the board
level references to the driver name. This change is understandable, as there
is a similarly named "davinci-mcasp" driver in davinci-mcasp.c.
There is probably no 'correct' name for this driver. The DM6446 datasheet
calls it the "ASP" and describes it as a "specialised McBSP". The DM355
datasheet calls it the "ASP" and describes it as a "specialised ASP". The
DM365 datasheet calls it the "McBSP". Rather than fix this problem by
reverting to "davinci-asp", I've elected to avoid future confusion with the
"davinci-mcasp" driver by changing it to "davinci-mcbsp", which is also
consistent with the names of the functions in the driver. There are other
fixes required, so it was never going to be as simple as a revert anyway.
--
The DM365 only has one McBSP port (of the McBSP platforms, only the DM355 has
2 ports), so I've changed the the id of the platform_device from 0 to -1.
--
In davinci-evm.c, the DM6446 EVM can no longer share a snd_soc_dai_link
structure with the DM355 EVM as they use different cpu DAI names (the DM355
has 2 ports and the EVM uses the second port, but the DM6446 only has 1 port).
This also means that the 2 boards need different snd_soc_card structures.
--
The codec_name entries in davinci-evm.c didn't match the i2c ids in the board
files. I have only checked and fixed the details of the names used for the
McBSP based platforms. Someone with a McASP based platform (eg DA8xx) should
check the others.
Signed-off-by: Chris Paulson-Ellis <chris@edesix.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 12 Nov 2010 15:14:55 +0000 (15:14 +0000)]
ASoC: Fix dapm_seq_compare() for multi-component
Ensure that we keep all widget powerups in DAPM sequence by making
the CODEC the last thing we compare on rather than the first thing.
Also fix the fact that we're currently comparing the widget pointers
rather than the CODEC pointers when we do the substraction so we
won't get stable results.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unfortunatelly, I misunderstood datasheet, and on s3c244x-iis
when MPLLin source for master clock is selected, prescaler has
no effect. Remove dividor calculation for 44100 rate; remove 88200
rate at all, rx1950 can't do it.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ryan Mallon [Wed, 10 Nov 2010 20:02:30 +0000 (09:02 +1300)]
Fix Atmel soc audio boards Kconfig dependency
Add Kconfig dependency on AT91_PROGRAMMABLE_CLOCKS for the Atmel SoC
audio SAM9G20-EK and PlayPaq boards. Fixes link errors on missing
clk_set_parent and clk_set_rate when building without
AT91_PROGRAMMABLE_CLOCKS.
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> Acked-by: Geoffrey Wossum <gwossum@acm.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Julia Lawall [Tue, 9 Nov 2010 22:00:41 +0000 (23:00 +0100)]
ALSA: sound/pci/ctxfi/ctpcm.c: Remove potential for use after free
In each function, the value apcm is stored in the private_data field of
runtime. At the same time the function ct_atc_pcm_free_substream is stored
in the private_free field of the same structure. ct_atc_pcm_free_substream
dereferences and ultimately frees the value in the private_data field. But
each function can exit in an error case with apcm having been freed, in
which case a subsequent call to the private_free function would perform a
dereference after free. On the other hand, if the private_free field is
not initialized, it is NULL, and not invoked (see snd_pcm_detach_substream
in sound/core/pcm.c). To avoid the introduction of a dangling pointer, the
initializations of the private_data and private_free fields are moved to
the end of the function, past any possible free of apcm. This is safe
because the previous calls to snd_pcm_hw_constraint_integer and
snd_pcm_hw_constraint_minmax, which take runtime as an argument, do not
refer to either of these fields.
In each function, there is one error case where apcm needs to be freed, and
a call to kfree is added.
The sematic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
*e->f = a
... when != e->f = e1
when any
if (...) {
... when != free1(...,e,...)
when != e->f = e2
* kfree(a)
... when != free2(...,e,...)
when != e->f = e3
}
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/673075
According to the datasheet of 92HD87B, there is a digital mic
at nid 0x11, so enable it in order to be able to use the mic.
Cc: stable@kernel.org Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jesper Juhl [Mon, 8 Nov 2010 23:11:03 +0000 (00:11 +0100)]
sound/oss: Remove unnecessary casts of void ptr
The [vk][cmz]alloc(_node) family of functions return void pointers which
it's completely unnecessary/pointless to cast to other pointer types since
that happens implicitly.
Mark Brown [Tue, 9 Nov 2010 14:38:58 +0000 (14:38 +0000)]
ASoC: Ensure sane WM835x AIF configuration by default
Ensure that whatever ran before us leaves the WM835x with a sane default
audio interface configuration as we do not override the companding,
loopback or tristate settings and do not reset the chip at startup (as it
is a PMIC).
Reported-by: Keiji Mitsuhisa <Keiji.Mitsuhisa@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Tue, 9 Nov 2010 12:00:11 +0000 (12:00 +0000)]
ASoC: Remove broken WM8350 direction constants
The WM8350 driver was using some custom constants to interpret the direction
of the MCLK signal which had the opposite values to those used as standard
by the ASoC core, causing confusion in machine drivers such as the 1133-EV1
board.
Reported-by: Tommy Zhu <Tommy.Zhu@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Marek Belisko [Mon, 8 Nov 2010 12:14:51 +0000 (13:14 +0100)]
ASoC: s3c24xx: Fix compilation problem for mini2440
When make mini2440_defconfig compilation end with undefined
references to DMA functions. There was missing selection
for S3C2410_DMA when compile ASoC audio for S3C24xx CPU.
Tested on mini2440 board.
Signed-off-by: Marek Belisko <marek.belisko@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Mon, 8 Nov 2010 05:24:58 +0000 (13:24 +0800)]
ASoC: Return proper error if snd_soc_register_dais fails in psc_i2s_of_probe
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The member reg_cache is not used at all and therefore it should be
removed. This member was usually needed for older versions of ASoC
that did not handle caching automatically and had to be done in the
driver itself.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sascha Hauer [Thu, 4 Nov 2010 16:05:42 +0000 (17:05 +0100)]
ASoC i.MX: register dma audio device
We have two different transfer methods on i.MX: FIQ and DMA. Since
the merge of the ASoC multicomponent support the DMA device is lost.
Add it again. Also, imx_ssi_dai_probe has to be called for !AC97
aswell.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Wed, 3 Nov 2010 14:39:00 +0000 (16:39 +0200)]
ASoC: tpa6130a2: Get rid of compile warning from tpa6130a2_power
Patch "ASoC: tpa6130a2: Fix unbalanced regulator disables" introduced a
compiler warning "‘ret’ may be used uninitialized in this function".
Initialize ret to zero to get rid of it and making sure that the function
does not return any random error code when the code is falling through.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Wed, 3 Nov 2010 11:26:35 +0000 (12:26 +0100)]
ALSA: oxygen: add HiFier Serenade support
Add support for the TempoTec/MediaTek HiFier Serenade sound card.
The PCI ID was already there, but the driver handled it like the
Fantasia model, which resulted in a dummy recording device. As
a stereo output-only card, this model is to be handled exactly
like the HG2PCI.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Wed, 3 Nov 2010 10:36:33 +0000 (11:36 +0100)]
ALSA: oxygen: reorganize PCI IDs
Sort the PCI IDs so that they make logical sense. Also move the card
name comments into this list because the model symbols should be (more)
self-explanationary.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Axel Lin [Wed, 3 Nov 2010 07:04:45 +0000 (15:04 +0800)]
ASoC: Fix snd_soc_register_dais error handling
kzalloc for dai may fail at any iteration of the for loop,
thus properly unregister already registered DAIs before return error.
The error handling code in snd_soc_register_dais() already ensure all the DAIs
are unregistered before return error, we can remove the error handling code
to unregister DAIs in snd_soc_register_codec().
Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clemens Ladisch [Tue, 2 Nov 2010 16:08:37 +0000 (17:08 +0100)]
ALSA: oxygen: merge HiFier driver into snd-oxygen
The snd-hifier driver contains more duplicated code than model-specific
code, so it does not make sense for it to be a separate driver.
Handling the two-channel output restriction can be easily done in the
generic driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mandar Joshi [Tue, 2 Nov 2010 14:43:19 +0000 (14:43 +0000)]
ALSA: usb-audio - Support for Power/Status LED on Creative USB X-Fi S51
This patch adds support for Power/Status LED on Creative USB X-Fi S51.
There is just one LED on the device. The LED can either be On or it
can be set to Blink. There doesn't seem to be a way to switch it off.
The control message to change LED status is similar to that of
audigy2nx except that the index is to be set to 0 and value is 1 for
Blink and 0 for On.
The 'Power LED' control in alsamixer when muted will cause the LED to
Blink continuously. When unmuted the LED will stay On. The Creative
driver under Windows sets the LED to blink whenever audio is muted.
This LED can be treated as the CMSS LED but I figured since there is
just one LED, it should be treated as the Power LED. Is that alright?
I've also changed the comment "Usb X-Fi" to "Usb X-Fi S51" as there
are other external X-Fi devices from Creative like Usb X-Fi Go and
Xmod. The volume knob and LED support patch doesn't apply to them.
Jesper Juhl [Fri, 29 Oct 2010 19:35:25 +0000 (21:35 +0200)]
ALSA: asihpi - Unsafe memory management when allocating control cache
I noticed that sound/pci/asihpi/hpicmn.c::hpi_alloc_control_cache() does
not check the return value from kmalloc(), which may fail.
If kmalloc() fails we'll dereference a null pointer and things will go bad
fast.
There are two memory allocations in that function and there's also the
problem that the first may succeed and the second may fail and nothing is
done about that either which will also go wrong down the line.
Joe Perches [Sat, 30 Oct 2010 21:08:56 +0000 (14:08 -0700)]
ASoC: Update WARN uses in wm_hubs
Add missing newlines.
Signed-off-by: Joe Perches <joe@perches.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Fri, 29 Oct 2010 13:47:44 +0000 (16:47 +0300)]
ASoC: Fix SND_SOC_ALL_CODECS typo for alc5623
Include alc5623.c in SND_SOC_ALL_CODECS when dependencies are met.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Fri, 29 Oct 2010 13:47:45 +0000 (16:47 +0300)]
ASoC: Include cx20442 to SND_SOC_ALL_CODECS
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Fri, 29 Oct 2010 13:47:44 +0000 (16:47 +0300)]
ASoC: Fix SND_SOC_ALL_CODECS typo for jz4740
Include jz4740.c to SND_SOC_ALL_CODECS when the dependencies are met.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 29 Oct 2010 22:41:17 +0000 (15:41 -0700)]
ASoC: Remove volatility from WM8900 POWER1 register
Not all bits can be read back from POWER1 so avoid corruption when using
a read/modify/write cycle by marking it non-volatile - the only thing we
read back from it is the chip revision which has diagnostic value only.
We can re-add later but that's a more invasive change than is suitable
for a bugfix.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
Linus Torvalds [Mon, 1 Nov 2010 11:50:43 +0000 (07:50 -0400)]
Merge branch 'next-spi' of git://git.secretlab.ca/git/linux-2.6
* 'next-spi' of git://git.secretlab.ca/git/linux-2.6:
spi/pl022: fix erroneous platform data in U300
spi: fixed odd static string conventions in core code
spi/bfin_spi: only request GPIO on first load
spi/bfin_spi: handle error/status changes after data interrupts
spi: enable spi_board_info to be registered after spi_master
Jesper Juhl [Fri, 29 Oct 2010 20:54:45 +0000 (22:54 +0200)]
ALSA: cs46xx memory management fixes for cs46xx_dsp_spos_create()
When reading through sound/pci/cs46xx/dsp_spos.c I noticed a couple of
things in cs46xx_dsp_spos_create().
It seems to me that we don't always free the various memory buffers we
allocate and we also do some work (structure member assignment) early,
that is completely pointless if some of the memory allocations fail and
we end up just aborting the whole thing.
I don't have hardware to test, so the patch below is compile tested only,
but it makes the following changes:
- Make sure we always free all allocated memory on failures.
- Don't do pointless work assigning to structure members before we know
all memory allocations, that may abort progress, have completed
successfully.
- Remove some trailing whitespace.
Jesper Juhl [Fri, 29 Oct 2010 18:40:23 +0000 (20:40 +0200)]
ALSA: usb - driver neglects kmalloc return value check and may deref NULL
sound/usb/pcm.c::snd_usb_pcm_check_knot() fails to check the return value
from kmalloc() and may end up dereferencing a null pointer.
The patch below (compile tested only) should take care of that little
problem.
Linus Walleij [Thu, 21 Oct 2010 19:05:25 +0000 (21:05 +0200)]
spi/pl022: fix erroneous platform data in U300
This fixes an erroneous use of LSB first in the U300 machine, the
PL022 used in U300 is a standard ARM core that doesn't support this
bit so it should never have been set.
Cc: Kevin Wells <wellsk40@gmail.com>OA Signed-off-by: Linus Walleij <linus.walleij@stericsson.com> Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
We were forgetting to set up proper return value in success path causing
ir_getkeycode() to fail intermittently:
drivers/media/IR/ir-keytable.c: In function 'ir_getkeycode':
drivers/media/IR/ir-keytable.c:363: warning: 'retval' may be used
uninitialized in this function
Linus Torvalds [Mon, 1 Nov 2010 01:03:23 +0000 (21:03 -0400)]
Merge branch 'i2c-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jdelvare/staging
* 'i2c-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jdelvare/staging:
i2c-i801: Add PCI idents for Patsburg 'IDF' SMBus controllers
i2c-i801: Handle multiple instances instead of keeping global state
i2c-i801: Add Intel Patsburg device ID
i2c: Drop unused I2C_CLASS_TV flags
Linus Torvalds [Mon, 1 Nov 2010 01:02:39 +0000 (21:02 -0400)]
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/dtor/input
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/dtor/input:
Input: appletouch - remove extra KERN_DEBUG use from dprintk
Input: bu21013_ts - fix null dereference in error handling
Input: ad7879 - prevent invalid finger data reports
projects / karo-tx-linux.git / log