Philby John [Fri, 26 Mar 2010 16:07:51 +0000 (21:37 +0530)]
ALSA: aaci - Fix alignment faults on ARM Cortex introduced by commit 29a4f2d3
The commit 29a4f2d3 used writel() at offset 0x26 which is
half-word aligned causing unaligned exceptions on a
Cortex-A8. The original patch solved the "aaci-pl041 fpga:04:
ac97 read back fail" issue on a soft reset. Reading from any
arbitrary aaci register seems to solve this issue.
Signed-off-by: Philby John <pjohn@mvista.com> Acked-by: Russell King <rmk+kernel@arm.linux.org.uk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Marek Vasut [Mon, 5 Apr 2010 04:13:38 +0000 (06:13 +0200)]
ASoC: Zipit Z2 WM8750 ASoC driver
This patch adds support for sound through the WM8750 codec on Zipit Z2.
Also, this patch incorporates support for detecting headset jack
insertion through the jack detection API.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Bill Gatliff [Fri, 9 Apr 2010 17:08:08 +0000 (18:08 +0100)]
ASoC: Use SNDRV_PCM_RATE_8000_96000 macro for WM8731
Signed-off-by: Bill Gatliff <bgat@billgatliff.com> Acked-by: Richard Purdie <rpurdie@rpsys.net> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sony VAIO models with ALC269 need to initialize the pin 0x19 to VREF
ground or Hi-Z to make the headphone working. Other than that, model=auto
works fine, so let's use model=auto with a specific fix-up table.
ALSA: hda - Enhance fix-up table for Realtek codecs
A few enhancement / fixes for fix-up table of some Realtek codecs:
- Apply fix-ups only for the auto model
- Apply additional verbs after normal init verbs
- Add a debug print to show the fix-up application
This is basically a preliminary work for the next fix for Sony VAIO.
usb-midi causes sometimes Oops at snd_usbmidi_output_drain() after
disconnection. This is due to the access to the endpoints which have
been already released at disconnection while the files are still alive.
This patch fixes the problem by checking disconnection state at
snd_usbmidi_output_drain() and by releasing urbs but keeping the
endpoint instances until really all freed.
ALSA: hda - Fix initial capture source connections of ALC880/260
The widget connections of ADC of ALC880 and ALC2260 aren't initialized,
thus it might point to invalid pin. This can be a problem when mode=auto
and there is only one input pin. Then user can't change the connection
at all.
This patch adds the code to initialize the input pin connection of these
codecs.
Marek Vasut [Thu, 8 Apr 2010 18:48:51 +0000 (20:48 +0200)]
ASoC: WM8750: Convert to new API
Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code
around. Hugely inspired by WM8753 which was already converted.
Also, this patch fixes the Jive and Spitz machine.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use a local mutex instead of BKL. This should suffice since each device
type has also its open_mutex.
Also, a bit of clean-up of the legacy device auto-loading code.
ALSA: hda - Fix ALC882 DAC connections in auto mode
Assign DACs properly to each output. Currently, the front output is bound
to HP/speaker outputs blindly, but they should be assigned to individual
DACs.
Dan Carpenter [Tue, 6 Apr 2010 16:31:26 +0000 (19:31 +0300)]
ALSA: mixart: range checking proc file
The original code doesn't take into consideration that the value of
MIXART_BA0_SIZE - pos can be less than zero which would lead to a large
unsigned value for "count".
Also I moved the check that read size is a multiple of 4 bytes below
the code that adjusts "count".
ALSA: hda - Fix a wrong array range check in patch_realtek.c
The commit 6a4f2ccb467e00281470cde2dee08fe5ecde62d1 introduced a wrong
comparision for the array range check, which effectively skips the whole
initialization of DAC connections. Fixed now.
Mark Brown [Mon, 29 Mar 2010 19:57:12 +0000 (20:57 +0100)]
ASoC: Implement interrupt based WM8994 microphone detection
Support interrupt based microphone bias detection. The WM8994 has two
microphone bias supplies, with detection supported on both. Detection
using GPIOs together with the standard GPIO based jack framework is
already supported via the platform data for the WM8994 core driver.
Note that as well as the microphone bias itself the system clock and
whichever AIF clock is supplying the system clock will need to be
enabled for detection to function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 26 Mar 2010 16:49:15 +0000 (16:49 +0000)]
mfd: Add WM8994 interrupt controller support
The WM8994 has an interrupt controller which supports interrupts for
both CODEC and GPIO portions of the chip. Support this using genirq,
while allowing for systems that do not have an interrupt hooked up.
Wrapper functions are provided for the IRQ request and free to simplify
the code in consumer drivers when handling cases where IRQs are not
set up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Daniel Mack [Mon, 22 Mar 2010 09:11:15 +0000 (10:11 +0100)]
ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
[Note that this is a backported version for 2.6.34.
Upstream commit is fd23b7dee]
Signed-off-by: Daniel Mack <daniel@caiaq.de> Reported-by: Sven Neumann <s.neumann@raumfeld.com> Reported-by: Michael Hirsch <m.hirsch@raumfeld.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tony Vroon [Mon, 5 Apr 2010 15:30:43 +0000 (16:30 +0100)]
ALSA: hda - Enable amplifiers on Acer Inspire 6530G
After more tests it appears that EAPD needs to be enabled
on both the 0x14 and 0x15 NIDs to enable the main speaker
and headphone amplifiers. The maximum volume setting is
now equal to what the machine achieves under other operating
systems.
Disabling Front or LFE playback triggers EAPD and disables
the amplifier. As such, these two playback switches have
been removed from the mixer.
Signed-off-by: Tony Vroon <tony@linx.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Mon, 29 Mar 2010 16:18:41 +0000 (17:18 +0100)]
ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
The DCS_DATAPATH_BUSY bit used to monitor the completion of DC servo
operations has been deprecated and with some more recente revisions
may perform incorrectly, especially when only analogue bypass paths
are in use. Switch to using readback from the DC servo command
register instead, which is supported for all devices. Without this
unacceptably long timeouts may be observed in some circumstances.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Mon, 29 Mar 2010 15:34:42 +0000 (16:34 +0100)]
ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
If we need to offset correct the DC servo then don't use runtime
recalibration since that is likely to introduce further offsets
which will be evident on powerdown.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Mon, 29 Mar 2010 16:09:45 +0000 (17:09 +0100)]
ASoC: Support second DC servo readback method for wm_hubs
More recent Wolfson hubs devices add the ability to read back the DC
servo calibration information from the register used to write offsets,
and later still ones remove the old readback registers. Add support
for the new scheme, and use it for WM8994 device revisions that
support it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
With recent (2.6.34) chnages in PCM handling, capture stopped working on my
OMAP1510 based Amstrad Delta videophone.
Using 2.6.34-rc2, I was able to correct the problem in 3 different ways:
1. reverting commit 7b3a177b0d4f92b3431b8dca777313a07533a710,
2. enabling additional jiffies check with
echo 4 >/proc/asound/card0/pcm0c0/xrun_debug
3. applying the patch below.
Since I wasn't able to reproduce the problem on my i686 PC, I guess the
problem is probably machine specific.
The patch reuses the method for software emulation of missing hardware
pointer, already implemented for playback on OMAP1510. It's possible that
event if a hardware pointer is available for capture on this machine, its
behaviour may be not compatible with what upper layer expects.
If you think the problem may be more general and should be solved differently,
on a higher level, I can try to work more on it if you give me a hint.
If the patch gets accepted, I suggest it goes as a fix in the current release
cycle.
Created and tested against linux-2.6.34-rc2.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tejun Heo [Mon, 29 Mar 2010 17:52:40 +0000 (02:52 +0900)]
mfd: update gfp/slab.h includes
Implicit slab.h inclusion via percpu.h is about to go away. Make sure
gfp.h or slab.h is included as necessary.
Signed-off-by: Tejun Heo <tj@kernel.org> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel T Chen [Tue, 30 Mar 2010 17:29:28 +0000 (13:29 -0400)]
ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
BugLink: https://launchpad.net/bugs/551606
The OR's hardware distorts at PCM 100% because it does not correspond to
0 dB. Fix this in patch_ad1981() for all models using the Thinkpad
quirk.
Reported-by: Jane Silber Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper, snd_hda_codec_update_cache(), for reducing the unneeded
verbs. This function checks the cached value and skips if it's identical
with the given one. Otherwise it works like snd_hda_codec_write_cache().
Barry Song [Mon, 29 Mar 2010 03:16:00 +0000 (11:16 +0800)]
ASoC: ad193x: move codec register/unregister to bus probe/remove
The way i've factored out the bus probe and removal functions so
that there's no code in the individual I2C and SPI functions means
that the register() and unregister() functions could just be squashed
into the bus_probe() and bus_remove() functions.
Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Graham Gower [Thu, 25 Mar 2010 00:22:12 +0000 (10:52 +1030)]
ASoC: Fix passing platform_data to ac97 bus users and fix a leak
[The issue is an attempt to write the pdata without the AC97 device
allocated when using ac97.c - also added a comment in soc-core.c for the
special case for ac97. -- broonie]
Signed-off-by: Graham Gower <graham.gower@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Mon, 29 Mar 2010 15:21:45 +0000 (17:21 +0200)]
ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
ALC269 codec has a few different variants, and each of them may have
different ADC and MUX widgets. For example, one model has ADC 0x08
with MUX 0x23 while others has ADC 0x09 or ADC 0x07 with MUX 022 or
0x24. The difference of ADC appears usually as the capability of
the digital mic pin (0x12), and the current driver sometimes misses
the internal mic pin due to the mismatching ADC.
This patch adds a bit more clever way to find the matching ADC instead
of the static list. Now the driver checks all active input pins and
fills only the ADC/MUX's that contain all of them.
Takashi Iwai [Mon, 29 Mar 2010 07:16:24 +0000 (09:16 +0200)]
ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
The mask and value parameters passed to snd_hda_codec_amp_stereo()
should be 8-bit values for mute and volume. Passing AMP_IN_MUTE() is
wrong, which is found in many places in patch_realtek.c as a left-over
from the conversion to snd_hda_codec_amp_stereo().
Reported-by: Dan Carpenter <error27@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reported-by: Carlos Laviola <claviola@debian.org> Tested-by: Carlos Laviola <claviola@debian.org> Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Daniel Chen [Sun, 28 Mar 2010 20:32:34 +0000 (13:32 -0700)]
ALSA: ac97: Add Toshiba P500 to ac97 jack sense blacklist
BugLink: https://launchpad.net/bugs/481058
The OR has verified that both 'Headphone Jack Sense' and 'Line Jack Sense'
need to be muted for sound to be audible, so just add the machine's SSID
to the ac97 jack sense blacklist.
Reported-by: Richard Gagne Tested-by: Richard Gagne Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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