Felix Homann [Mon, 23 Apr 2012 18:24:23 +0000 (20:24 +0200)]
ALSA: usb-audio: Unify M-Audio Fast Track Ultra and Ebox-44 mixer quirks.
Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().
[fixed minor checkpatch.pl warnings by tiwai]
Signed-off-by: Felix Homann <linuxaudio@showlabor.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: updates for 3.4
Slightly larger than normal - the DAPM fix is a "this should always have
worked" type of thing which is very clear and should have no impact on
systems that don't need it. The WM8994 fix is driver specific but
pretty important for that driver.
Some old codecs like ALC880 seem to give a bogus pin capability value 0
occasionally. This breaks the new sanity check in snd_hda_set_pin_ctl().
Skip the sanity checks in such a case.
ALSA: hda - Add snd_hda_get_default_vref() helper function
Add a new helper function to guess the default VREF pin control bits
for mic in. This can be used to set the pin control value safely
matching with the actual pin capabilities.
For setting the pin-control values more safely to match with the
actual pin capability bits, a copule of new helper functions,
snd_hda_set_pin_ctl() and snd_hda_set_pin_ctl_cache(), are
introduced. These are simple replacement of the codec verb write with
AC_VERB_SET_PIN_WIDGET but do more sanity checks and filter out
superfluous pin-control bits if they don't fit with the corresponding
pin capabilities.
Some codecs are screwed up or ignore the command when such a wrong bit
is set. These helpers will avoid such secret errors.
ALSA: HDA: Add external mic quirk for Asus Zenbook UX31E
According to the reporter, external mic starts to work if the
laptop-dmic model is used. According to BIOS pin config, all
pins are consistent with the alc269vb_laptop_dmic fixup, except
for the external mic, which is not present.
Merge fixes for Thinkpad docking-station regressions for 3.3 kernels
back to 3.4. These were committed in that branch to make the stable
merging easier.
ALSA: hda/conexant - Set up the missing docking-station pins
ThinkPad 410,420,510,520 and X201 with cx50585 & co chips have the
docking-station ports, but BIOS doesn't initialize for these pins.
Thus, like the former X200, we need to set up the pins manually in the
driver.
The odd part is that the same PCI SSID is used for X200 and T400, thus
we need to prepare individual fixup tables for cx5051 and others.
ALSA: workaround: change the timing of alsa_sound_last_init()
Current alsa_sound_last_init() was called as __initcall().
So, on current ALSA, only devices that had been properly
registered at this point were shown.
So, it will show "No soundcards found" if driver requests
probe deferment. it's often misleading.
This patch delays the timing of alsa_sound_last_init()
as workaround.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviwed-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
While refactoring the mute-LED handling for HP laptops, I messed up
the polarity check in a wrong way. The red (or the mute-LED if any)
should appear in the muted state, corresponding to GPIO on.
Reported-by: Mikko Vinni <mmvinni@yahoo.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Paul Mundt [Wed, 18 Apr 2012 02:13:04 +0000 (19:13 -0700)]
ASoC: fsi: update for dmaengine prep_slave_sg fallout.
Leading up to the ->device_prep_slave_sg change in 185ecb5f4fd43911c35956d4cc7d94a1da30417f 'dmaengine: add context
parameter to prep_slave_sg and prep_dma_cyclic' a generic wrapper was
added in place to guard against the API change, though the fsi driver
wasn't updated in the process (presumably its dmaengine support hadn't
been merged yet at the time). This trivially switches over to the new
wrapper and gets it building again.
Signed-off-by: Paul Mundt <lethal@linux-sh.org> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA: Add definitions for CEA-861 Audio InfoFrames
Along with the IEC-60958 channel status word, CEA-861 Audio InfoFrames
are used in HDMI and DisplayPort to describe the parameters of the audio
stream. Hence, drivers for such devices may use these definitions to, for
instance, fill a CEA-861 data structure and pass it to a display driver
to configure an IP.
ASoC: core: Fix card RTD count for deferred probe.
Currently we increment the number of RTD's per card during the DAI link
bind. This can cause an incorrect RTD count when we cannot find a component
and defer the probe (and hence perform the DAI link bind for the card again).
Fix the count so that it is cleared before every card registration
and bind attempt.
Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ep->fill_max is a 1 bit flag, thus it has to be boolean.
sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params':
sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion
If cs42l73_get_mclkx_coeff() returns < 0 (which it can) in
sound/soc/codecs/cs42l73.c::cs42l73_set_mclk(), then we'll be using
the (negative) return value as array index on the very next line of
code - that's bad.
Catch the negative return value and propagate it to the caller (which
checks for it) and things are a bit more sane :-)
Signed-off-by: Jesper Juhl <jj@chaosbits.net> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel Mack [Thu, 12 Apr 2012 11:51:14 +0000 (13:51 +0200)]
ALSA: snd-usb: add support for implicit feedback
Implicit feedback is a streaming mode that does not rely on dedicated
sync endpoints but uses the information provided by record streams to
clock output streams. Now that the streaming logic is decoupled from the
PCM streams, this is easy to implement.
Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Daniel Mack [Thu, 12 Apr 2012 11:51:12 +0000 (13:51 +0200)]
ALSA: snd-usb: switch over to new endpoint streaming logic
With the previous commit that added the new streaming model, all
endpoint and streaming related code is now in endpoint.c, and pcm.c
only acts as a wrapper for handling the packet's payload.
Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Daniel Mack [Thu, 12 Apr 2012 11:51:11 +0000 (13:51 +0200)]
ALSA: snd-usb: implement new endpoint streaming model
This patch adds a new generic streaming logic for audio over USB.
It defines a model (snd_usb_endpoint) that handles everything that
is related to an USB endpoint and its streaming. There are functions to
activate and deactivate an endpoint (which call usb_set_interface()),
and to start and stop its URBs. It also has function pointers to be
called when data was received or is about to be sent, and pointer to
a sync slave (another snd_usb_endpoint) that is informed when data has
been received.
A snd_usb_endpoint knows about its state and implements a refcounting,
so only the first user will actually start the URBs and only the last
one to stop it will tear them down again.
With this sort of abstraction, the actual streaming is decoupled from
the pcm handling, which makes the "implicit feedback" mechanisms easy to
implement.
In order to split changes properly, this patch only adds the new
implementation but leaves the old one around, so the the driver doesn't
change its behaviour. The switch to actually use the new code is
submitted separately.
Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace).
In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the
'for (;;)' loop, if the 'badness' value returned from
fill_and_eval_dacs() is negative, then we'll return from the function
without freeing the memory we allocated for 'best_cfg', thus leaking.
Fix the leak by kfree()'ing the memory when badness is negative.
While I was there I also noticed some trailing whitespace in the
function that I removed (along with all other trailing whitespace in
the file) - it didn't seem worth-while to do that as two patches, so I
hope it's OK that I just did it all as one patch.
Mark Brown [Thu, 12 Apr 2012 16:29:36 +0000 (17:29 +0100)]
ASoC: dapm: Ensure power gets managed for line widgets
Line widgets had not been included in either the power up or power down
sequences so if a widget had an event associated with it that event would
never be run. Fix this minimally by adding them to the sequences, we
should probably be doing away with the specific widget types as they all
have the same priority anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines
A user reported that setting model=imac24 used to allow sound to work on their
Mac Pro 5,1 machine. Commit 5671087ffa "Move ALC885 macpro and imac24 models
to auto-parser" removed this model option. All Mac machines are now explicitly
handled with a quirk and the auto-parser. This adds a quirk for the device
found on the Mac Pro 5,1 machines.
This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559
[sorted the new entry in the ID number order by tiwai]
ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co
Add GPIO1 setup explicitly for Acer Aspire 493x & co.
This could be set by alc_auto_init_amp(), but it's safer to set it
more explicitly in the fixup table.
ALSA: hda/realtek - Add a few ALC882 model strings back
Since there are still many Acer models that might not be covered by
the current fixup table, let's add back a few typical model names so
that user can test the fixup without recompiling.
At the point of this error-handling code, both regions and the dma have
been allocated, so free it as done in previous and subsequent
error-handling code.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Julia Lawall [Mon, 9 Apr 2012 08:16:32 +0000 (10:16 +0200)]
sound: sound/oss/msnd_pinnacle.c: add vfrees
At the point of this error-handling code, HAVE_DSPCODEH may be undefined,
so free INITCODE and PERMCODE as done elsewhere. A jump and label are
introduced to avoid code duplication.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge tag 'asoc-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: fixes for 3.4
A bunch of driver-specific fixes and one generic fix for the new support
for platform DAPM contexts - we were picking the wrong default for the
idle_bias_off setting which was meaning we weren't actually achieving
any useful runtime PM on platform devices.
Michael Karcher [Fri, 6 Apr 2012 13:34:18 +0000 (15:34 +0200)]
ALSA: hda - Remove CD control from model=benq for CX20549
The ID used for detection of the BenQ R55E actually identifies the
Quanta TW3 ODM design, which is also used for the Gigabyte W551 laptop
series. Schematics on the internet clearly indicate that the "Port C"
(analog input connected to record source #4 and mixer input #4) is
unconnected.
Playing an audio CD through analog playback (using cdplay from cdtools)
produces no sound, even with the mixer input labelled "CD" enabled, and
the volume control in the CD drive set to maximum. This indicates the
connection is really not present.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Michael Karcher [Fri, 6 Apr 2012 13:34:17 +0000 (15:34 +0200)]
ALSA: hda - fix record volume controls of CX20459 ("Venice")
The "input converter" widget of the CX20459 has only one input amplifier,
expose that one as "Capture Volume/Capture Switch". The actual record
source selection is already exposed through the separately installed
input mux.
Signed-off-by: Michael Karcher <kernel@mkarcher.dialup.fu-berlin.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stephen Warren [Fri, 6 Apr 2012 05:11:16 +0000 (23:11 -0600)]
ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS
Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev->id" changed
the prototype of tegra_i2s_debug_add, but didn't update the dummy inline
used when !CONFIG_DEBUG_FS. Fix that.
Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: <stable@vger.kernel.org> # 3.3
Stephen Warren [Thu, 5 Apr 2012 18:28:01 +0000 (12:28 -0600)]
ASoC: set idle_bias_off=1 for all platform DAPM contexts
The ASoC core currently defaults to using STANDBY rather than OFF for
idle ASoC platform devices, which causes a permanent pm_runtime_get() on
them. This keeps the device active unnecessarily. This can be especially
problematic when the ASoC platform device and DAI device are the same
device.
The distinction between OFF and STANDBY is likely not relevant for ASoC
platform drivers, since they aren't analog devices. So, solve this issue
by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this
turns out to be a problem, this value could be sourced from the
snd_soc_platform_driver, similarly to soc_probe_codec().
Note: Prior to this change, this caused a large (10) runtime_active count
for the Tegra I2S controller even when not in use, and a leak in that
value as streams were started and stopped. This change probably hides a
bug.
Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA: hda - Fix internal mic for Lenovo Ideapad U300s
The internal mic input is phase inverted on one channel.
To avoid people in userspace summing the channels together
and get zero result, use a separate mixer control for the
inverted channel.
BugLink: https://bugs.launchpad.net/bugs/903853 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the ssi port numbers in sysfs to fix this problem.
Reported-by: Joan Carles <joancarles@fqingenieria.es> Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute.
But current settings didn't care +1 step for mute.
This patch adds it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
Mark Brown [Wed, 4 Apr 2012 11:06:24 +0000 (12:06 +0100)]
MAINTAINERS: Don't list everyone working on Wolfson drivers
Rather than listing every single person who works on the drivers include
the mailing list where they can all be found. Leave myself as a human
contact.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Tue, 3 Apr 2012 06:45:43 +0000 (09:45 +0300)]
MAINTAINERS: Add missing ASoC OMAP co-maintainer
Peter Ujfalusi has been co-maintaining sound/soc/omap/ for years but
was missing from this MAINTAINERS entry.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Martin Jansa [Mon, 2 Apr 2012 08:24:08 +0000 (10:24 +0200)]
ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro
* fixes
sound/soc/pxa/pxa2xx-i2s.c:86:2: error: implicit declaration of function 'IOMEM' [-Werror=implicit-function-declaration]
sound/soc/pxa/pxa2xx-i2s.c:86:2: error: initializer element is not constant
after 23019a733bb83c8499f192fb428b7e6e81c95a34 removed IOMEM
definition from arch/arm/mach-pxa/include/mach/hardware.h
Signed-off-by: Martin Jansa <Martin.Jansa@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Linus Torvalds [Sat, 31 Mar 2012 22:11:39 +0000 (15:11 -0700)]
Merge branch 's3-for-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/amit/virtio-console
Pull virtio S3 support patches from Amit Shah:
"Turns out S3 is not different from S4 for virtio devices: the device
is assumed to be reset, so the host and guest state are to be assumed
to be out of sync upon resume. We handle the S4 case with exactly the
same scenario, so just point the suspend/resume routines to the
freeze/restore ones.
Once that is done, we also use the PM API's macro to initialise the
sleep functions.
A couple of cleanups are included: there's no need for special thaw
processing in the balloon driver, so that's addressed in patches 1 and
2.
Testing: both S3 and S4 support have been tested using these patches
using a similar method used earlier during S4 patch development: a
guest is started with virtio-blk as the only disk, a virtio network
card, a virtio-serial port and a virtio balloon device. Ping from
guest to host, dd /dev/zero to a file on the disk, and IO from the
host on the virtio-serial port, all at once, while exercising S4 and
S3 (separately) were tested. They all continue to work fine after
resume. virtio balloon values too were tested by inflating and
deflating the balloon."
Pulling from Amit, since Rusty is off getting married (and presumably
shaving people).
* 's3-for-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/amit/virtio-console:
virtio-pci: switch to PM ops macro to initialise PM functions
virtio-pci: S3 support
virtio-pci: drop restore_common()
virtio: drop thaw PM operation
virtio: balloon: Allow stats update after restore from S4