Commit Graph

9965 Commits

Author SHA1 Message Date
Clemens Ladisch f3f7c1837f ALSA: isight: fix locking
Lockdep complains about conflicts between isight->mutex,
ALSA's register_mutex, mm->mmap_sem, and pcm->open_mutex.

This can be fixed by moving the calls to isight_pcm_abort(),
snd_card_disconnect(), and fw_iso_resources_update() out of
isight->mutex.  These functions are designed to be called
asynchronously; the mutex needs to protect only the device
streaming state modified by isight_start/stop_streaming().

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 14:53:09 +02:00
Clemens Ladisch 3cabffd72c ALSA: isight: remove experimental status
Experiments have shown this driver to work now.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 14:53:09 +02:00
Clemens Ladisch aee7040018 ALSA: isight: fix hang when unplugging a running device
When aborting a PCM stream, the xrun is signaled only if the stream is
running.  When disconnecting a PCM stream, calling snd_card_disconnect()
too early would change the stream into a non-running state and thus
prevent the xrun from being noticed by user space.

To prevent this, move the snd_card_disconnect() call after the xrun.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 14:53:08 +02:00
Stefan Richter ac34dad26e ALSA: isight: wrap up register accesses
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
[cl: removed superfluous variable]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 14:52:54 +02:00
Stefan Richter 8839eedafd ALSA: isight: add AudioEnable register write
which is needed to get the iSight to talk.

Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 14:51:20 +02:00
Clemens Ladisch f2934cd499 ALSA: isight: fix divide error when queueing packets
Set the .header_size field when queueing packets to avoid a division by
zero.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 14:51:16 +02:00
Clemens Ladisch 898732d1f1 ALSA: isight: fix packet requeueing
After handling a received packet, we want to resubmit the same packet,
so do not increase the packet index too early.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 14:51:13 +02:00
Clemens Ladisch 03c29680d4 ALSA: isight: fix isight_pcm_abort() crashes
Fix crashes in isight_pcm_abort() that happen when the driver tries to
access isight->pcm->runtime which does not exist when the device is not
open.  Introduce a new field pcm_active to track this state.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 14:51:09 +02:00
Clemens Ladisch 3a691b28a0 ALSA: add Apple iSight microphone driver
This adds an experimental driver for the front and rear microphones of
the Apple iSight web camera.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 14:51:05 +02:00
Ondrej Zary d7ba858a7f ALSA: fm801: implement TEA575x tuner autodetection
Autodetect TEA575x tuner connection type during init. This allows tuner to
work out-of-the box.

tea575x_tuner module parameter remains functional to force tuner type.

Tested with SF256-PCP and SF64-PCR.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-11 10:52:24 +02:00
Joe Perches 4ef7e71444 pcmcia: Make struct pcmcia_device_id const, sound drivers edition
Make declarations of struct pcmcia_device_id const.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
2011-05-11 10:48:57 +02:00
Clemens Ladisch 13882a82ee firewire: optimize iso queueing by setting wake only after the last packet
When queueing iso packets, the run time is dominated by the two
MMIO accesses that set the DMA context's wake bit.  Because most
drivers submit packets in batches, we can save much time by
removing all but the last wakeup.

The internal kernel API is changed to require a call to
fw_iso_context_queue_flush() after a batch of queued packets.
The user space API does not change, so one call to
FW_CDEV_IOC_QUEUE_ISO must specify multiple packets to take
advantage of this optimization.

In my measurements, this patch reduces the time needed to queue
fifty skip packets from userspace to one sixth on a 2.5 GHz CPU,
or to one third at 800 MHz.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
2011-05-10 22:53:45 +02:00
Stefan Richter f30e6d3e41 firewire: octlet AT payloads can be stack-allocated
We do not need slab allocations anymore in order to satisfy
streaming DMA mapping constraints, thanks to commit da28947e7e
"firewire: ohci: avoid separate DMA mapping for small AT payloads".

(Besides, the slab-allocated buffers that firewire-core, firewire-sbp2,
and firedtv used to provide for 8-byte write and lock requests were
still not fully portable since they crossed cacheline boundaries or
shared a cacheline with unrelated CPU-accessed data.  snd-firewire-lib
got this aspect right by using an extra kmalloc/ kfree just for the
8-byte transaction buffer.)

This change replaces kmalloc'ed lock transaction scratch buffers in
firewire-core, firedtv, and snd-firewire-lib by local stack allocations.
Perhaps the most notable result of the change is simpler locking because
there is no need to serialize usages of preallocated per-device buffers
anymore.  Also, allocations and deallocations are simpler.

Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
2011-05-10 22:53:44 +02:00
Mark Brown 0f3c6af921 Merge branch 'for-2.6.39' into for-2.6.40 2011-05-10 15:58:17 +02:00
Stephen Warren 61bf35b9a3 ASoC: WM8903: Fix Digital Capture Volume range
Increase the range of the Digital Capture Volume control to be 120 steps.
Each step is 0.75dB, and the range starts at -72dB, giving a max setting
of 18dB, which matches the latest datasheet, to the precision of the step
size.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-10 11:48:33 +02:00
Ondrej Zary 938a1566b1 ALSA: fm801: convert TEA575x support to new interface
Use common functions to access TEA575x tuner - remove original read/write
functions and provide new pin manipulation functions instead.

Also convert the original triple implementation to a simple GPIO pin map.

Tested with SF256-PCP and SF64-PCR (added the GPIO pin for MO/ST signal
for them).
SF256-PCS untested (pin for MO/ST signal is a guess).

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-10 09:31:45 +02:00
Ondrej Zary 72587173cc ALSA: es1968: convert TEA575x support to new interface
Use common functions to access TEA575x tuner - remove original read/write
functions and provide new pin manipulation functions instead.

Tested with SF64-PCE2 card.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-10 09:31:29 +02:00
Ondrej Zary 14219d0659 ALSA: tea575x: unify read/write functions
Implement generic read/write functions to access TEA575x tuners. They're now
implemented 4 times (once in es1968 and 3 times in fm801).
This also allows mute to work on all cards.
Also improve tuner detection/initialization.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-10 09:29:42 +02:00
Takashi Iwai 1209842af4 Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2011-05-10 09:24:50 +02:00
Takashi Iwai f0a2b0cb71 Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2011-05-10 09:20:19 +02:00
Linus Torvalds 047ec4b5de Merge branch 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ASoC: Fix CODEC DAI names for Goni
  ASoC: Fix CODEC name in Goni
  davinci-mcasp: fix _CBM_CFS pin directions
  davinci-mcasp: fix _CBM_CFS hw_params
  davinci-mcasp: use bitfield definitions for PDIR
  ASoC: davinci-mcasp: correct tdm_slots limit
2011-05-09 09:13:10 -07:00
Lars-Peter Clausen f3eee00da3 ASoC: SSM2602: Provide dB ranges for the volume controls
Also fix the maximum value for the capture volume control.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-09 01:45:26 +02:00
Lars-Peter Clausen 2a43801a76 ASoC: SSM2602: Model power supply for the digital core as a DAPM widget
Model the power supply for the digital core as a DAPM_SUPPLY widget. This allows
to cleanup the code a bit.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-09 01:45:17 +02:00
Lars-Peter Clausen 7dcf2760bf ASoC: SSM2602: Add entry for the ssm2603 to the device id table
The SSM2603 is mostly register compatible with the SSM2602 and can be supported
by the current driver without any changes.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-09 01:45:09 +02:00
Lars-Peter Clausen b1f7b2b56b ASoC: SSM2602: Add SSM2604 support
The SSM2604 is basically a lightweight variant of the SSM2602 with a compatible
register layout. Thus we can easily support both devices by the same driver,
by providing a slightly set of controls, widgets and routes.

Compared to the SSM2602 the SSM2604 has no microphone input and no headphone
output.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-09 01:45:01 +02:00
Lars-Peter Clausen f6c1f2d5e5 ASoC: SSM2602: Do not power the codec up in probe
It is not required to have the codec powered at this stage and DAPM will power
the ADC and DAC down again after probe has run anyway.
Thus we avoid some unnecessary writes by this change.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-09 01:44:54 +02:00
Lars-Peter Clausen 7164bdb643 ASoC: SSM2602: Fix default register cache
Some of the values in the default register cache did not represent the codecs
state after reset. This patch fixes it.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-09 01:44:45 +02:00
Mark Brown afd8f37c80 Merge branch 'for-2.6.39' into for-2.6.40 2011-05-08 15:33:41 +01:00
Marek Belisko bf707de21f ASoC: UDA134x: Remove POWER_OFF_ON_STANDBY define.
Define POWER_OFF_ON_STANDBY cause trobles when trying to get some
sound from codec because code for bias setup was not compiled
(define wasn't defined). This define was removed in commit:
cc3202f5 but again introduced by commit: f0fba2ad1 which then
completely break codec functionality so remove it again.

Signed-off-by: Marek Belisko <marek.belisko@open-nandra.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-05-08 15:27:48 +01:00
Lars-Peter Clausen 5663940e2a ASoC: SSM2602: Remove unused struct and define
Those are leftovers from a pre-multicomponent era.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-08 14:44:11 +01:00
Lars-Peter Clausen ffd13c39c7 ASoC: SSM2602: Remove duplicate control
There are currently two controls which allow selecting the capture source, one
as a normal control, the other as part of a DAPM_MUX widget.
Remove the normal control.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-08 14:44:11 +01:00
Lars-Peter Clausen 0b4cd2e01c ASoC: SSM2602: Cleanup coeff handling
Drop unused field from the coeff struct, precalculate the srate register at
compile-time and cleanup up the naming.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-08 14:44:05 +01:00
Mark Brown 5e8bc53b7c Merge branch 'for-2.6.39' into for-2.6.40 2011-05-08 14:43:18 +01:00
Lars-Peter Clausen 8fc63fe941 ASoC: SSM2602: Fix reg_cache_size
reg_cache_size is supposed to be the number of elements in the register cache,
not the size in bytes.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-08 14:42:21 +01:00
Lars-Peter Clausen 36c90ab33f ASoC: SSM2602: Fix 'Mic Boost2' control
The 'Mic Boost2' control's shift was off by one and thus was not working.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-05-08 14:42:15 +01:00
Lars-Peter Clausen 04b894553f ASoC: SSM2602: Properly annotate i2c probe and remove functions
Annotate the i2c probe and remove functions with __devinit and __devexit.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-08 14:41:34 +01:00
Dimitris Papastamos 64d2706975 ASoC: soc-cache: Allow codec->cache_bypass to be used with snd_soc_hw_bulk_write_raw()
If we specifically want to write a block of data to the hw bypassing the
cache, then allow this to happen inside snd_soc_hw_bulk_write_raw().

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-08 14:38:31 +01:00
Lars-Peter Clausen 77530150fb ASoC: Create codec DAPM widgets before calling the codecs probe function
This allows to create DAPM routes depending on those widgets in the
codecs probe function.  This is helpful when supporting similar codecs
with minor differences in the DAPM routing with the same driver.

Something similar has already been done for cards in commit
a841ebb9 (ASoC: Create card DAPM widgets early so they can be used in
callbacks).

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-08 14:38:08 +01:00
Randy Dunlap f428c94c84 ALSA: lola - fix lola build
sound/pci/lola/Makefile was trying to build lola modules even
when PCI and SND_LOLA were not enabled, causing build errors:

ERROR: "snd_pcm_hw_constraint_step" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_period_elapsed" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_dma_alloc_pages" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_hw_constraint_integer" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_sgbuf_ops_page" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_set_ops" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_lib_free_pages" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_lib_ioctl" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_lib_malloc_pages" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_sgbuf_get_chunk_size" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_dma_free_pages" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_lib_preallocate_pages_for_all" [sound/pci/lola/snd-lola.ko] undefined!
ERROR: "snd_pcm_new" [sound/pci/lola/snd-lola.ko] undefined!

Fix the Makefile to build only when CONFIG_SND_LOLA is enabled.

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-04 19:55:13 +02:00
Takashi Iwai 447ee6a7cb ALSA: hda - Use position_fix=3 as default for AMD chipsets
AMD chipsets often behave pretty badly regarding the DMA position
reporting.  It results in the bad quality audio recording.
Using position_fix=3 works well in general for them, so let's enable
it as default for AMD.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-04 18:28:50 +02:00
Mark Brown 20ed0938bf Merge branch 'for-2.6.39' into for-2.6.40 2011-05-03 23:30:36 +01:00
xingchao 9ab88434e8 ASoC: sst_platform: add hw_free callback to fix resource leak
Signed-off-by: xingchao <xingchao.wang@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 23:29:54 +01:00
Mark Brown e1a0206608 ASoC: Remove outdated FIXME from WM8915
Actually the current code is perfectly sensible given the hardware.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-05-03 23:29:28 +01:00
Mark Brown abc9d5aa08 ASoC: Use shared controls for input signal path in WM8915
Gives finer grained power management.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-05-03 23:29:08 +01:00
Mark Brown ed77cc122a ASoC: Don't crash on PM operations
The move over to exposing snd_soc_register_card() let the initialisation
of the driver data we use to find the card in PM operations go AWOL. Fix
this by setting the driver data when we register the card.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-05-03 23:28:04 +01:00
Stephen Warren af46800b9a ASoC: Implement mux control sharing
Control sharing is enabled when two widgets include pointers to the
same kcontrol_new in their definition. Specifically:

static const struct snd_kcontrol_new adcinput_mux =
	SOC_DAPM_ENUM("ADC Input", adcinput_enum);

static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = {
  SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux),
  SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux),
};

This is useful when a single register bit or field affects multiple
muxes at once. The common case is to have separate control bits or
fields for each mux (channel). An alternative way of looking at this
is that the mux is a stereo (or even n-channel) mux, rather than
independant mono muxes.

Without this change, a separate kcontrol will be created for each
DAPM_MUX. This has the following disadvantages:

* Confuses the user/programmer with redundant controls that don't
  map to separate hardware.

* When one of the controls is changed, ASoC fails to update the DAPM
  logic for paths solely affected by the other controls impacted by
  the same register bits. This causes some paths not to be correctly
  powered up or down. Prior to this change, to work around this, the
  user or programmer had to manually toggle all duplicate controls away
  from the intended setting, and then back to it.

Control sharing implies that the control is named based on the
kcontrol_new itself, not any of the widgets that are affected by it.

Control sharing is implemented by: When creating kcontrols, if a
kcontrol does not yet exist for a particular kcontrol_new, then a new
kcontrol is created with a list of widgets containing just a single
entry. This is the normal case. However, if a kcontrol does already
exists for the given kcontrol_new, the current widget is simply added
to that kcontrol's list of affected widgets.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 19:29:15 +01:00
Stephen Warren fafd2176f7 ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol
A future change will allow multiple widgets to be affected by the same
control. For example, a single register bit that controls separate muxes
in both the L and R audio paths.

This change updates the code that handles relevant controls to be able
to iterate over a list of affected widgets. Note that only the put
functions need significant modification to implement the iteration; the
get functions do not need to iterate, nor unify the results, since all
affected widgets reference the same kcontrol.

When creating the list of widgets, always create a 1-sized list, since
the control sharing is not implemented in this change.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 19:29:05 +01:00
Stephen Warren fad598887d ASoC: Add w->kcontrols, and populate it
Future changes will need reference to the kcontrol created for a given
kcontrol_new. Store the created kcontrol values now.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 19:28:57 +01:00
Stephen Warren 82cfecdc03 ASoC: s/w->kcontrols/w->kcontrol_news/g
A future change will modify struct snd_soc_dapm_widget to store the
actual kcontrol pointers for each kcontrol_new in a field named
kcontrols. Rename the existing kcontrols field to enable this.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 19:28:47 +01:00
Mark Brown 65f7e32520 Merge branch 'for-2.6.39' into for-2.6.40 2011-05-03 19:07:45 +01:00
Lars-Peter Clausen 005967a1df ASoC: JZ4740: Fix i2s shutdown
The i2s shutdown callback has the check whether it should be disabled reversed.
Currently it is disabled if another stream is still active, but kept enabled if
the last stream is closed. This patch fixes it.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 18:48:24 +01:00
Lars-Peter Clausen 6c45e12656 ASoC: Remove DAPM debugfs entries before freeing widgets
Remove the DAPM debugfs entries before freeing the context's widgets, otherwise a
use after free situation might occur.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 18:43:52 +01:00
Lars-Peter Clausen d5d1e0bef4 ASoC: Move DAPM widget debugfs entry creation to snd_soc_dapm_new_widgets
Currently debugfs entries for a DAPM widgets are only added in
snd_soc_dapm_debugfs_init. If a widget is added later (for example in the
dai_link's probe callback) it will not show up in debugfs.
This patch moves the creation of the widget debugfs entry to
snd_soc_dapm_new_widgets where it will be added after the widget has been
properly instantiated.

As a side-effect this will also reduce the number of times the DAPM widget list
is iterated during a card's instantiation.

Since it is possible that snd_soc_dapm_new_widgets is invoked form the codecs or
cards probe callbacks, the creation of the debugfs dapm directory has to be
moved before these are called.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 18:43:44 +01:00
Lars-Peter Clausen 8eecaf6244 ASoC: Move DAPM debugfs directory creation to snd_soc_dapm_debugfs_init
Move the creation of the DAPM debugfs directory to snd_soc_dapm_debugfs_init
instead of having the same duplicated code in both codec and card DAPM setup.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 18:43:32 +01:00
Lars-Peter Clausen 0aaae527c7 ASoC: Free the card's DAPM context
Free the card's DAPM context when the card is removed.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 18:43:15 +01:00
Mike Rapoport 1307394afd ASoC: tegra: TrimSlice machine support
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Acked-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-03 18:42:44 +01:00
Takashi Iwai f2e0192519 ALSA: lola - Yet another linux/delay.h inclusion
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 18:48:29 +02:00
Takashi Iwai f044785d0a ALSA: lola - Add missing inclusion of linux/delay.h
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 18:21:01 +02:00
Takashi Iwai fe4af1b55e ALSA: lola - Implement polling_mode like hd-audio
Also protect the call of lola_update_rirb() with spinlock.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 17:06:53 +02:00
Takashi Iwai 2db3002029 ALSA: lola - Rename to Digital SRC Capture Switch
Renamed to Digial SRC Capture Switch for more correct representation.
Also fixed analog volume control on Lola161611 and lola881.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 17:05:08 +02:00
Takashi Iwai c7aad3c317 ALSA: lola - Add sync in loop implementation
For assuring the synchronized state with the pause operation,
loop over the all linked streams and waits until all get ready
in a loop.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 17:02:35 +02:00
Takashi Iwai 7e79f22676 ALSA: lola - Add SRC refcounting
Added the refcounting for the exclusive SRC control.
Also, fixed the possible stall after PCM pause operations.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:59:27 +02:00
Takashi Iwai 8bd172dc96 ALSA: lola - Allow granularity changes
Add some sanity checks.
Change PCM parameters appropriately per granularity.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:51:56 +02:00
Takashi Iwai 972505ccde ALSA: lola - Use SG-buffer
Completely switch to SG-buffer now, as it's working stably.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:50:51 +02:00
Takashi Iwai fe3d393eda ALSA: lola - Add Lola-specific module options
Added granularity and sample_rate_min module options.

The former controls the h/w access granularity.  As default, it's set
to the max value 32.

The latter controls the minimum sample rate in Hz, as default 16000.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:48:59 +02:00
Takashi Iwai 0f8f56c959 ALSA: lola - Fix PCM stalls
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:47:03 +02:00
Takashi Iwai 333ff3971f ALSA: lola - Use a single BDL
Use a single BDL for both buffers instead of allocating for each.

Also a few tune-up to avoid the stream stalls in the PCM code and
the prelimianry work for SG-buffer support are added, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:41:02 +02:00
Takashi Iwai a426c78723 ALSA: lola - Suppress the debug print
Use snd_printdd() for less important debug messages.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:40:53 +02:00
Takashi Iwai c772bbe69a ALSA: lola - Changes in proc file
The codec proc file becomes a read only that shows the codec widgets
in a text form.  A new proc file, codec_rw, is introduced instead for
accessing the Lola verb directly by reading and writing to it.

Also, regs proc file shows the contents of DSD, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:40:52 +02:00
Takashi Iwai 1c5d7b312f ALSA: lola - Make SRC helper global
Make lola_sample_rate_convert() global so that it can be accessed from
other files.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:40:52 +02:00
Takashi Iwai d43f3010b8 ALSA: Add the driver for Digigram Lola PCI-e boards
Added a new driver for supporting Digigram Lola PCI-e boards.

Lola has a similar h/w design like HD-audio but with extended verbs.
Thus the driver is written similarly like HD-audio driver in the bus
part.  The codec part is rather written in a fixed way specific to the
Lola board because of the verb incompatibility.

The driver provides basic PCM, supporting multi-streams and mixing.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 16:31:05 +02:00
Raymond Yau ce85c9ac8d ALSA: hda - fix NULL-dereference in patch_realtek
Fix NULL-dereference when try to use alt_playback since those codecs
which support multistreaming playback usually have more than 1 adc but
the driver should create alt_capture when spec->stream_analog_alt_capture
is also defined.

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 10:32:04 +02:00
Linus Torvalds c7bcecbe98 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix Realtek's chained fixup checks
  Revert "ALSA: hda - Fix pin-config of Gigabyte mobo"
  ALSA: HDA: Fix automute for Gateway NV79
  ALSA: hda: add beep quirk for Realtek 0x1043:831a
  ALSA: usb-audio - Terratec Aureon 7.1 USB ID as C-Media cm6206 quirks
  ALSA: hda - VIA: Fix notify_aa_path_ctls() invalid issue.
  ALSA - au88x0 - Add buffer bytes constraints
2011-05-02 09:07:27 -07:00
Takashi Iwai 20ec8b2463 Merge branch 'fix/hda' into topic/hda 2011-05-02 13:58:23 +02:00
Takashi Iwai 24af2b1cc4 ALSA: hda - Fix Realtek's chained fixup checks
The check of chained fixup list entry was done against the wrong element.
A stupid mistake during refactoring.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 13:55:36 +02:00
Takashi Iwai 90dd48a1a9 ALSA: hda - Constify fixup and other array data in patch_via.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:38:19 +02:00
Takashi Iwai 2b63536f0c ALSA: hda - Constify fixup and other array data in patch_sigmatel.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:33:43 +02:00
Takashi Iwai 9cf0aa9eba ALSA: hda - Constify fixup and other array data in patch_si3054.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:22:39 +02:00
Takashi Iwai fb79e1e0a2 ALSA: hda - Constify fixup and other array data in patch_hdmi.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:20:31 +02:00
Takashi Iwai 34cbe3a6fa ALSA: hda - Constify fixup and other array data in patch_conexant.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:20:31 +02:00
Takashi Iwai c42d47829a ALSA: hda - Constify fixup and other array data in patch_cirrus.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:20:30 +02:00
Takashi Iwai 728850a7f2 ALSA: hda - Constify fixup and other array data in patch_ca0110.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:20:29 +02:00
Takashi Iwai 779d065983 ALSA: hda - Constify fixup and other array data in patch_cmedia.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:20:28 +02:00
Takashi Iwai 498f5b175b ALSA: hda - Constify fixup and other array data in patch_analog.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:20:27 +02:00
Takashi Iwai 4c6d72d138 ALSA: hda - Constify fixup and other array data in patch_realtek.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:20:21 +02:00
Takashi Iwai dda144103c ALSA: hda - Constify some API function arguments
Also fixed the assignment of multiout.dac_nids to satisfy const.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 12:07:48 +02:00
Takashi Iwai a9111321f2 ALSA: hda - Constify fixup and other array data in patch_realtek.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 11:30:18 +02:00
Takashi Iwai 031024eea8 ALSA: hda - Constify some API function arguments
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 11:29:30 +02:00
Takashi Iwai a3ea8e8f24 Merge branch 'fix/hda' into topic/hda 2011-05-02 10:41:40 +02:00
Takashi Iwai ebb47241ea Revert "ALSA: hda - Fix pin-config of Gigabyte mobo"
This reverts commit c6b358748e.

It turned out that there are different pin configurations for this
PCI SSID, including multi-channel modes.  And more proper fix for
allowing line-out mutes will come up in 2.6.40 tree, so we won't need
this fixup any more there.

Reported-by: Andrew Clayton <andrew@digital-domain.net>
Reported-by: Emmanuel Benisty <benisty.e@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-02 10:37:29 +02:00
David Henningsson 94024cd1ae ALSA: HDA: Fix automute for Gateway NV79
The PCI SSID is 1025:031c and the codec SSID is 1025:031d,
so the driver mistakes this for a SKU value, but looking at
the numbers, this is obviously wrong.

Cc: stable@kernel.org (2.6.38+)
BugLink: http://bugs.launchpad.net/bugs/761861
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-29 14:19:31 +02:00
Takashi Iwai c2de187e5b ALSA: hda - Show the line-out type in snd_hda_parse_pin_def_config()
Helpful for debugging.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-29 13:01:33 +02:00
Daniel Cordero a7e985e18f ALSA: hda: add beep quirk for Realtek 0x1043:831a
PC Beep was not being reported as enabled on my EeePC 901:
        SKU: enable_pcbeep=0x0

Signed-off-by: Daniel Cordero <danielcordero@lavabit.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-29 08:18:06 +02:00
Wolfgang Breyha 8129e79ed7 ALSA: usb-audio - Terratec Aureon 7.1 USB ID as C-Media cm6206 quirks
This patch adds support for the Terratec Aureon 7.1 USB which uses a
C-Media cm6206 and needs all the quirks already found in the past.

Signed-off-by: Wolfgang Breyha <wbreyha@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 18:22:41 +02:00
Takashi Iwai ae8a60a598 ALSA: hda - Add Auto-Mute Mode enum for two-output cases
The Auto-Mute Mode control is useful even when only two outputs
(e.g. HP and speaker) are available.  Then user can enable/disable
the auto-mute behavior on the fly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 18:09:52 +02:00
Takashi Iwai 1daf5f46c6 ALSA: hda - More line-out auto-mute support for Realtek
Not only supporting the line-out automute as additional feature
to the existing headphone automute, now the headphone jack can
mute the line-out alone even without the speaker outs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 17:57:46 +02:00
Takashi Iwai 1a1455de10 ALSA: hda - Add support for Line-Out automute to Realtek auto-parser
By popular demands, I add the functionality to mute / unmute the
line-out jacks per the headphone plug / unplug.  For achieving this
and keeping the compatibility with the old behavior, the new mixer
enum "Auto-Mute Mode" is added.  With this, user can control the
auto-mute behavior either disabled, speaker-only or lineout+speaker.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 17:55:53 +02:00
Takashi Iwai 0f0f391c73 ALSA: hda - More reduction of redundant automute codes in Realtek parser
Removed the redundant codes by replacing with the common helper
functions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 16:26:24 +02:00
Takashi Iwai e9427969f5 ALSA: hda - Consolidate auto-mute with master-switch for Realtek
Yet another consolidation of auto-mute functions for the devices
controlling the output muts together with the master mixer switch,
typically found for ALC262 machines.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 15:46:07 +02:00
Takashi Iwai e6a5e1b709 ALSA: hda - Add support of line-out automute for Realtek
Add the common helper function and flags to support the auto-mute
per line-out jack detection, and also the mute of line-out jacks.

A few model-specific implementations are replaced with the common
helpers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 14:45:56 +02:00
Takashi Iwai 3b8510ce97 ALSA: hda - Add common automute support for mxier-amp on/off for Reatek
Some models do mute on/off the connected mixer widget for the automatic
muting, instead of controlling the pin widget itself.  This patch adds
the implementation of such type of auto-mute in the common helper
function, and reduces the redundant codes for each model preset.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 14:45:50 +02:00
Takashi Iwai d922b51dab ALSA: hda - Consolidate default automute functions for Realtek
There are two entry points for the headphone automute functions for
Realtek, alc_automute_amp() and alc_automute_pin().  These call the
same function in the end, so we can basically consolidate these
with a flag in spec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 14:45:19 +02:00
Mark Brown 9b1b937c77 ASoC: Don't specify the DMA driver for Goni baseband link
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-28 12:11:06 +01:00
Mark Brown 3784019af3 ASoC: Don't specify the DMA driver for OpenMoko baseband link
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-28 12:11:00 +01:00
Mark Brown dd4028c59e Merge branch 'for-2.6.39' into for-2.6.40 2011-04-28 12:10:25 +01:00
Mark Brown 69b91bc155 ASoC: Fix CODEC DAI names for Goni
Immediately after sending the last fix I realised that the CODEC DAI names
also don't correspond to the WM8994 driver. Update the DAI names to match.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-28 12:09:53 +01:00
Mark Brown 1270b01f75 ASoC: Fix CODEC name in Goni
This was typoed at some point in the multi-component merge, though the
driver was added along with that.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-28 12:09:41 +01:00
Mark Brown fb257897bf ASoC: Work around allmodconfig failure
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-28 12:09:06 +01:00
Lydia Wang 525566cb60 ALSA: hda - VIA: Fix notify_aa_path_ctls() invalid issue.
In notify_aa_path_ctls(), adds 'rear mic' item and confirms the A-A
path control existing before notifying card that the A-A path volume
is muted if smart5.1 is enabled.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 11:35:18 +02:00
Takashi Iwai 59bb7f0eeb ALSA: usb-audio - Don't expose broken dB ranges
Some crappy USB-audio devices give broken dB ranges, e.g. both min and max
are 0dB.  This confuses the volume control that prefers dB expression such
as alsactl or PulseAudio.  In such a case, it's much better not to expose
the broken dB information.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 09:58:43 +02:00
Mark Brown 6be449e53d ASoC: Implement WM8962 ADC high pass filter configuration
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-27 22:33:13 +01:00
Lars-Peter Clausen 91a5fca4b1 ASoC: Add dapm_find_widget helper
This patch adds a helper function for searching DAPM widgets by name.
This allows to streamline functions which operate on widgets by name.
It also allows to get rid of copy'n'pasted code which was added to fallback to
widgets from other contexts if the widget was not found in the current context.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-27 22:33:13 +01:00
Mark Brown b864a8c9dd ASoC: Don't specify the DMA driver for Speyside baseband link
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-27 22:33:12 +01:00
Mark Brown 848dd8beef ASoC: Add more natural support for no-DMA DAIs
Since we can now support multiple platforms allow machines to not specify
a platform in a DAI link. Since the rest of the code requires that we have
a struct device for all objects we do this by substituting in a dummy
device that we register automatically.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-27 22:33:11 +01:00
Mark Brown 8842c72afe ASoC: Allow platform drivers to have no ops structure
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-27 22:10:55 +01:00
Raymond Yau 54a96dadaa ALSA - au88x0 - Add buffer bytes constraints
This allow application such as gstreamer and wine which use
snd_pcm_hw_params_set_buffer_time_near() won't fail any more
since sound chips require special containt power 2 period bytes

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-27 17:00:00 +02:00
Takashi Iwai ce764ab22e ALSA: hda - Add channel-mode support to Realtek auto-parser
This patch adds the support of "Channel Mode" enum control to Realtek
auto-parser.  When line-in or mic-in jacks are capable to output and
free DACs are available, the driver allows to switch to multi-channel
mode via "Channel Mode" enum switch, as already implemented in some
preset cases.

Not implemented in all Realtek codecs.  Currently, ALC880, 882, 861,
662 and the compatible codecs are supported.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-27 16:39:00 +02:00
Takashi Iwai 604401a92c ALSA: hda - Minor update for alc662-parser functions
Allow alc662_dac_to_mix() and alc662_look_for_dac() to parse
down the selector widget that is found in ALC880-type codecs,
and rename them to alc_auto_*() accordingly.
This is for the next coming multi-io extensions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-27 15:46:40 +02:00
Lydia Wang cb34c207af ALSA: hda - VIA: Fix Smart5.1 isn't useful for 6 audio jacks motherboard.
For some motherboards with 5 or 6 audio jacks which had six or eight multiple
channels output, smart5.1 item is no useful and should be removed.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-27 11:55:23 +02:00
Lucas De Marchi e9c549998d Revert wrong fixes for common misspellings
These changes were incorrectly fixed by codespell. They were now
manually corrected.

Signed-off-by: Lucas De Marchi <lucas.demarchi@profusion.mobi>
2011-04-26 23:31:11 -07:00
Takashi Iwai d507cd668a ALSA: hda - Enable sync_write workaround for AMD generically
The workaround for AMD chipset via sync_write flag seems needed for
machines with Realtek codecs.  So, it's better to activate it
generically in hda_intel.c from the beginning.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-26 15:33:43 +02:00
Takashi Iwai 0da2692256 ALSA: hda - Move EAPD power-down into shutup callback for AD codecs
EAPD power-down should be called also for normal shutup cases.
Let's move to there.   This also fixes the compile warnings when
CONFIG_PM isn't set automatically.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-26 15:18:33 +02:00
Takashi Iwai 31d44b57c5 Merge branch 'fix/hda' into topic/hda 2011-04-26 15:05:39 +02:00
Mark Brown 5357e8f505 ASoC: Don't warn if the WM8962 SYSCLK FLL setting doesn't match reality
When bringing up audio low power modes boards may configure SYSCLK before
they actually start the FLL as we do much of the clocking setup prior to
the power up sequence.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-26 11:49:17 +01:00
Mark Brown e47ac37c01 ASoC: Implement WM8962 DMIC support
DMIC support is automatically disabled when none of the GPIOs are set up
to bring out the DMICCLK and DMICDAT pins at startup.

Note that there's no support for controlling DMIC routing except the power
control so the board DAPM configuration will need to manage DMIC enable and
disable if analogue mics (eg, a headset) also exist.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-26 11:49:09 +01:00
Mark Brown 92a4352cdb ASoC: Move WM8962 FLL configuration to CODEC
There's only one DAI anyway.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-26 11:48:54 +01:00
Mark Brown 3b8a6d80e5 ASoC: Support FLL lock interrupt on WM8962
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-26 11:48:37 +01:00
Mark Brown c5f336cc00 ASoC: Support 24.576MHz MCLKs in WM8915
We can safely divide these down to within the supported SYSCLK range.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-26 11:48:26 +01:00
Mark Brown f9f4b1c71d Merge branch 'for-2.6.39' into for-2.6.40 2011-04-26 11:46:47 +01:00
Ben Gardiner db92f43745 davinci-mcasp: fix _CBM_CFS pin directions
The current davinci_mcasp_set_dai_fmt() sets bits ACLKX and ACLKR in the PDIR
register for the codec clock-master/frame-slave mode; however, this results in
the ACLKX and ACLKR pins being outputs according to SPRUFM1 [1]  which
conflicts with "codec is clock master."

Similarly to the previous patch in this series, "fix _CBM_CFS hw_params" --
For codec clock-master/frame-slave mode (_CMB_CFS), clear bits ACLKX and ACLKR
in the PDIR register to set the pins as inputs and hence allow externally
sourced bit-clocks.

[1] http://www.ti.com/litv/pdf/sprufm1

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-26 11:43:53 +01:00
Ben Gardiner a90f549e25 davinci-mcasp: fix _CBM_CFS hw_params
The current davinci_mcasp_set_dai_fmt() sets bits ACLKXE and ACLKRE (CLKXM
and CLKRM as they are reffered to in SPRUFM1 [1]) for codec clock-slave/
frame-slave mode (_CBS_CFS) which selects internally generated bit-clock and
frame-sync signals; however, it does the same thing again for codec
clock-master/frame-slave mode (_CBM_CFS) in the very next case statement which
is incorrectly selecting internally generated bit-clocks in this mode.

For codec clock-master/frame-slave mode (_CBM_CFS), clear bits ACLKXE and
ACLKRE to select externally-generated bit-clocks.

[1] http://www.ti.com/litv/pdf/sprufm1

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-26 11:43:38 +01:00
Ben Gardiner 9595c8f035 davinci-mcasp: use bitfield definitions for PDIR
The current driver creates value for set/clr of PDIR using (x<<26) instead
of the #defines that are convieniently made available.

Update the driver to use the bitfield definitions of PDIR. There is no
functional change introduced by this patch.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-26 11:43:29 +01:00
Ben Gardiner 049cfaaa47 ASoC: davinci-mcasp: correct tdm_slots limit
The current check for the number of tdm-slots specified by platform data is
always true (x >= 2 || x <= 32); therefore the else branch that warns of an
incorrect number of slots can never be taken.

Check that the number of tdm slots specified by platform data is between 2
and 32, inclusive.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-26 11:43:19 +01:00
Kuninori Morimoto 1f5e2a319d ASoC: sh: fsi: Add module/port clock control function
The FIFO of each port were always working though it was not used
in current FSI driver.
This patch add module/port clock control function for fixing it.
This patch is also caring suspend/resume.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Simon Horman <simon@horms.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-26 11:42:11 +01:00
Kuninori Morimoto 106c79ecf2 ASoC: sh: fsi: add dev_pm_ops :: suspend/resume
Current FSI driver sets important settings when probing.
And it are not set again as long as driver is not bind again.
This mean FSI driver will lost it from register
if suspend/resume are happen.
This patch save important settings for suspend/resume.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Simon Horman <simon@horms.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-26 11:42:06 +01:00
Kuninori Morimoto 6a9ebad821 ASoC: sh: fsi: add fsi_is_clk_master function
If FSI port is clock master, it use set_rate function
which is callback from platform,
and it is not necessary to call it if FSI port is clock slave.
Current FSI driver called this callback if platform provide it.
This patch modify it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Simon Horman <simon@horms.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-26 11:40:55 +01:00
Raymond Yau 13eb4ab8ca ALSA: au88x0 - Use a better name for pcm devices of au88x0
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-26 12:27:21 +02:00
Mark Brown 5debd6c14c ASoC: Remove default settings from Tegra Kconfig
There needs to be a strong reason for overriding the Kconfig default.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-26 11:26:44 +01:00
Daniel Mack 8ae9572b5b ALSA: 6fire: use the kernel's built-in bit reverse table
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-26 12:26:12 +02:00
Risto Suominen 30282f96d1 ALSA: powermac - Correct lineout detection on PowerMac G4 DA
Correct lineout (Pro Speaker) detection on PowerMac G4 Digital Audio (Tumbler).

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-22 13:21:01 +02:00
Takashi Iwai 885f42e1f4 ALSA: hda - Enable sync_write for AMD chipset with IDT 92HD8x codecs
The AMD chipset seems unstable in the normal operation mode, and it
seems requring more sensible access for each verb.  Enabling sync_write
mode and allowing bus-reset is a sort of workaround for these chipset
stability issues.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-21 15:27:58 +02:00
Mark Brown a9e3de6f9f Merge branch 'tegra' into for-2.6.40
Fix up merge with Harmony driver rename.

Conflicts:
	sound/soc/tegra/Kconfig
2011-04-21 12:00:27 +01:00
Stephen Warren 47912a657e ARM: Tegra: select MACH_HAS_SND_SOC_TEGRA_WM8903
CONFIG_SND_SOC_TEGRA_WM8903 is useful for many Tegra boards. To avoid the
ASoC tegra/Kconfig enumerating them all, instead have the Tegra machine
Kconfig select MACH_HAS_SND_SOC_TEGRA_WM8903 where appropriate, and have
SND_SOC_TEGRA_WM8903 depend on this.

[Redid ASoC diff so it applies. -- broonie]

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Olof Johansson <olof@lixom.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-21 11:57:31 +01:00
Takashi Iwai 6a9a6f233b Merge branch 'fix/hda' into for-linus 2011-04-21 12:44:38 +02:00
Mike Waychison 1c7276cfc0 ALSA: hda - Fix unused warnings when !SND_HDA_NEEDS_RESUME
When SND_HDA_NEEDS_RESUME is not defined, the compiler identifies that
the following symbols are static but not used:

 restore_shutup_pins
 hda_cleanup_all_streams

Fix warnings by adding SND_HDA_NEEDS_RESUME guards.

Signed-off-by: Mike Waychison <mikew@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-20 21:24:31 +02:00
Seth Heasley d2edeb7c6f ALSA: hda - ALSA HD Audio patch for Intel Panther Point DeviceIDs
This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH.

Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-20 21:03:48 +02:00
Takashi Iwai e66d74ced1 ALSA: asihpi - Use %zd for size_t argument in error message (again)
This was reverted mistakenly in the recent update patch.
Fixed again.

Reported-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-20 21:02:27 +02:00
Stephen Warren 97945c46a2 ASoC: WM8903: Implement DMIC support
In addition to the currently supported analog capture path, the WM8903
also supports digital mics.

The analog and digital capture paths are exclusive; a mux is present to
select the capture source.

Logically, the mux exists to select the decimator's input, from either
the ADC or DMIC block outputs. However, the ADC power domain also
includes the DMIC interface. Consequently, this change represents the
mux as existing immediately before the ADC, and selecting between the
Input PGA and DMIC block outputs.

An alternative might be to represent the mux in its correct location,
and associate the ADC power enable controls with both the real ADC, and
a fake ADC for the DMIC?

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 14:00:35 +01:00
Peter Hsiang dad31ec133 ASoC: Add EQ and filter to max98095 CODEC driver
This patch adds the equalizer and biquad filter controls.

Signed-off-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:50:41 +01:00
Stephen Warren dea8b6eef0 ASoC: Tegra: wm8903: s/code/data/ for control/widget/maps
Replace calls to a variety of registration functions by updating
struct snd_soc_card snd_soc_tegra_wm8903 to directly point at the
various control/widget/map tables instead. The ASoC core now
performs any required registration based on these data fields.

(Applying Mark's TrimSlice review comments to the existing driver)

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:50:36 +01:00
Lu Guanqun a739362362 ASoC: fix two ident style problems
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:50:29 +01:00
Lu Guanqun f9861e17bd ASoC: remove unused comment
`type` parameter is not longer used in `snd_soc_codec_set_cache_io`,
so remove this line.

Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:50:16 +01:00
Lu Guanqun dc2bea616a ASoC: fix a simple coding style issue
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:50:11 +01:00
Stephen Warren a68b38ada5 ASoC: snd_soc_dapm_get_pin_status: Match other contexts too
Not all widgets on a card are within the codec's DAPM context. Fix
snd_soc_dapm_get_pin_status to search all contexts when looking for a
widget.

This change is required when modifying tegra_wm8903 to use
snd_soc_card.widgets rather than calling snd_soc_dapm_new_controls; the
former adds the widgets to the card's DAPM context, whereas tegra_wm8903
uses the codec's DAPM context when calling snd_soc_dapm_new_controls.

By code inspection, I suspect this also applies to Samsung Speyside.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:50:06 +01:00
Stephen Warren a32955dba2 ASoC: Tegra: Retrieve card from DAPM context not codec
Card widgets are created in the card's DAPM context, not any codec's DAPM
context. Hence, w->codec==NULL. Instead, find the card from the widget
through the DAPM context of the widget, not the codec of the widget.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:50:01 +01:00
Stephen Warren 075413966a ASoC: Tegra: Don't return mclk_changed from utils_set_rate
Only the clock programming code needs to know whether the clocks changed,
and that is encapsulated within tegra_asoc_utils_set_rate(). The machine
driver's call to snd_soc_dai_set_sysclk(codec_dai, ...) is safe
irrespective of whether the clocks changed.

(Applying Mark's TrimSlice review comments to the existing driver)

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:49:55 +01:00
Stephen Warren acb8303f15 ASoC: Tegra: wm8903: Remove redundant drvdata clears
When the driver is not initialized/registered, nothing should be touching
these fields anyway, so there's no point clearing them out.

(Applying Mark's TrimSlice review comments to the existing driver)

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:49:50 +01:00
Stephen Warren d9e3c4cc68 ASoC: Tegra: wm8903 probe: Don't call machine_is_*()
This machine driver is a platform driver, and hence will only be
instantiated on the correct machines. Hence, there is no need to
check the current machine during probe.

(Applying Mark's TrimSlice review comments to the existing driver)

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-20 13:49:35 +01:00
Raymond Yau b6a4840408 ALSA: emu10k1 - Remove "Front" controls only for STAC9758/59
Remove "Front Playback Volume" and "Front Playback Switch" from emu10k1 only
for STAC9758/59

Since commit 7eae36fbd5
      "Fix the confliction of 'Front' control",
the "Front Playback Volume" control created by commit
	edf8e4565c
	"emu10k1: Front channels via fxbus 8 and 9"
was removed

"Front Playback Volume" and "Surround Playback Volume" have same dB range
since I2S DAC of SB Live! and SB Live! Platinum does not has any hardware
volume control.

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-20 14:23:15 +02:00
Takashi Iwai 6981d18437 ALSA: hda - Add a fix-up for Acer dmic with ALC271x codec
Acer laptops with ALC271x needs a magic initialization for digital-mic
to make it working with mono streams (and PulseAudio).
Added a fix-up applied to Acer with ALC271x generically.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-19 16:45:31 +02:00
Stephen Warren 773b1d3d31 ASoC: Tegra: Support more boards
* Ventana is identical to Harmony.
* Seaboard, Kaen, and Aebl are all pretty similar, mainly with slightly
  different sets of GPIOs, and slightly different WM8903 pin connectivity.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-18 21:34:16 +01:00
Stephen Warren 3eb25f998d ASoC: Tegra: Don't store snd_soc_jack_gpio in an array
Storing the struct in an array makes the assignments to the GPIO member a
little non-obvious, and is pointless when there's only a single GPIO.

(I thought I fixed this during the review cycle when first submitting this
driver, but I guess I overlooked that)

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-18 21:34:03 +01:00
Stephen Warren 2ba9471b34 ASoC: Tegra: Rename Kconfig SND_TEGRA_SOC_* to SND_SOC_TEGRA_*
The previous commit renames SND_TEGRA_SOC_HARMONY to SND_TEGRA_SOC_WM8903.
While we're breaking people's .config files, rename all Tegra/SOC-related
Kconfig variables to be more consistent with at least the core codec
variables. Note that there exist machines that name their variables both
ways.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-18 21:33:54 +01:00
Stephen Warren dc0a50afa6 ASoC: Tegra: Rename harmony.c to tegra_wm8903.c
Soon, this machine driver will be updated to handle a number of Tegra boards
using the WM8903 codec. Rename the file in advance to reflect this.

Fix the content of tegra_wm8903.c to match the rename; replace references
to Harmony board with something more generic.

* s/struct tegra_harmony/struct tegra_wm8903/
* s/harmony/machine/ # variable name
* Similar rename for some functions
* Similar comment fix
* Similar MODULE_DESCRIPTION fix

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-18 21:33:42 +01:00
Mark Brown c6d46678a1 Merge branch 'tegra' into for-2.6.40 2011-04-18 18:08:22 +01:00
Mark Brown d5381e42f6 ASoC: Merge branch 'for-2.6.39' into for-2.6.40
Fix trivial conflict caused by silly spelling fix patch.

Conflicts:
	sound/soc/codecs/wm8994.c
2011-04-18 18:07:43 +01:00
Stephen Warren 7b33af252f ASoC: Tegra: Rename pdev tegra-snd-harmony to tegra-snd-wm8903
Soon, this machine driver will be updated to handle a number of Tegra boards
using the WM8903 codec. Rename the platform device in advance to reflect this.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-18 17:54:09 +01:00
Stephen Warren 4651d55668 ARM: Tegra: Rename harmony_audio.h -> tegra_wm8903_pdata.h
The audio driver will soon support more than just the Tegra Harmony board.
Rename the platform data header file and data type to reflect this.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-18 17:54:05 +01:00
Guennadi Liakhovetski b3c27b51db ASoC: add a module alias to the FSI driver
This patch enables FSI driver autoloading on sh-mobile systems.

Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Reviewed-by: Simon Horman <horms@verge.net.au
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-18 17:14:28 +01:00
Mark Brown fac56c2df5 Merge commit 'v2.6.39-rc3' into for-2.6.39 2011-04-18 17:12:14 +01:00
Andrew Morton 5b17b077eb ALSA: hda - sound/pci/hda/hda_codec.c: fix warning
sound/pci/hda/hda_codec.c: In function 'snd_hda_get_connections':
sound/pci/hda/hda_codec.c:332: warning: unused variable 'j'

Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-15 08:41:22 +02:00
Daniel Mack 9cdc352936 ALSA: usb-audio: Add quirks for Audio Kontrol 6
This new device by Native Instruments is also compliant to the USB
standard v2.0, but hides this detail at when connected.

It needs the same boot quirks than other models, and also has two
non-class-compliant mixer controls.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-14 12:06:02 +02:00
Lars-Peter Clausen 674479124f ASoC: codecs: JZ4740: Convert to table based controls and DAPM setup
Use the newly introduced dapm_widgets, dpam_routes and controls fields of the
snd_soc_dai_driver struct to setup controls and DAPM.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-13 10:35:03 -07:00
Lars-Peter Clausen 621206b768 ASoC: JZ4740: qi_lb60: Use the SND_SOC_DAPM_EVENT_OFF for the speakers status
Use SND_SOC_DAPM_EVENT_OFF for determining whether the speaker should be turned
on or off instead of open coding it.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-13 10:34:54 -07:00
Lars-Peter Clausen c6f0ede7c5 ASoC: JZ4740: qi_lb60: Use gpio_request_array to request and setup gpios
This patch changes the qi_lb60 setup code to use gpio_request_array instead of
manually calling gpio_request and gpio_direction_output for each gpio.
Doing so makes the code a bit more compact.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-13 10:34:43 -07:00
Lars-Peter Clausen 1331969911 ASoC: JZ4740: Convert qi_lb60 codec to table based DAPM setup
Use the newly introduced dapm_widgets, dpam_routes and fields of the
snd_soc_card struct to setup DAPM.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-13 10:34:36 -07:00
Mark Brown ec5af076f5 Merge branch 'for-2.6.39' into for-2.6.40 2011-04-13 10:33:52 -07:00
Lars-Peter Clausen 1fdf9b49e9 ASoC: codecs: JZ4740: Fix OOPS
Commit ce6120cc(ASoC: Decouple DAPM from CODECs) changed the signature of
snd_soc_dapm_widgets_new to take an pointer to a snd_soc_dapm_context instead of
a snd_soc_codec. The call to snd_soc_dapm_widgets_new in jz4740_codec_dev_probe
was not updated to reflect this change, which results in a compiletime warning
and a runtime OOPS.

Since the core code calls snd_soc_dapm_widgets_new after the codec has been
registered it can be dropped here.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-04-13 10:26:46 -07:00
Mark Brown b7a5d14c60 ASoC: Mark Speyside widgets as ignoring suspend
Allow audio paths through the Speyside system to be kept active while the
system is suspended (for example, when on a voice call) by marking all the
external widgets and the DAI link to the WM1250-EV1 baseband module as
ignoring suspend.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 10:02:23 -07:00
Mark Brown 556e4fb1d8 ASoC: Add stub baseband link on Speyside
Demonstrate the connection of a baseband to the system. We add a DAI for
the link to the baseband. This will become visible to the application
layer - audio should be started from the application layer using an
application such as this:

   http://opensource.wolfsonmicro.com/~gg/bluetooth-pcm/bluetooth_pcm.c

which starts up audio as for CPU based playback and record up to the point
where data is streamed.

Due to non-availability of baseband simulation hardware we reuse the
configuration for the CPU link with the CODEC acting as clock master,
allowing signals to be observed with a scope. A more standard system
would have separate configuration for the baseband with its own ops
structure and operations. Normally the baseband would be clock master
as the baseband audio will be synchronised to the external telephony
network.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 10:02:17 -07:00
Mark Brown ea0e60de38 ASoC: Add pin switches for fixed analogue inputs and outputs on Speyside
Pin switches enable direct control of the DAPM state from userspace,
enabling simple enabling and disabling of the path. This is especially
useful for outputs such as the speaker which are composed of several
physical devices as it allows them to be controlled as a group.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 10:02:12 -07:00
Mark Brown 68688e78ed ASoC: Add Speyside headset jack detection support
Speyside makes use of support the WM8915 has for detecting the polarity
of the microphone and ground connections on headsets, using a GPIO to
control the polarity of the ground connection and switching between the
two microphone bias supplies available on the device in order to do so.
As a result of this the detection support is more involved than for most
other CODECs, using a callback to configure the current polarity of the
jack and translate this into the board-specific connections required for
the current scenario.

On Android some additional work is required to hook this up to the
application layer as the Android HeadsetObserver monitors a custom
drivers/switch API rather than the standard Linux APIs.  This can be
done by either updating HeadsetObserver or modifying the ALSA core to
report via drivers/switch as well.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 10:02:06 -07:00
Mark Brown ea3e98e75a ASoC: Support the sub speaker driver on Speyside
Speyside includes a WM9081 configured as an external speaker driver taking
an analogue input from HPOUT2 on the WM8915 on the system. Add support for
this to the driver, using a prefix of "Sub" for the WM9081 controls to
ensure we avoid collisions with controls on the WM8915.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 10:02:01 -07:00
Mark Brown ea0a591a28 ASoC: Optimise clock management for WM8915 Speyside
Dynamically enable and disable the FLL on the WM8915, configuring the
system clock to 256fs for 48kHz when the device is active but reverting
to using the input 32.768kHz clock directly at other times to support
features such as jack detection with minimal power consumption.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 10:01:57 -07:00
Mark Brown ecfb1adf5f ASoC: Add basic widgets for WM8915 Speyside
Provide widgets for the basic widgets connected directly to the WM8915
on Speyside - the headphones, speaker, digital and analogue microphones.
For the outputs this is just documentation, for the inputs this ensures
that the relevant microphone biases are enabled when they are in use.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 10:01:52 -07:00
Mark Brown 9b8dc66fba ASoC: Initial audio support for Speyside on Cragganmore 6410
This is minimal code required to get audio out of the Speyside audio
subsystem on the Wolfson Cragganmore 6410 reference platform.  It sets
up the link between the CPU and AIF1 of the WM8915 on the system,
enabling audio playback via the headphone and speaker outputs of the
device (which require no further configuration except runtime).  It
allows verification of basic functionality of the system.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 10:01:30 -07:00
Mark Brown 9a841ebb9c ASoC: Create card DAPM widgets early so they can be used in callbacks
This helps with things like setting up the initial state.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 10:00:21 -07:00
Mark Brown 01b07e2d84 ASoC: Move WM8915 FLL operations onto the CODEC
Since the WM8915 FLL is not tied to a particular audio interface move it
to a CODEC wide operation.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-13 09:52:52 -07:00
Peter Ujfalusi 82a58a8b7f ASoC: tlv320dac33: Lower the OSC calibration time
To get correct calibration, we can decrease the time
needed for the OSC to calibrate itself.
With this change we can save ~15ms in the OSC
calibration phase.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-04-13 09:32:37 +01:00
Mark Brown 420dd718ad ASoC: Fix mis cherry-pick of wm1250-ev1 driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-11 21:44:43 -07:00
Mark Brown 4bb3f43c6e ASoC: Add initial WM1250-EV1 Springbank audio I/O module driver
The WM1250-EV1 Springbank audio I/O module for the Wolfson Glenfarclas
reference platform provides a simple audio I/O with an independant clock
domain, intended to simulate cellular modem and bluetooth subsystems
within the platform.

The card supports some limited GPIO based control but this is currently not
implemented.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-11 13:34:13 -07:00
Mark Brown c93993aca4 ASoC: Add WM8915 CODEC driver
The WM8915 is an ultra low power mobile CODEC designed for smartphones,
featuring a mixture of digital and analogue I/O with flexible mixing
options and advanced low power accessory detection functionality in a
compact package.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-04-11 13:33:50 -07:00
Kuninori Morimoto 0671fd8ef4 ASoC: Add soc_remove_dai_links
card->num_rtd should be 0 after soc_romve_dai_link

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-11 13:31:52 -07:00
Sangbeom Kim b8eeee68dc ASoC: SAMSUNG: Add WM8580 PCM Machine driver
This patch add WM8580 PCM machine driver to support PCM audio
on SMDKC110, SMDKV210, SMDK6450, SMDK6440 boards.
Playback and Capture supports 8kHz sampling rates.
and It is tested on SMDKC110, SMDKV210, SMDK6450

Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-11 13:31:31 -07:00
Mark Brown a19809875f Merge branch 'for-2.6.39' into for-2.6.40 2011-04-11 13:29:24 -07:00
Mark Brown 39cca168bd ASoC: Fix output PGA enabling in wm_hubs CODECs
The output PGA was not being powered up in headphone and speaker paths,
removing the ability to offer volume control and mute with the output
PGA.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
2011-04-11 13:28:56 -07:00
Lu Guanqun 90db8ece6a ASoC: sn95031: decorate function with __devexit_p()
According to the comments in include/linux/init.h:

"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the config
options."

Fix this issue in codecs sn95031.

Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-11 13:28:54 -07:00
Sangbeom Kim 68e0c6696c ASoC: SAMSUNG: Fix the inverted clocks handling for pcm driver
Fix the inverted clocks handling for pcm cpu driver.
By using SND_SOC_DAIFMT_NB_NF, Audio noise can be generated on SMDK.

Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-11 13:15:01 -07:00
Lu Guanqun d89b0a136e ASoC: sst_platform: Fix lock acquring
Fix the possible dead lock shown below:

spin_lock
sst_get_stream_status
sst_period_elapsed
intel_sst_interrupt
handle_IRQ_event
handle_fasteoi_irq
do_IRQ
common_interrupt
spin_lock
sst_set_stream_status
sst_platform_pcm_trigger

Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-11 13:14:51 -07:00
Kuninori Morimoto d985f27e13 ASoC: fsi: driver safely remove for against irq
free_irq and pm_runtime_disable should be called before
snd_soc_unregister_xxx

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-11 13:14:41 -07:00
Kuninori Morimoto b9c9f9675f ASoC: fsi: modify vague PM control on probe
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-11 13:14:33 -07:00
Kuninori Morimoto 0b5ec87d3e ASoC: fsi: take care in failing case of dai register
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-11 13:14:09 -07:00
Linus Torvalds 4263a2f1da Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Don't query connections for widgets have no connections
  ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
  ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums
  ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
  ALSA: HDA: Fix dock mic for Lenovo X220-tablet
  ASoC: format_register_str: Don't clip register values
  ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare
  ASoC: zylonite: set .codec_dai_name in initializer
2011-04-10 09:56:10 -07:00
Takashi Iwai 84f3b6dab9 Merge branch 'fix/hda' into for-linus 2011-04-09 10:05:53 +02:00
Takashi Iwai 664cee46e7 Merge branch 'fix/asoc' into for-linus 2011-04-09 10:05:30 +02:00
Mark Brown 0d86733cce ASoC: Allow DAPM pin operations to match any context
The DAPM pin operations currently require that the specific DAPM context
that the pin being operated in is contained in be specified. With multi
component and especially with the addition of a per-card DAPM context
this isn't ideal as it means that things like disabling unused pins on
CODECs require looking up the CODEC DAPM context.

Fix this by falling back to matching a widget in any context if there isn't
a match in the current context. The code isn't ideal currently but will do
the job.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-09 11:25:20 +09:00
Mark Brown 52ba67bf85 ASoC: Force all DAPM contexts into the same bias state
Currently we allow all DAPM contexts to determine their own bias level.
While this should in general work in most situations and will deliver the
lowest possible power it causes problems for our integration with the
card bias level as we're calling the card bias level functions for each
DAPM context even though they're card wide but don't say which CODEC
we're calling them for. Mitigate against this by forcing everything to
be in the same state.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-09 11:24:08 +09:00
Mark Brown d25b7c1ec7 ASoC: Remove special casing for registerless widgets
Since we recently explicitly set the register for registerless widgets
to no register there is no longer any need to special case power updates
for them, we can allow them to be handled with the register compression
code as other widgets are.

As this is the only remaining user of dapm_generic_apply_power() and
dapm_update_bits() also remove those functions.

Noticed-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-08 17:29:41 +09:00
Mark Brown faeede8cdc Merge branch 'for-2.6.39' into for-2.6.40 2011-04-08 09:31:02 +09:00
Mike Frysinger b39e285545 ASoC: SSM2602: add SPI support
The ssm2602 codec has a SPI interface as well as I2C, so add the simple
bit of glue to make it usable.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-08 09:24:24 +09:00
Mark Brown b7af1dafdf ASoC: Add data based control initialisation for CODECs and cards
Allow CODEC and card drivers to point to an array of controls from their
driver structure rather than explicitly calling snd_soc_add_controls().

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-08 09:18:11 +09:00
Dilan Lee 1b877cb57a ASoC: WM8903: HP and Line out PGA/mixer DAPM fixes
Update the headphone and line out mixers and PGAs use the same logical
set of register bits and sequencing as the speaker mixer/PGA.

This allows ALSA controls for mute and volume on headphone and line out
to operate correctly.

Per conversation on alsa-devel, earlier datasheets indicated that the
POWER_MANAGEMENT_* register bits 0 and 1 were aliases to ANALOG_* register
bits 0 and 4, and hence only one copy of those bits was programmed.
However, later datasheets corrected this.

From: Dilan Lee <dilee@nvidia.com>
[swarren: Applied same change to headphone widgets]
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-08 09:17:11 +09:00
Linus Torvalds 42933bac11 Merge branch 'for-linus2' of git://git.profusion.mobi/users/lucas/linux-2.6
* 'for-linus2' of git://git.profusion.mobi/users/lucas/linux-2.6:
  Fix common misspellings
2011-04-07 11:14:49 -07:00
Takashi Iwai a12d3e1e1c ALSA: hda - Remember connection lists
The connection lists are static and we can reuse the previous results
instead of querying via verb at each time.  This will reduce the I/O
in the runtime especially for some codec auto-parsers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 15:55:15 +02:00
Takashi Iwai cd9abc7a22 ALSA: hda - Don't query connections for widgets have no connections
Fixes the kernel warnings with IDT codecs like
    hda_codec: connection list not available for 0x1e

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 14:55:57 +02:00
Takashi Iwai 8e28e3b29f Merge branch 'fix/hda' into topic/hda 2011-04-07 12:57:53 +02:00
Takashi Iwai ad93ffe6e4 ALSA: hda - Fix unused variable warning in patch_realtek.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 12:49:26 +02:00
Takashi Iwai 35ffe11587 ALSA: hda - Remove superfluous inits for ALC662 auto-parser
Since we now set up the connections and mutes dynamically in the
auto-parser, all static initializations via alc662_init_verbs & co are
no longer needed.  Let's drop them.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 12:49:10 +02:00
Takashi Iwai 10696aa0e5 ALSA: hda - Mute ADC as default in ALC882 and other auto-parsers
Mute the ADC as default in the auto-parser dynamically instead of relying
on the static init verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 12:49:08 +02:00
Takashi Iwai 0e53f34409 ALSA: hda - Unmute mixer dynamically in alc662 auto-parser
Instead of static init array, better to determine the connection and
the mute status of the pin/mixer/DAC route dynamically.  This fixes the
uninitialized mixer 0x0f on ALC892.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 12:49:05 +02:00
David Henningsson 262ac22d21 ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
In cases where there is only one internal mic connected to ADC 0x11,
alc275_setup_dual_adc won't handle the case, so we need to add the
ADC node to the array of candidates.

Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/752792
Reported-by: Vincenzo Pii
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 12:12:00 +02:00
Aaron Plattner 1f34852284 ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums
The MCP7x hardware computes the audio infoframe channel count
automatically, but requires the audio driver to set the audio
infoframe checksum manually via the Nv_VERB_SET_Info_Frame_Checksum
control verb.

When audio starts playing, nvhdmi_8ch_7x_pcm_prepare sets the checksum
to (0x71 - chan - chanmask).  For example, for 2ch audio, chan == 1
and chanmask == 0 so the checksum is set to 0x70.  When audio playback
finishes and the device is closed, nvhdmi_8ch_7x_pcm_close resets the
channel formats, causing the channel count to revert to 8ch.  Since
the checksum is not reset, the hardware starts generating audio
infoframes with invalid checksums.  This causes some displays to blank
the video.

Fix this by updating the checksum and channel mask when the device is
closed and also when it is first initialized.  In addition, make sure
that the channel mask is appropriate for an 8ch infoframe by setting
it to 0x13 (FL FR LFE FC RL RR RLC RRC).

Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 12:04:00 +02:00
Takashi Iwai 5402e4cb80 ALSA: hda - Rewrite alc269_suspend to alc269_shutup
alc269_suspend is just calling the shut-up, so we can use the new shutup
callback for the purpose.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 10:39:25 +02:00
Takashi Iwai 1c716153a8 ALSA: hda - Introduce shutup callback to Realtek spec struct
Add shutup callback to be called codec-specifically for avoiding pop
noises at suspend or shutdown.  As a generic callback, just turn EAPD
off.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 10:37:16 +02:00
Takashi Iwai 691f1fccf7 ALSA: hda - Refactoring EAPD controls
Reduced the duplicated codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 10:31:43 +02:00
Takashi Iwai a7f2371f9e ALSA: hda - Split EAPD init to a separate array from alc662_init_verbs
So far, alc662_init_verbs[] is used for all ALC662-compatible chips,
but the EAPD controls for 0x15 in there is invalid for ALC892.
Also, since EAPDs should be set up in alc_auto_init_amp(), these static
elements aren't needed for auto-parser, too.

In this patch, the EAPD init verbs are split from alc662_init_verbs,
and applied only to static quirks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-07 10:24:23 +02:00
Mark Brown d9b3e4c515 Merge branch 'for-2.6.39' into for-2.6.40 2011-04-07 08:27:06 +09:00
Mark Brown baa8160382 ASoC: Set left channel volume update bits for WM8994
Ensures that we apply volume updates that don't affect the right channel.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-07 08:26:09 +09:00
Lu Guanqun c51def6598 ASoC: fix config error path
initialize ret to invalid value so that when we reach the config error path in
soc_pcm_open, it will return the correct error code. without this patch, though
config error path is executed, soc_pcm_open will return 0 in
snd_pcm_open_substream and then cause double release of substream.

Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-07 08:25:45 +09:00
Lu Guanqun b04cfcf70b ASoC: check channel mismatch between cpu_dai and codec_dai
Suppose we have:

	cpu_dai
		channels_min = 1
		channels_max = 1

	codec_dai
		channels_min = 2
		channels_max = 2

This is a mismatch that should not happen, however according to the current
code, the result of runtime->hw will be:

		channels_min = 2
		channels_max = 1

We better spot it early. This patch checks this mismatch.

Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-07 08:25:45 +09:00
Lu Guanqun fb631eae1f ASoC: sst_platform: unregister sst card when being closed
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-06 23:15:23 +09:00
Lu Guanqun 83a3fd3cf0 ASoC: sst_platform: free the resources on fail path
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-06 23:15:23 +09:00
Lu Guanqun 0d1d7ce951 ASoC: sst_platform: initialize module_name properly
module_name will be checked in register_sst_card.
It will fail to register sst card if it's not initialized.

Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-06 23:15:23 +09:00
Peter Hsiang 82a5a936f6 ASoC: Add max98095 CODEC driver
This patch adds the MAX98095 CODEC driver.

Signed-off-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-06 23:15:23 +09:00
Mark Brown fa88000468 Merge branch 'for-2.6.39' into for-2.6.40 2011-04-06 23:15:15 +09:00
Stephen Warren deb2607e6c ASoC: Tegra: Suspend/resume support
ASoC machine drivers that are their own platform_driver (as opposed to
those using the soc-audio platform_driver) need to explicitly set up
power-management operation callbacks.

To avoid cut/paste, snd_soc_pm_ops also needs to be exported.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-06 23:13:48 +09:00
Takashi Iwai 1304ac8993 ALSA: hda - Fix mix->DAC deduction for ALC892
The current alc662 parser doesn't set the DAC for the mixer 0x0f
properly for ALC892, which has 4 DACs while ALC662 has 3.
Fixed by implementing alc662_mix_to_dac() more genericly with the
dynamic widget list.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-06 15:16:21 +02:00
Takashi Iwai 1bc7cf99a9 ALSA: hda - Correct initial dac_nids for some ALC272-quirks
Some ALC272-quirks use alc662_dac_nids instead of alc272_dac_nids.
This patch fixes these entries.  No functional change since the first
two elements are identical in both arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-06 09:42:29 +02:00
Raymond Yau e217b960e4 ALSA: emu10k1 - Remove CLFE-related controls for SB Live! Platinum CT4760P
SB Live! Platinum CT4760P is just a 4 channels sound card with STAC9721 and
Philips UDA1334 DAC.

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-06 08:31:26 +02:00
Raymond Yau 4bf4a6c5b1 ALSA: hda - Fix alc662_dac_nid and change "6stack-dig" to "5stack-dig"
alc662 series only have 3 DAC, so it can only support 5stack-dig
instead of 6stack-dig.

[updated HD-Audio-Models.txt as well by tiwai]

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-06 08:18:39 +02:00
Tarek Soliman 49c039f071 ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
There are many USB MIDI cables out there that have buggy
firmware that reports it can do more than 4 bytes in a
packet when they can only properly handle 4

This patch adds the ID of yet another one of those cables

Signed-off-by: Tarek Soliman <tarek@bashasoliman.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-06 08:05:30 +02:00
Eliot Blennerhassett 42258daba2 ALSA: asihpi: Minor cleanups
Remove some unneeded defintions
Use %pR to print resources
Make some data const
Consistent braces for else

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-05 11:51:04 +02:00
Eliot Blennerhassett 6d0b898e9c ALSA: asihpi: Simplify driver unload cleanup
Replacing subsys_delete_adapter with adapter_delete
allows some special-case adapter lookup code to be removed.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-05 11:50:13 +02:00
Eliot Blennerhassett b0096a6567 ALSA: asihpi: Standardise substream name generation
Define and use pcm_debug_name if CONFIG_SND_DEBUG

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-05 11:47:54 +02:00
Eliot Blennerhassett 6027dfa46e ALSA: asihpi: Remove 2 unused functions
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-05 11:47:40 +02:00
Eliot Blennerhassett f3d145aac9 ALSA: asihpi: MMAP for non-busmaster cards
Allow older non DMA capable cards to use MMAP by
emulating the DMA using read and write functions,
and getting rid of copy & silence callbacks that
were used only by older cards.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-05 11:47:25 +02:00
Eliot Blennerhassett 0b7ce9e2bd ALSA: asihpi: Handle playback drained status better
Use the card drained status reporting for playback,
but allow it to persist for a few timer cycles before
signalling XRUN, to allow card to recover by itself.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-05 11:47:08 +02:00
Eliot Blennerhassett a6477134db ALSA: asihpi: Update debug printing
Debug print full substream ID.
Other minor debug print updates.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-05 11:46:48 +02:00
Eliot Blennerhassett 550ac6ba4e ALSA: snd-asihpi: Control naming
Clock source is neither capture nor playback,
so change 'Capture Clock' to 'Clock'.
Add spaces to control name string for consistency,
always 'PCM 0' , never 'PCM0'

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-05 11:46:06 +02:00
David Henningsson b2cb1292b1 ALSA: HDA: Fix dock mic for Lenovo X220-tablet
Without the "thinkpad" quirk, the dock mic in
Lenovo X220 tablet edition won't work.

BugLink: http://bugs.launchpad.net/bugs/751033
Cc: stable@kernel.org
Tested-by: James Ferguson <james.ferguson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-05 09:17:10 +02:00