Commit Graph

14892 Commits

Author SHA1 Message Date
Mark Brown 82e993fac4 ASoC: wm2200: Add controls for firmware enumeration
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-18 15:06:26 +09:00
Mark Brown c712326d6c ASoC: wm_adsp: Implement support for coefficeint file format 1
Implement support for a new revision of the coefficeint file format for
ADSP cores.

Since coefficient file format 0 has not been widely deployed and is very
unlikely to ever be used with this driver code support for it has been
removed.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-18 15:02:17 +09:00
Takashi Iwai 36c9db7a1a ALSA: hda - Use generic parser for STAC/IDT codec driver
Finally we reached here.  All codecs driver (except for CA0132, which
has really device-specific requirements) have been converted to use
the generic parser.

This patch appears bigger than others since it also involves with the
code shuffling, but mostly the cut-off of parser codes and adapt to
the generic parser flags.  Most of fixup codecs haven't been changed
but just removed a few unnecessary codes.

The only missing stuff is the SPDIF mux control.  It'll be added again
later.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-17 17:46:13 +01:00
Takashi Iwai 8f0fdc09aa Merge branch 'test/hda-gen-parser' into test/hda-migrate
* test/hda-gen-parser:
  ALSA: hda - Improve naming rule for primary output
  ALSA: hda - Add PCM capture hook to hda_gen_spec
  ALSA: hda - Record all detected ADCs in hda_gen_spec
  ALSA: hda - Move vmaster TLV parsing to snd_hda_gen_parse_auto_config()
  ALSA: hda - Add input jack mode enum controls to generic parser
  ALSA: hda - Give more comments to hda_gen_spec flags
  ALSA: hda - Add suppress_auto_mute flag to hda_gen_spec
  ALSA: hda - Record the current speaker / LO mute status in hda_gen_spec
  ALSA: hda - Properly call automute/switch hooks at init
2013-01-17 16:20:14 +01:00
Takashi Iwai 247d85ee06 ALSA: hda - Improve naming rule for primary output
When the volume or mute control of the primary output is shared with
other (headphone or speaker) outputs, we shouldn't name it as a
specific output type but rather name it with the channel name or a
generic name like "PCM".

Also, this check should be performed individually for the volume and
the mute controls because some codecs may have shared volumes but
separate mute controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-17 16:18:11 +01:00
Takashi Iwai ac2e87366c ALSA: hda - Add PCM capture hook to hda_gen_spec
Not only PCM playback, a hook for PCM capture would be required for
power controls in codec drivers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-17 15:57:10 +01:00
Takashi Iwai 0ffd534eb1 ALSA: hda - Record all detected ADCs in hda_gen_spec
Since the generic parser reduces the ADC list, copy the list of the
all detected ADCs and keep it.

This list can be later referred by the codec driver for finer power
controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-17 15:53:29 +01:00
Takashi Iwai 7a71bbf310 ALSA: hda - Move vmaster TLV parsing to snd_hda_gen_parse_auto_config()
Add vmaster_tlv[] to hda_gen_spec and store the suggested TLV data
in snd_hda_gen_parse_auto_config().  This allows the codec driver to
correct the TLV data (e.g. mute capability) before actually creating
vmaster instance.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-17 10:25:15 +01:00
Takashi Iwai 29476558de ALSA: hda - Add input jack mode enum controls to generic parser
Just like the jack mode enum ctls for output jacks, add the support
for similar enum ctls for input pins to control the bias Vref.
The new controls will be added when spec->add_in_jack_modes is set
either by the codec driver or by a hint string.

Note that ground and 100% vrefs are excluded from the list for
simplicity, currently.  We may add a new flag to allow them, too.
But I guess it's easier to put a value override in the pinfix in such
a case.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-17 09:55:21 +01:00
Takashi Iwai f6655d52a3 ALSA: hda - Minor cleanup/fixes for patch_sigmatel.c fixup transition
- spec->hp_detect has to be overridden in HDA_FIXUP_ACT_PARSE, not in
  PRE_PARSE.
- Remove err == 0 check but return directly -EINVAL from
  stac92xx_parse_auto_config()
- Set spec->default_polarity for 92HD71bxx
- Some code shuffles

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-17 08:49:01 +01:00
Lucas Stach 6995b8cb96 ASoC: tegra: add tegra machine driver using wm9712 codec
This adds a very simple machine driver using the Wolfson wm9712 AC97
codec.

Signed-off-by: Lucas Stach <dev@lynxeye.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-17 16:02:41 +09:00
Mark Brown 4706ccbbe8 Merge remote-tracking branch 'asoc/fix/arizona' into asoc-arizona
Conflicts:
	sound/soc/codecs/arizona.c
2013-01-17 15:31:54 +09:00
Mark Brown b59e0f82aa ASoC: arizona: Use actual rather than desired BCLK when calculating LRCLK
Otherwise we'll get the wrong LRCLK if we need to pick a higher BCLK than
is required.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-17 14:36:07 +09:00
Takashi Iwai acc47aafcf ALSA: hda - Give more comments to hda_gen_spec flags
Since we have many bit flags in hda_gen_spec, rearrange in sections
and give more comments there.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 18:28:38 +01:00
Takashi Iwai f72706be35 ALSA: hda - Add suppress_auto_mute flag to hda_gen_spec
A new flag to skip the auto-mute handling in the generic parser, just
like suppress_auto_mic flag.  It has to be set before calling
snd_hda_gen_parse_auto_config().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 18:22:37 +01:00
Takashi Iwai 47b9ddb83b ALSA: hda - Record the current speaker / LO mute status in hda_gen_spec
... to be referred by the codec driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 18:19:50 +01:00
Takashi Iwai a5cc25091c ALSA: hda - Properly call automute/switch hooks at init
... and a little bit of code refactoring.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 18:08:55 +01:00
Takashi Iwai ae127005fc Merge branch 'test/hda-gen-parser' into test/hda-migrate
* test/hda-gen-parser:
  ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecs
  ALSA: hda - force different capture controls if amp caps differ
  ALSA: hda - do not add non-existing Mic boost controls
  ALSA: hda - initialize channel counts correctly
  ALSA: hda - fix wrong adc_idx in generic parser
  ALSA: hda - Check array bounds in get_input_path
  ALSA: hda - Add prefer_hp_amp flag to hda_gen_spec
  ALSA: hda - fix OOPS in hda_mark_cmd_cache_dirty
  ALSA: hda - Check pincap while parsing the configuration
2013-01-16 16:25:24 +01:00
David Henningsson 6fc4cb97cb ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecs
Otherwise no PCM will be built for codecs without analog I/O.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 16:24:42 +01:00
David Henningsson 99a5592d6a ALSA: hda - force different capture controls if amp caps differ
Otherwise setting the capture volume for amps will be weird and
inconsistent (it will try to set values outside the range of the
second amp based on capabilities of the first amp).

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 16:24:00 +01:00
David Henningsson 02aba55053 ALSA: hda - do not add non-existing Mic boost controls
If the input node does not have any volume capable input amp,
don't add such a control.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 16:22:43 +01:00
Chris Rattray a80cc73428 ASoC: wm2200: correct mixer values and text
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-16 20:47:26 +09:00
Thierry Reding e43fc6af25 ASoC: fsi: Remove __devinitconst
__devinitconst and friends have recently been removed and must not be
used anymore.

Signed-off-by: Thierry Reding <thierry.reding@avionic-design.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-16 20:28:59 +09:00
David Henningsson c0f3b21643 ALSA: hda - initialize channel counts correctly
Even a single DAC can output two channels, so the channel count
is twice the number of DACs.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 11:57:00 +01:00
David Henningsson a053d1e3c4 ALSA: hda - fix wrong adc_idx in generic parser
We use knew->index for adc_idx when we create "Capture Volume" and
"Capture Switch", so use the same to retrieve adc_idx.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 11:56:50 +01:00
David Henningsson b56fa1ed09 ALSA: hda - Check array bounds in get_input_path
This gives us some additional safety.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 11:56:36 +01:00
Mark Brown c98137bfcb ASoC: arizona: Don't request FLL lock IRQ
We only log the result and since the interrupt triggers on loss of lock
during shutdown this may lead to spurious interrupts during shutdown
delaying the process.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-16 19:28:59 +09:00
Takashi Iwai ccd7bd3d07 ALSA: hda/ca0132 - Make some symbols static
sound/pci/hda/patch_ca0132.c:387:19: sparse: symbol 'ca0132_voicefx' was not declared. Should it be static?

Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 07:56:02 +01:00
Adrian Knoth 49ba4f94bd ALSA: hdsp - Remove obsolete settings functions
With HDSP_TOGGLE_SETTING in place, these functions are no longer
required. Removing them makes the code DRY and considerably shorter.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 07:49:17 +01:00
Adrian Knoth 4833c673de ALSA: hdsp - Use HDSP_TOGGLE_SETTING to alter settings
HDSP_TOGGLE_SETTING and its corresponding functions allow to change
settings in the control register. Instead of using the specialised
functions, use the generic code to make the code DRY.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 07:49:06 +01:00
Adrian Knoth 66d9244ec7 ALSA: hdsp - Implement generic function to toggle settings
The driver contains multiple similar functions that change only a single
bit in the control register, only the bit position varies.

This patch implements a generic function to toggle a certain bit
position that will be used to replace the old code.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 07:48:51 +01:00
Adrian Knoth 0c2bc7c7d8 ALSA: hdsp - Fix detection for RME RPM/Multiface/Digiface ioboxes
The current iobox detection code reportedly fails for various users, so
simply do what the Win32 driver does instead.

Patch originally by Karl Grill <kgrill@chello.at> and then modified to
comply with kernel coding guidelines + current HEAD.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-16 07:48:38 +01:00
Mark Brown 5851cb3daf ASoC: wm2200: Initialise the ADSPs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-16 10:25:24 +09:00
Mark Brown 2ce4616e4f Merge remote-tracking branch 'asoc/topic/adsp' into asoc-wm2200 2013-01-16 10:24:08 +09:00
Mark Brown 5e7a7a221f ASoC: wm_adsp: Add initialisation function for ADSP1
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-16 10:21:57 +09:00
Takashi Iwai ea46c3c87c ALSA: hda - Add prefer_hp_amp flag to hda_gen_spec
Add a new flag to indicate whether HP amp is turned on as default for
speaker or line-outs, and enable this for ALC260 codec, as many
machines with this codec require the HP amp even for speakers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 18:45:53 +01:00
Takashi Iwai dea500c7c6 ALSA: hda/ca0132 - Fix a wrong comma in snd_printdd() call
sound/pci/hda/patch_ca0132.c: In function ‘ca0132_effects_set’:
sound/pci/hda/patch_ca0132.c:3391:2: warning: too many arguments for
  format [-Wformat-extra-args]

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:49:56 +01:00
Takashi Iwai 7a527edee4 ALSA: hda/ca0132 - Declare firmware only when really built
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:49:45 +01:00
Takashi Iwai 8ae3124b8f ALSA: hda/ca0132 - Fix possible invalid DMA channel deallocation
... in the error path in dspxfr_image().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:49:38 +01:00
Takashi Iwai 549e8292a1 ALSA: hda/ca0132 - Fix possible NULL dereference
Spotted by smatch,
  sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: potential
  null dereference 'dma_engine'.  (kzalloc returns null)
  sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: we
  previously assumed 'dma_engine' could be null (see line 1857)

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:42:15 +01:00
Takashi Iwai 425a7880e8 ALSA: hda/ca0132 - Fix another smatch warning
sound/pci/hda/patch_ca0132.c:1781 dspxfr_one_seg() info: why not
propagate 'status' from dsp_dma_stop() instead of (-5)?

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:41:21 +01:00
Takashi Iwai b645d79619 ALSA: hda/ca0132 - Fix superfluous unsigned check
Fix a warning by smatch,
 sound/pci/hda/patch_ca0132.c:714 dspio_send() warn: always true
 condition '(res >= 0) => (0-u32max >= 0)'

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:39:29 +01:00
Takashi Iwai a0c041cb6f ALSA: hda/ca0132 - Use snd_hda_set_pin_ctl() helper again
The recent update of ca0132 driver replaced the pinctl setup to the
direct write via snd_hda_codec_write() again.  This should be covered
by snd_hda_set_pin_ctl() to be safer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:13:31 +01:00
Takashi Iwai 15e4ba666c Revert "ALSA: hda - Add firmware caching to CA0132 codec"
This reverts commit c3b4eea262.

Since the recent firmware loader code supports caching at S3/S4 by
itself, we don't have to handle f/w caching in the driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:09:27 +01:00
Ian Minett 406261ce99 ALSA: hda/ca0132: Fix potential init errors and update module description
Handle a potential dma_engine alloc error and fix the possible use of an
uninitialized status variable in dspxfr_one_seg(). Also correct the initial
sampling rate for Mic 1.
Update the module description.

Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:01:16 +01:00
Ian Minett 441aa6a016 ALSA: hda/ca0132: Shuffle to group together related code
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:01:01 +01:00
Ian Minett e90f29e442 ALSA: hda/ca0132: Code shuffle to group similar functions.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:00:45 +01:00
Ian Minett 44f0c9782c ALSA: hda/ca0132: Add tuning controls
This patch adds the controls used for tuning the DSP effects.

Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 17:00:31 +01:00
Ian Minett a73d511c48 ALSA: hda/ca0132: Add unsol handler for DSP and jack detection
This patch adds the unsolicited response handler for incoming DSP responses and
jack detection reporting, and routines for reading the incoming DSP response.

Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 16:59:56 +01:00
Ian Minett 825315bc5b ALSA: hda/ca0132: Add PCM enhancements
Remove the playback PCM open callback.
PCM stream setup and cleanup functions are added for use by PCM callbacks.
Delay stream cleanup if effects are on, to allow time for any effects tail to
finish.
Add the analog capture PCM callbacks.
Change the max channels of analog playback to 6.
Add two new PCMs: AMic2 and What-U-Hear.

Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 16:59:21 +01:00
Ian Minett a7e76271bd ALSA: hda/ca0132: Add DSP mixer controls and helpers
This patch adds the kcontrols for the DSP effects, playback and recording
source selection.
ca0132_is_vnode_effective() checks whether virtual node settings have
taken effect.
The control change helpers ca0132_pe_switch_set(), ca0132_voicefx_set()
and ca0132_cvoice_switch_set() are added to toggle playback / capture
DSP effects, ca0132_voicefx_info(), _get() and _put() are added for
input path DSP effect value access. The volume helpers are updated to
volume_info(), _get() and _set() to use the virtual nodes.
The redundant headphone and speaker switches and ct_extension function
are removed.

Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 16:58:12 +01:00
Ian Minett 5aaca44d8d ALSA: hda/ca0132: Init chip, DSP effects and mixer settings
This patch adds the framework to set effect parameters: ca0132_effects_set()
and ca0132_setup_defaults() are general functions for parameter setting and
initializing to default values. dspio_set_param() and dspio_set_uint_param()
are lower-level fns to simplify setting individual DSP parameters via an
SCP buffer transfer to the firmware.
The CA0132 chip parameter init code is added in ca0132_init_params().
In chipio_[write,read]_data(), the current chip address is auto-incremented
if no error has occurred.
ca0132_select_out() selects the current output. If autodetect is enabled,
use headphones (if jack detected) or speakers (if no jack).
ca0132_select_mic() selects the current mic in. If autodetect is enabled,
use exterior mic (if jack detected) or built-in mic (if no jack).
Init digital mic and switch between dmic and amic with ca0132_init_dmic(),
ca0132_set_dmic(). amic2 is initialized in ca0132_init_analog_mic2().
Finally, add verb tables for configuring DSP firmware.

Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 16:57:56 +01:00
Ian Minett ef6b2eada3 ALSA: hda/ca0132: Add new definitions and structs for DSP
This patch adds definitions and structs used for configuring DSP effects,
virtual nodes, effect tuning controls, and mixer control helpers.
The effect structs are also initialized.

Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 16:57:42 +01:00
David Henningsson f038fcaca8 ALSA: hda - fix OOPS in hda_mark_cmd_cache_dirty
Obvious copy-paste error.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 15:32:46 +01:00
Takashi Iwai 6f54c36132 ALSA: hda/hdmi - Work around "alsactl restore" errors
When "alsactl restore" is performed on HDMI codecs, it tries to
restore the channel map value since the channel map controls are
writable.  But hdmi_chmap_ctl_put() returns -EBADFD when no PCM stream
is assigned yet, and this results in an error message from alsactl.
Although the error is harmless, it's certainly ugly and can be
regarded as a regression.

As a workaround, this patch changes the return code in such a case to
be zero for making others happy.  (A slight excuse is: when the chmap
is changed through the proper alsa-lib API, the PCM status is checked
there anyway, so we don't have to be too strict in the kernel side.)

Cc: <stable@vger.kernel.org> [v3.7+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 14:55:16 +01:00
Takashi Iwai 9b473e8516 ALSA: hda/sigmatel - Remove superfluous fields from sigmatel_spec
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 11:22:52 +01:00
Takashi Iwai 8c698fe210 ALSA: hda/sigmatel - Move w/a for HP Mini 110 LED to fixup table
Instead of checking the codec SSID in find_mute_led_cfg() for HP Mini
110, set the proper spec->default_polairty in the fixup table.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 11:22:36 +01:00
Takashi Iwai 89bb3e74b1 ALSA: hda/sigmatel - Remove PCI id check in find_mute_led_cfg()
The PCI vendor ID check in find_mute_led_cfg() is now superfluous
because the function is called in the fixup table entries of HP
machines.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 11:12:18 +01:00
Takashi Iwai 372f8c7502 ALSA: hda - Use standard fixup table for IDT92HD83xxx
Finally all codecs in patch_sigmatel.c have been converted to use the
standard fixup helpers.  This change also includes trivial cleanups
like the call of common setup for GPIO LED or the removal of unused
function.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 11:09:26 +01:00
Takashi Iwai 55e30141d8 ALSA: hda - Use standard fixup table for IDT92HD73xx
This one is rather a simple conversion.  The fixups for Dell machines
are implemented by fixup functions in the end.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 11:09:24 +01:00
Takashi Iwai 0f6fcb73c0 ALSA: hda - Use standard fixup table for IDT92HD71Bxx
This time, the only intrusive changes are for HP machines.
As the mute LED fixup and the bass speaker switch are required only
for HP machines, we can move these checks into the fixup entries; the
former is applied generically to all HP machines while the latter for
only certain models.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 11:09:16 +01:00
Takashi Iwai 52fd5cbc9b ALSA: hda - Check pincap while parsing the configuration
Sometimes (or rather often) BIOS sets the pin default configurations
obviously wrongly.  Looking through these failures, one common pattern
is to enable some dead pins that are usually marked as speaker pins.
In such a case, we can skip them if the pins don't have the output
capability.

In this patch, add a check for the valid pin cap bit for each parsed
pin, and filter out when it's invalid.

The fix was originally suggested by Raymond Yau.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 08:49:09 +01:00
Takashi Iwai 29ac83635f ALSA: hda - Use standard fixup table for STAC927x
This conversion is a bit tricky.  Since STAC927x may take two
different volume-knob initialization values depending on the model, a
new flag, spec->volknob_init, is introduced to indicate whether it's
the standard volume-knob initialization or not.

Also, Dell BIOS model is now directly mapped onto the fixup table
instead of parsing in the function.  This resulted in a new model ref,
STAC_927X_DELL_BIOS_SPDIF, which is a chained entry.

Also, for reducing the fixups, virtual entries like
STAC_927X_DELL_DMIC and STAC_D965_VERBS are introduced.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 08:21:50 +01:00
Takashi Iwai 0a4278464e ALSA: hda - Use standard fixup table for STAC922x
Rather straightforward conversion, except for ones for Intel Mac.
As Intel Mac have only unique codec SSIDs, we need to remap the fixup
again for the codec SSID and call the new fixup there.

Also, we can reduce model enums like STAC_MACMINI, which are model
aliases for backward compatibility, since they can be pointed directly
via hda_model_fixup table.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-15 08:20:06 +01:00
Takashi Iwai fe6322ca66 ALSA: hda - Use standard fixup table for STAC9205
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 18:14:58 +01:00
Takashi Iwai fc268c10ca ALSA: hda - Use standard fixup table for STAC9872
Now for STAC9872.  It has a small fixup table, fortunately.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 18:14:49 +01:00
Takashi Iwai d2077d40cb ALSA: hda - Use standard fixup table for STAC925x
Similar like the previous commit, convert patch_stac925x() to use the
standard fixup table.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 14:21:16 +01:00
Takashi Iwai d39a3ae821 ALSA: hda - Use standard fixup table for STAC9200
Convert patch_stac9200() to use the standard fixup table instead of
manual switch-case with board_config.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 14:21:10 +01:00
Takashi Iwai ac06e2298d Merge branch 'test/hda-gen-parser' into test/hda-migrate
* test/hda-gen-parser:
  ALSA: hda - Add capture_switch_hook to generic parser
2013-01-14 12:36:18 +01:00
Takashi Iwai ae177c3fd0 ALSA: hda - Add capture_switch_hook to generic parser
Add a hook for the capture mixer switch.  This will be used by IDT
codecs for controlling the mic-mute LED.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 12:13:06 +01:00
Eldad Zack 39e95156b9 ALSA: usb-audio: selector map for M-Audio FT C400
Add names of the clock sources for the M-Audio Fast Track
C400.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:06:11 +01:00
Eldad Zack 83e3acd494 ALSA: usb-audio: M-Audio FT C400 skip packet quirk
Attain constant real-world latency by skipping 16 data packets.
The number of packets to be skipped was found by trial and error.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:06:03 +01:00
Eldad Zack 2aad272b3f ALSA: usb-audio: correct M-Audio C400 clock source quirk
Taking another look at the C400 descriptors, I see now that there is
a clock selector (0x80) for this device.
Right now, the clock source points to the internal clock (0x81), which
is also valid. When the external clock source (0x82) is selected in the
mixer, and the rates mismatch (if it's free-running it is fixed to
48KHz), xruns will occur.

Set the clock ID to the clock selector unit (0x81), which then
allows the validation code to function correctly.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:05:57 +01:00
David Henningsson b98ae2729d ALSA: usb - fix race in creation of M-Audio Fast track pro driver
A patch in the 3.2 kernel caused regression with hotplugging the
M-Audio Fast track pro, or sound after suspend. I don't have the
device so I haven't done a full analysis, but it seems userspace
(both udev and pulseaudio) got confused when a card was created,
immediately destroyed, and then created again.

However, at least one person in the bug report (martin djfun)
reports that this patch resolves the issue for him. It also leaves
a message in the log:
"snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is
a bit misleading. It is better than non-working audio, but maybe
there's a more elegant solution?

BugLink: https://bugs.launchpad.net/bugs/1095315
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:03:03 +01:00
Stephen Rothwell c890caee54 ASoC: ak4642: remove __devinitconst annotation
CONFIG_HOTPLUG is always true now and the __dev* macros have been removed.

Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-14 13:52:21 +09:00
Kuninori Morimoto 9e7b6d60d8 ASoC: fsi: add device tree support
Support for loading the Renesas FSI driver via devicetree.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-14 08:27:18 +09:00
Lucas Stach 609dad9bdf ASoC: tegra: add ac97 host driver
This adds the driver for the Tegra 2x AC97 host controller.

Signed-off-by: Lucas Stach <dev@lynxeye.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-14 08:21:04 +09:00
Kuninori Morimoto a4a2992c53 ASoC: simple-card: add asoc_simple_dai for initializing
Current simple-card driver calls asoc_simple_card_dai_init()
if platform had a asoc_simple_card_dai_init pointer.
And, this initialization function works only
when platform has an applicable initial value for each dai settings.
And basically, almost all sound card requires certain initialization.
This means that almost all platform has initialization settings,
and driver do nothing if it doesn't have settings.

And additionally, current simple-card supports sysclk settings but it was
only for codec.  In order to abolish deviation between cpu and codec,
and in order to simplify processing,
this patch adds asoc_simple_dai, and removed pointless
struct asoc_simple_dai_init_info which was trigger of
calling asoc_simple_card_dai_init().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-14 06:55:43 +09:00
Mark Brown 2eebcef31a Merge branch 'topic/fsi' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-simple-card 2013-01-14 06:55:33 +09:00
Peter Ujfalusi f4d8ada2a0 ASoC: tlv320dac33: Remove suspend/resume soc driver operations
With idle_bias_off these are no longer needed.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-13 21:34:47 +09:00
Mark Brown f48aa39221 Merge remote-tracking branch 'asoc/fix/arizona' into asoc-arizona 2013-01-13 21:34:01 +09:00
Mark Brown d37fb92326 Merge remote-tracking branch 'asoc/topic/adsp' into asoc-arizona 2013-01-13 21:33:03 +09:00
Mark Brown 57a10a1fc3 ASoC: wm5110: Provide MICSUPP widget for regulator driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-13 21:31:04 +09:00
Mark Brown 55e7276e93 ASoC: wm5102: Provide MICSUPP widget for regulator driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-13 21:31:03 +09:00
Mark Brown 1023dbd90c ASoC: wm_adsp: Add basic firmware selection support
There are many firmwares available for ADSP devices. Add basic support
for selecting between them, including a couple of feature sets in the
set of available firmware to start off with.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-12 18:36:07 +00:00
Takashi Iwai b3f6008f2d ALSA: hda - Use generic parser for VIA codec driver
Yet another step forward.  As all features for VIA codecs have been
implemented in the generic driver, we can move on to migrate the VIA
codec parser, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:45:02 +01:00
Takashi Iwai 78bb3cb0e2 ALSA: hda - Add generic parser support to Analog Device codec driver
This patch adds the support for the generic auto-parser to AD codec
driver.  For AD1988, the old code is replaced simply with the new
generic parser.  For other codecs, new model "auto" is added and
directed to use the generic parser.

No fixup codes have been implemented yet as of now.  Eventually we'd
replace each static quirk with the generic parser + fixup.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:45:00 +01:00
Takashi Iwai bf92d1d503 ALSA: hda - Rearrange for dropping static quirk codes in Coexant driver
Just shuffle the codes and add ifdefs for testing to drop the static
quirk codes from patch_conexant.c.

By commenting out ENABLE_CXT_STATIC_QUIRKS define at the beginning of
the file, you can disable the whole static codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:58 +01:00
Takashi Iwai aed523f193 ALSA: hda - Use generic parser in Conexant codec driver
... and drop most of own parser code.

It doesn't replace any present static quirks, though.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:56 +01:00
Takashi Iwai 1077a02481 ALSA: hda - Use generic parser for Cirrus codec driver
This time, the target is Cirrus codec.  Its parser is a subset of
generic parser, so we can migrate fully with it now.

The only tricky part is the handling of SPDIF automute.
Cirrus driver sets the SPDIF out plug over the headphone.  As a
workaround, set spec->gen.master_mute for toggling the headphone (and
other) mute.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:55 +01:00
Takashi Iwai 8fadf1da3f ALSA: hda - Use generic parser for CA0110 codec
CA0110 codec is a fairly straightforward hardware implementation,
and we can use the generic parser almost as is.
Just set spec->multi_cap_vol flag to follow the current behavior.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:53 +01:00
Takashi Iwai b060fb0eef ALSA: hda - Use generic codec parser for C-Media codecs
Replace the old parser code for C-Media auto-parser with the latest
generic parser.  For compatibility reason, the static bindings are
still left, but they could be cleaned up in future.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:52 +01:00
Takashi Iwai 84721e81fa ALSA: hda - Remove superfluous kconfig depends
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:50 +01:00
Takashi Iwai 1c70a58341 ALSA: hda - Allow user to give hints for codec parser behavior
Through the hints via sysfs or patch, user can set specific behavior
flags for the generic parser now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:48 +01:00
Takashi Iwai bc759721fb ALSA: hda - Add snd_hda_get_int_hint() helper function
It'll be used in hda_generic.c, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:47 +01:00
Takashi Iwai 09b70e8509 ALSA: hda - Protect user-defined arrays via mutex
The pincfgs, init_verbs and hints set by sysfs or patch might be
changed dynamically on the fly, thus we need to protect it.
Add a simple protection via a mutex.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:45 +01:00
Takashi Iwai 08fb0d0ee1 ALSA: hda/realtek - Generic mute LED implementation for HP laptops
As David Henningsson recently suggested, some HP laptops use an unused
mic pin for controlling a mute LED, and this information is provided
via DMI string "HP_Mute_LED_X_Y" string.  This patch adds the generic
support for such cases, as we've already done in patch_sigmatel.c.
This is applied generically to all devices with ID 0x103c.

But as we don't know whether the device 103c:1586 really contains
HP_Mute_LED_X_Y DMI string, still keep the static setup for this
device using the mic2 pin 0x19.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:43 +01:00
Takashi Iwai 9bb1f06fe0 ALSA: hda/realtek - Fix the timing for some fixups
Some fixups such as setting the flags influencing on the parser
behavior should be applied before actually parsing the tree.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:42 +01:00
Takashi Iwai 39aedee7a1 ALSA: hda/realtek - Add a fixup for FSC S7020 laptop
Try to recover from the regression: set the HP amp for the speaker and
add the hp jack mode enum as default.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:40 +01:00
Takashi Iwai 978e77e78c ALSA: hda - Add output jack mode enum controls
Add the enum controls for changing the headphone amp bits of output
jacks, such as "Headphone Jack Mode".  This feature isn't enabled as
default, so far, unless spec->add_out_jack_modes flag is set.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:38 +01:00
Takashi Iwai a365fed980 ALSA: hda - Update automute / automic upon jack retasking
When a multi-io jack is switched to another direction, call the
automute and autoswitch update functions, as this jack won't be used
as the headphone or the mic jack that may turn off others.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:37 +01:00
Takashi Iwai fd1082159d ALSA: hda - Add a new fixup type to override pinctl values
Add a new fixup type, HDA_FIXUP_PINCTLS, for overriding the pinctl
values of the given pins.  It takes the same array of struct pintbl
like HDA_FIXUP_PINS, but each entry contains the pinctl value instead
of the pin default config value.

This patch also replaces the corresponding codes in patch_realtek.c.
Without this change, the direct call of verbs may be overridden again
by the later call of pinctl restoration by the driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:35 +01:00
Takashi Iwai d3f02d60ee ALSA: hda/realtek - Read the cached pinctl value in fixups
... instead of reading the value from the codec at each time.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:33 +01:00
Takashi Iwai 1727a771b4 ALSA: hda/realtek - Drop aliases for old fixups
Now the whole codebase has been changed from the earlier kernels, it
makes little sense to keep these aliases.  Simply replace with the
official names.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:32 +01:00
Takashi Iwai 0b4df931ce ALSA: hda - Avoid auto-mute or auto-mic of retasked jacks
When a jack is retasked as a different direction (e.g. a mic jack is
used as a CLFE output), such a jack shouldn't be counted as the target
for the automatic jack switching.  Skip the automute or the autoswitch
when the current pinctl direction is different from what we suppose.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:30 +01:00
Takashi Iwai 2c12c30d3f ALSA: hda - Manage current pinctl values in generic parser
Use the new pin target accessors for managing the current pinctl
values in the generic parser.  The pinctl values of all active pins
are once determined at the initialization phase, and stored via
snd_hda_codec_set_pin_target().  This will be referred again in the
codec init or resume phase to set the actual pinctl.

This value is kept while the auto-mute.  When a line-out or a speaker
pin is muted by auto-mute, the driver simply disables the pin, but it
doesn't touch the cached pinctl target value.  Upon unmute, this value
is used to restore the original pinctl in return.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:28 +01:00
Takashi Iwai 62f3a2f718 ALSA: hda - More strict correction of invalid pinctl bits
Check more strictly about the validity of pinctl values in
snd_hda_set_pin_ctl() and correct the wrong bits automatically.
Also provide the helper function to correct pinctl bits to codec
drivers.

This automatically fixes the invalid pinctl writes that are found in
a few Realtek fixups for NID 0x0f amp like ASUS A6Rp.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:27 +01:00
Takashi Iwai d7fdc00ae5 ALSA: hda - Add helper functions to cache the current pinctl target
We already have the list of whole pin widgets and there is an unused
space in the list; let's use it for caching the current pinctl value.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:25 +01:00
Takashi Iwai 980428cecc ALSA: hda - Clear the dropped paths properly
When a DAC is reassigned from surrounds to front or ADCs are reduced
due to incomplete imux, we clear the path indices but the path
instances remain as is.  Since the paths might be still referred
through the whole path list parsing (e.g. is_active_nid()), we should
clear these path instances as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:23 +01:00
Takashi Iwai f3fc0b0b1f ALSA: hda - Allow aamix as a capture source
Since some codecs can choose the aamix as a capture source, we should
support it as well.  When spec->add_stereo_mix_input flag is set, the
parser checks the availability of aamix as the input source, and adds
the paths automatically when possible.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:21 +01:00
Takashi Iwai 3a65bcdc57 ALSA: hda - Fix inconsistent input_paths after ADC reduction
In the current parser code, the input_paths[] may become inconsistent
when some of detected ADCs are dropped due to incomplete inputs, since
the driver rearranges only adc_nids[] but doesn't touch input_paths[].

This patch fixes the issue, and also it optimizes the reachability
checks by simply referring to the parsed input_paths[] instead of
calling is_reachable() again for each connection.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:20 +01:00
Takashi Iwai 54d778b31c ALSA: hda - Return "Headphone Mic" from hda_get_autocfg_input_label()
Instead of handling special cases in the caller side, give a proper
name string "Headphone Mic" from hda_get_autocfg_input_label() when
the headhpone jack pin is specified as an input.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:18 +01:00
Takashi Iwai ca29683bd6 ALSA: hda - Exclude aamix from capture paths
The capture paths shouldn't contain the analog loopback mixer.
Pass a proper argument to exclude the aamix NID.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:16 +01:00
Takashi Iwai d12daf6f41 ALSA: hda - Add a flag to suppress mic auto-switch
Add a new flag spec->suppress_mic_auto_switch for codecs that don't
support unsol events properly like VT1708.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:15 +01:00
Takashi Iwai fb690cf582 ALSA: hda - Handle BOTH jack port as a fixed output
When the default config value shows the connection AC_JACK_PORT_BOTH,
it's better to handle it as a speaker pin.  This makes the behavior
consistent in snd_hda_get_pin_label() and snd_hda_parse_pin_defcfg().

There are only few old machines showing this attribute, and all of
them are actually fixed speaker pins, as far as I know.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:13 +01:00
Takashi Iwai 3ca529d339 ALSA: hda - Re-define snd_hda_parse_nid_path()
This commit modifies the definition of snd_hda_parse_nid_path()
slightly, now with_aa_mix argument is changed to anchor_nid, so that
it can handle any NID generically as an anchor point to include or
exclude.

The with_aa_mix field in struct nid_path is removed again by this
change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:11 +01:00
Takashi Iwai c697b71685 ALSA: hda - Manage input paths via path indices
... like we did for output and loopback paths.
It makes the code slightly easier.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:09 +01:00
Takashi Iwai a07a949be6 ALSA: hda - Fix multi-io channel mode management
The multi-io channels can vary not only from 1 to 6 but also may vary
from 6 to 8 or such.  At the same time, there are more speaker pins
available than the primary output pins.  So, we need three variables
to check: the minimum channel counts for primary outputs, the current
channel counts for primary outputs, and the minimum channel counts for
all outputs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:08 +01:00
Takashi Iwai affdb62b81 ALSA: hda - Don't set up active streams twice
We don't have to set up a stream that has been already set up
previously.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:06 +01:00
Takashi Iwai 50b1548775 ALSA: hda - Remove unused dac reference in create_multi_out_ctls()
Remove useless code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:04 +01:00
Takashi Iwai 0e614dd058 ALSA: hda - Use direct path reference in assign_out_path_ctls()
Instead of looking through paths with the dac -> pin connection at
each time, just pass the already parsed path index to
assign_out_path_ctls().  This simplifies the code a bit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:03 +01:00
Takashi Iwai cd5be3f9de ALSA: hda - Clear path indices properly at each re-evaluation
The path indices must be reset at each evaluation of DAC assignment.
Otherwise the badness value will be wrongly calculated and mixers may
be inconsistently assigned.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:44:01 +01:00
Takashi Iwai 5187ac168d ALSA: hda - Add brief comments to exported snd_hda_gen_*_() functions
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:59 +01:00
Takashi Iwai dd5e720304 ALSA: hda - Remove dead HDA_CTL_BIND_VOL and HDA_CTL_BIND_SW codes
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:57 +01:00
Takashi Iwai fce52a3bb1 ALSA: hda - Add snd_hda_gen_free() and snd_hda_gen_check_power_status()
Just to remove duplicated codes.
Also fixed EXPORT_SYMBOL() in hda_generic.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:56 +01:00
Takashi Iwai 76a19c69d9 ALSA: hda - Allow jack detection when polling is enabled
Let is_jack_detectable() return true when the jack polling is enabled
for the codec.  VT1708 uses the polling explicitly so we need to allow
it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:54 +01:00
Takashi Iwai e6b85f3c9d ALSA: hda - Add pcm_playback_hook to hda_gen_spec
The new hook which is called at each PCM playback ops.
It can be used to control the codec-specific power-saving feature in
each codec driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:52 +01:00
Takashi Iwai c2c803830a ALSA: hda - Drop bind-volume workaround
The bind-volume workaround was introduced for simplifying the mixer
abstraction in the case where one or more pins of multiple outputs
lack of individual volume controls.  This was essentially the case
like Acer Aspire 5935, which has 5.1 speakers and 5.1 (multi-io)
jacks although there are 5 DACs, so some of them must share a DAC.

However, the recent code rewrite changed the DAC assignment policy to
share with the same channel instead of binding to the front, thus
binding the volumes for all channels makes little sense now, rather
it's confusing.  So in this patch, the ugly workaround is finally
dropped and simply create the volume control corresponding to the
parsed path position.

For dual headphones or 2.1 speakers with a shared volume control, it's
anyway bound to the same DAC if needed, so this change shouldn't bring
any practical difference.

And, as a good bonus, we can cut off the whole code handling the bind
volume elements.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:51 +01:00
Takashi Iwai d4156930b2 ALSA: hda - Drop unneeded pin argument from set_output_and_unmute()
Just a minor refactoring.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:49 +01:00
Takashi Iwai ee79c69ac7 ALSA: hda - Add missing slave names for Speaker Surround, etc
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:47 +01:00
Takashi Iwai 7385df6134 ALSA: hda - Prefer binding the primary CLFE output
When 5.1 or more multiple speakers with found but not enough DACs are
available, it's better to bind such pins to the DACs of the primary
outputs with the same channels rather than binding to the first DAC
(i.e. the front channel).  For the cases with two speaker pins, it's
rather regarded as front + bass combination, thus it's more practical
to still bind to the front, though.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:45 +01:00
Takashi Iwai 5abd4888f6 ALSA: hda - Fix truncated control names
... like "Speaker Surround Playback Switch".
This fix had been already applied to patch_conexant.c but was
forgotten in other places, then migrated to hda_generic.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:44 +01:00
Takashi Iwai c30aa7b242 ALSA: hda - Add Loopback Mixing control
For codecs that have individual routes going through a loopback mixer
(like VIA codecs), we need to provide an explicit switch to choose
whether the output goes through mixer or directly from DAC.

This patch adds the check for such paths and creates "Loopback Mixing"
enum control when available.

It won't influence on codecs like Realtek or others where the loopback
mixer is connected independently from the primary output routes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:42 +01:00
Takashi Iwai 117688a9c1 ALSA: hda - Correct aamix output paths
The output paths including aamix should be parsed only for the first
output.  The surround paths including aamix must be wrong, since it
would mix all streams, i.e. all channels would be mixed into a single
and multiplexed again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:40 +01:00
Takashi Iwai 2430d7b78b ALSA: hda - Initialize digital-input path properly
Call the path activation for the digital input pin properly, not only
setting the pin control.  Also add spec->digin_path for keeping the
path index.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:38 +01:00
Takashi Iwai 196c176680 ALSA: hda - Manage using output/loopback path indices
Instead of search for the path with the certain route at each time,
keep the path index for each output and loopback, and just use it when
referred.

In this implementation, the path index number begins with one, not
zero (although I've been writing in C over decades).  It's just to
make the check for uninitialized values easier.

So far, the input paths aren't handled with indices yet, but still
picked up via snd_hda_get_nid_path() at each time.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:37 +01:00
Takashi Iwai 05453b7e97 ALSA: hda - Fix multi-io pin assignment in create_multi_out_ctls()
The multi-io pins are calculated with a blind assumption of
cfg->line_outs = 1.  This isn't always true.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:35 +01:00
Takashi Iwai e22aab7dcf ALSA: hda - Simplify the multi-io assignment with multi speakers
When speakers are chosen as the the primary output during evaluation,
we did some tricks to assign the possible multi-io jacks with a
certain offset value to multi_out dacs.  This was a workaround for the
case with multiple speakers like Acer Aspire.  But this is quite ugly
at the same time and the resultant code is hard to understand.  More
badly, it works wrongly for 2.1 speakers like Apple iMac91.

In this patch, instead of fiddling with the offset to multi_out dacs,
simply add a certain badness number if headphone(s) + multi-ios are
possible.  This simplify the code a bit, and it's more robust.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:33 +01:00
Takashi Iwai f5172a7ed9 ALSA: hda - Check the existing path in snd_hda_add_new_path()
If the requested path has been already added, return the existing path
instance instead of adding a duplicated instance.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:31 +01:00
Takashi Iwai 1e0b528696 ALSA: hda - Avoid duplicated path creations
When the paths are created in map_singles(), we don't have to
re-create new paths in try_assign_dacs().  Just evaluate the badness
and skip to the next item.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:30 +01:00
Takashi Iwai e1284af730 ALSA: hda - Initialize output paths with current active states
Set path->active flag at the path creation time and let the paths
initialized according to the current path->active state in
set_output_and_unmute().  This allows to modify the active flag of
some output paths dynamically, e.g. switching the front output route
with or without aamix like patch_via.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:28 +01:00
Takashi Iwai 985803ca91 ALSA: hda - Don't skip amp init for activated paths
activate_amp() in the generic parser checks whether the given NID is
included in any active paths and skips it if found.  This was a
workaround for avoiding disabling the widgets in the active paths when
one path is disabled, thus it shouldn't be applied to the case for
path activation.  Due to this wrong check, some analog loopback paths
haven't been initialized correctly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:26 +01:00
Takashi Iwai 2e03e9528d ALSA: hda - Add hooks for HP/line/mic auto switching
... as a preliminary work for migrating patch_sigmatel.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:24 +01:00
Takashi Iwai ee8e765b0b ALSA: hda - Revive snd_hda_get_conn_list()
Manage the connection list cache using linked lists instead of
snd_array, and revive snd_hda_get_conn_list() again, so that we don't
have to keep the expanded values locally.
This will reduce the stack usage by recursive call of
snd_hda_get_conn_index() or parse_nid_path() of the generic parser.

The list management doesn't include any mutex protection, thus the
caller needs to take care of race appropriately.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:23 +01:00
Takashi Iwai 9cc159c664 ALSA: hda - Add codec->inv_jack_detect flag
Yet another broken hardware workaround: there are hardware indicating
the inverted jack detection bit result.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:21 +01:00
Takashi Iwai ecac3ed174 ALSA: hda - Add inv_eapd flag to struct hda_codec
Add the new flag, codec->inv_eapd, indicating that the EAPD
implementation is inverted.

There are always broken hardware in the world.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:43:19 +01:00
Takashi Iwai 38cf6f1a41 ALSA: hda - Implement independent HP control
Similar like the implementation in patch_analog.c and patch_via.c,
the generic parser can provide the independent HP PCM stream now.
It's enabled when spec->indep_hp is set by the caller while parsing.

Currently no dynamic PCM switching as in patch_via.c is implemented
yet.  The control returns -EBUSY when the value is changed during PCM
operations.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:42:56 +01:00
Takashi Iwai b3a8c74522 ALSA: hda - Allow aamix in the primary output path
Allow the path including the loopback mixer widget in the primary
output channel as an alternative in the generic codec parser.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:30 +01:00
Takashi Iwai 4ac0eefa76 ALSA: hda - Define HDA_PARSE_* for snd_hda_parse_nid_path() argument
... instead of numbers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:28 +01:00
Takashi Iwai 708122e836 ALSA: hda - Fix typos in debug_show_configs()
It never showed the 4th line out and headphone pins since quite ago.
Oh well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:27 +01:00
Takashi Iwai 0c8c0f56e6 ALSA: hda - Add more debug prints about new paths
Add a better debug print code to show the new assigned paths in
generic parser.  It appears only with CONFIG_SND_DEBUG_VERBOSE=y.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:25 +01:00
Takashi Iwai 545502de54 ALSA: hda - Drop spec->channel_mode field from hda_gen_spec
It's never used in the generic parser.  It was there from the old
Realtek code, which has been dropped quite ago, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:24 +01:00
Takashi Iwai f873e536b6 ALSA: hda - Fix PCM name string for generic parser
When a PCM name string is generated from the chip name, it might
become strange like "CX20549 (Venice) Analog".  In this patch, the
parser tries to drop the invalid words like "(Venice)" in the PCM name
string.  Also, when the name string is given beforehand by the caller,
respect it and use it as is.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:22 +01:00
Takashi Iwai 7594aa3396 ALSA: hda - Use cached version for changing pins in hda_generic.c
There is no reason to avoid snd_hda_set_pin_ctl_cache() there.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:20 +01:00
Takashi Iwai d5a9f1bb38 ALSA: hda - Dynamically turn on/off EAPD in generic codec driver
When spec->own_eapd_ctl isn't set, try to turn on/off EAPD on demand
for each pin.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:19 +01:00
Takashi Iwai 64049c81df ALSA: hda - Fix initialization of primary outputs in hda_generic.c
There were some old codes that look not stable enough, which was
derived from the old Realtek code.  The initialization for primary
output in init_multi_out() needs to consider the case of shared DAC.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:17 +01:00
Takashi Iwai db23fd193d ALSA: hda - Refactor init_extra_out() in hda_generic.c
Just a small clean up by splitting a function.
No functional changes at all.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:16 +01:00
Takashi Iwai 973e4972f9 ALSA: hda - Clear unsol enable bits on unused pins in generic parser
For preliminary works to migrate the generic parser for Conexant
codecs: the same function is ported to hda_generic.c.
But now it looks through the jack detect table so that it can cover
better.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:14 +01:00
Takashi Iwai fd25a97a97 ALSA: hda - Add spec->vmaster_mute_enum flag to generic parser
Add a flag to indicate whether the vmaster mute hook enum is exposed
or not.  Conexant codecs may want not to expose the control depending
on the model.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:12 +01:00
Takashi Iwai 406b285da3 ALSA: hda - Begin HDA_GEN_* event tag from 1
... to distinguish from the invalid event type.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:11 +01:00
Takashi Iwai d94ddd85b1 ALSA: hda - Increase the max depth of widget connections
Old codecs like AD1986A tend to have long paths as they were just made
to be the way like AC97.  The current max depth 5 can be too short for
such devices.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:09 +01:00
Takashi Iwai 2ce4886abc ALSA: hda - Avoid access of amp cache element outside mutex
The access to a cache array element could be invalid outside the
mutex, so copy locally for the later references.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:07 +01:00
Takashi Iwai 8565f052c5 ALSA: hda - Fix wrong dirty check in snd_hda_codec_resume_amp()
The dirty entry has to be checked at the beginning in the loop, not in
the inner loop for channels.  This caused a regression that the right
channel isn't properly written.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:06 +01:00
Takashi Iwai 3bbcd274c2 ALSA: hda - Do sequential writes in snd_hda_gen_init()
This would reduce the number of actually executed verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:04 +01:00
Takashi Iwai 47d46abba2 ALSA: hda - Add / fix comments about capture vol/sw controls in hda_generic.c
A bit of details won't hurt.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:02 +01:00
Takashi Iwai 84e3908dc8 ALSA: hda - Add missing amp cache flush for bound capture vol/sw ctls
The bound capture volume and switch controls use the cached amp
updates, but it's missing the flushing at the end.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:34:01 +01:00
Takashi Iwai 0c3d47b007 ALSA: hda - Add snd_hda_codec_flush_*_cache() aliases
It makes easier to understand although these are identical with
snd_hda_codec_resume_*() functions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:33:59 +01:00
Takashi Iwai c4f3ebed3c ALSA: hda - Flush dirty amp caches before writing inv_dmic fix
The inverted dmic fix overwrites the right channel amp value, but it
would work only when the amp values have been already actually
written.  Put snd_hda_codec_resume_amp() before the amp write for
flushing caches.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:33:57 +01:00
Takashi Iwai 3bcce5c0d9 ALSA: hda - Check CORB overflow
Add an overflow check of CORB in HD-audio controller and codec drivers
so that flood of sequential writes would work properly.
In the controller side, add a check of CORB read-pointer to make
returning -EAGAIN when it's full.  Meanwhile in the codec side, when
-EAGAIN error is received, it retries the write after flushing the
pending verbs (calling get_response() essentially does it).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:33:56 +01:00
Takashi Iwai aa88a3553e ALSA: hda - Clear cached_write flag in snd_hda_codec_resume_*()
These functions are supposed to be called at finishing the cached
sequential writes, so clear the flag properly for lazy developers who
often forget details.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:33:54 +01:00
Takashi Iwai de1e37b7d0 ALSA: hda - Clear dirty flag upon cache write
When verbs or amps are actually written to the hardware, we can clear
dirty flag so that the later snd_hda_codec_resume_*() calls can skip
these verbs / amps.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:33:43 +01:00
Takashi Iwai 5fdaecdb0d ALSA: hda - Allow one chance for zero NID in connection list
The commit [2e9bf24: ALSA: hda_codec: Check for invalid zero
connections] trims the whole connection list when an invalid value is
reported by the hardware.  But some codecs (at least AD1986A) may give
a zero NID in the middle of the connection list, so dropping the whole
list isn't good for such cases.

In this patch, as a workaround, allow a single zero NID in the read
connection list.  If it hits zero twice, it's handled as an error, so
that we can avoid "too many connections" errors.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:31:06 +01:00
Takashi Iwai 624d914d09 ALSA: hda - Use "Capture Source" for single sources
In general we prefer "Capture Source" to "Input Source".
The latter was chosen in many places just because "Capture Source"
label doesn't work well with the current alsa-lib mixer abstraction
when multiple instances are present.  But when we know that there is a
single input-source element, we can safely choose "Capture Source"
label.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:31:04 +01:00
Takashi Iwai 08c189f2c5 ALSA: hda - Use generic parser codes for Realtek driver
The next migration step is to use the common code in generic driver
for Realtek driver.  This is no drastic change and there should be no
real functional changes, as the generic parser code comes from Realtek
driver originally.

As Realtek driver requires the generic parser code, it needs a
reverse-selection of CONFIG_SND_HDA_GENERIC kconfig.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:31:03 +01:00
Takashi Iwai 5d550e15be ALSA: hda - Export standard jack event handlers for generic parser
These handlers are supposed to be called externally from the codec
drivers once when they need to handle own jack events.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:31:01 +01:00
Takashi Iwai 36502d0200 ALSA: hda - Fix NULL dereference in snd_hda_gen_build_controls()
When no controls are assigned in the parser (e.g. no analog path),
spec->kctls.list is still NULL.  We need to check it before passing to
snd_hda_add_new_ctls().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:59 +01:00
Takashi Iwai 9eb413e5e4 ALSA: hda - Move the call of snd_hda_parse_pin_defcfg() from snd_hda_gen_parse_auto_config()
In some cases, we want to manipulate the auto_pin_cfg table before
passing to snd_hda_gen_parse_auto_config() (e.g. Realtek SSID check
code fiddles with the headphone pin).   Also passing ignore_pins just
for snd_hda_parse_pin_defcfg() isn't good.

In this patch, snd_hda_gen_parse_auto_config() is changed to receive
the auto_pin_cfg table to be parsed.  The passed auto_pin_cfg table
must have been initialized (typically by calling
snd_hda_gen_parse_auto_config()) beforehand by the caller.

Also together with this change, spec->parse_flags is also removed.
Since this was referred only at the place calling
snd_hda_parse_pin_defcfg(), no longer needed to be kept in spec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:58 +01:00
Takashi Iwai 12c93df60c ALSA: hda - Export snd_hda_gen_add_kctl()
It may be used in other codec drivers, so let it free.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:56 +01:00
Takashi Iwai 731dc3019c ALSA: hda - Add EAPD control to generic parser
Enable EAPD in output path initializations automatically unless the
new flag spec->own_eapd_ctl is set.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:55 +01:00
Takashi Iwai 352f7f914e ALSA: hda - Merge Realtek parser code to generic parser
Finally the whole generic parser code in Realtek driver is moved into
hda_generic.c so that it can be used for generic codec driver.
The old dumb generic driver is replaced.  Yay.

The future plan is to adapt this generic parser for other codecs,
i.e. the codec driver calls the exported functions in generic driver
but adds some codec-specific fixes and setups.

As of this commit, the complete driver code is still duplicated in
Realtek codec driver.  The big code reduction will come from now on.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:53 +01:00
Takashi Iwai fdf52cab88 ALSA: hda/realtek - Remove redundant argument from alc_mux_select()
The argument "force" is always false in the recent code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:51 +01:00
Takashi Iwai ab16c6dd79 ALSA: hda - More generic auto-mic switching for Realtek codecs
This patch extends the capability of the auto-mic feature.
Instead of limiting the automatic input-source selection only to the
mics (internal, external and dock mics), allow it for generic inputs,
e.g. switching between the rear line-in and the front mic.

The logic is to check the attribute and location of input pins, and
enable the automatic selection feature only if all such pins are in
different locations (e.g. internal, front, rear, etc) and line-in or
mic pins.  That is, if multiple input pins are assigned to a single
location, the feature isn't enabled because we don't know the
priority.

(You may wonder why this restriction doesn't exist for the headphone
 automute.  The reason is that the output case is different from the
 input: the input source is an exclusive selection while the output
 can be multiplexed.)

Note that, for avoiding regressions, the line-in auto switching
feature isn't activated as default.  It has to be set explicitly via
spec->line_in_auto_switch flag in a fixup code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:50 +01:00
Takashi Iwai 5ec16d12c8 ALSA: hda - Rearrange INPUT_PIN_ATTR_*
Put INPUT_PIN_ATTR_FRONT after INPUT_PIN_ATTR_REAR, and define
INPUT_PIN_ATTR_LAST to point to the last element.

This is a preliminary work for cleaning up Realtek auto-mic parser.
In the auto-mic implementation, the front panel is preferred over the
rear panel.  By arranging the attr definitions like in this commit, we
can simply use sort() for figuring out the priority order.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:48 +01:00
Takashi Iwai 7e35dd3d6b ALSA: hda/realtek - Fix split stereo dmic code
The previous commit passed an utterly wrong value for checking the
split inv dmic pin.  This patch fixes it and also tries to remove
inv_dmic_split_idx field.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:46 +01:00
Takashi Iwai c9ce6b260b ALSA: hda - Move fixup code into struct hda_codec
Since the fixup code is used commonly, it's worth to move it to the
common place, struct hda_codec, instead of keeping in hda_gen_spec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:45 +01:00
Takashi Iwai 81fede89ed ALSA: hda/realtek - Add conexant-style inverted dmic handling
To make the parser more generic, a few codes to handle the inverted
stereo dmic in a way Conexant parser does is added in this patch.

The caller should set spec->inv_dmic_split flag appropriately.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:43 +01:00
Takashi Iwai 9bf387b612 ALSA: hda/realtek - Allow multiple individual capture volume/switch controls
So far we create only "Capture Volume" and "Capture Switch" controls
for binding all possible amps, but we'd prefer creating individual
capture volume and switch controls per input in some cases
(e.g. conexant parser does it).

Add a new flag, spec->multi_cap_vol, to follow that policy.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:41 +01:00
Takashi Iwai bc54976721 ALSA: hda/realtek - Allow passing name=NULL to alc_kcontrol_new()
This prevents stupid typos.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:40 +01:00
Takashi Iwai 2eab694a6c ALSA: hda/realtek - Merge a few split functions
Merge a few functions that have been split due to historical reasons
to single functions.  Splitting too much (and placing too far away)
actually worsens the readability.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:38 +01:00
Takashi Iwai 52a8efab10 ALSA: hda/realtek - Assign Master mixer when possible
There are a few more cases where we can assign "Master" mixer element
safely, e.g. when a single DAC is used in the whole output paths.

Also, when vmaster hook is present, avoid "Master" but assign "PCM"
instead.  Otherwise vmaster hook won't work properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:36 +01:00
Takashi Iwai 3bd7b644d0 ALSA: hda/realtek - Handle vmaster hook in the parser side
... so that the fixup just needs to set the hook function in
FIXUP_ACT_PROBE.  This will make easier to port for other codecs,
too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:35 +01:00
Takashi Iwai 20c18f562a ALSA: hda/realtek - Remove unused fields and macro definitions
Also arranged alc_spec definitions to optimize bit fields.
Use a bit field for spec->need_dac_fix, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:33 +01:00
Takashi Iwai 480967db6c ALSA: hda/realtek - Drop auto_mic_valid_imux flag
This flag is superfluous now and it's always as same as
spec->auto_mic.  Let's drop.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:32 +01:00
Takashi Iwai 37c0420765 ALSA: hda/realtek - Allow different pins for shared hp/mic vref check
Add a new field to indicate the possible pin NID for alternative vref
setup for the shared hp/mic.  Although 0x18 is valid for all Realtek
codecs, it'll be different on other vendor's codecs.

Also, drop the sanity check in update_shared_mic_hp() since the
reference pin is set explicitly in the caller side.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:30 +01:00
Takashi Iwai df1d1fb09a ALSA: hda/realtek - Parse digital input path
This was the last forgotten path.  Now it's parsed via the same path
parser.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:28 +01:00
Takashi Iwai 965ccebccd ALSA: hda/realtek - Rename add_new_out_path() with add_new_nid_path()
Make the function more generic for both input and output directions,
and returns the assigned path pointer.  The argument order is changed
to follow the standard (from, to) way.

Now this new function is used for analog input and loopback path
parser codes, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:27 +01:00
Takashi Iwai 62343997e4 ALSA: hda/realtek - Remove superfluous input amp init
The amps will be initialized via activate_path(), thus it's
superfluous to set in alc_auto_init_analog_input().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:25 +01:00
Takashi Iwai 27d3153651 ALSA: hda/realtek - Clean up some spec fields
Remove some fields from struct alc_spec, and clean up the usage.
Namely,
- spec->input_mux becomes a single element, private_imux[] is removed
- spec->adc_nids becomes an array by itself, and private_adc_nids[]
  gets removed, too

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:23 +01:00
Takashi Iwai 666a70d42b ALSA: hda/realtek - Make input path parser more generic
Now we reached to the final big piece of parser rewrite: the input
paths.  While the old parser code assumes the more-or-less direct and
similar connections from input pin to ADC, the new code handles the
complete input paths.  The capture source is switched by simple calls
of activate_path() function.

The parsing of capture volume and capture switches is, however, not
fully generalized.  It assumes that amps are available in the vicinity
of ADCs (in three depth).  This isn't perfect but it should cover all
codecs I know of.

Also, this commit removes some NID mapping of capture-related controls
temporarily for simplicity.  It'll be restored in later commits.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:22 +01:00
Takashi Iwai 183a444a6d ALSA: hda/realtek - Don't change connection at path deactivation
The widget connection selection must be changed only when the path is
enabled.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:20 +01:00
Takashi Iwai 829f69ea59 ALSA: hda/realtek - Initialize loopback paths properly
Now we have a complete list of loopback paths, thus we can initialize
the paths more completely based on it, instead of assuming a direct
connection from pin to mixer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:18 +01:00
Takashi Iwai 8dd4867858 ALSA: hda/realtek - Add boost volumes to path list
Don't forget to take boost volumes into account in the managed path
list.  Since it's an additional volume, we need to extend the ctls[]
array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:17 +01:00
Takashi Iwai 3ebf1e940a ALSA: hda/realtek - Add missing initialization of multi-io routes
The paths used for multi-io haven't been initialized properly, so
far.  It's usually no big matter because the pins are set to input as
default, but it's still cleaner to initialize the paths properly.

Now with the path active/inactive check, we can do it easily.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:15 +01:00
Takashi Iwai 0250f7cbea ALSA: hda/realtek - Fix the initialization of pin amp-in
The pin widget has only a single amp value for the input even if it
has multiple "sources".  Handle the situation in activate_path().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:14 +01:00
Takashi Iwai 6518f7ac51 ALSA: hda/realtek - Rename get_out_path() to get_nid_path()
The function can be used not only for output paths but generically.
Also swap the argument order.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:12 +01:00
Takashi Iwai fef7fbbc5d ALSA: hda/realtek - Use path-based parser for digital outputs
Similar like analog output paths, use the path list for parsing and
initializing digital outputs as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:10 +01:00
Takashi Iwai c9967f1cba ALSA: hda/realtek - Consolidate to a single path list
We don't have to keep three individual path lists for input, output
and loopback.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:09 +01:00
Takashi Iwai 9c64076e54 ALSA: hda/realtek - Consolidate is_reachable_path()
alc_auto_is_dac_reachable() can be replaced fully with
is_reachable_path().  The only difference is the order of arguments.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:07 +01:00
Takashi Iwai 130e5f0642 ALSA: hda/realtek - Add path active flag
... and rewrite the initialization of output paths as a generic
function that is applicable for both i/o directions.

The new flag, active, is introduced to each nid_path entry.  This
indicates whether the given path is active, and it's used for checking
whether a certain widget can be turned off or changed when a path is
no longer used or newly enabled.

It's still used only in the output paths.  More wider adaption for
input and loopback paths will be achieved in the later patch.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:05 +01:00
Takashi Iwai b8a47c79b2 ALSA: hda/realtek - Remove non-standard automute mode
We are using only AUTOMUTE_MODE_PIN in patch_realtek.c and all others
have been already dropped.  Let's remove the old superfluous codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:04 +01:00
Takashi Iwai 280e57d544 ALSA: hda - Introduce snd_hda_codec_amp_init*()
The new function snd_hda_codec_amp_init() (and the stereo variant)
initializes the amp value only once at the first access.  If the amp
was already initialized or updated, this won't do anything more.

It's useful for initializing the input amps that are in the part of
the path but never used.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:30:02 +01:00
Takashi Iwai c370dd6e9f ALSA: hda - Introduce cache & flush cmd / amp writes
For optimizing the verb executions, a new mechanism to cache the verbs
and amp update commands is introduced.  With the new "write to cache
and flush" way, you can reduce the same verbs that have been written
multiple times.

When codec->cached_write flag is set, the further
snd_hda_codec_write_cache() and snd_hda_codec_amp_stereo() calls will
be performed only on the command or amp cache table, but not sent to
the hardware yet.  Once after you call all commands and update amps,
call snd_hda_codec_resume_amp() and snd_hda_codec_resume_cache().
Then all cached writes and amp updates will be written to the
hardware, and the dirty flags are cleared.

In this implementation, the existing cache table is reused, so
actually no big code change is seen here.  Each cache entry has a new
dirty flag now (so the cache key is now reduced to 31bit).

As a good side-effect by this change, snd_hda_codec_resume_*() will no
longer execute verbs that have been already issued during the resume
phase by checking the dirty flags.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-12 08:29:17 +01:00
Mark Brown 7d5cb4f710 ASoC: wm5110: Correct AEC loopback mask
The generated defines in the header are pre-shifted.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-12 00:16:30 +00:00
Mark Brown 7f39bb9e9f ASoC: wm5102: Correct AEC loopback mask
The generated defines in the header are pre-shifted.

Reported-by: Heather Lomond <Heather.Lomond@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-12 00:16:23 +00:00
Mark Brown 8784c77a6c ASoC: dapm: Fix sense of regulator bypass mode
Enable bypass when the regulator is idle, not when it is in use. This is
consistent with what the few existing users actually want.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-12 00:11:47 +00:00
Dan Carpenter fffc0ca29f ASoC: pcm: delete some dead code
I've removed several unreachable returns.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-12 00:05:58 +00:00
Shawn Guo 25b8d31488 ASoC: fsl: fix multiple definition of init_module
With commit f2818d0 (ASoC: fsl: fix miscompilation of snd-soc-imx-pcm),
we will see the following build error when building modules with
CONFIG_SND_IMX_SOC=m in imx_v6_v7_defconfig.

  CC [M]  sound/soc/fsl/phycore-ac97.o
  LD [M]  sound/soc/fsl/snd-soc-fsl-ssi.o
  LD [M]  sound/soc/fsl/snd-soc-fsl-utils.o
  LD [M]  sound/soc/fsl/snd-soc-imx-ssi.o
  LD [M]  sound/soc/fsl/snd-soc-imx-audmux.o
  LD [M]  sound/soc/fsl/snd-soc-imx-pcm.o
sound/soc/fsl/imx-pcm-dma.o: In function `init_module':
imx-pcm-dma.c:(.init.text+0x0): multiple definition of `init_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.init.text+0x0): first defined here
sound/soc/fsl/imx-pcm-dma.o: In function `cleanup_module':
imx-pcm-dma.c:(.exit.text+0x0): multiple definition of `cleanup_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.exit.text+0x0): first defined here
make[4]: *** [sound/soc/fsl/snd-soc-imx-pcm.o] Error 1

Instead of using bool for SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA
to fix the original issue, we should completely remove SND_SOC_IMX_PCM
and have imx-pcm.o statically linked with imx-pcm-fiq.o or imx-pcm-dma.o.

Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-12 00:05:14 +00:00
Ricardo Neri a88fedfd34 ASoC: OMAP: HDMI: Initialize IEC-60958 channel status word
As the IEC-60958 channel status word is set by ANDing and ORing with
the appropriate definitions, the word bytes need to be initialized
to zero to avoid misconfiguration due to previous hw_params calls.

Signed-off-by: Ricardo Neri <rneri@dextratech.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-11 23:58:37 +00:00
Peter Ujfalusi 85becda62c ASoC: twl6040: Remove leftover code from hs/hf ramp implementation
The code to do the ramp has been removed a long time ago. Remove the
remaining code as well since this is not needed.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-11 23:55:12 +00:00
Peter Ujfalusi da2107d1e4 ASoC: twl6040: Switch to use system workqueue for jack reporting
There's no need to create a queue for this anymore

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-11 23:54:57 +00:00
Peter Ujfalusi 9523fcdcc0 ASoC: twl6040: Convert to use devm_* when possible
In this way we can clean up the probe and remove paths

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-11 23:54:57 +00:00
Peter Ujfalusi 156db9f3bb ASoC: twl6040: Only set the bias_level once in twl6040_resume()
No need to set the bias_level twice to _STANDBY - since this is the only
state the device could be at suspend time. The driver do not support
idle_bias_off yet.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-11 23:54:57 +00:00
Peter Ujfalusi 09a8b6719c ASoC: twl4030: Remove suspend/resume soc driver operations
With idle_bias_off these are no longer needed.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-11 23:54:56 +00:00
Misael Lopez Cruz 8d61f4901f ASoC: twl6040: Convert PLUGINT to no-suspend irq
Convert headset PLUGINT interrupt to NO_SUSPEND type in order to
allow handling of insertion/removal events while device is suspended.

Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-11 23:54:47 +00:00
Takashi Iwai 31be5425d7 ALSA: usb-audio: Fix NULL dereference by access to non-existing substream
The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for
audioformat mismatch] introduced the correction of parameters to be
set for sync EP.  But since the new code assumes that the sync EP is
always paired with the data EP of another direction, it triggers Oops
when a device only with a single direction is used.

This patch adds a proper check of sync EP type and the presence of the
paired substream for avoiding the crash.

Reported-and-tested-by: Jens Axboe <axboe@kernel.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-11 11:12:17 +01:00
Kukjin Kim 232910d6bf ARM: S3C24XX: make h1940.h and h1940-latch.h local
The headers can be local in mach-s3c24xx/.

Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
2013-01-10 10:45:35 -08:00
Kukjin Kim b2ca78717c ARM: S3C24XX: make gta02.h local
The header can be local in mach-s3c24xx/ and sort out inclusions.
Accordingly, the GTA02_ macro in driver can be replaced.

Cc: Sangbeom Kim <sbkim73@samsung.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
2013-01-10 10:45:35 -08:00
Takashi Iwai c18ab0bac4 ASoC: Fixes for v3.8
Nothing terribly exciting here except for the DOUBLE_RANGE fix which
 just hadn't worked before, nobody noticed due to lack of use.
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Merge tag 'asoc-fix-3.8-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.8

Nothing terribly exciting here except for the DOUBLE_RANGE fix which
just hadn't worked before, nobody noticed due to lack of use.
2013-01-10 17:41:54 +01:00
Mark Brown 49a170bcf2 Merge remote-tracking branch 'asoc/fix/wm5100' into tmp 2013-01-10 12:22:30 +00:00
Mark Brown 921c038d87 Merge remote-tracking branch 'asoc/fix/wm2200' into tmp 2013-01-10 12:22:29 +00:00
Mark Brown 28f2675db8 Merge remote-tracking branch 'asoc/fix/wm2000' into tmp 2013-01-10 12:22:26 +00:00
Mark Brown 92a9d1524e Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmp 2013-01-10 12:22:25 +00:00
Mark Brown a883eae513 Merge remote-tracking branch 'asoc/fix/sta529' into tmp 2013-01-10 12:22:22 +00:00
Mark Brown fd2eab87a2 Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmp 2013-01-10 12:22:17 +00:00
Mark Brown 87fee06c5b Merge remote-tracking branch 'asoc/fix/pxa' into tmp 2013-01-10 12:22:16 +00:00
Mark Brown c31b71de6f Merge remote-tracking branch 'asoc/fix/lm49453' into tmp 2013-01-10 12:22:15 +00:00
Mark Brown fa17cb4a02 Merge remote-tracking branch 'asoc/fix/cs42l52' into tmp 2013-01-10 12:22:14 +00:00
Mark Brown 587691ea39 Merge remote-tracking branch 'asoc/fix/cs4271' into tmp 2013-01-10 12:22:11 +00:00
Mark Brown a18a31a161 Merge remote-tracking branch 'asoc/fix/core' into tmp 2013-01-10 12:21:50 +00:00
Mark Brown ae1abb0c3b Merge remote-tracking branch 'asoc/fix/arizona' into tmp 2013-01-10 12:21:42 +00:00
Kuninori Morimoto bbf1453e28 ASoC: ak4642: add Device Tree support
Support for loading the ak4642 codec module via devicetree.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-10 12:19:39 +00:00
Takashi Iwai 8092e60654 ALSA: hda - Remove snd_hda_codec_amp_update() call from patch_*.c
It's used only in one place in patch_analog.c, and it can be replaced
with others better.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:30 +01:00
Takashi Iwai 9366ede7fd ALSA: hda/realtek - Fix initialization of input amps in output paths
When initializing the output paths, we assumed the input amps have
almost two inputs blindly.  It's not only generic but even incorrect
for some codecs like ALC268 & co.  Also, the same assumption (two
sources) exists for the bind input-amp controls.

This patch changes the codes in these places to handle the input
connections in a more generic way.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:30 +01:00
Takashi Iwai bd32f782b9 ALSA: hda/realtek - Check amp capabilities of aa-mixer widget
For handling the analog-loopback paths more generically, check the amp
capabilities of the aa-mixer widget, and create only the appropriate
mixer elements.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:29 +01:00
Takashi Iwai c2fd19c2fc ALSA: hda/realtek - Parse analog loopback paths more generically
Improve the parser of analog loopback paths and handle in a more
generic way.  The following changes are included in this patch:

- Instead of assuming direct connections between pins and
  the mixer widget, track the whole path between them.  This fixes
  some missing connections like ALC660.

- Introduce the path list for loopback paths like input and output
  path lists.  Currently it's not used for any real purposes, yet.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:29 +01:00
Takashi Iwai 36f0fd540e ALSA: hda/realtek - Parse input paths
Just like the output paths, parse the whole paths for inputs as well
and store in a path list.  For that purpose, rewrite the output parser
code to be generically usable.

The input path list is not referred at all in this patch.  It'll be
used to replace the fixed adc/capsrc array in later patches for more
flexible input path selections.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:28 +01:00
Takashi Iwai 95e960cece ALSA: hda/realtek - Make path->idx[] and path->multi[] consistent
So far, idx[i] and multi[i] indicate the attribute of the widget
path[i - 1].  This was just for simplifying the code in
__parse_output_path(), but this is rather confusing for later use.
It's more natural if both idx[i] and multi[i] point to the same widget
of path[i].  This patch changes to that way.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:28 +01:00
Takashi Iwai 78e635c93b ALSA: hda/realtek - Simplify the output volume initialization
Simplify the output path initialization using the existing path
information instead of assuming the topology specific to Realtek
codecs.  This is also implicitly a fix for some amp values on output
pins where the old parser missed (e.g. ALC260 output pins).

The same function alc_auto_set_output_and_unmute() can be used now for
the multi-io activation, since the output selection means nothing but
activating the given output path.

And, finally at this stage, we can get rid of alc_go_down_to_selector()
and other functions that are codec really specifically to Realtek
codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:28 +01:00
Takashi Iwai 792cf2fa2e ALSA: hda/realtek - Reduce vol/mute ctl lookups at parsing codec
So far, Realtek codec driver evaluates the NIDs for volume and mute
controls twice, once while parsing the DACs and evaluating the
assignment, and another while creating the mixer elements.  This is
utterly redundant and even fragile, as it's assuming that the ctl
element evaluation is identical between both parsing DACs and creating
mixer elements.

This patch simplifies the code flow by doing the volume / mute
controls evaluation only once while parsing the DACs.  The patch ended
up in larger changes than expected because of some cleanups became
mandatory.

As a gratis bonus, this patch also fixes some cases where the stereo
channels are used wrongly for mono amps.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:27 +01:00
Takashi Iwai 2f179721c4 ALSA: hda - Fix mono amp values in proc output
The mono widget is always connected to the left channel, thus the left
channel amp value also should be referred for mono widgets instead of
the right channel.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:27 +01:00
Takashi Iwai ba8111276f ALSA: hda/realtek - Manage mixer controls in out_path list
As we parse the output paths more precisely now, we can use this path
list for parsing the widgets for volume and mute mixer controls.
The spec->vol_ctls[] and sw_ctls[] bitmasks are replaced with the
ctls[] in each output path instance.

Interestingly, this move alone automagically fixes some bugs that the
conflicting volume or mute NIDs weren't properly detected.
Also, by parsing the whole path, there are more chances to get a free
widget for volume/mute controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:27 +01:00
Takashi Iwai 30dcd3b404 ALSA: hda/realtek - Add output path parser
Add the output path parser to Realtek codec driver as we already have
in patch_via.c.  The nid_path struct represents the complete output
path from a DAC to a pin.  The alc_spec contains an array of these
paths, and a new path is added at each time when a new DAC is
assigned.

So far, this path list is used only in limited codes: namely in this
patch, only alc_is_dac_already_used() checks the list instead of dac
arrays in all possible outputs.  In the later development, the path
list will be referred from more places, such as the mixer control
assignment / check, the mute/unmute of active routes, etc.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:26 +01:00
Takashi Iwai 463419de86 ALSA: hda/realtek - List up all available DACs
In the probing phase, create a list of all available DACs in the codec
and use it for checking the single DAC connections.
This list will be used in more other places in the later commits, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:26 +01:00
Takashi Iwai 6a84c305f0 ALSA: hda/realtek - Simplify alc_auto_is_dac_reachable()
Use the helper function snd_hda_get_conn_index() instead of open
codes.  This also improves the detection of some routes to DAC on
ALC260 (although the difference doesn't influence on the end
results of the mapping).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:34:26 +01:00
Kailang Yang 065380f088 ALSA: hda - Add support of new codec ALC284
Added the support for a new codec ALC284, which is compatible with
ALC269.  Also add more codec variants to handle the SSID check
properly.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:27:46 +01:00
Sachin Kamat e8e7da23c9 ALSA: usb-audio: Make ebox44_table static
Fixes the following sparse warning:
sound/usb/mixer_quirks.c:1209:23: warning:
symbol 'ebox44_table' was not declared. Should it be static?

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:22:25 +01:00
Andre Schramm 56bde0f328 ALSA: hdspm - Fix wordclock status on AES32
Use correct bitmask for AES32 cards to determine wordclock lock state,
add missing bitmask for sync check and make output of the corresponding
control and /proc coherent.

Signed-off-by: Andre Schramm <andre.schramm@iosono-sound.com>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 16:59:24 +01:00
Masanari Iida c46d5c04f3 sound: soc: Fix typo in sound/codecs
Correct spelling typo in sound/soc/codecs

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2013-01-09 11:44:56 +01:00
Takashi Iwai 6ab317419c ALSA: hda - Allow power_save_controller option override DCAPS
Change the power_save_controller option to bint from bool so that user
can override the runtime PM capability bit and force to enable or
disable the runtime PM.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 11:15:13 +01:00
David Henningsson 7ed4165e2d Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"
This reverts commit 697c373e34.

The original patch was meant to remove clicking, but in fact caused even
more clicking instead.

Thanks to c4pp4 for doing most of the work with this bug.

BugLink: https://bugs.launchpad.net/bugs/886975
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 11:03:38 +01:00
Takashi Iwai d7dab4dbbb ALSA: hda - Disable runtime D3 for Intel CPT & co
We've got a few bug reports that the runtime D3 results in the dead
HD-audio controller.  It seems that the problem is in a deeper level
than the sound driver itself, so as a temporal solution, disable the
feature for these controllers again.

Reported-and-tested-by: Vincent Blut <vincent.debian@free.fr>
Reported-and-tested-by: Maurizio Avogadro <mavoga@gmail.com>
Cc: <stable@vger.kernel.org> [v3.7]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 11:00:08 +01:00
Mark Brown 471f488583 ASoC: wm_adsp: Implement support for algorithm-specific coefficient blocks
WMDR coefficient files can specify coefficients in terms of algorithm
specific data regions. Record the start addresses of these regions while
parsing the algorithms and then use them to handle coefficients with
these formats.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-08 20:47:34 +00:00
Mark Brown d62f4bc665 ASoC: wm_asdp: Validate sanity of algorithm count
If we run into I/O problems the algorithm count may be crazy, validate it
before we proceed and dump the read data for diagnostic purposes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-08 20:47:32 +00:00
Mark Brown 45b9ee72d0 ASoC: wm_adsp: Factor out calculation of memory base addresses
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-08 20:47:30 +00:00
Mark Brown db40517c75 ASoC: wm_adsp: Add support for parsing algorithms
ADSP devices report information on the algorithms loaded on them.  Parse
this data and use it to allow coefficients to be configured for specific
algorithms.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-08 20:47:29 +00:00
Charles Keepax e31c194672 ASoC: arizona: Disable free-running mode on FLL1
The free running mode can cause problems when attempting to bring up the
FLL running from a defined clock source. This patch disables
free-running mode.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-08 18:01:17 +00:00
Fabio Estevam 324a7fb02b ASoC: mxs-saif: Use a signed integer for error value
saif->id and saif->master_id are unsigned, so they can not be negative.

Fix the following warning when building with W=1 option:

sound/soc/mxs/mxs-saif.c: In function 'mxs_saif_probe':
sound/soc/mxs/mxs-saif.c:676:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
sound/soc/mxs/mxs-saif.c:688:3: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
sound/soc/mxs/mxs-saif.c:692:2: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]

Use a signed variable 'ret' to handle the error values.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-08 17:58:32 +00:00
David Henningsson f4f0a8c478 ALSA: hda - print power state for AFG node in proc file
It seems useful, and power states are required for AFG nodes,
so I see no reason not to print it. As a bonus, also print the
AFG nid.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-08 17:02:27 +01:00
Mike Dunn 053fe0f166 ALSA: pxa27x: rename pxa27x_assert_ac97reset()
This patch does nothing functionally, it just gives the function a new name and
modifies the prototype slightly in order to clarify what the function is doing
(which is not necessarily asserting the reset).
Some commentary also added.

Tested on a palm treo 680 machine.

Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Acked-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-08 11:30:08 +00:00
Mark Brown 07afa01813 Merge remote-tracking branch 'asoc/fix/pxa' into asoc-pxa 2013-01-08 11:29:45 +00:00
Mike Dunn 3b4bc7bccc ALSA: pxa27x: fix ac97 warm reset
This patch fixes some code that implements a work-around to a hardware bug in
the ac97 controller on the pxa27x.  A bug in the controller's warm reset
functionality requires that the mfp used by the controller as the AC97_nRESET
line be temporarily reconfigured as a generic output gpio (AF0) and manually
held high for the duration of the warm reset cycle.  This is what was done in
the original code, but it was broken long ago by commit fb1bf8cd
    ([ARM] pxa: introduce processor specific pxa27x_assert_ac97reset())
which changed the mfp to a GPIO input instead of a high output.

The fix requires the ac97 controller to obtain the gpio via gpio_request_one(),
with arguments that configure the gpio as an output initially driven high.

Tested on a palm treo 680 machine.  Reportedly, this broken code only prevents a
warm reset on hardware that lacks a pull-up on the line, which appears to be the
case for me.

Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Signed-off-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-08 11:27:35 +00:00
Mike Dunn 41b645c862 ALSA: pxa27x: fix ac97 cold reset
Cold reset on the pxa27x currently fails and

     pxa2xx_ac97_try_cold_reset: cold reset timeout (GSR=0x44)

appears in the kernel log.  Through trial-and-error (the pxa270 developer's
manual is mostly incoherent on the topic of ac97 reset), I got cold reset to
complete by setting the WARM_RST bit in the GCR register (and later noticed that
pxa3xx does this for cold reset as well).  Also, a timeout loop is needed to
wait for the reset to complete.

Tested on a palm treo 680 machine.

Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Acked-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-08 11:27:27 +00:00
Fabio Estevam 4498a3cae5 ASoC: mxs-saif: Remove platform data
All MXS users have been converted to device tree and the board files have been
removed.

No need to keep platform data in the driver.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Acked-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-08 11:23:26 +00:00
Mark Brown a76fefab5c ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses
Otherwise we won't run correctly on systems that require this for larger
data transfers.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-07 19:13:35 +00:00
David Henningsson 6d3cd5d444 ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
The mute LED is in this case connected to the Mic1 VREF.

The machine also exposes the following string in BIOS:
"HP_Mute_LED_0_A", so if more machines are coming, it probably
makes sense to try to do something more generic, like for the
IDT codec.

Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1096789
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-07 17:29:55 +01:00
Nickolai Zeldovich 61ed1dca16 ALSA: au88x0: fix incorrect left shift
vortex_wt_setdsout performs bit-negation on the bit position (wt&0x1f)
rather than on the resulting bitmask.  This code is never actually
invoked (vortex_wt_setdsout is always called with en=1), so this does
not currently cause any problem, and this patch is simply cleanup.

Signed-off-by: Nickolai Zeldovich <nickolai@csail.mit.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-07 09:33:56 +01:00
Mark Brown b272efc860 ASoC: arizona: Factor out rate selection code
In preparation for more advanced sample rate managment move the existing
code out of the main hw_params() function.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-04 21:30:21 +00:00
Mark Brown 66b6eaf23a Merge branch 'fix/arizona' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-arizona 2013-01-04 21:30:16 +00:00
Mark Brown bc9ab6d31c ASoC: arizona: Allow runtime reconfiguration of the output mode
Some systems use external analogue switches to connect more analogue
devices to the CODEC than are supported by the device.  In some systems
this requires changing the switched output from single ended to
differential mode dynamically at runtime. Add a new function
arizona_set_output_mode() to support this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-04 21:20:59 +00:00
Mark Brown 267f8fa2e1 ASoC: wm2000: Fix sense of speech clarity enable
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 21:19:42 +00:00
Mark Brown 5f960294e2 ASoC: wm5100: Remove DSP B and left justified formats
These are not supported

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 21:06:08 +00:00
Mark Brown 91660bd65c ASoC: wm5102: Implement routing and power management for ISRCs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-04 20:55:55 +00:00
Mark Brown d71753e22b ASoC: arizona: Remove DSP B and left justified AIF modes
These are not supported.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 11:33:22 +00:00
Mark Brown 0cc411b934 ASoC: wm2200: Remove DSP B and left justified AIF modes
These are not supported.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 11:31:57 +00:00
Asim Kadav dc30a43690 sound: oss/pas2: Fix possible access out of array
Added a fix for hardware dependence bug where a sound card failure
should not result in reading beyond array memory index.

Signed-off-by: Asim Kadav <kadav@cs.wisc.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-04 10:38:27 +01:00
Damien Zammit b7b435e81b ALSA: usb-audio: Fix kernel panic of Digidesign Mbox2 quirk
This patch is based on 3.8-rc1. It fixes two things:
1) A kernel panic caused by incorrect allocation of a u8 variable
   "bootresponse".
2) A noisy dmesg (urb status -32) caused by broken pipe to an
   invalid midi endpoint.

It is also a little cleaner because there is no need for a new
QUIRK_MIDI type as suggested by kernel developers, since the device
follows exactly the MIDIMAN protocol.

Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-04 09:53:17 +01:00
Philippe De Muyter 9a32299394 powerpc, dma: move bestcomm driver from arch/powerpc/sysdev to drivers/dma
The bestcomm dma hardware, and some of its users like the FEC ethernet
component, is used in different FreeScale parts, including non-powerpc
parts like the ColdFire MCF547x & MCF548x families.  Don't keep the
driver hidden in arch/powerpc where it is inaccessible for other arches.
.c files are moved to drivers/dma/bestcomm, while .h files are moved to
include/linux/fsl/bestcomm.  Makefiles, Kconfigs and #include directives
are updated for the new file locations.

Tested by recompiling for MPC5200 with all bestcomm users enabled.

Signed-off-by: Philippe De Muyter <phdm@macqel.be>
Signed-off-by: Anatolij Gustschin <agust@denx.de>
2013-01-03 15:41:20 +01:00
Alexander Schremmer 8f7f3ab15e ALSA: usb-audio: Add support for Creative BT-D1 via usb sound quirks
Support the Creative BT-D1 Bluetooth USB audio device. Before this
patch, Linux had trouble finding the correct USB descriptors and bailed
out with these messages:

 no or invalid class specific endpoint descriptor

Now it still prints these messages on hotplug:

 snd-usb-audio: probe of ...:1.0 failed with error -5
 snd-usb-audio: probe of ...:1.2 failed with error -5
 snd-usb-audio: probe of ...:1.3 failed with error -5

But the device works correctly, including the HID support.

The patch is diff'ed against 3.8-rc1 but should apply to older kernels
as well.

Signed-off-by: Alexander Schremmer <alex@alexanderweb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-03 14:26:48 +01:00
David Henningsson c86c2d440c ALSA: hda - Switch "On" and "Off" for "Mute-LED Mode" kcontrol
The vmaster hook sends 1 for enabled/unmuted and 0 for disabled/muted,
but "Mute-LED Mode" being "On" refers to the LED being on, not the
volume being on.
Therefore "On" and "Off" should be switched.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-03 14:22:34 +01:00
Kuninori Morimoto fd974e52db ASoC: fsi: don't use platform info pointer on probe()
Current FSI driver is using platform info pointer,
but it is not good design for DT support.
This patch made it not to use platform info pointer.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-03 12:49:54 +00:00
Mark Brown 1b8d52e63c ASoC: wm5102: Improve speaker enable performance
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:08:42 +00:00
Mike Dunn 01a61f490c ASoC: palm27x: register card in platform_driver probe
Remove creation of an soc-audio device from the machine platform_driver probe
function, and add a call to snd_soc_register_card() instead.

The current code still works, but this mechanism has been deprecated, if I'm not
mistaken.  The ASoC core code produces the warning "ASoC: machine Palm/PXA27x
should use snd_soc_register_card()"

Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:07:01 +00:00
Mike Dunn 016fb39c98 ASoC: palm27x: fix widgets and routes in dai_link init
ASoC core code now handles creation of controls and routing based on contents of
struct snd_soc_card, so remove calls to snd_soc_dapm_new_controls() and
snd_soc_dapm_add_routes() from the snd_soc_dai_link init function, and add
widget and route definitions to struct snd_soc_card.

Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:07:00 +00:00
Peter Ujfalusi 57d61b9d2d ASoC: OMAP: Remove obsolete machine drivers for Zoom2 and SDP3430
These boards are using the common omap-twl4030 machine driver, no need for
separate machine drivers anymore.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:04:25 +00:00
Peter Ujfalusi bd0b286e83 ASoC: omap-twl4030: Add support for routing, voice port and jack detect
Update the common machine driver to support more boards including Zoom2 and
SDP3430.
- Support for voice port of twl4030
- HS jack plug detection support
- The audio routing can be fine tuned via pdata or via provided routing
  table from DT.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:04:25 +00:00
Peter Ujfalusi fff3dd4013 ASoC: sdp3430: No need to configure pin mux for extmute
The codec driver takes care of this.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:04:25 +00:00
Peter Ujfalusi 5712ded9cf ASoC: twl4030: Configure extmute pinmux when the dedicated pin is in use
When HS extmute is enabled without custom GPIO we should configure the mux
to allow the pin to be used as extmute signal.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:04:25 +00:00
Peter Ujfalusi e04d6e55fe ASoC: twl4030: Convert MICBIAS to SUPPLY widget
In order to avoid breakage update the machine drivers at the same time using
twl4030: omap3pandora, sdp3430 and zoom2

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:04:25 +00:00
Peter Ujfalusi 57296cc28c ASoC: sdp3430: No need to configure the Voice port anymore
The codec driver takes care of these bits.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:04:24 +00:00
Peter Ujfalusi 01df26edaf ASoC: zoom2: No need to configure the Voice port anymore
The codec driver takes care of these bits.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:04:24 +00:00
Peter Ujfalusi 927a77476e ASoC: twl4030: Correct the support for Voice port
In order to be able to use the Voice port of twl4030 three bits need to be
handled in VOICE_IF register:
VIF_EN: to enable the voice port (needed for both playback and capture)
VIF_DIN_EN: Need to be enabled for playback only (input to the codec)
VIF_DOUT_EN: Need to be enabled for capture only (output from codec)

Use DAPM_SUPPLY for the VIF_EN bit and add DAPM_AIF_IO/OUT widget to handle
the playback/capture bit.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:04:24 +00:00
Kuninori Morimoto f89983ef61 ASoC: simple-card: use struct device pointer for dev_xxx()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27 16:51:16 +00:00
Fabio Estevam 5f3d25c08d ASoC: wm8985: Refactor set_pll code to avoid gcc warnings
Refactor set_pll code to avoid the following warnings:

sound/soc/codecs/wm8985.c:852:50: warning: 'pll_div.k' may be used uninitialized in this function
sound/soc/codecs/wm8985.c:849:9: warning: 'pll_div.n' may be used uninitialized in this function
sound/soc/codecs/wm8985.c:848:23: warning: 'pll_div.div2' may be used uninitialized in this function

Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid
GCC warnings).

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27 16:49:00 +00:00
Axel Lin e958f8b806 ASoC: cs42l52: Convert to devm_input_allocate_device()
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27 16:46:38 +00:00
Chuansheng Liu d3bf156125 ASoC: core: fix the memory leak in case of remove_aux_dev()
When probing aux_dev, initializing is as below:
device_initialize()
device_add()

So when remove aux_dev, we need do as below:
device_del()
device_put()
Otherwise, the rtd_release() will not be called.

So here using device_unregister() to replace device_del(),
like the action in soc_remove_link_dais().
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27 16:14:43 +00:00
Chuansheng Liu 865df9cb12 ASoC: core: fix the memory leak in case of device_add() failure
After called device_initialize(), even device_add() returns
error, we still need use the put_device() to release the reference
to call rtd_release(), which will do the free() action.

Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27 16:14:43 +00:00
Tejun Heo 8a47ca957a ASoC: wm8350: don't use [delayed_]work_pending()
There's no need to test whether a (delayed) work item in pending
before queueing, flushing or cancelling it.  Most uses are unnecessary
and quite a few of them are buggy.

Remove unnecessary pending tests from wm8350.  Only compile tested.

Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 16:10:22 +00:00
Axel Lin 3271a4fc7d ASoC: cs42l52: Catch no-match case in cs42l52_get_clk
In the case of no-match, return -EINVAL instead of 0.

Since we assign i to ret in the for loop, ret always less than
ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 16:02:19 +00:00
Lucas Stach 15fab58507 ASoC: tegra: setup DAP3<->DAC3 connection by default
This connection is used by the AC97 controller.

Signed-off-by: Lucas Stach <dev@lynxeye.de>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 16:01:08 +00:00
Lucas Stach 919ad49c21 ASoC: tegra: add function to set ac97 rate
AC97 uses a fixed rate, unrelated to the sample rate. Add a function to
make the setup more trivial.

Signed-off-by: Lucas Stach <dev@lynxeye.de>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:59:10 +00:00
Kuninori Morimoto abca75814a ASoC: fsi: remove SH_FSI_xxx_INV flags
3449f5fab8
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
added clock inversion support via snd_soc_dai_set_fmt().
Thus, this patch removed SH_FSI_xxx_INV and fsi_get_info()
from fsi driver, and modified platform settings to use new style.
Then, it cleaned up meaningless settings from platform.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Simon Horman <horms+renesas@verge.net.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:57:09 +00:00
Kuninori Morimoto 6cbdbffba1 ASoC: fsi: remove platform depended .set_rate() callback support
ab6f6d8521
(ASoC: fsi: add master clock control functions)
added driver level clock control functions.
And now, platform depended .set_rate() is no longer needed.
This patch removed unnecessary .set_rate() platform callback support.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:56:55 +00:00
Fabio Estevam 6757d8cc0c ASoC: wm8993: Refactor set_pll code to avoid GCC warnings
Refactor set_pll code to avoid the following warnings:

sound/soc/codecs/wm8983.c:873:40: warning: 'pll_div.k' may be used uninitialized in this function [-Wuninitialized]
sound/soc/codecs/wm8983.c:870:9: warning: 'pll_div.n' may be used uninitialized in this function [-Wuninitialized]
sound/soc/codecs/wm8983.c:869:23: warning: 'pll_div.div2' may be used uninitialized in this function [-Wuninitialized]

Do the same as in commit 86ce6c9a (ASoC: WM8804: Refactor set_pll code to avoid
GCC warnings).

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:55:40 +00:00
Fabio Estevam 1edbd35667 ASoC: wm8804: Remove redundant check
The condition "if (!freq_in || !freq_out)" has already been tested previously,
so no need to do it again.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:54:27 +00:00
Daniel Mack fd23fb9f6b ALSA: ASoC: cs4271: add optional soft reset workaround
The CS4271 requires its LRCLK and MCLK to be stable before its RESET
line is de-asserted. That also means that clocks cannot be changed
without putting the chip back into hardware reset, which also requires
a complete re-initialization of all registers.

One (undocumented) workaround is to assert and de-assert the PDN bit
in the MODE2 register.

This patch adds a new flag to both the DT bindings as well as to the
platform data to enable that workaround.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:53:28 +00:00
Mark Brown 133d2e6188 Merge branch 'asoc-fix-cs4271' into asoc-cs4271 2012-12-24 15:52:48 +00:00
Joachim Eastwood 153f5a18e4 ASoC: atmel-soc: make it buildable on other architectures
Not very useful on non AT91/AVR32 platforms but it provides
more build coverage and prepares for ARM multiplatform.

Also fixes a couple of format type warnings.

Signed-off-by: Joachim Eastwood <manabian@gmail.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:48:25 +00:00
MR.Swami.Reddy@ti.com 9dc754dfa7 ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B
Update lm49453_reg_defs values as per LM49453 HW revision-B

Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:45:10 +00:00
MR.Swami.Reddy@ti.com 88ac43924b ASoC: lm49453: Fix adc, mic and sidetone volume ranges
Add adc, mic, sidetone volume ranges and appropriately added the controls.
Fix the DAC HP/EP/LS/LO/HA maximum gain values.

Signed-off-by: MR Swami Reddy <mr.swami.reddy@ti.com>
Tested-by: Vinod Koul <vinod.koul@intel.com>

--
 sound/soc/codecs/lm49453.c |   43 ++++++++++++++++++++++++-------------------
 1 files changed, 24 insertions(+), 19 deletions(-)
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:43:56 +00:00
Mark Brown d61100bbd1 ASoC: wm2000: Use clock API integration to configure MCLK divisor
Since we are now using the clock API integration to manage MCLK we can now
use clk_get_rate() to determine if we need to divide MCLK without relying
on platform data.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:42:35 +00:00
Mark Brown 514cfd6dd7 ASoC: wm2000: Integrate with clock API
Request MCLK as a clock and then enable it when carrying out a state
transtion and while ANC is active, minimising system power consumption
in idle modes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:42:33 +00:00
Mark Brown a8c02db029 ASoC: arizona: Correct FLL source definitions
The FLL source constants were numbered as a simple enumeration but were
being used in the code as direct values to be written to the registers.
Renumber the constants to reflect the usage.

Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-24 15:41:44 +00:00
Axel Lin 7110a287ff ASoC: arizona: Do proper shift for setting AIF rate
ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */
Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-24 15:41:44 +00:00
Mark Brown 01df259f59 ASoC: arizona: Implement tristate support
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:39:23 +00:00
Mark Brown bd7fe24bc4 ASoC: wm5110: Add noise gate control
The references used for the noise gates and parameters for their triggering
are configurable, expose that to users.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:39:11 +00:00
Mark Brown 5057126372 ASoC: wm5102: Add noise gate control
The references used for the noise gates and parameters for their triggering
are configurable, expose that to users.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:39:09 +00:00
Mark Brown 845571cce6 ASoC: arizona: Add noise gate hold time enumeration
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:39:07 +00:00
Mark Brown 02482da46e ASoC: wm5110: Split input PGA controls
Though the controls are named as stereo controls in the part the main use
case for the analogue inputs to the WM5102 is mono. Reflect this in the
controls exposed to userspace, providing a series of mono controls rather
than stereo ones.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:38:51 +00:00
Mark Brown c63f650c0d ASoC: wm5102: Split input PGA controls
Though the controls are named as stereo controls in the part the main use
case for the analogue inputs to the WM5102 is mono. Reflect this in the
controls exposed to userspace, providing a series of mono controls rather
than stereo ones.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:38:39 +00:00
Mark Brown 346f1d4083 ASoC: wm8962: Unconditionally wait for the FLL to lock
If the FLL is being shut down we will exit early so there is no need to
check here and in fact we're checking the wrong thing anyway.

Reported-by: Graeme Gregory <gg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:38:10 +00:00
Mark Brown a2ce64750e ASoC: wm8962: Convert to devm_input_allocate_device()
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:38:02 +00:00
Fabio Estevam 5ce568329e ASoC: wm8962: Add device tree support
Add device tree support.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:36:57 +00:00
Chuansheng Liu ff541f4b2a ASoC: core: giving WARN when device starting from non-off bias with idle_bias_off
Just found some cases that some codec drivers set the bias to _STANDBY and
set idle_bias_off to 1 during probing.
It will cause unpaired runtime_get_sync/put() issue. Also as Mark suggested,
there is no reason to start from _STANDBY bias with idle_bias_off == 1.

So here giving one warning when detected (dapm.idle_bias_off == 1) and
(dapm.bias_level != SND_SOC_BIAS_OFF) just after driver->probe().

Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:35:34 +00:00
Axel Lin ec20f2f8d3 ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()
The mode variable is either 0 or 1.
To update mode setting, the mask should be BIT(0) rather than BIT(1).

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Omair M. Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:32:28 +00:00
Fabio Estevam b50684da6c ASoC: sgtl5000: Fix maximum value for microphone gain
sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3.

From Eric Nelson:
"We also found that for the microphones we have here (commodity PC boom mics) a
default value of 2 for the gain gives the best results."

So change the default microphone gain as well.

Signed-off-by: Eric Nelson <eric.nelson@boundarydevices.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:31:25 +00:00
Fabio Estevam 5db1bc1892 ASoC: soc-core: Remove unused 'ret' variable
commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced
the following build warning:

sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable]

Remove the unused 'ret' variable.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:20:16 +00:00
Pierre-Louis Bossart e4cc615340 ALSA: usb-audio: support delay calculation on capture streams
Enable delay report on capture path. The delay is reset when an
URB is retired and increment at each call to .pointer based
on frame counter changes. The precision of the delay
information is limited to 1ms as in the playback case.

This reverts commit 3f94fad095.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-24 10:53:57 +01:00
Axel Lin 2a5f431592 ASoC: wm2200: Fix setting dai format in wm2200_set_fmt
According to the defines in wm2200.h:
/*
 * R1284 (0x504) - Audio IF 1_5
 */

We should not left shift 1 bit for fmt_val when setting dai format.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-21 09:32:20 +00:00
Mark Brown 9bde4f0b1c ASoC: core: Fix SOC_DOUBLE_RANGE() macros
Although we've had macros defining double _RANGE controls for a while now
they've not actually been backed up properly by the implementation, it's
treated everything as mono. Fix that by implementing the handling in the
stereo controls, ensuring that the mono controls don't mistakenly get
treated as stereo.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-12-20 17:46:55 +00:00
Axel Lin ad1937cdd5 ASoC: sta529: Fix update register bits in sta529_set_dai_fmt
Both the mask and mode settings are wrong in current code.

According to the datasheet:

S2PCFG0 (0x0A)
BIT[3:1] DATA_FORMAT
        serial interface protocol format:
        000: left Justified
        001: I2S (default)
        010: right justified
        100: PCM no delay
        101: PCM delay
        111: DSP

Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and
RIGHT_J_DATA_FORMAT.
Also adds define for DATA_FORMAT_MSK.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-20 16:01:26 +00:00
Patrick Lai 08b27848da ASoC: pcm: allow backend hardware to be freed in pause state
When front-end PCM session is in paused state, back-end
PCM session will be put in paused state as well if given
front-end PCM session is the only client of given back-end.
Then, application closes front-end PCM session, DPCM
framework will not allow back-end enters HW_FREE state
so back-end will never get shutdown completely.

Signed-off-by: Patrick Lai <plai@codeaurora.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-20 15:59:46 +00:00
Linus Torvalds 03c850ec32 Sound fixes for 3.8-rc1
This update contains overall only driver-specific fixes.
 Slightly large LOC are seen in usb-audio driver for a couple of new
 device quirks and cs42l71 ASoC driver for enhanced features.
 The others are a few small (regression) fixes HD-audio, and yet other
 small / trival ASoC fixes.
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Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This update contains overall only driver-specific fixes.  Slightly
  large LOC are seen in usb-audio driver for a couple of new device
  quirks and cs42l71 ASoC driver for enhanced features.  The others are
  a few small (regression) fixes HD-audio, and yet other small / trival
  ASoC fixes."

* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
  ALSA: HDA: Fix sound resume hang
  ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
  ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup
  ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs
  ASoC: atmel-ssc: change disable to disable in dts node
  ASoC: Prevent pop_wait overwrite
  ALSA: usb-audio: ignore-quirk for HP Wireless Audio
  ALSA: hda - Always turn on pins for HDMI/DP
  ALSA: hda - Fix pin configuration of HP Pavilion dv7
  ASoC: core: Fix splitting of log messages
  ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT
  ASoC: cs42l73: Add DAPM events for power down.
  ASoC: cs42l73: Add DMIC's as DAPM inputs.
  ASoC: sigmadsp: Fix endianness conversion issue
  ASoC: tpa6130a2: Use devm_* APIs
2012-12-20 07:52:13 -08:00
Damien Zammit cb99864d40 ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
This patch is the result of a lot of trial and error, since there are no specs
available for the device.

Full duplex support is provided, i.e. playback and recording in stereo.
The format is hardcoded at 48000Hz @ 24 bit, which is the maximum that the
device supports.  Also, MIDI in and MIDI out both work.

Users will notice that the S/PDIF light also flashes when playback or recording
is active.  I believe this means that S/PDIF input/output is simultaneously
activated with the analogue i/o during use.
But this particular functionality remains untested.

Note that this particular version of the patch is so far untested on the
physical hardware because I have not compiled a full kernel with the changes.
However, extensive testing has been done by many users of the hardware
who believe other versions of my patch have worked since circa 2009.

[Modified to make a function static by tiwai]

Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-19 11:27:22 +01:00
Daniel J Blueman 44728e97c3 ALSA: HDA: Fix sound resume hang
Resuming a switcheroo'd HDA controller hangs since the completion
is one-shot (thus works the first time). Fix by using completions
that explictly need rearming, so remain fired before.

Signed-off-by: Daniel J Blueman <daniel@quora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-18 17:07:11 +01:00
Mengdong Lin 6ffe168f82 ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
Haswell HDMI codec pins may report invalid connection list entries, which
will cause failure to play audio via HDMI or Display Port.

So this patch adds fixup for Haswell to workaround this hardware issue:
enable DP1.2 mode and override the pins' connection list entries with proper
value.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Xingchao Wang <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-18 11:05:36 +01:00
Tao Ma 8f6e604196 sound: remove reference to feature-removal-schedule.txt
In commit 9c0ece069b ("Get rid of Documentation/feature-removal.txt"),
Linus removed feature-removal-schedule.txt from Documentation, but there
is still some reference to this file.  So remove them.

Signed-off-by: Tao Ma <boyu.mt@taobao.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2012-12-17 17:15:12 -08:00
Takashi Iwai b78562b10f ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup
The workaround to force VREF50 for dallas/hp model with ALC861VD
was introduced in commit 8fdcb6fe42,
but it contained wrong pincap override bits.

This patch fixes to exclude VREF80 pincap bit correctly.

Cc: <stable@vger.kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-17 20:10:50 +01:00
Takashi Iwai 1098b7c228 ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs
It turned out that Realtek codecs (ALC260, etc) with input amps in
audio-input widgets don't handle the multiple individual input amps.
Thus we need to set codec->single_adc_amp flag for them in general.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-17 20:03:15 +01:00
Takashi Iwai 6be7f5344b ASoC: More updates for v3.8
Nothing terribly exciting here, just small localised changes.
 
 As well as fixes there are a couple of Cirrus changes and one devm_
 change which were in prior to the merge window but got missed from the
 original pull to Takashi.
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Merge tag 'asoc-3.8p1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: More updates for v3.8

Nothing terribly exciting here, just small localised changes.

As well as fixes there are a couple of Cirrus changes and one devm_
change which were in prior to the merge window but got missed from the
original pull to Takashi.
2012-12-17 15:40:55 +01:00
Mark Brown 8246b5b03e Merge remote-tracking branch 'asoc/topic/tpa6130a2' into asoc-next 2012-12-15 23:56:46 +09:00