Commit Graph

14892 Commits

Author SHA1 Message Date
Takashi Iwai a607148ff3 ALSA: hda - Set individual name to secondary analog PCM stream
It'd be better to give another name to the secondary (alt) analog PCM
stream, which is dedicated for the independent HP out and extra
inputs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-22 14:12:40 +01:00
Takashi Iwai f2f8be43c5 ALSA: hda - Add aamix NID to AD codecs
The aamix NIDs are also missing for AD codecs.  All AD codecs seem to
have a (more or less) working aamix widget.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-22 14:12:08 +01:00
Mark Brown a8c136d2eb Merge remote-tracking branch 'asoc/fix/wm2200' into tmp 2013-01-22 16:26:33 +08:00
Mark Brown 32eca984f6 Merge remote-tracking branch 'asoc/fix/fsl' into tmp 2013-01-22 16:26:21 +08:00
Mark Brown 05780d7771 Merge remote-tracking branch 'asoc/fix/core' into tmp 2013-01-22 16:26:15 +08:00
Mark Brown bc04c93bbc Merge remote-tracking branch 'asoc/fix/arizona' into tmp 2013-01-22 16:26:06 +08:00
Charles Keepax a4cdbec758 ASoC: wm_adsp: Release firmware on error
This patch correctly releases the firmware if the magic string in the
firmware header does not match.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-22 16:40:26 +09:00
Charles Keepax f63d944b71 ASoC: wm_adsp: Release firmware on error
This patch correctly releases the firmware if the magic string in the
firmware header does not match.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-22 16:27:21 +09:00
Takashi Iwai 42c364ace5 ALSA: hda - Add Conexant CX20755/20756/20757 codec IDs
These are just compatible with other CX2075x codecs.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-21 16:53:37 +01:00
Takashi Iwai 2748746f40 ALSA: hda - Add aamix NID to IDT 92HD codecs
IDT codecs have analog-loopback mixer widgets, but we haven't cared
about it, so far.  Let's set them.  This will avoid also possible
wrong routes for the input paths.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-21 16:25:18 +01:00
Takashi Iwai 6efcc52653 ALSA: hda - Remove superfluous header inclusions
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-21 16:10:56 +01:00
Takashi Iwai 139611705a ALSA: hda - Enable parsing the independent HP mode as default for VIA codecs
The original VIA codec parser enabled it as default, so let's keep the
behavior as it was.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-21 15:15:55 +01:00
Takashi Iwai a1e908edcc ALSA: hda - Fix conflicts between Loopback Mixing and Independent HP
This patch eventually fixes two issues:
- Handle the case where the primary output is a headphone and can have
  independent HP mode;
  so far we checked only the case where the headphone is the secondary
  output.

- Fix the conflict of HP independent mode and aamix mode;
  when switched to aamix mode, the DAC might be also switched to
  another widget shared with other outputs.  Then even if we disable
  the DAC for the original output, it doesn't change -- because the
  active route is from another (shared) DAC to HP pin through aamix.
  So, in such a case, we have to prohibit the switch to aamix for HP
  routes.

This fixes issues appearing on VT codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-21 15:11:25 +01:00
Takashi Iwai f87498b651 ALSA: hda - Check aamix-output paths from other DACs, too
Many codecs provide routes to multiple output pins through an aamix
widget, but most of them do it only from a single DAC.  However, the
current generic parser checks only the aamix paths from the original
(directly bound) DACs through aamix NID, and miss the path:
  primary DAC -> aamix -> target out pin

This patch adds a more check for the routes like the above.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-21 14:32:53 +01:00
Takashi Iwai 1fa335b0b7 ALSA: hda - Add missing badness evaluation for unresolved paths
When a patch couldn't be resolved in try_assign_dacs() although the
target DAC is expected, we forgot to add a proper badness value but
continued.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-21 11:47:07 +01:00
Takashi Iwai 9314a5813f ALSA: hda - Set the pin targets after deciding output config
Since fill_and_eval_dacs() may be called repeatedly with different
configurations, setting pinctls at each time there isn't optimal.
We can set it better only once after deciding the output configuration
in parse_output_paths().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-21 11:09:03 +01:00
Takashi Iwai a769409cf3 ALSA: hda - Improve debug prints for output paths
Print the information of outputs in a bit more details and concisely
in a single place instead of printing the path at each time when
detected.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-21 11:08:52 +01:00
Mark Brown b5a8fe439a ASoC: core: Ensure SND_SOC_BYTES writes are from DMA safe memory
With some buses the transfers may DMAed, especially for larger blocks.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-21 17:49:52 +09:00
Mark Brown c94aa30eda ASoC: arizona: Allow number of channels clocked to be restricted
Place a cap on the number of channels clocks are generated for. This is
intended for use with systems which have the WM5102 master an I2S bus with
multiple data lines.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-21 17:47:39 +09:00
Mark Brown 20da6d5ac0 ASoC: wm_adsp: Provide explicit trace of coefficient writes
Helpful for debugging.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-21 17:46:35 +09:00
Mark Brown 69485d3e6d Merge remote-tracking branch 'asoc/fix/adsp' into asoc-adsp 2013-01-21 17:46:16 +09:00
Mark Brown f2c26d48d9 ASoC: arizona: Support clearing clocks
Some systems may wish to support switching between telephony and CD audio
clock rates but this is restricted by enforcement of constraints on the
current DAI clock. Support setting clocks to zero and don't enforce any
constraints in that case in order to facilitate this use case.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-21 17:38:54 +09:00
Rusty Russell 373d4d0997 taint: add explicit flag to show whether lock dep is still OK.
Fix up all callers as they were before, with make one change: an
unsigned module taints the kernel, but doesn't turn off lockdep.

Signed-off-by: Rusty Russell <rusty@rustcorp.com.au>
2013-01-21 17:17:57 +10:30
Mark Brown f2a93e2a4c ASoC: wm_adsp: Use GFP_DMA for algorithm readback
Normally kmalloc() returns things that are DMA safe so not visible on all
platforms but we do need to explicitly request DMA safe memory.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-20 22:17:30 +09:00
Mark Brown 7881fd0fb3 ASoC: wm_adsp: Use GFP_DMA for things that may be DMAed
Normally kmalloc() returns things that are DMA safe so not visible on all
platforms but we do need to explicitly request DMA safe memory.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-20 22:14:34 +09:00
Mark Brown 25c62f7e70 ASoC: wm_adsp: Make region identification errors more informative
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-20 22:13:24 +09:00
Mark Brown 908a5741ab ASoC: wm2200: Implement EQ and LHPF coefficient configuration
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-20 22:10:25 +09:00
Mark Brown 20fc48632f ASoC: wm5100: Implement DRC, EQ and LHPF coefficient configuration
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-20 22:09:40 +09:00
Takashi Iwai ec50b4cea6 ALSA: hda - Add fixup for Acer AO725 laptop
Acer AO725 needs the same fixup as AO756.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-19 12:17:54 +01:00
Joe Millenbach 4f73bc4dd3 tty: Added a CONFIG_TTY option to allow removal of TTY
The option allows you to remove TTY and compile without errors. This
saves space on systems that won't support TTY interfaces anyway.
bloat-o-meter output is below.

The bulk of this patch consists of Kconfig changes adding "depends on
TTY" to various serial devices and similar drivers that require the TTY
layer.  Ideally, these dependencies would occur on a common intermediate
symbol such as SERIO, but most drivers "select SERIO" rather than
"depends on SERIO", and "select" does not respect dependencies.

bloat-o-meter output comparing our previous minimal to new minimal by
removing TTY.  The list is filtered to not show removed entries with awk
'$3 != "-"' as the list was very long.

add/remove: 0/226 grow/shrink: 2/14 up/down: 6/-35356 (-35350)
function                                     old     new   delta
chr_dev_init                                 166     170      +4
allow_signal                                  80      82      +2
static.__warned                              143     142      -1
disallow_signal                               63      62      -1
__set_special_pids                            95      94      -1
unregister_console                           126     121      -5
start_kernel                                 546     541      -5
register_console                             593     588      -5
copy_from_user                                45      40      -5
sys_setsid                                   128     120      -8
sys_vhangup                                   32      19     -13
do_exit                                     1543    1526     -17
bitmap_zero                                   60      40     -20
arch_local_irq_save                          137     117     -20
release_task                                 674     652     -22
static.spin_unlock_irqrestore                308     260     -48

Signed-off-by: Joe Millenbach <jmillenbach@gmail.com>
Reviewed-by: Jamey Sharp <jamey@minilop.net>
Reviewed-by: Josh Triplett <josh@joshtriplett.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2013-01-18 16:15:27 -08:00
Takashi Iwai 164a7adac9 ALSA: hda/conexant - Set mixer NID 0x19 for CX20551 codec
Conexant CX20551 codec has a mixer in NID 0x19 and a few outputs have
to take the input through this widget.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 18:27:29 +01:00
Takashi Iwai cf799aa300 ALSA: hda - Correct more array rooms in hda_gen_spec
Looking through the whole definitions, some fields have inappropriate
array sizes, especially about the capture.  The array assigned to each
input (pin) should have HDA_MAX_NUM_INPUTS entries while the array
assigned to each ADC should have AUTO_CFG_MAX_INS entries.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 16:38:08 +01:00
Takashi Iwai 2a8d53916b ALSA: hda - Fix the wrong adc_idx for capture source
The patch "ALSA: hda - fix wrong adc_idx in generic parser" fixed the
adc_idx for the capture volume and capture switch controls.  But also
modified the adc_idx retrieval for the capture source controls
wrongly.  As multiple capture source controls are created in a single
shot with counts > 1, the id.index doesn't contain the real value.
The real index has to be taken via snd_ctl_get_ioffidx() as in the
original code.

This patch reverts the fixes partially to recover from the
regression.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 16:26:37 +01:00
David Henningsson 3f25dcf691 ALSA: hda - Don't add unnecessary indices on HDMI and SPDIF
If there's one each of HDMI and SPDIF, we should not add an index
on the one that comes second.

[slight code refactoring by tiwai]

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 15:56:07 +01:00
David Henningsson 8e8db7f123 ALSA: hda - don't compare with yourself in fill_input_pin_labels
Just stumbled over this one while reading the code.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 15:49:04 +01:00
David Henningsson d3d982f744 ALSA: hda - make sure there are enough input labels and paths
I found a codec configuration which had six inputs, so the max of
five was not appropriate.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 15:48:13 +01:00
Takashi Iwai 7513e6dae5 ALSA: hda - Fix speaker pin of FSC Lifebook S7110 laptop
Some BIOS version of FSC Lifebook S7110 laptop seems to give a wrong
default pin config for NID 0x15, which confuses the parser.  Give a
fixup to correct the value.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 15:41:34 +01:00
Takashi Iwai 1799cdd51a ALSA: hda - Add boost to line inputs, too
Although I commented that boost volumes would be added only for
line-in and mic pins in the source code, the actual code excludes but
for mic-in.  Fix it to accept the line-ins, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 14:37:16 +01:00
Takashi Iwai 49920427ec ALSA: hda/sigmatel - Add bass speaker support for HP ENVY Spectre XT
The pin configuration for the bass speaker needs to be corrected in a
fixup.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 14:34:30 +01:00
Takashi Iwai a90229e051 ALSA: hda - Consolidate cap_sync_hook and capture_switch_hook
Two hooks in hda_gen_spec, cap_sync_hook and capture_switch_hook, play
very similar roles.  The only differences are that the former is
called more often (e.g. at init or switching capsrc) while the latter
can take an on/off argument.

As a more generic implementation, consolidate these two hooks, and
pass snd_ctl_elem_value pointer as the second argument.  If the
secondary argument is non-NULL, it can take the on/off value, so the
caller handles it like the former capture_switch_hook.  If it's NULL,
it's called in the init or capsrc switch case.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 14:34:22 +01:00
Takashi Iwai a35bd1e3e6 ALSA: hda - Fix missing call of capture_switch_hook
When a standard capture switch without multiple binding is used, the
call for capture_switch_hook isn't called properly.  Replace the put
ops to add the hook call in that case.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 14:02:51 +01:00
David Henningsson e04340375a ALSA: hda - Fix mute led for another HP machine
This machine also has the "HP_Mute_LED_0_A" string in DMI information.

Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1096789
Tested-by: Tammy Yang <tammy.yang@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 12:29:10 +01:00
Takashi Iwai 6f7c83afc6 ALSA: hda - Look for boost controls more deeply
In the current generic parser code, we look for the (mic) boost
controls only on input pins.  But many codecs assign the boost volume
to a widget connected to each input pin instead of the input amp of
the pin itself.

In this patch, the parser tries to look through more widgets connected
to the pin and find a boost amp.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 11:07:15 +01:00
Takashi Iwai 8999bf0af0 ALSA: hda - Fix invalid mute in path activation
When an amp in the activation path is associated with mixer controls,
activate_amp() tries to skip the initialization.  It's good, but only
if the mixer really initializes both mute and volume.  Otherwise,
either the mute of the volume is left uninitialized.

This patch adds this missing check and properly initialize the
partially controlled amps in an activation path.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 11:01:33 +01:00
Takashi Iwai c970042c12 ALSA: hda - Unify input label creations in generic parser
There are a few places creating the labels and indices of kctls for
each input pin in the current generic parser code.  This is redundant
and makes harder to maintain.  Let's create the labels and indices at
once and keep them in hda_gen_spec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 10:17:30 +01:00
Takashi Iwai 9dba205b48 ALSA: hda - Keep autocfg.input idx value in imux table
Since the imux table entries can be a subset of autocfg.input table,
the indices of these aren't always same.  For passing the proper index
value of autocfg.input at creating input ctl labels (via
snd_hda_autocfg_input_label()), keep the corresponding autocfg.input
idx value in the index field of each imux item, which isn't used in
the generic driver.

Also, this makes easier to check the invalid imux pin for stereo mix.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 10:01:15 +01:00
Mark Brown ba3b8cd26d Merge remote-tracking branch 'asoc/topic/adsp' into asoc-wm2200 2013-01-18 17:54:36 +09:00
Chris Rattray 94e205bfb7 ASoC: wm_adsp: Set ADSP1 clock rate to match sys clock
Sets the ADSP1 clock rate to match the system clock
rate. To support this the codec driver provides
details of register containing the system clock
control bits.

Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-18 17:46:28 +09:00
Takashi Iwai 8a6c21aee8 ALSA: hda - Fix missing unsol event handler in some codec drivers
This resulted in non-working auto-mute behavior, of course...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-18 07:57:46 +01:00
Mark Brown e5ddd30321 ASoC: wm5102: Add controls for firmware selection
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-18 15:07:49 +09:00
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