The digital-mic unit on ASUS Eee PC gives PDM signals instead of the
normal stereo PCM, thus you can't record a mono stream from the stereo
stream as is; the summed stereo signal results in almost zero level, and
you'll hear only soft noise.

As a workaround, use ALC269-specific COEF to manipulate the dmic route
for mono, like used for ALC271x.  This is implemented as a fix-up, thus
it works only with model=auto or without REALTEK_QUIRKS Kconfig.
Reported-and-tested-by: NPavel Roskin <proski@gnu.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Avoids assigning possibly invalid address to pa, even if it
is never dereferenced.
Correct error response to reflect request object/function ids.
Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

We leak the memory allocated to 'firmware' when we fail to
release_firmware() after a kmalloc() failure in hpi_dsp_code_open().
This patch should take care of the leak.
Signed-off-by: NJesper Juhl <jj@chaosbits.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

rtctimer.c uses interfaces from linux/module.h, so it should
include that file.  This fixes build errors.
Signed-off-by: NRandy Dunlap <rdunlap@xenotime.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

552d1ef6 [ASoC: core - Optimise and refactor
pcm_new() to pass only rtd] breaks compilation of txx9aclc.c:

  CC [M]  sound/soc/txx9/txx9aclc.o
/home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c: In function 'txx9aclc_pcm_new':
/home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: error: 'card' undeclared (first use in this function)
/home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: note: each undeclared identifier is reported only once for each function it appears in
make[5]: *** [sound/soc/txx9/txx9aclc.o] Error 1

Fixed by providing a definition for card.
Signed-off-by: NRalf Baechle <ralf@linux-mips.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Apparently, there are multiple old firmware revisions in the wild for
the PCI RME MADI cards. Just add them to the list of supported devices
and treat them like their modern counterparts.
Signed-off-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In slave mode, the card can only detect the base frequency (32..48kHz)
on the MADI link (exception: 96k frames), so the real external sample
rate is this base frequency multiplied by 1, 2 or 4 depending on the
speed mode.

This patch enables 64..192kHz sample rates in clock slave mode, which
failed before due to an alleged sample rate mismatch between the MADI
link (e.g., 48kHz) and the application in DS/QS mode (e.g., 96kHz,
192kHz).
Signed-off-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When running in slave mode (no clock master), there is no way to
determine the real wirespeed on the MADI link (single/double/quad
speed). Like physical gear, simply provide the user with a tristate
switch to select the appropriate format.
Signed-off-by: NAdrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Convert array index from the loop bound to the loop index.

A simplified version of the semantic patch that fixes this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
@@
expression e1,e2,ar;
@@

for(e1 = 0; e1 < e2; e1++) { <...
  ar[
- e2
+ e1
  ]
  ...> }
// </smpl>
Signed-off-by: NJulia Lawall <julia@diku.dk>
Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This collides with powerpc exported functions from bitops.h.  Rename the
local copy in the oss soundblaster mixer and ad1848 driver.
Signed-off-by: NAndy Whitcroft <apw@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Copying hp_pins and speaker_pins from line_out_pins may confuse the
parser, and it can lead to duplicated initializations for the same pin
with a wrong DAC assignment.  The problem appears in 3.0 kernel code.

Cc: <stable@kernel.org> (for 3.0)
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

"adapter" is used as an array index in the adapters[] array so
the off by one would make us read past the end.

1c073b67 "ALSA: asihpi - Remove spurious adapter index check"
reverted Dan Rosenberg's check that would have prevented the
overflow here.
Signed-off-by: NDan Carpenter <error27@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Somce quirk models don't set adc_nids but let the parser filling it.
But the recent code has unnecessary NULL-checks of spec->input_mux,
and it resulted in NULL dereferences.
This patch fixes that regression.
Reported-and-tested-by: NOliver Neukum <oneukum@suse.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fixes bug introduced by 1c073b67.
Also declare pa local to block in which it is used.
Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch also registers all necessary callbacks to support mute LED
only when such control is enabled. And it keeps codec AFG in D0 or D1
state all the time when aggressive power managemnt is enabled for vref-out
control (and mute LED) work correctly.
Signed-off-by: NVitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This update includes the changes necessary for supporting the
CS421x family of codecs.  Previously this file only supported
the CS420x family of codecs.

This file also contains init verbs to correct several issues in
the CS421x hardware.

Behavior between the CS421x and CS420x codec families is similar,
so several functions have been reused with "if" statements to
determine which codec family (CS421x or CS420x) is present.

Also, this file will be updated sometime in the near future in
order to add support for a system using CS421x that requires
mono mix on the speaker output only.

[Fix const usages and adaption for new APIs by tiwai]
Signed-off-by: NTim Howe <tim.howe@cirrus.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Move the macros depending on snd_mask_min() and co out of pcm.h into
pcm_params.h.  Otherwise using some params_*() macros will give comiple
errors without inclusion of pcm_params.h.

Also use hw_param_interval_c() and hw_param_mask_c() for const pointer.
Reported-by: NTim Blechmann <tim@klingt.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The procedure for codec D-state change may have exceptional cases
depending on the codec chip, such as a longer delay or suppressing D3.

This patch adds a new codec ops, set_power_state() to override the system
default function.  For ease of porting, snd_hda_codec_set_power_to_all()
helper function is extracted from the default set_power_state() function.

As an example, the Conexant codec-specific delay is removed from the
default routine but moved to patch_conexant.c.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a new ops, post_suspend(), which is called after suspend() ops is
performed.  This is called only in the case of the real PM suspend, and
the codec driver can use this for further changing of D-state or
clearing the LED, etc.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

It makes little sense to enable power-saving without PM.
This removes SND_HDA_NEEDS_RESUME define so that we can use CONFIG_PM
in all places.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds checking of mute state on all outputs besides just
speakers to calculate the master mute state for mute led support.
It also renames and splits the function that does it for better code
clarity.
Signed-off-by: NVitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Codec state is not restored immediately on resume but on the first
access when power-save is enabled.  That leads to an invalid mute led
state after resume until either sound is played or some control is
changed.  This patch adds a possibility for a vendor specific patch to
restore codec state immediately after resume if required.  And it adds
code to restore IDT codecs state immediately on resume on HP systems
with mute led support.
Signed-off-by: NVitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This reverts commit d7c3e952 as it does
not currently build due to missing dependencies in the Samsung tree.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NJassi Brar <jassisinghbrar@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

Signed-off-by: NVitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Formatting a PCM name is useful for module debug too.
Add snd_prefix when making function public.

[minor coding-style fixes by tiwai]
Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix a regression in the DAC filling code in patch_realtek.c.  The already
filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0,
thus always pointed to the first DAC.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NTakashi Iwai <tiwai@suse.de>

HPI Version is used to check for firmware compatibility.
This version  will accept 4.08.xx released firmware,
and will also accept 4.09.xx beta firmware
Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Mute functionality was recently added to the DSP firmware
Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add names corresponding to new HPI node types.
Shorten some names so that constructed names don't overflow the
maximum name length.
Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Because mutex is used in adapter struct defined here.
Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fixes a sparse warning.
Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NEliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>