BugLink: https://launchpad.net/bugs/677652

The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers.  Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality.  Testing was done
with an alsa-driver build from 2010-11-21.

Reported-and-tested-by: Joan Creus
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

After clk_get() pclk is checked second time instead of sample_clk check.
Signed-off-by: NVasiliy Kulikov <segoon@openwall.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

. Fix PulseAudio "ALSA driver bug" issue
  (if we have two alternated areas within a 64k DMA buffer, then max
  period size should obviously be 32k only).
  Back references:
   http://pulseaudio.org/wiki/AlsaIssues
   http://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction

When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.

PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.

Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: NAndreas Mohr <andi@lisas.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

BugLink: https://launchpad.net/bugs/677830

The original reporter states that the subwoofer does not mute when
inserting headphones.  We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).

Reported-and-tested-by: i-NoD
Cc: <stable@kernel.org>
Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

BugLink: http://launchpad.net/bugs/669092

ALC887 does not have any volume control ability on the mixer NIDs,
so put the volume controls on the dac NIDs instead. Without this
patch, ALC887 users cannot use alsamixer at all.

Cc: stable@kernel.org
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Using %pR standardizes the struct resource output.
Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

BugLink: https://launchpad.net/bugs/669279

The original reporter states: "The Master mixer does not change the
volume from the headphone output (which is affected by the headphone
mixer). Instead it only seems to control the on-board speaker volume.
This confuses PulseAudio greatly as the Master channel is merged into
the volume mix."

Fix this symptom by applying the hp_only quirk for the reporter's SSID.
The fix is applicable to all stable kernels.
Reported-and-tested-by: NBen Gamari <bgamari@gmail.com>
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: NDaniel T Chen <crimsun@ubuntu.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The Atmel SSC can divide by even numbers, not only powers of two.
Signed-off-by: NPeter Rosin <peda@axentia.se>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In each function, the value apcm is stored in the private_data field of
runtime.  At the same time the function ct_atc_pcm_free_substream is stored
in the private_free field of the same structure.  ct_atc_pcm_free_substream
dereferences and ultimately frees the value in the private_data field.  But
each function can exit in an error case with apcm having been freed, in
which case a subsequent call to the private_free function would perform a
dereference after free.  On the other hand, if the private_free field is
not initialized, it is NULL, and not invoked (see snd_pcm_detach_substream
in sound/core/pcm.c).  To avoid the introduction of a dangling pointer, the
initializations of the private_data and private_free fields are moved to
the end of the function, past any possible free of apcm.  This is safe
because the previous calls to snd_pcm_hw_constraint_integer and
snd_pcm_hw_constraint_minmax, which take runtime as an argument, do not
refer to either of these fields.

In each function, there is one error case where apcm needs to be freed, and
a call to kfree is added.

The sematic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)

// <smpl>
@@
expression e,e1,e2,e3;
identifier f,free1,free2;
expression a;
@@

*e->f = a
... when != e->f = e1
    when any
if (...) {
  ... when != free1(...,e,...)
      when != e->f = e2
* kfree(a)
  ... when != free2(...,e,...)
      when != e->f = e3
}
// </smpl>
Signed-off-by: NJulia Lawall <julia@diku.dk>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If the platform already provides a definition for these accessors
do not redefine them. The warning was caught on MIPS.
Signed-off-by: NFlorian Fainelli <florian@openwrt.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

BugLink: http://launchpad.net/bugs/673075

According to the datasheet of 92HD87B, there is a digital mic
at nid 0x11, so enable it in order to be able to use the mic.

Cc: stable@kernel.org
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The [vk][cmz]alloc(_node) family of functions return void pointers which
it's completely unnecessary/pointless to cast to other pointer types since
that happens implicitly.

This patch removes such casts from sound/oss/
Signed-off-by: NJesper Juhl <jj@chaosbits.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Patch "ASoC: tpa6130a2: Fix unbalanced regulator disables" introduced a
compiler warning "‘ret’ may be used uninitialized in this function".
Initialize ret to zero to get rid of it and making sure that the function
does not return any random error code when the code is falling through.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add support for the TempoTec/MediaTek HiFier Serenade sound card.

The PCI ID was already there, but the driver handled it like the
Fantasia model, which resulted in a dummy recording device.  As
a stereo output-only card, this model is to be handled exactly
like the HG2PCI.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Sort the PCI IDs so that they make logical sense.  Also move the card
name comments into this list because the model symbols should be (more)
self-explanationary.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add support for the Kuroutoshikou CMI8787-HG2PCI sound card.

[replaced non-latin letters in the patch by tiwai]
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The snd-hifier driver contains more duplicated code than model-specific
code, so it does not make sense for it to be a separate driver.
Handling the two-channel output restriction can be easily done in the
generic driver.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch add support for the MacBookAir3,1 and MacBookAir3,2 to the alsa
sound system.
Signed-off-by: NEdgar (gimli) Hucek <gimli@dark-green.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds support for Power/Status LED on Creative USB X-Fi S51.
There is just one LED on the device. The LED can either be On or it
can be set to Blink. There doesn't seem to be a way to switch it off.
The control message to change LED status is similar to that of
audigy2nx except that the index is to be set to 0 and value is 1 for
Blink and 0 for On.

The 'Power LED' control in alsamixer when muted will cause the LED to
Blink continuously. When unmuted  the LED will stay On. The Creative
driver under Windows sets the LED to blink whenever audio is muted.
This LED can be treated as the CMSS LED but I figured since there is
just one LED, it should be treated as the Power LED. Is that alright?

I've also changed the comment "Usb X-Fi" to "Usb X-Fi S51" as there
are other external X-Fi devices from Creative like Usb X-Fi Go and
Xmod. The volume knob and LED support patch doesn't apply to them.
Signed-off-by: NMandar Joshi <emailmandar@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

I noticed that sound/pci/asihpi/hpicmn.c::hpi_alloc_control_cache() does
not check the return value from kmalloc(), which may fail.
If kmalloc() fails we'll dereference a null pointer and things will go bad
fast.
There are two memory allocations in that function and there's also the
problem that the first may succeed and the second may fail and nothing is
done about that either which will also go wrong down the line.
Signed-off-by: NJesper Juhl <jj@chaosbits.net>
Acked-by: NEliot Blennerhassett <linux@audioscience.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add missing newlines.
Signed-off-by: NJoe Perches <joe@perches.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Include jz4740.c to SND_SOC_ALL_CODECS when the dependencies are met.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Not all bits can be read back from POWER1 so avoid corruption when using
a read/modify/write cycle by marking it non-volatile - the only thing we
read back from it is the chip revision which has diagnostic value only.
We can re-add later but that's a more invasive change than is suitable
for a bugfix.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org

converts a 1 bit signed bitfield to an unsigned.
Reported-by: NDr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: NTim Blechmann <tim@klingt.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When reading through sound/pci/cs46xx/dsp_spos.c I noticed a couple of
things in cs46xx_dsp_spos_create().

It seems to me that we don't always free the various memory buffers we
allocate and we also do some work (structure member assignment) early,
that is completely pointless if some of the memory allocations fail and
we end up just aborting the whole thing.

I don't have hardware to test, so the patch below is compile tested only,
but it makes the following changes:

- Make sure we always free all allocated memory on failures.
- Don't do pointless work assigning to structure members before we know
  all memory allocations, that may abort progress, have completed
  successfully.
- Remove some trailing whitespace.
Signed-off-by: NJesper Juhl <jj@chaosbits.net>
Tested-by: NOndrej Zary <linux@rainbow-software.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

sound/usb/pcm.c::snd_usb_pcm_check_knot() fails to check the return value
from kmalloc() and may end up dereferencing a null pointer.
The patch below (compile tested only) should take care of that little
problem.
Signed-off-by: NJesper Juhl <jj@chaosbits.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This driver has unbalanced regulator_disable when doing module loading and
unloading. This is because tpa6130a2_probe followed by tpa6130a2_remove
calls twice tpa6130a2_power(0). Fix this by implementing a state checking
in tpa6130a2_power.
Signed-off-by: NJarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Do not allow invalid (too big) nSample value, when FIFO Mode1
and automatic fifo configuration has been selected.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Limit the time window to maximum 1s in the macro.
The driver deals with much shorter times (<200ms).
This will fix a rare division by zero bug in Mode1.
This could happen, when the work is not executed in
time (within mode1_latency) after the interrupt.
In this case the DAC33 will not receive the needed
nSample command in time, and enters to an unknown
state, and won't recover.
In such event the time window will increase, and
eventually going to be bigger than 1s, resulting
devision by zero.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Correct/Implement handling of broken chip.
Fail the soc_prope if the communication with the chip
fails (can not read chip ID).
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

strict_strtoul() has just been made must check so do so.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

There are two USB Audio Class specifications (v1 and v2), but neither of
them clearly defines the feedback format for high-speed UAC v1 devices.
Add to this whatever the Creative and M-Audio firmware writers have been
smoking, and it becomes impossible to predict the exact feedback format
used by a particular device.

Therefore, automatically detect the feedback format by looking at the
magnitude of the first received feedback value.

Also, this allows us to get rid of some special cases for E-Mu devices.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The ctrl_xxx routines are deprecated, switch over to the __raw_xxx
versions.
Signed-off-by: NPaul Mundt <lethal@linux-sh.org>

This fixes the following warning:

sound/soc/codecs/wm9090.c:668:12: warning: 'wm9090_i2c_remove' defined but not used
Signed-off-by: NArnaud Lacombe <lacombar@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This fixes the following warning:

sound/soc/codecs/max98088.c:2054:12: warning: 'max98088_i2c_remove' defined but not used
Signed-off-by: NArnaud Lacombe <lacombar@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>