This is compatible with ALC255.
It is use for Lenovo.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

These two new pin tables can fix headset mic problems for several
new Dell machines.

And also delete some machines from old quirk table since the existing
pin talbes already cover them.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

New codec support for ALC891.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Sometimes PORT_EXIT messages are lost when a process is exiting.
This happens if you subscribe to the announce port with client A,
then subscribe to the announce port with client B, then kill client A.
Client B will not see the PORT_EXIT message because client A's port is
closing and is earlier in the announce port subscription list. The
for each loop will try to send the announcement to client A and fail,
then will stop trying to broadcast to other ports. Killing B works fine
since the announcement will already have gone to A. The CLIENT_EXIT
message does not get lost.

How to reproduce problem:

*** termA
$ aseqdump -p 0:1
  0:1   Port subscribed            0:1 -> 128:0

*** termB
$ aseqdump -p 0:1

*** termA
  0:1   Client start               client 129
  0:1   Port start                 129:0
  0:1   Port subscribed            0:1 -> 129:0

*** termB
  0:1   Port subscribed            0:1 -> 129:0

*** termA
^C

*** termB
  0:1   Client exit                client 128
   <--- expected Port exit as well (before client exit)
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

snd_bebob_stream_map() is not defined.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Currently mutex_destroy() is called in module's cleanup function. But after
cleaned up, this mutex is automatically released. So this function call
is meaningless.

[fixed a typo in changelog by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The constants of enum snd_efw_grp_type is for struct snd_efw_phys_grp.type.
But this member is 1 byte. Although the value is between 0x00-0xff, a constant
has 0x10000. This constant is meaningless.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

It includes descriptions to cause misreading.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

To reverse a pointer for the ring buffer, subtraction by buffer
size is better than assignment to the beginning of the buffer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

All assignment for local variables in these functions are not related to
critical section.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

snd_seq_event_dup returns -ENOMEM in some buffer-full conditions,
but usually returns -EAGAIN. Make -EAGAIN trigger the overflow
condition in snd_seq_fifo_event_in so that the fifo is cleared
and -ENOSPC is returned to userspace as stated in the alsa-lib docs.
Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The commit [e1d4d3c8: ASoC: free jack GPIOs before the sound card is
freed] introduced snd_soc_card remove callbacks to a few drivers, but
they are implemented with a wrong argument type.  The callback should
receive snd_soc_card pointer instead of snd_soc_pcm_runtime.

Fixes: e1d4d3c8 ('ASoC: free jack GPIOs before the sound card is freed')
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Merge tag 'asoc-v3.16-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: Final updates for v3.16

A few more updates from the last week of development, nothing too
exciting.  Highlights include:

- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers.
- New drivers for Analog Devices ADAU1361, ADAU1381, ADAU1761 and
  ADAU1781, and Realtek RT5677.

This is the same change as commit fb6b8e71 "ASoC: tegra: free jack
GPIOs before the sound card is freed", but applied to all other ASoC
machine drivers where code inspection indicates the same problem exists.

That commit's description is:
==========
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.

snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.

To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, guard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
==========

Note that I have not even compile-tested this in most cases, since most
of the drivers rely on specific mach-* support I don't have enabled, and
don't support COMPILE_TEST. Testing by the relevant board maintainers
would be useful.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next

Just to catch up a few small fixes for HD-audio and DMA engine.

The comment for fcp_avc_transaction() describes it doesn't support this type
of operation. But it was already supported by this commit.

00a7bb81
ALSA: firewire-lib: Add support for deferred transaction
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

It is not an error to have no cache so we shouldn't return an error code
and cause our callers to fail, just silently do nothing instead.  Thanks
to Jarkko for identify the problematic commit.
Reported-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Reported-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

The conversion to a fixup table for Replacer model with ALC260 in
commit 20f7d928 took the wrong widget NID for COEF setups.  Namely,
NID 0x1a should have been used instead of NID 0x20, which is the
common node for all Realtek codecs but ALC260.

Fixes: 20f7d928 ('ALSA: hda/realtek - Replace ALC260 model=replacer with the auto-parser')
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Correcion of wrong fixup entries add in commit ca8f0424 to replace
static model quirk for PB V7900 laptop (will model).

[note: the removal of ALC260_FIXUP_HP_PIN_0F chain is also needed as a
 part of the fix; otherwise the pin is set up wrongly as a headphone,
 and user-space (PulseAudio) may be wrongly trying to detect the jack
 state -- tiwai]

Fixes: ca8f0424 ('ALSA: hda/realtek - Add the fixup codes for ALC260 model=will')
Signed-off-by: Ronan Marquet <ronan.marquet@orange.fr>
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>

According to AM824 in IEC 61883-6:2002, 2 bits in LSB of label for Raw Audio
data means Valid Length Code (VBL). Ths value is:
- b00 for 24 bits sample (label is 0x40)
- b01 for 20 bits sample (label is 0x41)
- b10 for 16 bits sample (label is 0x42)

But current firewire-lib apply 24 bits label for both of 16/24 bits samples.

As long as developers investigate BeBoB/Fireworks/OXFW/Dice, all of them
have a behaviour to ignore the label. They can generate correct sound even
if firewire-lib gives wrong label (i.e. 0xff). On BeBoB, this is not only
for Raw Audio data channel, but also for IEC 60958 Conformant data channel.

So there is little possibility of regression.
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

This patch adds the Realtek ALC5677 codec driver.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

Signed-off-by: Mark Brown <broonie@linaro.org>

The patch adds the function "get_clk_info" to RL6231 shared support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

The patch adds the function of the PLL clock calculation to RL6231 shared
support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

The patch adds the RL6231 class device shared support for RT5640, RT5645 and
RT5651. The function of the DMIC clock calculation can be shared by RL6231
shared support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>

Merge branches 'topic/rt5640', 'topic/rt5645' and 'topic/rt5651' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rl6231