More HP machine need mute led support.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

According to the bug reporter (Данило Шеган), the external mic
starts to work and has proper jack detection if only pin 0x19
is marked properly as an external headset mic.

AlsaInfo at https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1328587/+attachment/4128991/+files/AlsaInfo.txt

Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1328587Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The fixup value for codec alc293 was set to
ALC269_FIXUP_DELL1_MIC_NO_PRESENCE by a mistake, if we don't fix it,
the Dock mic will be overwriten by the headset mic, this will make
the Dock mic can't work.

Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

do_posix_clock_monotonic_gettime() is a leftover from the initial
posix timer implementation which maps to ktime_get_ts() and returns
the monotonic time in a timespec.

Use ktime based ktime_get() and use the ktime_delta_us() function to
calculate the delta instead of open coding the timespec math.
Signed-off-by: NThomas Gleixner <tglx@linutronix.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

do_posix_clock_monotonic_gettime() is a leftover from the initial
posix timer implementation which maps to ktime_get_ts().
Signed-off-by: NThomas Gleixner <tglx@linutronix.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch will verify the pin's coverter selection for an active stream
when an unsol event reports this pin becomes available again after a display
mode change or hot-plug event.

For Haswell+ and Valleyview: display mode change or hot-plug can change the
transcoder:port connection and make all the involved audio pins share the 1st
converter. So the stream using 1st convertor will flow to multiple pins
but active streams using other converters will fail. This workaround
is to assure the pin selects the right conveter and an assigned converter is
not shared by other unused pins.
Signed-off-by: NMengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Cancel the optimization of compiler for struct snd_compr_avail
which size will be 0x1c in 32bit kernel while 0x20 in 64bit
kernel under the optimizer. That will make compaction between
32bit and 64bit. So add packed to fix the size of struct
snd_compr_avail to 0x1c for all platform.
Signed-off-by: NZhang Dongxing <dongxing.zhang@intel.com>
Signed-off-by: Nxiaoming wang <xiaoming.wang@intel.com>
Acked-by: NVinod Koul <vinod.koul@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Bios does not set up the pin config default correctly (everything
is set to zero). Reporter claims that 6stack-dig and 6stack-automute
solve the problem.

Alsa-info at http://www.alsa-project.org/db/?f=376c0804cbdde90bcd2cb94799407cb1cacf5d05
BugLink: https://bugs.launchpad.net/bugs/1319291Reported-by: NStefano Statuti <stefano.statuti@hotmail.it>
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This reverts commit 7189eb9b.

It will use LPIB to get the DMA position on Broadwell HDMI Audio.
Signed-off-by: NLibin Yang <libin.yang@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Broadwell HDMI can't use position buffer reliably, force to use LPIB
Signed-off-by: NLibin Yang <libin.yang@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

New codec suooprt of ALC667.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Some vendor has special bonding options.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This is compatible with ALC255.
It is use for Lenovo.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Signed-off-by: NTakashi 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: NHui Wang <hui.wang@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

New codec support for ALC891.
Signed-off-by: NKailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi 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: NAdam Goode <agoode@google.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

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

All assignment for local variables in these functions are not related to
critical section.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NAdam Goode <agoode@google.com>
Signed-off-by: NTakashi 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: NMark Brown <broonie@linaro.org>
Signed-off-by: NTakashi 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: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NMark Brown <broonie@linaro.org>

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