bacon: Disable JTAG support

 * This has no place in production.

Change-Id: Iacd267982143e6783cf4e948dd3187d549e16156
Signed-off-by: franciscofranco <franciscofranco.1990@gmail.com>
net: guard tcp_set_keepalive() to tcp sockets

Its possible to use RAW sockets to get a crash in
tcp_set_keepalive() / sk_reset_timer()

Fix is to make sure socket is a SOCK_STREAM one.

Change-Id: Ieeb498a3e623cfcb54e1c865a3c0229e4acf1e87
Reported-by: Dave Jones <davej@redhat.com>
Signed-off-by: Eric Dumazet <edumazet@google.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
Signed-off-by: franciscofranco <franciscofranco.1990@gmail.com>
ALSA: snd-usb: add support for DSD DOP stream transport

In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for the official document.

The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.

To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.

The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: add DSD formats

This patch adds two formats for Direct Stream Digital (DSD), a
pulse-density encoding format which is described here:
https://en.wikipedia.org/wiki/Direct_Stream_Digital

DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit
stream.

The two new types added by this patch describe streams that are capable
of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8
or x16 data rate, respectively).

DSD itself specifies samples in *bit*, while DOP and ALSA handle them
as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample
rare configuration, according to the following table:

                                                  configured hardware
        176.4KHz   352.8kHz   705.6KHz     <----       sample rate

8-bit                2.8MHz     5.6MHz
16-bit    2.8Mhz     5.6MHz    11.2MHz

         `-----------------------------'
             actual DSD sample rates

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: snd-usb: add support for bit-reversed byte formats

There is quite some confusion around the bit-ordering in DSD samples,
and no general agreement that defines whether hardware is supposed to
expect the oldest sample in the MSB or the LSB of a byte.

ALSA will hence set the rule that on the software API layer, bytes
always carry the oldest bit in the most significant bit of a byte, and
the driver has to translate that at runtime in order to match the
hardware layout.

This patch adds support for this by adding a boolean flag to the
audio format struct.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: snd-usb: add quirks handler for DSD streams

Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.

That quirks handler can augment the already parsed formats bit-field,
by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop
flag in the audio format, if the driver should take care for the DOP
byte stuffing.

The only devices that are known to work with this are the ones with
a 'Playback Designs' vendor id.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb: Work around CM6631 sample rate change bug

The C-Media CM6631 USB receiver doesn't respond to changes in sample rate
while the interface is active. The same behavior is observed in other UAC2
hardware like the VIA VT1731.

Reset the interface after setting the sampling frequency on sample rate
changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is
used. Otherwise, the device will try to use the sample rate of the previous
stream, causing distorted sound on sample rate changes.

The reset is performed for all UAC2 devices, as it should not affect a
standards compliant device, but it is only necessary for C-Media CM6631,
VIA VT1731 and possibly others.

Failure to read sample rate from the device is not handled as an error in
set_sample_rate_v2(), as (permanent or intermittent) failure to read sample
rate isn't essential for a successful sample rate set.

Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Clean up the code in set_sample_rate_v2()

Just for cleaning up, introduce a new function get_sample_rate_v2()
for replacing two identical calls in set_sample_rate_v2().

No functional change.

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

ALSA: usb-audio: use endianness macros

Replace the endianness conversions with the kernel-wide swabbing macros
in get/set_sample_rate_v2.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: UAC2: do clock validity check earlier

Move the check that parse_audio_format_rates_v2() do after
receiving the clock source entity ID directly into the find
function and add a validation flag to the function.

This patch does not introduce any logic flow change.

It is provided to allow introducing automatic clock switching
easier later. By moving this uac_clock_source_is_valid callsite,
2 additional callsites can be avoided.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: UAC2: try to find and switch to valid clock

If a selector is available on a device, it may be pointing to a
clock source which is currently invalid.
If there is a valid clock source which can be selected, switch
to it.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: UAC2: auto clock selection module param

Add a module param to disable auto clock selection.
This is provided for users that expect the audio stream to
fail when the clock source is invalid (e.g., the word clock
was unintentionally disconnected).

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Conflicts:
	sound/usb/card.c

ALSA: usb-audio: show err in set_sample_rate_v2 debug

Show the error code returned from the USB subsystem in
the debug messages.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: UAC2: support read-only freq control

Some clocks might be read-only, e.g., external clocks (see also
UAC2 4.7.2.1).

In this case, setting the sample frequency will always fail
(even if the rate is equal to the current clock rate),
therefore do not write, but read the value and compare to the
requested rate.
If the clock is read only, avoid reading it twice.

If it doesn't match, return -ENXIO since the clock is invalid for
this configuration.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources

The Scarlett 2i2 seems to take almost 500 ms to set the sample rate,
even if the clock is currently set to that value. This patch speeds
up prepare of the device, by avoiding setting the clock to something
it already is.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

sound: usb: endpoint: Add dev state check in completion handler

When a USB Headset is unplug from usb host port during audio is playing,
the system is very possible to clear the urb buffer before urb completion
handler is done. Then, clear_bit call in the completion handler will access
the freed pointer and caused the system crash. Check the dev state and skip
the clear_bit call when dev stat is USB_STATE_NOTATTACHED.

CRs-fixed: 647140
Signed-off-by: Guang Jian Zeng <gzeng@codeaurora.org>

ALSA: usb-audio: spelling correction

Correct spelling of snd_usb_endpoint_implict_feedback_sink in all
occurances.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: do not trust too-big wMaxPacketSize values

commit 57e6dae1087bbaa6b33d3dd8a8e90b63888939a3 upstream.

The driver used to assume that the streaming endpoint's wMaxPacketSize
value would be an indication of how much data the endpoint expects or
sends, and compute the number of packets per URB using this value.

However, the Focusrite Scarlett 2i4 declares a value of 1024 bytes,
while only about 88 or 44 bytes are be actually used.  This discrepancy
would result in URBs with far too few packets, which would not work
correctly on the EHCI driver.

To get correct URBs, use wMaxPacketSize only as an upper limit on the
packet size.

Reported-by: James Stone <jamesmstone@gmail.com>
Tested-by: James Stone <jamesmstone@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

ALSA: usb-audio: use bitmap_weight

Use bitmap_weight to count the total number of bits set in bitmap.

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

ALSA: snd-usb: handle raw data format of UAC2 devices

UAC2 compliant audio devices may announce the capability to transport
raw audio data on their endpoints. Catch this and handle it as
'special' stream on the ALSA side.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: pcm_format_to_bits strong-typed conversion

Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.

Change such conversions to use this function and silence sparse
warnings.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Conflicts:
	include/sound/pcm.h

ALSA: usb-audio: Add missing kconfig dependecy

commit 4fa71c1550a857ff1dbfe9e99acc1f4cfec5f0d0 upstream.

The commit 44dcbbb1cd61 introduced the usage of bitreverse helpers but
forgot to add the dependency.  This patch adds the selection for
CONFIG_BITREVERSE.

Fixes: 44dcbbb1cd61 ('ALSA: snd-usb: add support for bit-reversed byte formats')
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

ALSA: usb: Parse UAC2 extension unit like for UAC1

UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in
the same way when parsing the unit. Otherwise parse_audio_unit() fails when it
sees an extension unit on a UAC2 device.

UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1.

Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: snd-usb: mixer: propagate errors up the call chain

In check_input_term() and parse_audio_feature_unit(), propagate the
error value that has been returned by a failing function instead of
-EINVAL. That helps cleaning up the error pathes in the mixer.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*

The usb_control_msg() function expects __u16 types and performs
the endianness conversions by itself.
However, in three places, a conversion is performed before it is
handed over to usb_control_msg(), which leads to a double conversion
(= no conversion):
* snd_usb_nativeinstruments_boot_quirk()
* snd_nativeinstruments_control_get()
* snd_nativeinstruments_control_put()

Caught by sparse:

sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:512:38:    expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:512:38:    got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types)
sound/usb/mixer_quirks.c:543:35:    expected unsigned short [unsigned] [usertype] value
sound/usb/mixer_quirks.c:543:35:    got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:543:56:    expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:543:56:    got restricted __le16 [usertype] <noident>
sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types)
sound/usb/quirks.c:502:35:    expected unsigned short [unsigned] [usertype] value
sound/usb/quirks.c:502:35:    got restricted __le16 [usertype] <noident>

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb: Add quirk for 192KHz recording on E-Mu devices

When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.

Userspace expected:  L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1

Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.

Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.

Signed-off-by: Calvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Trust fields given in the quirk

The maxpacksize field is given in some quirks, but it gets ignored (in
favour of wMaxPacketSize from the first endpoint.) This patch favours
the one in the quirk.

Digidesign Mbox and Mbox 2 are the only affected quirks and the devices
are assumed to be working without this patch. So for safety against the
values in the quirk being incorrect, remove them.

The datainterval is also ignored but there are not currently any quirks
which choose to override this.

Cc: Damien Zammit <damien@zamaudio.com>
Cc: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: Mark Hills <mark@xwax.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT

The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually
stuffed directly after the standard USB endpoint descriptor, and this is
where the driver currently expects it to be.

There are, however, devices in the wild that have it the other way
around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes
*before* the standard enpoint. Devices known to implement it that way
are "Sennheiser BTD-500" and Plantronics USB headsets.

When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to
change sample rates, as the bitmask for the validity of this command is
storen in bmAttributes of that descriptor.

Fix this by searching the entire interface instead of just the extra
bytes of the first endpoint, in case the latter fails.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Torstein Hegge <hegge@resisty.net>
Reported-and-tested-by: Yves G <alsa-user@vivigatt.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: store protocol version in struct audioformat

Instead of reading bInterfaceProtocol from the descriptor whenever it's
needed, store this value in the audioformat structure.  Besides
simplifying some code, this will allow us to correctly handle vendor-
specific devices where the descriptors are marked with other values.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>

ALSA: usb-audio: fix regression for fixed stream quirk

Commit 8f898e92aea2c24c7f379ee265d178f69ebb9c07 removed the redundant
reads of bInterfaceProtocol from the descriptors, but introduced a
regression to devices with quirks of type QUIRK_AUDIO_FIXED_ENDPOINT,
since fp->protocol is not set in setup process.

As a consequence, audio streams would not get initialized, as the
following logs show:

[   48.923043] setting usb interface 3:1
[   48.923056] Creating new capture data endpoint #81
[   48.923484] 4:3:1: cannot set freq 48000 to ep 0x81

This patch sets fp->protocol in create_fixed_stream_quirk() and
resolves the regression.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: rename alt_idx to altsetting

As Clemens Ladisch kindly explained:
 "Please note that there are two methods to identify alternate settings:
  the number, which is the value in bAlternateSetting, and the index,
  which is the index in the descriptor array.  There might be some wording
  in the USB spec that these two values must be the same, but in reality,
  [insert standard rant about firmware writers], bAlternateSetting
  must be treated as a random ID value."

This patch changes the name to express the correct usage semantics.
No functional change.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Fix races at disconnection and PCM closing

commit 92a586bdc06de6629dae1b357dac221253f55ff8 upstream.

When a USB-audio device is disconnected while PCM is still running, we
still see some race: the disconnect callback calls
snd_usb_endpoint_free() that calls release_urbs() and then kfree()
while a PCM stream would be closed at the same time and calls
stop_endpoints() that leads to wait_clear_urbs().  That is, the EP
object might be deallocated while a PCM stream is syncing with
wait_clear_urbs() with the same EP.

Basically calling multiple wait_clear_urbs() would work fine, also
calling wait_clear_urbs() and release_urbs() would work, too, as
wait_clear_urbs() just reads some fields in ep.  The problem is the
succeeding kfree() in snd_pcm_endpoint_free().

This patch moves out the EP deallocation into the later point, the
destructor callback.  At this stage, all PCMs must have been already
closed, so it's safe to free the objects.

Reported-by: Alan Stern <stern@rowland.harvard.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jiri Slaby <jslaby@suse.cz>

ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()

commit a5065eb6da55b226661456e6a7435f605df98111 upstream.

BugLink: http://bugs.launchpad.net/bugs/1305133

Malfunctioning or slow devices can cause a flood of dmesg SPAM.

I've ignored checkpatch.pl complaints about the use of printk_ratelimit() in favour
of prior art in sound/usb/pcm.c.

WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit
+	if (printk_ratelimit() &&

Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Eldad Zack <eldad@fogrefinery.com>
Cc: Daniel Mack <zonque@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jiri Slaby <jslaby@suse.cz>

ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while DEBUG not defined

This (widely used) construction:

if(printk_ratelimit())
	dev_dbg()

Causes the ratelimiting to spam the kernel log with the "callbacks suppressed"
message below, even while the dev_dbg it is supposed to rate limit wouldn't
print anything because DEBUG is not defined for this device.

[  533.803964] retire_playback_urb: 852 callbacks suppressed
[  538.807930] retire_playback_urb: 852 callbacks suppressed
[  543.811897] retire_playback_urb: 852 callbacks suppressed
[  548.815745] retire_playback_urb: 852 callbacks suppressed
[  553.819826] retire_playback_urb: 852 callbacks suppressed

So use dev_dbg_ratelimited() instead of this construction.

Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Resume mixer values properly

Implement reset_resume callback so that the mixer values are properly
restored.  Still no boot quirks are called, so it might not work well
on some devices.

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

ALSA: usb-audio: Save mixer status only once at suspend

The suspend callback of usb-audio driver may be called multiple times
per suspend when multiple USB interfaces are bound to a single sound
card instance.  In such a case, it's superfluous to save the mixer
values multiple times.  This patch fixes it by checking the counter.

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

ALSA: usb-audio: Fix deadlocks at resuming

The recent addition of the USB audio mixer suspend/resume may lead to
deadlocks when the driver tries to call usb_autopm_get_interface()
recursively, since the function tries to sync with the finish of the
other calls.  For avoiding it, introduce a flag indicating the resume
operation and avoids the recursive usb_autopm_get_interface() calls
during the resume.

Reported-and-tested-by: Bryan Quigley <gquigs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: improve buffer size computations for USB PCM audio

This patch changes the way URBs are allocated and their sizes are
determined for PCM playback in the snd-usb-audio driver.  Currently
the driver allocates too few URBs for endpoints that don't use
implicit sync, making underruns more likely to occur.  This may be a
holdover from before I/O delays could be measured accurately; in any
case, it is no longer necessary.

The patch allocates as many URBs as possible, subject to four
limitations:

	The total number of URBs for the endpoint is not allowed to
	exceed MAX_URBS (which the patch increases from 8 to 12).

	The total number of packets per URB is not allowed to exceed
	MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is
	decreased from 20 to 6.

	The total duration of queued data is not allowed to exceed
	MAX_QUEUE, which is decreased from 24 ms to 18 ms.

	The total number of ALSA frames in the output queue is not
	allowed to exceed the ALSA buffer size.

The last requirement is the hardest to implement.  Currently the
number of URBs needed to fill a buffer cannot be determined in
advance, because a buffer contains a fixed number of frames whereas
the number of frames in an URB varies to match shifts in the device's
clock rate.  To solve this problem, the patch changes the logic for
deciding how many packets an URB should contain.  Rather than using as
many as possible without exceeding an ALSA period boundary, now the
driver uses only as many packets as needed to transfer a predetermined
number of frames.  As a result, unless the device's clock has an
exceedingly variable rate, the number of URBs making up each period
(and hence each buffer) will remain constant.

The overall effect of the patch is that playback works better in
low-latency settings.  The user can still specify values for
frames/period and periods/buffer that exceed the capabilities of the
hardware, of course.  But for values that are within those
capabilities, the performance will be improved.  For example, testing
shows that a high-speed device can handle 32 frames/period and 3
periods/buffer at 48 KHz, whereas the current driver starts to get
glitchy at 64 frames/period and 2 periods/buffer.

A side effect of these changes is that the "nrpacks" module parameter
is no longer used.  The patch removes it.

Signed-off-by: Alan Stern <stern@rowland.harvard.edu>
CC: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Daniel Mack <zonque@gmail.com>
Tested-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Conflicts:
	sound/usb/card.c

ALSA: usb-audio: remove unused parameter from sync_ep_set_params

Since the format is not actually used in sync_ep_set_params(),
there is no need to pass it down.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: don't deactivate URBs on in-use EP

If an endpoint in use, its associated URBs should not be
deactivated.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback data

The TEAC UD-H01 firmware sends wrong feedback frequency values, thus
causing the PC to send the samples at a wrong rate, which results in
clicks and crackles in the output.

Add a workaround to detect and fix the corruption.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
[mick37@gmx.de: use sender->udh01_fb_quirk rather than
 ep->udh01_fb_quirk in snd_usb_handle_sync_urb()]
Reported-and-tested-by: Mick <mick37@gmx.de>
Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: add a workaround for the NuForce UDH-100

The NuForce UDH-100 numbers its interfaces incorrectly, which makes the
interface associations come out wrong, which results in the driver
erroring out with the message "Audio class v2 interfaces need an
interface association".

Work around this by searching for the interface association descriptor
also in some other place where it might have ended up.

Reported-and-tested-by: Dave Helstroom <helstroom@google.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: add channel map support

Add the support for channel maps of the PCM streams on USB audio
devices.  The channel map information is already found in
ChannelConfig descriptor entries, which haven't been referred until
now.

Each chmap entry is added to audioformat list entry and copied to TLV
dynamically instead of creating a whole chmap array.

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

Conflicts:
	sound/usb/card.h

ALSA: usb: supply channel maps even when wChannelConfig is unspecified

If wChannelconfig is given for some formats but not others, userspace
might not be able to set the channel map.

This is RFC because I'm not sure what the best behaviour is - to guess
the channel map from the given number of channels (it's quite likely
that one channel is MONO and two channels is FL FR), or just to supply
UNKNOWN for all channels.

But the complete lack of channel map for a format leads userspace to
believe that the format is not available at all. Or am I
misunderstanding how this should be used?

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb - For class 2 devices, use channel map from altsettings

The channel config from the streaming descriptor is probably a
better indicator of the channel map than the input terminal.
Use the input terminal's channel map as fallback only.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb - Don't trust the channel config if the channel count changed

In case the channel count of the input terminal is not the same as
the channel count of the streaming descriptor, the channel config of
the input terminal can not be trusted. Instead fall back to a default
(guessed) channel map.

This was found on a Logitech USB Headset.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb: Fix wrong mapping of RLC and RRC channels

According to USB Audio spec v2 bits 25 and 26 of bmChannelConfig are
"Back Left of Center - BLC" and "Back Right of Center - BRC",
respectively.

They are currently assigned to ALSA channels BLC/BRC. However, the ALSA
BLC/BRC are actually the rather nonsensical "bottom left center" and
"bottom right center", so the channels will be assigned wrongly. The
comments in the USB code are also similarly wrong, so this is not
readily apparent without looking at the actual specification.

Fix the channel mapping by mapping bits 25 and 26 to RLC (Rear Left
Center) and RRC (Rear Right Center), respectively, instead.

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: add front jack channel selector for EMU0204

Add support for front jack channel selector which is present on EMU0204.
It allows to get 4 channels out of this soundcard.

Tested-by: Yury Bushmelev <jay@jay-tech.ru>
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Add a quirk for Plantronics Gamecom 780

Plantronics Gamecom 780 headset has a firmware problem, and when the
FU 0x09 volume is changed, it results in either too loud or silence
except for a very narrow range.  This patch provides a workaround,
ignoring the node, initialize the volume in a sane value and keep
untouched.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65251
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usx2y: Don't peep the card internal object

Avoid traversing the device object list of the card instance just for
checking the PCM streams.  The driver's private object already
contains the array of substream pointers, so it can be simply looked
through.  The card internal may be restructured in future, thus better
not to rely on it.

Also, this fixes the possible deadlocks in PCM mutex.  Instead of
taking multiple PCM mutexes, just take the common mutex in all
places.  Along with it, rename prepare_mutex as pcm_mutex.

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

ALSA: Define more channel map positions

For following the standard, define more channel map positions and
shuffle the items a bit:

- As both PulseAudio and gstreamer define MONO channel position
  explicitly, we should follow that, too.  The mono streams point to
  this channel position unless they are explicitly assigned to certain
  channel positions.

- Top-front-* and Top-rear-* positions are added, carried from
  PulseAudio's definitions.

- Move NA and MONO definitions at the top of table right after
  UNKNOWN, since these are more abstract in comparison with other
  practical positions.

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

Conflicts:
	Documentation/sound/alsa/Channel-Mapping-API.txt
	sound/pci/ctxfi/ctpcm.c

ALSA: Extend chmap definitions for UAC2

USB audio class 2 has more channel map positions than we currently
have.  Let's add missing definitions.

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

ALSA: snd-usb: add delay quirk for "Playback Design" products

"Playback Design" products need a 50ms delay after setting the USB
interface.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locations

It turns out the devices from Playback Design need the delay quirk
after usb_set_interface from clocks.c as well. Make it a proper
quirks function and factor out the code to quirks.c.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error

If setting the interface fails, the SUBSTREAM_FLAG_SYNC_EP_STARTED
should be cleared.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: remove assignment from if condition

Following general kernel style.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: detect implicit feedback on Roland devices

All the Roland/Edirol/BOSS USB audio devices that need implicit feedback
show this unambiguously in their descriptors, so it might be a good idea
to let the driver detect this.

This should make playback work correctly (at least with Jack) with the
following devices:
- BOSS GT-100
- BOSS JS-8 Jam Station
- Edirol M-16DX
- Roland GAIA SH-01

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>

ALSA: usb-audio: separate sync endpoint setting from set_format

Setting the sync endpoint currently takes up about half of set_format().
Move it to a dedicated function.
No functional change.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: move implicit fb quirks to separate function

Separate setting implicit feedback quirks from setting
a sync endpoint (which may also be explicit feedback or async).

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: reverse condition logic in set_sync_endpoint

Reverse logic on the conditions required to qualify for a sync endpoint
and remove one level of indendation.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: do not initialize and check implicit_fb

Since implicit_fb is not changed, !implicit_fb will always
be true - it is set only after these checks.
Similarly, there's also no need to set it at the top of the function.

Change the type of implicit_fb to bool (more appropriate).

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: remove is_playback from implicit feedback quirks

An implicit feedback endpoint can only be a capture source. The
consumer (sink) of the implicit feedback endpoint is therefore limited
to playback EPs.
Check if the target endpoint is a playback first and remove redundant
checks.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: remove implicit_fb from quirk

Since the quirks all apply to implicit feedback (the source endpoint
is always a data endpoint), there's no need to set and check
a flag for it.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: WARN_ON when alts is passed as NULL

Prevent NULL dereference in snd_usb_add_endpoints(), when
alts is passed as NULL. In this case, WARN (since this is
a non-fatal bug) and return NULL ep. Call sites treat a NULL
return value as an error.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: 6fire: fix DMA issues with URB transfer_buffer usage

Patch fixes 6fire not to use stack as URB transfer_buffer. URB buffers need to
be DMA-able, which stack is not. Furthermore, transfer_buffer should not be
allocated as part of larger device structure because DMA coherency issues and
patch fixes this issue too.

Cc: stable@vger.kernel.org
Signed-off-by: Jussi Kivilinna <jussi.kivilinna@iki.fi>
Tested-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: don't push static constants on stack for %*ph

There is no need to pass constants via stack. The width may be explicitly
specified in the format.

Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: Fix assignment of 0/1 to bool variables

Convert 0 to false and 1 to true when assigning values to bool
variables. Inspired by commit 3db1cd5c05f35fb43eb134df6f321de4e63141f2.

The simplified semantic patch that find this problem is as
follows (http://coccinelle.lip6.fr/):

@@
bool b;
@@
(
-b = 0
+b = false
|
-b = 1
+b = true
)

Signed-off-by: Peter Senna Tschudin <peter.senna@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Use standard printk helpers

Convert with dev_err() and co from snd_printk(), etc.
As there are too deep indirections (e.g. ep->chip->dev->dev),
a few new local macros, usb_audio_err() & co, are introduced.

Also, the device numbers in some messages are dropped, as they are
shown in the prefix automatically.

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

Conflicts:
	sound/usb/pcm.c

ALSA: usb-audio: work around KEF X300A firmware bug

When the driver tries to access Function Unit 10, the KEF X300A
speakers' firmware apparently locks up, making even PCM streaming
impossible.  Work around this by ignoring this FU.

Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Add Audio Advantage Micro II

This patch is adding extensive support (beside standard usb audio class)
for Audio Advantage Micro II usb sound card.
Features included:
- Access to AES bits (so now sending the IEC61937 compliant stream is
possible).
- Mixer SPDIF control added to turn on/off the optical transmitter.

Signed-off-by: Przemek Rudy <prudy1@o2.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format

freqshift is only set for the data endpoint and syncmaxsize is only set
for the sync endpoint. This results in a syncmaxsize of zero used in the
proc output feedback format calculation, which gives a feedback format
incorrectly shown as 8.16 for UAC2 devices.

As neither the data nor the sync endpoint gives all the relevant
content, output the two combined.

Also remove the sync_endpoint "packet size" which is always zero
and the sync_endpoint "momentary freq" which is constant.

Tested with UAC2 async and UAC1 adaptive, not tested with UAC1 async.

Reported-by: B. Zhang <bb.zhang@free.fr>
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: add support for many Roland/Yamaha devices

Add quirks to detect the various vendor-specific descriptors used by
Roland and Yamaha in most of their recent USB audio and MIDI devices.

Together with the previous patch, this should add audio/MIDI support for
the following USB devices:
- Edirol motion dive .tokyo performance package
- Roland MC-808 Synthesizer
- Roland BK-7m Synthesizer
- Roland VIMA JM-5/8 Synthesizer
- Roland SP-555 Sequencer
- Roland V-Synth GT Synthesizer
- Roland Music Atelier AT-75/100/300/350C/500/800/900/900C Organ
- Edirol V-Mixer M-200i/300/380/400/480/R-1000
- BOSS GT-10B Effects Processor
- Roland Fantom G6/G7/G8 Keyboard
- Cakewalk Sonar V-Studio 20/100/700 Audio Interface
- Roland GW-8 Keyboard
- Roland AX-Synth Keyboard
- Roland JUNO-Di/STAGE/Gi Keyboard
- Roland VB-99 Effects Processor
- Cakewalk UM-2G MIDI Interface
- Roland A-500S Keyboard
- Roland SD-50 Synthesizer
- Roland OCTAPAD SPD-30 Controller
- Roland Lucina AX-09 Synthesizer
- BOSS BR-800 Digital Recorder
- Roland DUO/TRI-CAPTURE (EX) Audio Interface
- BOSS RC-300 Loop Station
- Roland JUPITER-50/80 Keyboard
- Roland R-26 Recorder
- Roland SPD-SX Controller
- BOSS JS-10 Audio Player
- Roland TD-11/15/30 Drum Module
- Roland A-49/88 Keyboard
- Roland INTEGRA-7 Synthesizer
- Roland R-88 Recorder

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>

ALSA: usb-audio: claim autodetected PCM interfaces all at once

snd_card_register() registers all devices newly added since the last
call.  However, the playback/capture streams are handled as one ALSA
device, so the second /dev device will not be registered if the PCM
streams are added in two steps.

QUIRK_AUTODETECT caused the probe callback to be called once for each
interface, which triggered this problem.  Work around this by handling
this like the composite quirk, i.e., autodetecting all other interfaces
that might be used for PCM or MIDI.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>

ALSA: usb-audio: fix automatic Roland/Yamaha MIDI detection

Commit aafe77cc45a5 (ALSA: usb-audio: add support for many Roland/Yamaha
devices) had several logic errors that prevented create_auto_midi_quirk
from enumerating any MIDI ports.

Reported-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: snd-usb: handle the bmFormats field as unsigned int

This field may use up to 32 bits, so it should be handled as unsigned
int.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: snd-usb: mixer: coding style fixups

Shorten some over-long lines, multi-line comments, spurious whitespaces,
curly brakets etc.  No functional change.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: snd-usb: mixer: remove error messages on failed kmalloc()

If kmalloc() fails, warnings will be loud enough. We can safely just
return -ENOMEM in such cases.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: improve dmesg source grepability

This improves messages from commit 80acefff3bc7bd53d46c16c683ab66c0ece20810.

Cc: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Respond to suspend and resume callbacks for MIDI input

sound/usb/card.c registers USB suspend and resume but did not previously
kill the input URBs. This means that USB MIDI devices left open across
suspend/resume had non-functional input (output still usually worked,
but it looks like that is another issue). Before this change, we would
get ESHUTDOWN for each of the input URBs at suspend time, killing input.

Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb: use multiple packets per urb for Wireless USB inbound audio

For Wireless USB audio devices, use multiple isoc packets per URB for
inbound endpoints with a datainterval < 5.  This allows the WUSB host
controller to take advantage of bursting to service endpoints whose
logical polling interval is less than the 4ms minimum polling interval
limit in WUSB.

Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Whitespace cleanups for sound/usb/midi.*

Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Adjust Gamecom 780 volume level

Original patch fixed the original problem, but the sound was far too low
for most users. This patch references a compare matrix to allow the
volume levels to act normally. I personally tested this patch myself,
and volume levels returned to normal. Please see this discussion for
more details: https://bugzilla.kernel.org/show_bug.cgi?id=65251

Signed-off-by: Paul S McSpadden <fisch602@gmail.com>
Cc: <stable@vger.kernel.org> [v3.14+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: usb-audio: Use SNDRV_DEV_CODEC for mixer objects

Instead of SNDRV_DEV_LOWLEVEL, use SNDRV_DEV_CODEC type for mixer
objects so that they are managed in a proper release order.
No functional change at this point.

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

ALSA: usb-audio: Fix format string mismatch in mixer.c

Fix format string mismatch in parse_audio_selector_unit().

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

ALSA: ua101: Fix unlocked snd_pcm_stop() call

snd_pcm_stop() must be called in the PCM substream lock context.

Change-Id: I0a1305936cb22245fc9f391b2d933ee2cbd8001c
Cc: <stable@vger.kernel.org>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: dapm: Fix _PRE and _POST events for DAPM performance improvements

Ever since the DAPM performance improvements we've been marking all widgets
as not dirty after each DAPM run. Since _PRE and _POST events aren't part
of the DAPM graph this has rendered them non-functional, they will never be
marked dirty again and thus will never be run again.

Fix this by skipping them when marking widgets as not dirty.

Change-Id: I5aa40ea131921c3f1d309a6567ddff4fce6ab220
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@vger.kernel.org
ASoC: msm: qdsp6v2: check proxy port is open when closing

The current logic opens the afe proxy port only if the
proxy instance virtual port is opened. If only
pcm instance virtual port is opened and a close of the
afe proxy port is attempted, this causes the device to
crash due to a wait event timeout as the driver tries
to close an unopened afe port.
The change ensures that only if both the virtual ports
are opened, which in effect starts the proxy port,
the afe close commands for the proxy port is sent to DSP,
else return immediately.

Change-Id: I9450001c608208f3d8da22b2881a88db1c21d950
Signed-off-by: Aravind Kumar <akumark@codeaurora.org>
ASoC: msm: qdsp6v2: requeue read buffers in mmap device switch

When device switch happens, ASM might return a 0 length
buffer. For MMAP style capture cases, buffers need to be
re-queued at the event handler for this case, otherwise
ASM will have no buffers to read data into after the
device is over successful.

CRs-fixed: 700614
Change-Id: I2b8930a3058bbe3979405d5de04d3371291db42d
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
ASoC: msm: audio-effects: improve logging

Make logs more meaningful by logging values where ever
applicable.

Change-Id: I0fd0e62e543480762096d64c7ee7c92eb1204126
Signed-off-by: Dhananjay Kumar <dhakumar@codeaurora.org>
ASoC: msm: qdsp6v2: Fix bit alignment in snd_codec params

Pointer member variables in snd_codec params break bit alignment and
causes data corruption. By changing these pointers to fixed size
array variables, the bit alignment is rectified. The size has been
set to max possible size. Also, remove params which are not required
anymore.

Change-Id: Ib87bbeb07b0df1ce8a81166b319976fe54c0f013
Signed-off-by: Amit Shekhar <ashekhar@codeaurora.org>
ASoC: msm: qdsp6v2: Add the matrix mixer call during ASM loopback

Add support for matrix mixer command. The render window start and
end are set up before RUN CMD for ASM loopback is issued. This
will ensure that audio data is not dropped and audio quality
is good.

CRs-fixed: 667925
Change-Id: I406d86e9805d9f57613c0349f11cfc729ef7a24c
Signed-off-by: Damir Didjusto <damird@codeaurora.org>
ASoC: msm: qdsp6v2: Cleanup of audio drivers

Using the format strings will remove need for typecasting.
This change cleans up audio msm drivers to use the kernel
defined data types and format specifiers for printk.

Change-Id: Ie1b3d5c8dcf868aa1cc7407d3e679976dffafc76
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
ASoC: msm: qdsp6v2: Cleanup of compress offload drivers

Clean up the format specifiers for compress offload drivers
and add compat ioctl support

Change-Id: I0829830336ec62c66658d6a5199d73b61249f791
Signed-off-by: Damir Didjusto <damird@codeaurora.org>
Signed-off-by: Gopikrishnaiah Anandan <agopik@codeaurora.org>
ASoC: qdsp6v2: update perf mode flag for LL capture cases

Read perf mode flag from platform data and update to the
audio client structure, so that the right settings can be
applied to ADM for Low-latency capture use-case.
Also, update the correct bit value for ASM perf mode flag
sent to DSP.

Change-Id: If1f8b9e657a409fd9d2af19ff004a797fd987788
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
ASoC: msm: q6dspv2: add pause/resume support for audio record

Add support for pause/resume functionality for audio capture
path. When application calls stop()for for pausing, suspend
the ASM stream, so that hardware resources can be released.
Again when resuming capture streams with prepare()->start(),
ASM open and the routing call should not be called again.

CRs-fixed: 692084
Change-Id: I59e014696ab0a8c473b87361ac9b739adf538d32
Signed-off-by: Banajit Goswami <bgoswami@codeaurora.org>
ASoC: wcd9320: Fix micbias setting for MAD

For Microphone Activity Detection (MAD) to work properly, it is required
to update the micbias setting correctly to indicate the micbias is
external or internal. Correct the MAD micbias settings to account for
type of micbias (internal or external).

CRs-fixed: 684818
Change-Id: If209f9e4ad971a25fd28848ba7ba297c4f3dbf06
Signed-off-by: Bhalchandra Gajare <gajare@codeaurora.org>
ASoC: msm: qdsp6v2: Change client names for ion alloc

ASM and AFE are using the same client name when requesting
ion alloc. This is making it harder to debug which client
requested the memory.  Changing the client names to more
descriptive names.

CRs-fixed: 714707
Change-Id: Id774ce06ff5300da1de5b653812a02a22a47147c
Signed-off-by: Damir Didjusto <damird@codeaurora.org>
ASoC: msm: qdsp6v2: update int to size_t

Int aren't same size as size_t in 64bit kernel. This actually cause part of
the stack to be overwritten

Change-Id: I0ebff1fb9030fdb93a6889ee02329599ae2800fb
Signed-off-by: Fred Oh <fred@codeaurora.org>
arm/dts: Scope pic16F setup for Oppo board properly

 * OP doesn't have this hardware, and it's causing and interrupt storm
   on the i2c bus. Scope it to the devices which actually have it.

Change-Id: Ic28f73026c8ded44bf6ae84a939c26e4379e4b87
msm: kgsl: Fix gpu idle check

Fix the mask being used to check whether gpu has a pending
irq or not and check whether gpu status is idle before checking
for a pending irq. If pending irq is checked before the gpu
status then there is a possibility of the irq being set after
it has been checked but before the gpu status is checked which
will result in the logic detecting that gpu is idle with a
pending irq.

Change-Id: I5f97b6f3f4c6b58c95e91604ae87330d06685962
CRs-Fixed: 647176
Signed-off-by: Shubhraprakash Das <sadas@codeaurora.org>

msm: kgsl: Change check for gpu idle

Check whether gpu is idle by checking if any of its bits in
RBBM_STATUS is set except for AHB and HI bits.

Change-Id: I3dac8131a637c046d4c5cfde5935483cefaf8b53
Signed-off-by: Shubhraprakash Das <sadas@codeaurora.org>

msm: kgsl: Clean-up unneeded power code

No platforms with direct ebi1 connections exist, nor
is idling before changing the frequency needed.

Change-Id: I80911c81ef7303e229494ce6ba3df57ee25b24a5
Signed-off-by: Lucille Sylvester <lsylvest@codeaurora.org>

msm: kgsl: Set the GX rail voltage before enabling it

Make sure the rail turns on with a consistent voltage.  Also
this sequence moves an RPM call from after the rail is on to
before it is powered up which saves a bit of power.

Change-Id: Ided5f9777879fa48fd594c84b31cec9cdf513098
Signed-off-by: Lucille Sylvester <lsylvest@codeaurora.org>

msm: kgsl: Add traces for the constraints API

This interface helps with debugging when an application
requests for a pwrlevel constraint to be set/unset. It
also traces when the constraints actually come into affect
and when they could be revoked.

Change-Id: I638cee89253a4c2383cf082193d735fd1e279ca0
Signed-off-by: Suman Tatiraju <sumant@codeaurora.org>

Conflicts:
	drivers/gpu/msm/kgsl_pwrctrl.h

msm: kgsl: fix MIN power constraints in non-trivial cases

Switching from a MAX to a MIN power constraint in the same
context is broken.  So is switching from a MAX context constraint
to a MIN constraint on a different context.  Clarify constraint
ownership, update the timestamps accordingly, and reset constraints
when changed by the context.

Change-Id: Id27dbc20e2aac994101817af3857451c1703219e
Signed-off-by: Lucille Sylvester <lsylvest@codeaurora.org>

msm: kgsl: Remove the busy waiting loop from idle_check

Previously there were issues with extra long GPU sequences
running during the idle_check NAP attempt thwarting the
transition to NAP.  With the dispatcher architecture this
is much less likely to occur.  In addition the dispatcher
increases chances of a very short pause between a command
coming in and being submitted increasing the occurence of
"micro-NAPs" being caught by this loop.  These extremely
short NAP periods are actually a power loss on the CX rail.
Given that hw clock gating will take care of any long GPU
command tails remove the busy wait loop for power savings.

Change-Id: I4efc6215689c42ab03d4e478ad97839208d04004
Signed-off-by: Lucille Sylvester <lsylvest@codeaurora.org>

gpu: msm: Add missing trace include

Change-Id: Iefaa252a8cef43e5037f4bf3232560a44b11cb3f

msm: kgsl: Wait for dispatcher on adreno idle

When waiting for the adreno device to become idle wait for
the dispatcher to complete processing commands that are
already submitted to the ringbuffer and prevent dispatcher
from submitting more commands while a thread is waiting for
dispatcher to become idle. This is required to support mutiple
ringbuffers where it is not sufficient to just wait for current
ringbuffer to become idle. All ringbuffers need to be idle and
to do that dispatcher needs to run without submitting more
commands but retiring commands that have been submitted.

Change-Id: Idd36a8da3c230981476348682cd22aed6dab113c
Signed-off-by: Shubhraprakash Das <sadas@codeaurora.org>

Conflicts:
	drivers/gpu/msm/adreno_dispatch.c

msm: kgsl: Add CFF capture for IBs

If CFF is enabled, when an IB is submitted to hardware
parse it to find all objects in it and dump them.

Change-Id: I2caecb6a9c912a59be4c0d2acdd3949c96196653
Signed-off-by: Shubhraprakash Das <sadas@codeaurora.org>

msm: kgsl: Use the correct method for exporting trace event functions

Use EXPORT_TRACEPOINT_SYMBOL instead of my lazy "trace" wrappers for
trace_kgsl_regwrite and trace_kgsl_issueibcmds thus saving lots of
code and a shred or two of dignity along the way.

Change-Id: Ic0dedbad4f54a7522edd2f9c3d017ba1e9dba505
Signed-off-by: Jordan Crouse <jcrouse@codeaurora.org>

Conflicts:
	drivers/gpu/msm/adreno_ringbuffer.c
	drivers/gpu/msm/kgsl.h

msm: kgsl: Idle the dispatcher before writing perfcounter select

The GPU must be idle before writing to the perfcounter select
registers to avoid an AHB timeout. Idle the GPU before writing
to the register.

Change-Id: Ic0dedbadf8ad3f10f78ddf68520afe6cf98880e4
Signed-off-by: Jordan Crouse <jcrouse@codeaurora.org>

Conflicts:
	drivers/gpu/msm/adreno.c

msm: kgsl: Avoid race conditions with GPU halt variable

1. Take the device mutex before we check for GPU halt.
It's quite possible that after checking gpu halt this
thread might be preempted by kernel, other thread can
update the gpu halt. This will lead to a situation
where other thread has requested to stop sending cmds
and this thread might be sending cmds to ringbuffer.

2. Handle multiple threads changing the halt variable.
It's possible that one thread has set the halt value
and other thread clears that value.

3. init_completion() cannot be used to reinitialize
completion when reusing completion object,
INIT_COMPLETION macro should be used instead.

CRs-Fixed: 667179
Change-Id: Ie41c69be35bfd3ca14f3ce782ab45efb70aaf46a
Signed-off-by: Prakash Kamliya <pkamliya@codeaurora.org>

Conflicts:
	drivers/gpu/msm/adreno_dispatch.c

msm: kgsl: Avoid another race condition with GPU halt

A race condition may occur while clearing GPU halt
from _adreno_start() in one thread and at the same
time decrementing it inside adreno_put_gpu_halt()
from adreno_dispatcher_idle_unsafe() in the other
thread. This can cause BUG_ON as "halt" value would
become negative.

Clearing GPU halt inside _adreno_start() should not
be required explicitly for below reasons:
1. "halt" is member of static global variable, so it
would be initialized to zero at the beginning.
2. Commands submission to GPU from the thread executing
_adreno_start() should be blocked until "halt" is reset
back to zero from adreno_dispatcher_idle_unsafe().

CRs-Fixed: 678633
Change-Id: I06bbf4ad751a72b2e943365c1ac1952f67ac299c
Signed-off-by: Prabhat Awasthi <pawasthi@codeaurora.org>
msm: kgsl: Keep the GPU from going into a slumber after a touch event

If the GPU is on the tail end of the nap period when a new touch event
occurs the system could go into slumber before the first render which
increases the startup latency. If a touch event occurs while in nap
just kick the idle timer back out to the interval timeout - this should
keep the system from transitioning to the lower power state only to
come right back up.

Change-Id: Ic0dedbad07cac877d0a71c69516b7c01841be1bd
Signed-off-by: Jordan Crouse <jcrouse@codeaurora.org>
sched: Remove get_online_cpus() usage

Remove get_online_cpus() usage from the scheduler; there's 4 sites that
use it:

 - sched_init_smp(); where its completely superfluous since we're in
   'early' boot and there simply cannot be any hotplugging.

 - sched_getaffinity(); we already take a raw spinlock to protect the
   task cpus_allowed mask, this disables preemption and therefore
   also stabilizes cpu_online_mask as that's modified using
   stop_machine. However switch to active mask for symmetry with
   sched_setaffinity()/set_cpus_allowed_ptr(). We guarantee active
   mask stability by inserting sync_rcu/sched() into _cpu_down.

 - sched_setaffinity(); we don't appear to need get_online_cpus()
   either, there's two sites where hotplug appears relevant:
    * cpuset_cpus_allowed(); for the !cpuset case we use possible_mask,
      for the cpuset case we hold task_lock, which is a spinlock and
      thus for mainline disables preemption (might cause pain on RT).
    * set_cpus_allowed_ptr(); Holds all scheduler locks and thus has
      preemption properly disabled; also it already deals with hotplug
      races explicitly where it releases them.

 - migrate_swap(); we can make stop_two_cpus() do the heavy lifting for
   us with a little trickery. By adding a sync_sched/rcu() after the
   CPU_DOWN_PREPARE notifier we can provide preempt/rcu guarantees for
   cpu_active_mask. Use these to validate that both our cpus are active
   when queueing the stop work before we queue the stop_machine works
   for take_cpu_down().

CRs-fixed: 647141
Change-Id: Id41e66659574f716de0e7c29f477e56a86db9404
Signed-off-by: Peter Zijlstra <peterz@infradead.org>
Cc: "Srivatsa S. Bhat" <srivatsa.bhat@linux.vnet.ibm.com>
Cc: Paul McKenney <paulmck@linux.vnet.ibm.com>
Cc: Mel Gorman <mgorman@suse.de>
Cc: Rik van Riel <riel@redhat.com>
Cc: Srikar Dronamraju <srikar@linux.vnet.ibm.com>
Cc: Andrea Arcangeli <aarcange@redhat.com>
Cc: Johannes Weiner <hannes@cmpxchg.org>
Cc: Linus Torvalds <torvalds@linux-foundation.org>
Cc: Andrew Morton <akpm@linux-foundation.org>
Cc: Steven Rostedt <rostedt@goodmis.org>
Cc: Oleg Nesterov <oleg@redhat.com>
Link: http://lkml.kernel.org/r/20131011123820.GV3081@twins.programming.kicks-ass.net
Signed-off-by: Ingo Molnar <mingo@kernel.org>
Git-commit: 6acce3ef84520537f8a09a12c9ddbe814a584dd2
Git-repo: git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
[kaushalk@codeaurora.org: get_online_cpus has only 3 sites of usage in
 kernel/sched/core.c of msm-3.10 so migrate_swap changes are not
 applicable here. stop_two_cpus related change is not applicable to
 msm-3.10, so skip it.]
Signed-off-by: Kaushal Kumar <kaushalk@codeaurora.org>

Conflicts:
	kernel/sched/core.c

sched: Remove extra put_online_cpus() inside sched_setaffinity()

Commit 6acce3ef8:

	sched: Remove get_online_cpus() usage

has left one extra put_online_cpus() inside sched_setaffinity(),
remove it to fix the WARN:

   ------------[ cut here ]------------
   WARNING: CPU: 0 PID: 3166 at kernel/cpu.c:84 put_online_cpus+0x43/0x70()
   ...
   [<ffffffff810c3fef>] put_online_cpus+0x43/0x70 [
   [<ffffffff810efd59>] sched_setaffinity+0x7d/0x1f9 [
   ...

CRs-fixed: 647141
Change-Id: I33f799f30a963db3e9459832832e9c786931c8c2
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Tested-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Michael Wang <wangyun@linux.vnet.ibm.com>
Cc: Peter Zijlstra <peterz@infradead.org>
Link: http://lkml.kernel.org/r/526DD0EE.1090309@linux.vnet.ibm.com
Signed-off-by: Ingo Molnar <mingo@kernel.org>
Git-commit: ac9ff7997b6f2b31949dcd2495ac671fd9ddc990
Git-repo: git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
Signed-off-by: Kaushal Kumar <kaushalk@codeaurora.org>

sched: Fix endless sync_sched/rcu() loop inside _cpu_down()

Commit 6acce3ef8:

	sched: Remove get_online_cpus() usage

tries to do sync_sched/rcu() inside _cpu_down() but triggers:

	INFO: task swapper/0:1 blocked for more than 120 seconds.
	...
	[<ffffffff811263dc>] synchronize_rcu+0x2c/0x30
	[<ffffffff81d1bd82>] _cpu_down+0x2b2/0x340
	...

It was caused by that in the rcu boost case we rely on smpboot thread to
finish the rcu callback, which has already been parked before sync in here
and leads to the endless sync_sched/rcu().

This patch exchanges the sequence of smpboot_park_threads() and
sync_sched/rcu() to fix the bug.

CRs-fixed: 647141
Change-Id: Ia8b325d7c3c778f428d9fb51271977c849df32ad
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Tested-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Michael Wang <wangyun@linux.vnet.ibm.com>
Signed-off-by: Peter Zijlstra <peterz@infradead.org>
Link: http://lkml.kernel.org/r/5282EDC0.6060003@linux.vnet.ibm.com
Signed-off-by: Ingo Molnar <mingo@kernel.org>
Git-commit: 106dd5afde3cd10db7e1370b6ddc77f0b2496a75
Git-repo: git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git
Signed-off-by: Kaushal Kumar <kaushalk@codeaurora.org>

sched: Remove synchronize rcu/sched calls from _cpu_down

There is no need for sync_sched() in _cpu_down as stop_machine()
provides that barrier implicitly. Removing it also helps improve
hot-unplug latency.
The sync_sched/rcu were earlier removed for the same reason by the
commit 9ee349ad6d326df3633d43f54202427295999c47, but recently got
added as part of commit c4575f83b9fe87bb57b737bc8a667f746a20320a.

CRs-Fixed: 667325
Change-Id: I97763004454d082d3cc2d9d9dbef7da923608600
Signed-off-by: Kaushal Kumar <kaushalk@codeaurora.org>