Existing code only updates the audio delay when URBs were
submitted/retired. This can introduce an uncertainty of 8ms
on the number of samples played out with the default settings,
and a lot more when URBs convey more packets to reduce the
interrupt rate and power consumption.

This patch relies on the USB frame counter to reduce the
uncertainty to less than 2ms worst-case. The delay information
essentially becomes independent of the URB size and number of
packets. This should help applications like PulseAudio which
require accurate audio timing. Clemens Ladisch reported
a decrease of mplayer's A-V difference from nrpacks down to at
most 1ms.

Thanks to Clemens for also pointing out that the implementation
of frame counters varies between different HCDs. Only the
8 lowest-bits are used to estimate the delay.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
[clemens: changed debug code]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add support for Starr Labs USB MIDI devices such as the Z7S, which are
based on an FTDI serial UART chip.

Based on a patch by Daniel Mack.
Signed-off-by: Kristian Amlie <kristian@amlie.name>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

There are references in the code to 256 sources, so I tested it with 256 aplays,
of which the first and last with real data and the rest playing /dev/zero .

Also increase amount of page tables, so the default aplay size works.
Signed-off-by: Maarten Lankhorst <m.b.lankhorst@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Suppose the ALSA card already has a number of MAX_USER_CONTROLS controls, and
the user wants to replace one, it should not fail at this condition check.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

remove trailing tab on the line.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When the initial check of dB-range failed due to the read error, try to
check again at the later read, too.  When an invalid dB range is found,
remove TLV flags and notify the mixer info change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The recent fix for testing dB range at the mixer creation time seems
to cause regressions in some devices.  In such devices, reading the dB
info at probing time gives an error, thus both dBmin and dBmax are still
zero, and TLV flag isn't set although the later read of dB info succeeds.

This patch adds a workaround for such a case by assuming that the later
read will succeed.  In future, a similar test should be performed in a
case where a wrong dB range is seen even in the later read.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>

The new parser may use "PCM" volume, but it was missing the vmaster
slave list, thus "Master" volume didn't control it.

Reference: https://bugzilla.kernel.org/show_bug.cgi?id=41342Signed-off-by: Takashi Iwai <tiwai@suse.de>

Fix the duplicated creation of capture-mixer elements for some static
ALC268 configurations.  The capture mixers must be put to cap_mixer field
instead of mixers array.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Refactoring the code using snd_pcm_hw_constraint_pow2() helper function.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

AES32 supports the non-standard 128kHZ, and this is enabled only when
SNDRV_PCM_RATE_KNOT is set in hw.rates field.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Some modesl can support up to 8192 frames per period.
Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

On the Xonar Essence ST/STX, the connector J14 has been confirmed to be
a digital input, so enable it in the driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Remove a field that is not used at all.  This remained from
earlier tests, but the current driver has decided not to handle
iris notifications.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

BugLink: https://bugs.launchpad.net/bugs/826081

The original reporter needs 'Headphone Jack Sense' enabled to have
audible audio, so add his PCI SSID to the whitelist.

Reported-and-tested-by: Muhammad Khurram Khan
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Newer RME cards like RayDAT and AIO support 32 samples per period. This
value is encoded as {1,1,1} in the HDSP_LatencyMask bits in the control
register.

Since {1,1,1} is also the representation for 8192 samples/period on
older RME cards, we have to special case 32 samples and 32768 bytes
according to the actual card.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Currently, hdspm_decode_latency is called several times, violating the
DRY principle. Given that we need to distinguish between old and new
cards when decoding the latency bits in the control register, introduce
hdspm_get_latency() to provide the required functionality.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

On newer RME cards like RayDAT and AIO, the 8192 samples per period size
are no longer supported. Instead, setting all three bits of
HDSP_LatencyMask to one ({1,1,1}) now corresponds to 32 samples per
period.

To make this more obvious to future developers, let's reorder the array
according to their bit representation, starting at 64 ({0,0,0}) up to
4096 ({1,1,0}) and finally 32 ({1,1,1}).

Note that this patch doesn't change semantics.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

On newer RME cards like RayDAT and AIO, the lower bound is 32 samples
per period in contrast to 64 samples as seen on older cards.

We hence lower period_bytes_min to 32 * 4. Four bytes per sample.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Older RME cards like MADI and AES support period sizes of 8192 samples.
The original hdspm driver already featured this value, apparently, it
was lost during the rewrite.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The snd_usb_caiaq driver currently assumes that output urbs are serviced
in time and doesn't track when and whether they are given back by the
USB core. That usually works fine, but due to temporary limitations of
the XHCI stack, we faced that urbs were submitted more than once with
this approach.

As it's no good practice to fire and forget urbs anyway, this patch
introduces a proper bit mask to track which requests have been submitted
and given back.

That alone however doesn't make the driver work in case the host
controller is broken and doesn't give back urbs at all, and the output
stream will stop once all pre-allocated output urbs are consumed. But
it does prevent crashes of the controller stack in such cases.

See http://bugzilla.kernel.org/show_bug.cgi?id=40702 for more details.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Matej Laitl <matej@laitl.cz>
Cc: Sarah Sharp <sarah.a.sharp@linux.intel.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>

sound/soc/codecs/wm8750.c:784:2: warning: missing braces around initializer
sound/soc/codecs/wm8750.c:784:2: warning: (near initialization for ‘wm8750_spi_ids[2].name’)

It's because struct spi_device_id.name is a char array, not a pointer,
while the driver initializes explicitly with 0.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

My gmail account got disabled and I'm not going to reopen it.
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

I2S in Exynos4 and S5PC110(S5PV210) has a internal dma.
It can be used low power audio mode and 2nd channel transfer.
This patch can support idma.

[Reapplied after dependencies propagated through in 3.1-rc1. --broonie]
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>

This error would have no effect on current silicon revisions, the fall
through case has the same behaviour.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

This fixes faulty outbount packets in case the inbound packets
received from the hardware are fragmented and contain bogus input
iso frames. The bug has been there for ages, but for some strange
reasons, it was only triggered by newer machines in 64bit mode.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: William Light <wrl@illest.net>
Reported-by: Pedro Ribeiro <pedrib@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>

snd_azf3328_dbgcallenter is called at the very beginning of the function,
so it could be useful to call snd_azf3328_dbgcallleave at all exit points.
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

In commit 45eebda7, it add new function stac_vrefout_set, but it
is only used in code between CONFIG_SND_HDA_POWER_SAVE macro, so
add the macro to avoid such warning:

  sound/pci/hda/patch_sigmatel.c:676:12: warning: 'stac_vrefout_set' defined but not used
Signed-off-by: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Kazutomo Yoshii <kazutomo.yoshii@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

As we had no id_table only the driver name would be matched against
meaning that WM8987 devices wouldn't be bound.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>

The I2C address is misformatted and would never match.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org

Without this, request_irq on subsequent device initialization fails, and
the codec cannot be used.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Two issues were preventing module snd-soc-tegra-wm8903.ko from being
removed and re-inserted:

a) The speaker-enable GPIO is hosted by the WM8903 chip. This GPIO must
   be freed before snd_soc_unregister_card() is called, because that
   triggers wm8903.c:wm8903_remove(), which calls gpiochip_remove(), which
   then fails if any of the GPIOs are in use. To solve this, free all GPIOs
   first, so the code doesn't care where they come from.

b) We need to call snd_soc_jack_free_gpios() to match the call to
   snd_soc_jack_add_gpios() during initialization. Without this, the
   call to snd_soc_jack_add_gpios() fails during any subsequent modprobe
   and initialization, since the GPIO and IRQ are already registered. In
   turn, this causes the headphone state not to be monitored, so the
   headphone is assumed not to be plugged in, and the audio path to it is
   never enabled.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Cc: stable@kernel.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Not all PCM devices have all sub-streams. Specifically, the SPDIF driver
only supports playback and hence has no capture substream. Check whether
a substream exists before dereferencing it, when de-allocating DMA
buffers in tegra_pcm_deallocate_dma_buffer.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org

Merge branch 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-3.1