It seems git has been getting confused by the very similar contexts
for the speaker DAIs and has been applying patches to the wrong places
causing all sorts of confusion. Fix this up by hand.
Reported-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent change for USB-audio disconnection race fixes introduced a
mutex deadlock again. There is a circular dependency between
chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a
device is opened during the disconnection operation:
A. snd_usb_audio_disconnect() ->
card.c::register_mutex ->
chip->shutdown_rwsem (write) ->
snd_card_disconnect() ->
pcm.c::register_mutex ->
pcm->open_mutex
B. snd_pcm_open() ->
pcm->open_mutex ->
snd_usb_pcm_open() ->
chip->shutdown_rwsem (read)
Since the chip->shutdown_rwsem protection in the case A is required
only for turning on the chip->shutdown flag and it doesn't have to be
taken for the whole operation, we can reduce its window in
snd_usb_audio_disconnect().
Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a precedence bug because | has higher precedence than ?:. This
code was cut and pasted and I fixed a similar bug a few days ago.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dereference snd_pcm_plug_stream(plug) should come after the NULL
check snd_BUG_ON(!plug).
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dereference snd_pcm_plug_stream(plug) should come after the NULL
check snd_BUG_ON(!plug).
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is no mixer attached to the ASRC on the wm5110 only a multiplexer
to select the source for the single input line. This change correctly
defines this in the wm5110 CODEC driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no mixer attached to the ASRC on the wm5102 only a multiplexer
to select the source for the single input line. This change correctly
defines this in the wm5102 CODEC driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Asynchronous Sample Rate Converters on the wm5102/wm5110 have no
mixer attached to their input, but they do allow the input to be
selected from a number of sources via a multiplexer. Currently the
platform assumes the presence of 4 multiplexers and a mixer for each
block.
This patch adds support multiplexed single input blocks into the Arizona
platform.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Probing this device currently fails in snd_usb_audio_probe() because
the call to snd_usb_create_mixer() fails. This is due to unknown or
non-standard interface descriptor subtypes in parse_audio_unit():
usbaudio: unit 51: unexpected type 0x09
snd-usb-audio: probe of 1-8:1.0 failed with error -5
Some people are working around this by recompiling usb-audio with the
call to snd_usb_create_mixer() commented out. It would be nice to
avoid that.
While the best idea would be to look into the mixer creation failure,
a reasonable short-term solution is to use quirks to only probe the
trouble-free interfaces. This allows audio and MIDI interfaces to be
used without any obvious issues.
Interface 0 is the main one to ignore. It contains lots of
control-fu, including the unexpected interface descriptor subtypes.
Interface 5 is for firmware updates and I'm not sure how to get
support for this. Interface 3 is some sort of control interface that
I don't understand:
Interface Descriptor:
bLength 9
bDescriptorType 4
bInterfaceNumber 3
bAlternateSetting 0
bNumEndpoints 0
bInterfaceClass 1 Audio
bInterfaceSubClass 1 Control Device
bInterfaceProtocol 0
iInterface 0
AudioControl Interface Descriptor:
bLength 9
bDescriptorType 36
bDescriptorSubtype 1 (HEADER)
bcdADC 1.00
wTotalLength 9
bInCollection 1
baInterfaceNr( 0) 1
Signed-off-by: Martin Schwenke <martin@meltin.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I don't think this works as intended. '|' higher precedence than ?: so
the bitwize OR "0 | (val & STR_MOST)" is a no-op.
I have re-written it to be more clear.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: Fixes for v3.7
A few small fixes plus a large but simple change for WM5102 which writes
out a bunch of register updates to the device when we enable the clock
as recommended following chip evaluation.
In case of probe deferral, the allocated GPIO line is not freed, which
prevents it from being claimed and properly asserted in later attempts.
Fix this by using devm_gpio_request().
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Michael Hirsch <hirsch@teufel.de>
Cc: Alexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a MIDI driver for the Stanton FireWire DJ controllers.
Tested-by: Sean M. Pappalardo - D.J. Pegasus <spappalardo@mixxx.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Assume that unknown ICE1724-based cards are AC97-only that can suspend
without any additional card-specific code.
This fixes suspend on Gainward Hollywood@Home 7.1.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DAPM shutdown incorrectly uses "list" field of codec struct while
iterating over probed components (codec_dev_list). "list" field
refers to codecs registered in the system, "card_list" field is
used for probed components.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
A closer look shows that the name is not even used and can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When 2.1 speakers are detected, use the corresponding channel map
instead of the standard map with front+rear surrounds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When two built-in speakers are found on the machine, we can suppose
it's rather a 2.1 speaker system with a bass output instead of
front/surround channels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc269_toggle_power_output() was only use in ALC269VB. I rename it to
alc269vb_toggle_power_output().
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback. It turned out that the problem is that we don't
wait until all URBs are killed.
This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181
Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The RayDAT reports the sync status of its inputs in consecutive bit
positions, so all we do in hdspm_s1_sync_check is to iterate over idx:
status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
lock = (status & (0x1<<idx)) ? 1 : 0;
sync = (status & (0x100<<idx)) ? 1 : 0;
The index is given in kcontrol->private_value:
HDSPM_SYNC_CHECK("WC SyncCheck", 0),
HDSPM_SYNC_CHECK("AES SyncCheck", 1),
HDSPM_SYNC_CHECK("SPDIF SyncCheck", 2),
HDSPM_SYNC_CHECK("ADAT1 SyncCheck", 3),
HDSPM_SYNC_CHECK("ADAT2 SyncCheck", 4),
HDSPM_SYNC_CHECK("ADAT3 SyncCheck", 5),
HDSPM_SYNC_CHECK("ADAT4 SyncCheck", 6),
HDSPM_SYNC_CHECK("TCO SyncCheck", 7),
HDSPM_SYNC_CHECK("SYNC IN SyncCheck", 8),
The patch corrects the indicated sync flags by passing the proper index
value to hdspm_s1_sync_check().
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To parse properly the subwoofer outputs on ASUS G75 laptop with VT1802
codec, correct the default configurations of speaker pins 0x24 and
0x33.
Reported-by: Massimo Del Fedele <max@veneto.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1802 codec provides the invalid connection lists of NID 0x24 and
0x33 containing the routes to a non-exist widget 0x3e. This confuses
the auto-parser. Fix it up in the driver by overriding these
connections.
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In via_auto_fill_adc_nids(), the parser tries to fill dac_nids[] at
the point of the current line-out (i). When no valid path is found
for this output, this results in dac = 0, thus it creates a hole in
dac_nids[]. This confuses is_empty_dac() and trims the detected DAC
in later reference.
This patch fixes the bug by appending DAC properly to dac_nids[] in
via_auto_fill_adc_nids().
Reported-by: Massimo Del Fedele <max@veneto.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From Michal Simek:
* 'arm-next' of git://git.xilinx.com/linux-xlnx:
zynq: move static peripheral mappings
zynq: remove use of CLKDEV_LOOKUP
zynq: use pl310 device tree bindings
zynq: use GIC device tree bindings
+ Linux 3.7-rc3
From Linus Walleij:
This patchset will:
- Move all remaining headers out of arch/arm/plat-nomadik/include/plat
out to e.g. include/linux/platform_data
- Delete arch/arm/plat-nomadik
- Convert Nomadik and Ux500 to SPARSE_IRQ
* tag 'kill-plat-sparse-irq' of git://git.kernel.org/pub/scm/linux/kernel/git/linusw/linux-pinctrl:
ARM: plat-nomadik: convert platforms to SPARSE_IRQ
mfd/db8500-prcmu: use the irq_domain_add_simple()
mfd/ab8500-core: use irq_domain_add_simple()
ARM: plat-nomadik: move MTU, kill plat-nomadik
ARM: plat-nomadik: move DMA40 header to <linux/platform_data>
ARM: plat-nomadik: use DIV_ROUND_CLOSEST()
ARM: plat-nomadik: pass IRQ to timer driver
clk/ux500: explicitly include register header
pinctrl/nomadik: merge old pincfg header
pinctrl/nomadik: move the platform data header
ARM: plat-nomadik: move NMK_GPIO_PER_CHIP into gpio-nomadik.h
ARM: plat-nomadik: Introduce new DB8540 GPIO registers
Current FSI driver required set_rate() platform callback function
to set audio clock if it was master mode,
because it seemed that CPG/FSI-DIV clocks calculation depend on
platform/board/cpu.
But it was calculable regardless of platform.
This patch supports audio clock calculation method,
but the sampling rate under 32kHz is not supported at this point.
Old type set_rate() is still supported now,
but it will be deleted on next version
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When MCLK is supplied externally and BCLK and LRC are configured as outputs
(codec is master), the PLL values are only calculated correctly on the first
transmission. On subsequent transmissions, at differenct sample rates, the
wrong PLL values are used. Test for f_opclk instead of f_pllout to determine
if the PLL values are needed.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
