| Age | Commit message (Collapse) | Author |
|---|
|
commit 8974bd51a77824d91010176f9a5da28513c2e1f5 upstream.
When the system has only the headphone and the line-out jacks without
speakers, the current auto-mute code doesn't work. It's because the
spec->automute_lines flag is wrongly referred in update_speakers().
This flag must be meaningless when spec->automute_hp_lo isn't set, thus
they should be always coupled.
The patch fixes the problem and add a comment to indicate the
relationship briefly.
BugLink: http://bugs.launchpad.net/bugs/851697
Reported-by: David Henningsson <david.henningsson@canonical.com>
Tested-By: Jayne Han <jayne.han@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 763437a9e7737535b2fc72175ad4974048769be6 upstream.
wait_for_avail() in pcm_lib.c has a race in it (observed in practice by an
Intel validation group).
The function is supposed to return once space in the buffer has become
available, or if some timeout happens. The entity that creates space (irq
handler of sound driver and some such) will do a wake up on a waitqueue
that this function registers for.
However there are two races in the existing code
1) If space became available between the caller noticing there was no
space and this function actually sleeping, the wakeup is missed and the
timeout condition will happen instead
2) If a wakeup happened but not sufficient space became available, the
code will loop again and wait for more space. However, if the second
wake comes in prior to hitting the schedule_timeout_interruptible(), it
will be missed, and potentially you'll wait out until the timeout
happens.
The fix consists of using more careful setting of the current state (so
that if a wakeup happens in the main loop window, the schedule_timeout()
falls through) and by checking for available space prior to going into the
schedule_timeout() loop, but after being on the waitqueue and having the
state set to interruptible.
[tiwai: the following changes have been added to Arjan's original patch:
- merged akpm's fix for waitqueue adding order into a single patch
- reduction of duplicated code of avail check
]
Signed-off-by: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 2e1210bc3d065a6e26ff5fef228a9a7e08921d2c upstream.
This patch fixes "Surround Speaker Playback Volume" being cut off.
(Commit b4dabfc452a10 was probably meant to fix this, but it fixed
only the "Switch" name, not the "Volume" name.)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit c5d2e650bd805a00ff9af537d5b5dede598a198c upstream.
Fix the codec_name field of the dai_link to match the actual device name
of the codec. Otherwise the card won't be instantiated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 747da0f80e566500421bd7760b2e050fea3fde5e upstream.
We need to report the entire jack state to the core jack code, not just
the bits that were being updated by the caller, otherwise the status
reported by other detection methods will be omitted from the state seen
by userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 3bdf28feafc52864bd7f17b39deec64833a89d19 upstream.
'struct of_device' no longer exists, and its functionality has been merged
into platform_device. Update the MPC5200 audio DMA driver (mpc5200_dma)
accordingly. This fixes a build break.
Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 95c93d8525ebce1024bda7316f602ae45c36cd6f upstream.
dac word len value should left shift before setting
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Acked-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit bf545ed72f2eeac664695a8ea2199d9ddaef6020 upstream.
fix dac word len mask and adc tdm fmt shift value
Signed-off-by: Scott Jiang <scott.jiang.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit d2b4c7bd7eabfaa2e3e5b8107d5eeb56ac879813 upstream.
request_any_context_irq() returns a negative value on failure.
On success, it returns either IRQC_IS_HARDIRQ or IRQC_IS_NESTED.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit eade7b281c9fc18401b989c77d5e5e660b25a3b7 upstream.
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
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit da6094ea7d3c2295473d8f5134279307255d6ebf upstream.
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>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 38b65190c6ab0be8ce7cff69e734ca5b5e7fa309 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 29591ed4ac6fe00e3ff23b5be0cdc7016ef9c47e upstream.
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>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit a96edd59b2bc88b3d1ea47e0ba48076d65db9302 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 6678050442e90a4e9511a9ed14b9bdfc5e393323 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 15439bde3af7ff88459ea2b5520b77312e958df2 upstream.
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>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 824818b148db42173446707df4cbd61cd7133272 upstream.
The Focusrite Scarlett 18i6 USB has them that way, which is probably a
bug. Anyway, the driver should simply ignore this fact.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Nicolai Krakowiak <nicolai.krakowiak@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@suse.de>
|
|
commit 1faa5d07a93fc5b0a4a5254fc940a79e20b55540 upstream.
When creating the mixers for an USB audio device, the current code looks
at the host interface stored in mixer->chip->ctrl_if. Change this and
rather keep a local pointer to the interface that was given when
snd_usb_create_mixer() was called.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com>
Reported-by: Lean-Yves LENHOF <jean-yves@lenhof.eu.org>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 60c961a9e1ed879a4d151df6076bf1203f595f73 upstream.
Signed-off-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit f4389489b5cbe60b3441869c68bb4afe760969c4 upstream.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Renato <naretobh@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 0584ffa548b6e59aceb027112f23a55f0133400e upstream.
A slave-timer instance has no timer reference, and this results in
NULL-dereference at stopping the timer, typically called at closing
the device.
Reference: https://bugzilla.kernel.org/show_bug.cgi?id=40682
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 151798f872d6b386d82cd1707ad703e981fef8f2 upstream.
Cache handling in this driver is broken. The chip has 16-bit registers, yet the
register numbers also increase by 2 per register, i.e. there are only
even-numbered registers. The cache in this driver, though, simply increments
register numbers, so it does need some mapping as seen in
sgtl5000_restore_regs(), note the '>> 1':
snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
cache[SGTL5000_CHIP_LINREG_CTRL >> 1]);
That, of course, won't work with snd_soc_update_bits(). (Thus, we won't even
notice the missing register 0x1c in the default regs which shifted all follwing
registers to wrong values.) Noticed on the MX28EVK where enabling the regulators
simply locked up the chip.
Refactor the routines and use a properly sized default_regs array which matches
the register layout of the underlying chip, i.e. create a truly flat cache.
This also saves some code which should make up for the bigger array a little.
When soc-core will somewhen have another cache type which handles a step size,
this conversion will also ease the transition.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit ca9380fd68514c7bc952282c1b4fc70607e9fe43 upstream.
Convert array index from the loop bound to the loop index.
A simplified version of the semantic patch that fixes this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e1,e2,ar;
@@
for(e1 = 0; e1 < e2; e1++) { <...
ar[
- e2
+ e1
]
...> }
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit c48a8fb0d31d6147d8d76b8e2ad7f51a2fbb5c4d upstream.
Copying hp_pins and speaker_pins from line_out_pins may confuse the
parser, and it can lead to duplicated initializations for the same pin
with a wrong DAC assignment. The problem appears in 3.0 kernel code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit c81c6b356b52d3fcb4d531d149573fc100aad643 upstream.
Commit dd203fa97bd5 (ALSA: virtuoso: remove non-working controls on
Essence ST Deluxe) made it impossible to adjust the volume after the
driver initialized it to muted.
Ensure that those DACs that can be accessed with I2C are initialized
to the same volume that is the reset default of the DAC without I2C.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 7be4ba24a3ea53bc8ade841635e4d4a59e98ceb5 upstream.
Since quite a few drivers are not managing to flag the cache as needing
to be resynced after suspend and it's a reasonable thing to do flag the
cache as needing sync automatically when suspending.
The expectation is that systems will mainly only keep the CODEC powered
when doing audio through the CODEC so we won't actually suspend the
device anyway; drivers which want to can override this behaviour when
they resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 3012f43eaf7592d8121426918e43e3b5db013aff upstream.
According to DM365 voice codec data sheet at [1], before starting
recording or playback, ADC/DAC modules should follow a reset and
enable cycle. Writing a 1 to the ADC/DAC bit in the register resets
the module and clearing the bit to 0 will enable the module. But the
driver seems to be doing the reverse of it.
[1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
commit 82d1d521036eb3f5aae48b847f939d99a44c18bb upstream.
In davinci_vcif_trigger() function, a break() statement was missing
causing the davinci_vcif_stop() function to be called as a fallback
after calling davinci_vcif_start().
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: Correct WM8994 MICBIAS supply widget hookup
ASoC: Fix shift in WM8958 accessory detection default implementation
ASoC: sh: fsi-hdmi: fixup snd_soc_card name
ASoC: sh: fsi-da7210: fixup snd_soc_card name
ASoC: sh: fsi-ak4642: fixup snd_soc_card name
|
|
The WM8994 and WM8958 series of devices have two MICBIAS supplies rather
than one, the current widget actually manages the microphone detection
control register bit (which is managed separately by the relevant API).
Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2
widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
|
|
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
|
|
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
|
|
It's harmless but annyoing.
sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc
|
|
mask didn't cover update-data
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
|
|
When the dual-adc switching mode is active in Realtek auto-parser,
we need to couple all ADCs as a single capture-volume. Currently, the
volume control changes only the first ADC, thus others may remain silent.
This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
|
|
The snd_card->driver field contains a driver name string, and in
general it shouldn't contain space or special letters. The commit
2b39535b9e54888649923beaab443af212b6c0fd changed the string copy from
card->name, but the long name string may contain such letters, thus
it may still lead to a segfault.
A temporary fix is not to copy the long name string but just keep it
empty as the earlier version did.
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc
|
|
git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc
|
|
This delay is very conservative.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
|
|
The I2S controller needs a clock to respond to register writes. Without
this, register writes will at worst hang the CPU. In practice, I've only
observed writes being dropped.
Luckily, the dropped register writes historically had no effect:
TEGRA_I2S_TIMING: The value we wrote was the reset default.
TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data
when one slot was empty. The requested value was for the FIFOs to request
when four slots were empty. The DMA controller in the mainline kernel is
configured to burst a single entry at a time into the FIFO, hence there
was no issue. The only negative effect was on bus efficiency losses due
to an increased number of arbitration attempts.
However, in various non-upstream changes, the DMA controller now bursts
four entries at a time into the FIFO. If there is only space for one
entry, the data is simply dropped. In practice, this resulted in 3/4 of
samples being dropped, and playback at 4x the expected rate and pitch.
By fixing the clocking issue, this is solved.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
|
|
The char can be unsigned on some architectures. Since the code checks
the negative values, they should be declared as signed char explicitly.
sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type
sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
Fix the wrongly converted short values:
sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type
sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
The Blackfin DMA controller can report one frame beyond the end of the
buffer in the wraparound case but ALSA requires that the pointer always
be in the buffer. Do the wraparound to handle this. A similar bug is
likely to apply to the other Blackfin PCM drivers but the code is less
obvious to inspection and I don't have a user to test.
Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
|
|
I noticed that the last character of the ELD monitor name is lost,
this fixes the issue.
This fix should be confirming to the HDA spec, and works together with
the DRM part of the ELD patch.
The HDA spec does not mention that Monitor_Name_String is an '\0'
ending string, and it allows NML to be 1, which is only valid when MNL
does not count the possible ending '\0'.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
|
This patch updates the email address of the sound drivers supported by me to an
email account I will use on a more regular basis in the future.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
