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commit 01f58153aefc158fd690b337d29ad140e963959d upstream.
Previously arizona_calc_fll() was checking if the target frequency is
exactly divisible by reference frequency, but should have been product
of the ratio and the reference frequency.
Also scale down the Lamba and Theta coefficients be under 16-bits in
order to match the registers.
Signed-off-by: Ryo Tsutsui <ryo.tsutsui@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 0d2b6422529a26ac4dee06196524ba9da70cf735 upstream.
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit bbfd8a19b6913f50a362457c34d49bfafe5e456e upstream.
Currently, eld_valid is never set to false, except at kernel module
load time. This patch makes sure that eld is no longer valid when
the cable is (hot-)unplugged.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit ea9b43addc4d90ca5b029f47f85ca152320a1e8d upstream.
The commit [dcda58061: ALSA: hda - Add workaround for conflicting
IEC958 controls] introduced a workaround for cards that have both
SPDIF and HDMI devices for giving device=1 to SPDIF control elements.
It turned out, however, that this workaround doesn't work well -
- The workaround checks only conflicts in a single codec, but SPDIF
and HDMI are provided by multiple codecs in many cases, and
- ALSA mixer abstraction doesn't care about the device number in ctl
elements, thus you'll get errors from amixer such as
% amixer scontrols -c 0
ALSA lib simple_none.c:1551:(simple_add1) helem (MIXER,'IEC958
Playback Switch',0,1,0) appears twice or more
amixer: Mixer hw:0 load error: Invalid argument
This patch fixes the previous broken workaround. Instead of changing
the device number of SPDIF ctl elements, shift the element indices of
such controls up to 16. Also, the conflict check is performed over
all codecs found on the bus.
HDMI devices will be put to dev=0,index=0 as before. Only the
conflicting SPDIF device is moved to a different place. The new place
of SPDIF device is supposed by the updated alsa-lib HDA-Intel.conf,
respectively.
Reported-by: Stephan Raue <stephan@openelec.tv>
Reported-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 12e31a78c70dc12897fda2489113f445c0e94a18 upstream.
Some Vaio all-in-one desktop PCs (for example VGC-LN51JGB) are affected by
the same issue that caused Vaio Z laptops to become silent: the speaker pin
must be connected to the first DAC even though the codec itself advertises
flexible routing through any of the DACs.
Use the no-primary-hp fixup for choosing the speaker pin as the primary so
that the right DAC is assigned on this device.
Signed-off-by: Fernando Luis Vazquez Cao <fernando@oss.ntt.co.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 20608731f479d48be6bcb88e727f360ddf98ddaf upstream.
Commit d45e6889ee69456a4d5b1bbb32252f460cd48fa9 ("ALSA: hda - Provide
the proper channel mapping for generic HDMI driver") added support for
custom channel maps in the HDA HDMI driver. Due to a mistake in an
'if' condition the custom map is always used even when no such map has
been set. This causes incorrect channel mapping for multichannel audio
by default.
Pass per_pin->chmap_set to hdmi_setup_channel_mapping() as a parameter
so that it can use it for detecting if a custom map has been set instead
of checking if map is NULL (which is never the case).
Reported-by: Staffan Lindberg <pike@xbmc.org>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2ad779b7329d6894a80df94e693e72eaa0d56790 upstream.
If the driver detects and invalid ELD, it gives an open error.
But it forgot to release the assigned pin, converter and spdif ctls
before returning.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2c1350fdeaefefe1a149d3b083383409f43f0daa upstream.
We've got a regression report wrt the IRQ issue related with the
power-save on a Dell machine, and disabling runtime PM works around.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=53441
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 4909a0caabb8b4352efcea223e58b86f8bc1f98b upstream.
bootresponse in snd_usb_mbox2_boot_quirk is only 12 (decimal) u8's
long, but i9s passed to snd_usb_ctl_msg as it would be 0x12 (hexa)
long. Fix that by having proper size of the array, i.e. 0x12.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 7da58046482fceb17c4a0d4afefd9507ec56de7f upstream.
The quirk for the Roland/Cakewalk A-PRO keyboards accidentally used the
wrong interface number, which prevented the driver from attaching to the
device.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit edac894389f9c9de2a1368c78809c824b343f3a5 upstream.
snd-aloop driver has no proper PM implementation, thus the PM resume
may trigger Oops due to leftover timer instance. This patch adds the
missing suspend/resume implementation.
Reported-and-tested-by: El boulangero <elboulangero@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f49a59c4471d81a233e09dda45187cc44fda009d upstream.
According to the other code in this driver and similar
code in rme96 it seems, that spin_lock_irq in
snd_rme32_capture_close function should be paired
with spin_unlock_irq.
Found by Linux Driver Verification project (linuxtesting.org).
Signed-off-by: Denis Efremov <yefremov.denis@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit dacae5a19b4cbe1b5e3a86de23ea74cbe9ec9652 upstream.
snd_ali_pointer function is called with local
interrupts disabled. However it seems very strange to
reenable them in such way.
Found by Linux Driver Verification project (linuxtesting.org).
Signed-off-by: Denis Efremov <yefremov.denis@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus-uncursed
We need this for fixing build error regressions in soc/fsl.
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For non-snoop mode, we fiddle with the page attributes of CORB/RIRB
and the position buffer, but also the ring buffers. The problem is
that the current code blindly assumes that the buffer is contiguous.
However, the ring buffers may be SG-buffers, thus a wrong vmapped
address is passed there, leading to Oops.
This patch fixes the handling for SG-buffers.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=800701
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently we use LPIB forcibly for both playback and capture for
Poulsbo and Oaktrail devices, and this seems rather problematic.
The recent fix for LPIB delay count seems working well with these
devices, so let's enable it instead.
Reported-by: Martin Weishart <martin.weishart@telosalliance.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch enables internal mic input on the machine.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1107477
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When building modules with CONFIG_SND_IMX_SOC=m in imx_v6_v7_defconfig,
we will see the following link error.
LD [M] sound/soc/fsl/snd-soc-fsl-ssi.o
LD [M] sound/soc/fsl/snd-soc-fsl-utils.o
LD [M] sound/soc/fsl/snd-soc-imx-ssi.o
LD [M] sound/soc/fsl/snd-soc-imx-audmux.o
LD [M] sound/soc/fsl/snd-soc-imx-pcm.o
sound/soc/fsl/imx-pcm-dma.o: In function `init_module':
imx-pcm-dma.c:(.init.text+0x0): multiple definition of `init_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.init.text+0x0): first defined here
sound/soc/fsl/imx-pcm-dma.o: In function `cleanup_module':
imx-pcm-dma.c:(.exit.text+0x0): multiple definition of `cleanup_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.exit.text+0x0): first defined here
make[4]: *** [sound/soc/fsl/snd-soc-imx-pcm.o] Error 1
The module snd-soc-imx-pcm is designed to link imx-pcm.o with
imx-pcm-dma.o or imx-pcm-fiq.o depending on if option SND_SOC_IMX_PCM_DMA
or SND_SOC_IMX_PCM_FIQ is enabled. Both imx-pcm-dma and imx-pcm-fiq
register their own module_platform_driver. However, these two options
are not mutually exclusive and can be enabled together. And that's
why we see above multiple init_module definition error.
Instead of having both imx-pcm-dma and imx-pcm-fiq register their
own platform_driver, we should do only once in imx-pcm.c. Using
platform_device_id to distinguish between imx-pcm-dma and imx-pcm-fiq,
we can run-time call imx-pcm-dma/fiq specific initialization in .probe
hook to have module snd-soc-imx-pcm work for both cases.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This reverts commit 25b8d31488a3fb3611651991969526b2ea475764.
While the commit fixes multiple init_module definition error with
module build, it breaks build when both imx-pcm-fiq and imx-pcm-dma
are built in as below.
LD sound/soc/fsl/snd-soc-fsl-ssi.o
LD sound/soc/fsl/snd-soc-fsl-utils.o
LD sound/soc/fsl/snd-soc-imx-ssi.o
LD sound/soc/fsl/snd-soc-imx-audmux.o
LD sound/soc/fsl/snd-soc-imx-pcm-fiq.o
LD sound/soc/fsl/snd-soc-imx-pcm-dma.o
LD sound/soc/fsl/snd-soc-eukrea-tlv320.o
LD sound/soc/fsl/snd-soc-imx-sgtl5000.o
LD sound/soc/fsl/snd-soc-imx-mc13783.o
LD sound/soc/fsl/built-in.o
sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_free':
imx-pcm.c:(.text+0x464): multiple definition of `imx_pcm_free'
sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x1a8): first defined here
sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `snd_imx_pcm_mmap':
imx-pcm.c:(.text+0x35c): multiple definition of `snd_imx_pcm_mmap'
sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0xa0): first defined here
sound/soc/fsl/snd-soc-imx-pcm-dma.o: In function `imx_pcm_new':
imx-pcm.c:(.text+0x3dc): multiple definition of `imx_pcm_new'
sound/soc/fsl/snd-soc-imx-pcm-fiq.o:imx-pcm-fiq.c:(.text+0x120): first defined here
make[4]: *** [sound/soc/fsl/built-in.o] Error 1
Let's revert the commit and find a proper fix for multiple init_module
definition error later.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Commit 23caaf19b11e (ALSA: usb-mixer: Add support for Audio Class v2.0)
forgot to adjust the length check for UAC 2.0 feature unit descriptors.
This would make the code abort on encountering a feature unit without
per-channel controls, and thus prevented the driver to work with any
device having such a unit, such as the RME Babyface or Fireface UCX.
Reported-by: Florian Hanisch <fhanisch@uni-potsdam.de>
Tested-by: Matthew Robbetts <wingfeathera@gmail.com>
Tested-by: Michael Beer <beerml@sigma6audio.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: 2.6.35+ <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.8-rc4
The usual set of driver updates, nothing too thrilling in here - one
core change for the regulator bypass mode which was just not doing the
right thing at all and a bunch of driver specifics.
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A Packard-Bell desktop machine gives no proper pin configuration from
BIOS. It's almost equivalent with the 6stack+fp standard config, just
take the existing fixup.
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=901846
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The commit [26a6cb6c: ALSA: hda - Implement a poll loop for jacks as a
module parameter] introduced the polling jack detection code, but it
also moved the call of snd_hda_jack_set_dirty_all() in the resume path
after resume/init ops call. This caused a regression when the jack
state has been changed during power-down (e.g. in the power save
mode). Since the driver doesn't probe the new jack state but keeps
using the cached value due to no dirty flag, the pin state remains
also as if the jack is still plugged.
The fix is simply moving snd_hda_jack_set_dirty_all() to the original
position.
Reported-by: Manolo Díaz <diaz.manolo@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch correctly releases the firmware if the magic string in the
firmware header does not match.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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These are just compatible with other CX2075x codecs.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Normally kmalloc() returns things that are DMA safe so not visible on all
platforms but we do need to explicitly request DMA safe memory.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Acer AO725 needs the same fixup as AO756.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This machine also has the "HP_Mute_LED_0_A" string in DMI information.
Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1096789
Tested-by: Tammy Yang <tammy.yang@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Otherwise we'll get the wrong LRCLK if we need to pick a higher BCLK than
is required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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When "alsactl restore" is performed on HDMI codecs, it tries to
restore the channel map value since the channel map controls are
writable. But hdmi_chmap_ctl_put() returns -EBADFD when no PCM stream
is assigned yet, and this results in an error message from alsactl.
Although the error is harmless, it's certainly ugly and can be
regarded as a regression.
As a workaround, this patch changes the return code in such a case to
be zero for making others happy. (A slight excuse is: when the chmap
is changed through the proper alsa-lib API, the PCM status is checked
there anyway, so we don't have to be too strict in the kernel side.)
Cc: <stable@vger.kernel.org> [v3.7+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add names of the clock sources for the M-Audio Fast Track
C400.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Attain constant real-world latency by skipping 16 data packets.
The number of packets to be skipped was found by trial and error.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Taking another look at the C400 descriptors, I see now that there is
a clock selector (0x80) for this device.
Right now, the clock source points to the internal clock (0x81), which
is also valid. When the external clock source (0x82) is selected in the
mixer, and the rates mismatch (if it's free-running it is fixed to
48KHz), xruns will occur.
Set the clock ID to the clock selector unit (0x81), which then
allows the validation code to function correctly.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A patch in the 3.2 kernel caused regression with hotplugging the
M-Audio Fast track pro, or sound after suspend. I don't have the
device so I haven't done a full analysis, but it seems userspace
(both udev and pulseaudio) got confused when a card was created,
immediately destroyed, and then created again.
However, at least one person in the bug report (martin djfun)
reports that this patch resolves the issue for him. It also leaves
a message in the log:
"snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is
a bit misleading. It is better than non-working audio, but maybe
there's a more elegant solution?
BugLink: https://bugs.launchpad.net/bugs/1095315
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The generated defines in the header are pre-shifted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The generated defines in the header are pre-shifted.
Reported-by: Heather Lomond <Heather.Lomond@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Enable bypass when the regulator is idle, not when it is in use. This is
consistent with what the few existing users actually want.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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With commit f2818d0 (ASoC: fsl: fix miscompilation of snd-soc-imx-pcm),
we will see the following build error when building modules with
CONFIG_SND_IMX_SOC=m in imx_v6_v7_defconfig.
CC [M] sound/soc/fsl/phycore-ac97.o
LD [M] sound/soc/fsl/snd-soc-fsl-ssi.o
LD [M] sound/soc/fsl/snd-soc-fsl-utils.o
LD [M] sound/soc/fsl/snd-soc-imx-ssi.o
LD [M] sound/soc/fsl/snd-soc-imx-audmux.o
LD [M] sound/soc/fsl/snd-soc-imx-pcm.o
sound/soc/fsl/imx-pcm-dma.o: In function `init_module':
imx-pcm-dma.c:(.init.text+0x0): multiple definition of `init_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.init.text+0x0): first defined here
sound/soc/fsl/imx-pcm-dma.o: In function `cleanup_module':
imx-pcm-dma.c:(.exit.text+0x0): multiple definition of `cleanup_module'
sound/soc/fsl/imx-pcm-fiq.o:imx-pcm-fiq.c:(.exit.text+0x0): first defined here
make[4]: *** [sound/soc/fsl/snd-soc-imx-pcm.o] Error 1
Instead of using bool for SND_SOC_IMX_PCM_FIQ and SND_SOC_IMX_PCM_DMA
to fix the original issue, we should completely remove SND_SOC_IMX_PCM
and have imx-pcm.o statically linked with imx-pcm-fiq.o or imx-pcm-dma.o.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for
audioformat mismatch] introduced the correction of parameters to be
set for sync EP. But since the new code assumes that the sync EP is
always paired with the data EP of another direction, it triggers Oops
when a device only with a single direction is used.
This patch adds a proper check of sync EP type and the presence of the
paired substream for avoiding the crash.
Reported-and-tested-by: Jens Axboe <axboe@kernel.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.8
Nothing terribly exciting here except for the DOUBLE_RANGE fix which
just hadn't worked before, nobody noticed due to lack of use.
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