Takashi Iwai [Mon, 29 Jun 2015 06:38:02 +0000 (08:38 +0200)]
ALSA: hda - Fix the dock headphone output on Fujitsu Lifebook E780
Fujitsu Lifebook E780 sets the sequence number 0x0f to only only of
the two headphones, thus the driver tries to assign another as the
line-out, and this results in the inconsistent mapping between the
created jack ctl and the actual I/O. Due to this, PulseAudio doesn't
handle it properly and gets the silent output.
The fix is to ignore the non-HP sequencer checks.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99681
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Sat, 27 Jun 2015 08:21:13 +0000 (10:21 +0200)]
ALSA: hda - Add headset support to Acer Aspire V5
Acer Aspire V5 with ALC282 codec needs the similar quirk like Dell
laptops to support the headset mic. The headset mic pin is 0x19 and
it's not exposed by BIOS, thus we need to fix the pincfg as well.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=96201
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Hui Wang [Fri, 26 Jun 2015 04:35:17 +0000 (12:35 +0800)]
ALSA: hda - restore the MIC FIXUP for some Dell machines
Those FIXUPs were applied to the machines through pin quirks, but
recently the PCI_QUIRK makes them can't apply to the machines.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99851
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org> # v4.1
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 26 Jun 2015 04:59:57 +0000 (06:59 +0200)]
ALSA: jack: Fix endless loop at unique index detection
While the commit [
d0a601c278de: ALSA: jack: Fix the id uniqueness
check] fixes the wrong string check, it leads to a worse result -- the
loop in get_available_index() goes into an endless loop. The cause is
that snd_ctl_find_id() returns the object assigned to the numid if
it's set. Thus it points to the previous entry again.
This patch clears the numid field for the next call properly.
Reported-and-tested-by: Tomáš Pružina <pruzinat@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Alex Deucher [Wed, 24 Jun 2015 18:37:18 +0000 (14:37 -0400)]
ALSA: hda - set proper caps for newer AMD hda audio in KB/KV
Fixes audio problems on newer asics.
Noticed by: Kelly Anderson <kelly@xilka.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 25 Jun 2015 06:48:54 +0000 (08:48 +0200)]
ALSA: hda - Disable widget power-save for VIA codecs
The widget power-save that was enabled in 4.1 kernel seems resulting
in the silent output on VIA codecs by some reason. Some widgets get
wrong power states.
As a quick fix, turn this flag off while keeping power_down_unused
flag. This will bring back to the state of 4.0.x.
Fixes:
688b12cc3ca8 ('ALSA: hda - Use the new power control for VIA codecs')
Reported-and-tested-by: Harald Dunkel <harri@afaics.de>
Cc: <stable@vger.kernel.org> # v4.1
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Wed, 24 Jun 2015 08:46:33 +0000 (10:46 +0200)]
ALSA: hda - Fix Dock Headphone on Thinkpad X250 seen as a Line Out
Thinkpad X250, when attached to a dock, has two headphone outs but
no line out. Make sure we don't try to turn this into one headphone
and one line out (since that disables the headphone amp on the dock).
Alsa-info at http://www.alsa-project.org/db/?f=
36f8764e1d782397928feec715d0ef90dfddd4c1
Cc: stable@vger.kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 23 Jun 2015 09:56:22 +0000 (11:56 +0200)]
ALSA: pcm: Fix pcm_class sysfs output
The pcm_class sysfs of each PCM substream gives only "none" since the
recent code change to embed the struct device. Fix the code to point
directly to the embedded device object properly.
Fixes:
ef46c7af93f9 ('ALSA: pcm: Embed struct device')
Cc: <stable@vger.kernel.org> # v4.0+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 22 Jun 2015 11:25:34 +0000 (13:25 +0200)]
Merge tag 'asoc-v4.2-3' of git://git./linux/kernel/git/broonie/sound into for-next
ASoC: A couple more updates for v4.2
These were sitting on my laptop.
Mark Brown [Mon, 22 Jun 2015 10:19:46 +0000 (11:19 +0100)]
Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next
Mark Brown [Mon, 22 Jun 2015 10:19:45 +0000 (11:19 +0100)]
Merge remote-tracking branch 'asoc/topic/arizona' into asoc-next
Mark Brown [Mon, 22 Jun 2015 10:19:39 +0000 (11:19 +0100)]
Merge tag 'asoc-v4.2-2' into asoc-next
ASoC: Further updates for v4.2
There's a bunch of additional updates and fixes that came in since my
orignal pull request here, including DT support for rt5645 and fairly
large serieses of cleanups and improvements to tas2552 and rcar.
# gpg: Signature made Mon 22 Jun 2015 10:24:48 BST using RSA key ID
5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
Mark Brown [Mon, 22 Jun 2015 10:19:31 +0000 (11:19 +0100)]
Merge tag 'asoc-v4.2' into asoc-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
# gpg: Signature made Mon 08 Jun 2015 18:48:37 BST using RSA key ID
5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
Mark Brown [Mon, 22 Jun 2015 10:19:29 +0000 (11:19 +0100)]
Merge remote-tracking branches 'asoc/fix/wm8737', 'asoc/fix/wm8903', 'asoc/fix/wm8955' and 'asoc/fix/wm8960' into asoc-linus
Mark Brown [Mon, 22 Jun 2015 10:19:27 +0000 (11:19 +0100)]
Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fmtbit', 'asoc/fix/intel', 'asoc/fix/max98925', 'asoc/fix/rcar' and 'asoc/fix/ux500' into asoc-linus
Mark Brown [Mon, 22 Jun 2015 10:19:27 +0000 (11:19 +0100)]
Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus
Takashi Iwai [Mon, 22 Jun 2015 09:32:41 +0000 (11:32 +0200)]
Merge tag 'asoc-v4.2-2' of git://git./linux/kernel/git/broonie/sound into for-next
ASoC: Further updates for v4.2
There's a bunch of additional updates and fixes that came in since my
orignal pull request here, including DT support for rt5645 and fairly
large serieses of cleanups and improvements to tas2552 and rcar.
Mark Brown [Mon, 22 Jun 2015 09:24:37 +0000 (10:24 +0100)]
Merge remote-tracking branch 'asoc/topic/wm8995' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:36 +0000 (10:24 +0100)]
Merge remote-tracking branches 'asoc/topic/wm8523' and 'asoc/topic/wm8741' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:35 +0000 (10:24 +0100)]
Merge remote-tracking branch 'asoc/topic/tas2552' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:35 +0000 (10:24 +0100)]
Merge remote-tracking branch 'asoc/topic/simple' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:33 +0000 (10:24 +0100)]
Merge remote-tracking branches 'asoc/topic/qcom', 'asoc/topic/rcar', 'asoc/topic/rt286' and 'asoc/topic/rt5640' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:31 +0000 (10:24 +0100)]
Merge remote-tracking branches 'asoc/topic/mediatek', 'asoc/topic/ml26124' and 'asoc/topic/omap' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:30 +0000 (10:24 +0100)]
Merge remote-tracking branches 'asoc/topic/intel' and 'asoc/topic/max98090' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:30 +0000 (10:24 +0100)]
Merge remote-tracking branch 'asoc/topic/fsl' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:30 +0000 (10:24 +0100)]
Merge remote-tracking branch 'asoc/topic/davinci' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:28 +0000 (10:24 +0100)]
Merge remote-tracking branches 'asoc/topic/adsp' and 'asoc/topic/atmel' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:28 +0000 (10:24 +0100)]
Merge remote-tracking branch 'asoc/topic/rt5670' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:27 +0000 (10:24 +0100)]
Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next
Mark Brown [Mon, 22 Jun 2015 09:24:19 +0000 (10:24 +0100)]
Merge tag 'asoc-v4.2' into asoc-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
# gpg: Signature made Mon 08 Jun 2015 18:48:37 BST using RSA key ID
5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
Mark Brown [Mon, 22 Jun 2015 09:24:17 +0000 (10:24 +0100)]
Merge remote-tracking branches 'asoc/fix/wm8737', 'asoc/fix/wm8903', 'asoc/fix/wm8955' and 'asoc/fix/wm8960' into asoc-linus
Mark Brown [Mon, 22 Jun 2015 09:24:15 +0000 (10:24 +0100)]
Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fmtbit', 'asoc/fix/intel', 'asoc/fix/max98925', 'asoc/fix/rcar' and 'asoc/fix/ux500' into asoc-linus
Mark Brown [Mon, 22 Jun 2015 09:24:15 +0000 (10:24 +0100)]
Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus
Matt Ranostay [Sat, 20 Jun 2015 06:46:16 +0000 (23:46 -0700)]
ALSA: hda-beep: Update authors dead email address
Signed-off-by: Matt Ranostay <mranostay@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Richard Fitzgerald [Thu, 18 Jun 2015 12:43:19 +0000 (13:43 +0100)]
ASoC: wm_adsp: Move DSP Rate controls into the codec
The rate controls are codec-specific, it's not possible to
generically say what the range or the meaning of each control
is (or even if they exist at all) - that depends on the
particular codec.
This is currently being handled for Arizona codecs by putting
an Arizona-specific table of controls inside the wm_adsp driver.
This creates a dependency between wm_adsp and arizona.c, and is an
awkward solution if the ADSP is used in another family of codecs
Fix this by moving the Arizona-specific rate controls into the
Arizona codec drivers.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown [Fri, 19 Jun 2015 10:17:19 +0000 (11:17 +0100)]
Merge branches 'topic/adsp' and 'topic/dapm' of git://git./linux/kernel/git/broonie/sound into asoc-arizona
Axel Lin [Tue, 16 Jun 2015 11:39:09 +0000 (19:39 +0800)]
ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case
Fix trivial typo.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Vinod Koul [Wed, 17 Jun 2015 05:50:18 +0000 (11:20 +0530)]
ALSA: hda: provide default bus io ops extended hdac
A typical io ops use simple io accessors which can be common for most
drivers, so provide a default ops which will be used if driver doesn't
provide one
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vinod Koul [Wed, 17 Jun 2015 05:50:17 +0000 (11:20 +0530)]
ALSA: hda: add hda link cleanup routine
In HDA extended bus the HDA link objects are created when multilink
capabilities are parsed. We need a routine which free up these link objects
for a bus. So add snd_hdac_link_free_all routine
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vinod Koul [Wed, 17 Jun 2015 05:50:16 +0000 (11:20 +0530)]
ALSA: hda: add hdac_ext stream creation and cleanup routines
HDAC extended core should create streams for an extended bus and also free
up those on cleanup. So introduce snd_hdac_ext_stream_init_all and
snd_hdac_stream_free_all routines
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Kuninori Morimoto [Wed, 17 Jun 2015 00:36:48 +0000 (00:36 +0000)]
ASoC: rsrc-card: remove unused ret
'
047000278da3a17f8("ASoC: rsrc-card: cleanup for DPCM")'
cleanuped rsrc-card driver, but then, unused ret was left.
Below warning happen without this patch
${LINUX}/sound/soc/sh/rcar/rsrc-card.c: In function 'rsrc_card_startup':
${LINUX}/sound/soc/sh/rcar/rsrc-card.c:78:6: warning: unused variable \
'ret' [-Wunused-variable]
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Vinod Koul [Tue, 16 Jun 2015 15:30:22 +0000 (21:00 +0530)]
ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
Since this is common option for HDA driver to specfiy pre-allocated
buffer, we should make this option availble to all HDA driver by
moving this to HDA core
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Koro Chen [Mon, 15 Jun 2015 14:38:04 +0000 (22:38 +0800)]
ASoC: mediatek: Add machine driver for rt5650 rt5676 codec
This is the DPCM based machine driver with rt5650 and rt5676
Signed-off-by: Nicolas Boichat <drinkcat@chromium.org>
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Koro Chen [Mon, 15 Jun 2015 14:38:03 +0000 (22:38 +0800)]
ASoC: mediatek: Add machine driver for MAX98090 codec
This is the DPCM based machine driver with MAX98090
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Koro Chen [Mon, 15 Jun 2015 14:38:02 +0000 (22:38 +0800)]
ASoC: mediatek: Add AFE platform driver
This is the DPCM based platform driver of AFE (Audio Front End) unit.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:28:03 +0000 (06:28 +0000)]
ASoC: rsnd: remove io from rsnd_mod
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many path
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. rsnd_mod_to_io() is no longer needed. Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:27:47 +0000 (06:27 +0000)]
ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working()
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship.
This patch checks module working status via io instead of mod
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:27:28 +0000 (06:27 +0000)]
ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from snd_kcontrol
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:27:13 +0000 (06:27 +0000)]
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx()
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_src_xxx()
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:26:56 +0000 (06:26 +0000)]
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx()
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_ssi_xxx()
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:26:25 +0000 (06:26 +0000)]
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx()
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_dma_xxx()
and related function
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:26:08 +0000 (06:26 +0000)]
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr()
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_get_adinr()
and its related function
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:25:48 +0000 (06:25 +0000)]
ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. Then, interrupt handler can't use rsnd_mod_to_io().
This patch adds SSI/SRC/DMA common interrupt handler frame
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:25:20 +0000 (06:25 +0000)]
ASoC: rsnd: adds struct rsnd_dai_stream as on each fuction as parameter
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This means we can't call rsnd_mod_to_io() any more.
This patch adds struct rsnd_dai_stream to each function as parameter.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:24:54 +0000 (06:24 +0000)]
ASoC: rsnd: rsnd_mod has rsnd_priv
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This means we can't use rsnd_mod_to_io() in SSI/SRC/DMA
interrupt handler. In such case, we need to check all io in interrupt
handler, and then, "priv" is needed.
This patch adds rsnd_priv pointer in rsnd_mod for prepare it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:24:34 +0000 (06:24 +0000)]
ASoC: rsnd: count each mod (SSI/SRC/DVC)
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. Then, we don't need to re-call each mod function
that had been called. This patch count each mod status.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:24:15 +0000 (06:24 +0000)]
ASoC: rsrc-card: cleanup for DPCM
rsrc-card driver is based on simple-card driver which is caring about
CPU / Codec connection. OTOH, rsrc-card is used for DPCM system.
FE portion is constituted by CPU and dummy Codec, and BE is constituted
by dummy CPU and Codec in DPCM system.
Because of this, current rsrc-card is doing pointless method. It works well
if FE/BE was 1:1, but not good for multi FE/BE.
This patch cleanups rsrc-card driver for DPCM. and this is prepare for
MIX support for Renesas sound driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:23:52 +0000 (06:23 +0000)]
ASoC: rsrc-card: tidyup return value of clock error
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:23:30 +0000 (06:23 +0000)]
ASoC: rsrc-card: move rsrc_card_parse_daifmt() to upper side
This is prepare for DPCM cleanup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:23:08 +0000 (06:23 +0000)]
ASoC: rsnd-card: tidyup rsrc_priv_to_()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:22:47 +0000 (06:22 +0000)]
ASoC: rsrc-card: remove unused name
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:22:30 +0000 (06:22 +0000)]
ASoC: rsrc-card: enable multi cpu/codec for DPCM
Current rsrc-card is assuming 1 FE (= CPU), 1 BE (= codec) on card.
But, it will support multi FE/BE card. This is prepare for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:22:05 +0000 (06:22 +0000)]
ASoC: rsrc-card: tidyup priv->snd_card setup timing
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:21:34 +0000 (06:21 +0000)]
ASoC: rsrc-card: used "fe.xxx"/"be.xxx" name for dai_link
Current dai_link name is using "cpu_dai_name + codec_dai_name",
but one of them is always "snd-soc-dummy-dai" when DPCM.
This patch uses "fe.xxx" for cpu, "be.xxx" for codec.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:21:15 +0000 (06:21 +0000)]
ASoC: rsnd: don't call snd_pcm_period_elapsed() under spin lock
'
a9e1ac1a9e4585b5("ASoC: rsnd: spin lock for interrupt handler")'
added spin lock under interrupt handler to solve HW restart issue.
OTOH, current rsnd driver calls snd_pcm_period_elapsed() from
rsnd_dai_pointer_update(). but, it will be called under spin lock
if SSI was PIO mode.
If it was called under spin lock, it will call
snd_pcm_update_state() -> snd_pcm_drain_done().
Then, it calls rsnd_soc_dai_trigger() and will be dead-lock.
This patch doesn't call rsnd_dai_pointer_update() under spin lock
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Kuninori Morimoto [Mon, 15 Jun 2015 06:20:54 +0000 (06:20 +0000)]
ASoC: rsnd: don't care under/over run error when PIO
PIO is used only for checking data path / codec settings. And underrun
is very normal when PIO mode. Let's don't care about under/over run
error when PIO case. Otherwise, 1) too many HW restart happens, 2) some
sounds which need much data transfer can't play since it falls into
error detection method which was created for DMA transfer
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Tue, 16 Jun 2015 10:47:34 +0000 (12:47 +0200)]
Merge branch 'for-linus' into for-next
Yet another non-trivial conflicts for HDA legacy stuff.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 16 Jun 2015 10:23:36 +0000 (12:23 +0200)]
ALSA: hda - Fix unused label skip_i915
When CONFIG_SND_HDA_I915=n, we get a compile warning:
sound/pci/hda/hda_intel.c: In function ‘azx_probe_continue’:
sound/pci/hda/hda_intel.c:1882:2: warning: label ‘skip_i915’ defined but not used [-Wunused-label]
Fix it by putting again ifdef to it. Sigh.
Fixes:
bf06848bdbe5 ('ALSA: hda - Continue probing even if i915 binding fails')
Reported-by: Borislav Petkov <bp@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Michele Curti [Mon, 15 Jun 2015 02:44:11 +0000 (10:44 +0800)]
ASoC: rt5645: move RT5650 muxes to rt5650_specific_dapm_widgets
Developing a driver for an Asus X205TA laptop I get these dmesg
errors:
rt5645 i2c-
10EC5648:00: ASoC: mux RT5650 IF1 ADC1 Swap Mux has no paths
rt5645 i2c-
10EC5648:00: ASoC: mux RT5650 IF1 ADC2 Swap Mux has no paths
rt5645 i2c-
10EC5648:00: ASoC: mux RT5650 IF1 ADC3 Swap Mux has no paths
rt5645 i2c-
10EC5648:00: ASoC: mux RT5650 IF1 ADC Mux has no paths
rt5645 i2c-
10EC5648:00: ASoC: mux RT5650 IF1 DAC1 L Mux has no paths
rt5645 i2c-
10EC5648:00: ASoC: mux RT5650 IF1 DAC1 R Mux has no paths
rt5645 i2c-
10EC5648:00: ASoC: mux RT5650 IF1 DAC2 L Mux has no paths
rt5645 i2c-
10EC5648:00: ASoC: mux RT5650 IF1 DAC2 R Mux has no paths
so, move these muxes to the rt5650_specific_dapm_widgets[] list.
Signed-off-by: Michele Curti <michele.curti@gmail.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Mon, 15 Jun 2015 18:36:12 +0000 (20:36 +0200)]
ALSA: hda - Fix noisy outputs on Dell XPS13 (2015 model)
The new Dell XPS13 also requires the similar quirk for fixing the
noisy outputs. (But, as the codec was changed, now the fixup for
Latitude is used instead.)
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=99851
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:37 +0000 (12:49 +0900)]
ALSA: bebob: add support for Behringer FCA 610/1616
They're based on DM1500 (ArchWave produced), and BeBoB version 3 is
installed.
$ cat /proc/asound/FCA610/firewire/firmware
Manufacturer: bridgeCo
Protocol Ver: 3
Build Ver: 0
GUID: 0x001564000002AD73
Model ID: 0x03
Model Rev: 0
Firmware Date:
20121102
Firmware Time: 153431
Firmware ID: 0x610
Firmware Ver: 8348
Base Addr: 0x400C0080
Max Size:
1422624
Loader Date:
20121015
Loader Time: 104710
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:36 +0000 (12:49 +0900)]
ALSA: bebob: keep duplex streams always to keep internal multiplexer properly
Behringer FCA610 transmits packets with periodic noisy PCM samples
when receiving no streams, and generates a bit noisy sound.
ALSA BeBoB driver is programmed to establish both in/out connections
when starting streaming, then transfers packets as userspace applications
requested. This means that there's a case that one of incoming/outgoing
streams is running, to save CPU and bandwidth usage. Although, it's natural
to start transferring packets in both direction.
This commit makes this driver to keeps duplex streams always.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:35 +0000 (12:49 +0900)]
ALSA: bebob: loosen up severity of checking continuity for BeBoB v3 quirk
PrismSound Orpheus, Behringer UFX1604 and FCA610 work with BeBoB v3, and
they're confirmed to transmit discontinuous packets in the beginning of
streaming.
payload CIP headers
8 0x00070000 0x9002FFFF
8 0x00070000 0x9002FFFF
8 0x00070000 0x9002FFFF
8 0x00070008 0x9002FFFF <-
8 0x00070008 0x9002FFFF
8 0x00070008 0x9002FFFF
8 0x00070008 0x9002FFFF
8 0x00070008 0x9002FFFF
8 0x00070008 0x9002FFFF
232 0x00070000 0x9002E798 <-
232 0x00070008 0x9002FB99
232 0x00070010 0x90021398
8 0x00070018 0x9002FFFF
(This sample was got with Behringer FCA610 and FFADO library.)
This commit sets CIP_EMPTY_HAS_WRONG_DBC and CIP_SKIP_DBC_ZERO_CHECK to
ignore these discontinuities.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:34 +0000 (12:49 +0900)]
ALSA: bebob: expand timeout for DM1500 quirk
Behringer FCA610 and UFX1604 is confirmed to require more time till
transmitting packets after establishing connections. This seems to
be a quirk of DM1500 ASIC which ArchWave produced.
For this quirk, this commit extends the time to wait up to 2 seconds.
As a result, in worst cases, below userspace functions require 2 seconds
to return.
- snd_pcm_prepare()
- snd_pcm_hw_params()
- snd_pcm_recover()
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:33 +0000 (12:49 +0900)]
ALSA: bebob: add 'version' member for BeBoB protocol version
BeBoB installed devices have BeBoB register area. This area stores
basic information about its firmware. A register has its protocol
version.
This commit adds 'version' member and store the device's protocol
version to handle v3 quirks in following commits.
Tested-by: Kim Tore Jensen <kim@incendio.no>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:32 +0000 (12:49 +0900)]
ALSA: bebob: add SYT-Match support
In previous commits, this driver can detect the source of clock as mush
as possible. SYT-Match mode is also available.
This commit purge the restriction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:31 +0000 (12:49 +0900)]
ALSA: bebob: obsolete string literal expression for clock source
The old string literals were completely replaced by new normalized
representation.
This commit obsoletes it.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:30 +0000 (12:49 +0900)]
ALSA: bebob: use normalized representation for the type of clock source
This commit changes function prototype and its processing. As a result,
function caller can execute additional processing according to detected
clock source.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:29 +0000 (12:49 +0900)]
ALSA: bebob: preparation for replacing string literals by normalized representation for model-dependent structures
Previous commit adds a enumerator as a normalized representation of
clock source, while model-dependent structures still use string literals
for this purpose.
This commit is a preparation for replacement.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:28 +0000 (12:49 +0900)]
ALSA: bebob: apply new enumerator to normalize the type of clock source
Previous commit allows this driver to detect several types of clock
source, while there's no normalized expression for it.
This commit adds a new enumerator for this purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Sakamoto [Sun, 14 Jun 2015 03:49:27 +0000 (12:49 +0900)]
ALSA: bebob: improve signal mode detection for clock source
With BeBoB version 3, current ALSA BeBoB driver detects the type of
current clock signal source wrongly. This is due to a lack of proper
implementation to parse the information.
This commit renews the parser. As a result, this driver detects
SYT-Match clock signal, thus it can start streams with two modes;
SYT-Match mode and the others. SYT-Match mode will be supported in future
commits.
There's a constrain about detected internal/external clock source.
When detecting external clock source, this driver allows userspace
applications to use current sampling rate only. This is due to consider
abour synchronization to external clock sources such as S/PDIF, ADAT or
word-clock.
According to several information from some devices, I guesss that the
internal clock of most devices synchronize to IEEE 1394 cycle start
packet. In this case, by a usual way, it's detect as 'Sync type
of output Music Sub-Unit' connected to 'Sync type of PCR output Unit
(oPCR)', and this driver judges it as internal clock. Therefore,
userspace applications is allowed to request arbitrary supported sampling
rates.
On the other hand, several devices based on BeBoB version 3 have
additional internal clock. In this case, by a usual way, it's detect as
'Sync/Additional type of External input Unit'. Unfortunately, there's no
way to distinguish this sync type from the other external clock sources
such as word-clock. In this case, this driver handles it as external and
userspace applications is forced to use current sampling rate.
I note that when the source of clock is detected as 'Isochronous stream
type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the
synchronization clock is generated according to SYT-series in received
packets. In this case, this driver generates the series by myself. I
experienced this mode often make the device silent suddenly during
playbacking. This means that the mode is easy to lost synchronization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nicholas Mc Guire [Sun, 14 Jun 2015 17:16:59 +0000 (19:16 +0200)]
ALSA: mips: let SND_SGI_O2 select SND_PCM
Fix the missing dependency on PCM stuff.
[Add the same fix for HAL2, too -- tiwai]
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 15 Jun 2015 11:15:48 +0000 (13:15 +0200)]
Merge branch 'for-linus' into for-next
Conflicts:
sound/pci/hda/patch_realtek.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mathias Krause [Sat, 13 Jun 2015 12:25:16 +0000 (14:25 +0200)]
ASoC: intel: Constify ACPI device ids
Constify the ACPI device ID array, it doesn't need to be writable at
runtime.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Mathias Krause [Sat, 13 Jun 2015 12:25:15 +0000 (14:25 +0200)]
ASoC: rt5670: Constify ACPI device ids and register defaults
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime. Also drop the unneeded RT5670_INIT_REG_LEN
define.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Mathias Krause [Sat, 13 Jun 2015 12:25:13 +0000 (14:25 +0200)]
ASoC: max98090: Constify ACPI device ids and register defaults
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Mathias Krause [Sat, 13 Jun 2015 12:25:14 +0000 (14:25 +0200)]
ASoC: rt5640: Constify ACPI device ids and register defaults
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime. Also drop the unneeded RT5640_INIT_REG_LEN
define.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Axel Lin [Sun, 14 Jun 2015 02:28:01 +0000 (10:28 +0800)]
ASoC: ml26124: Remove duplicate code
Current code has duplicate code for 16000, 32000 and 48000 sample rates.
get_srate() returns negative error code for unsupported rate, so we can
remove the duplicate code in the swith cases by calling get_srate() first.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Takashi Iwai [Mon, 15 Jun 2015 09:59:32 +0000 (11:59 +0200)]
ALSA: hda - Fix audio crackles on Dell Latitude E7x40
We still got a report that the audio crackles and noises occur with
the recent 4.1 kernels on Dell machines. These machines seem to need
similar workarounds that have been applied to the recent Dell XPS 13
models. Since the codec of these machines (Dell Latitute E7240 and
E7440) is different from XPS 13's one, we need a new fixup entry.
Also, it was confirmed that the previous workaround to disable the
widget power-save (commit [
219f47e4f964: ALSA: hda - Disable widget
power-saving for ALC292 & co]) is no longer needed after this fix.
So, this patch includes the partial revert of the commit, too.
Reported-and-tested-by: Mihai Donțu <mihai.dontu@gmail.com>
Tested-by: Jonathan McDowell <noodles@earth.li>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fang, Yang A [Wed, 3 Jun 2015 22:07:41 +0000 (15:07 -0700)]
ASoC: Intel: fixed TI button detection
In order to make TI button interrupt working max98090 codec
Need provide mic bias all the time as long as mic is present
so SHDN and micbias pin are forced on.we also need set max98090
codec bias close or lower than TI bias.We set them in bios/coreboot
kernel reads them from device property
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Rasmus Villemoes [Thu, 4 Jun 2015 08:13:53 +0000 (10:13 +0200)]
ASoC: Intel: mrfld: simplify sst_fill_widget_module_info
Some tiny improvements, cutting 180 bytes off the generated code.
- use strchr() for single-character needle
- compute index using pointer subtraction instead of two strlen()
calls
- factor out the common check for whether the initial part of
kctl->id.name (before the space) is identical to w->name.
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Hui Wang [Mon, 15 Jun 2015 09:43:39 +0000 (17:43 +0800)]
ALSA: hda - adding a DAC/pin preference map for a HP Envy TS machine
On a HP Envy TouchSmart laptop, there are 2 speakers (main speaker
and subwoofer speaker), 1 headphone and 2 DACs, without this fixup,
the headphone will be assigned to a DAC and the 2 speakers will be
assigned to another DAC, this assignment makes the surround-2.1
channels invalid.
To fix it, here using a DAC/pin preference map to bind the main
speaker to 1 DAC and the subwoofer speaker will be assigned to another
DAC.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Linus Torvalds [Mon, 15 Jun 2015 01:51:10 +0000 (15:51 -1000)]
Linux 4.1-rc8
Linus Torvalds [Mon, 15 Jun 2015 01:48:26 +0000 (15:48 -1000)]
Merge branch 'fixes' of git://git.infradead.org/users/vkoul/slave-dma
Pull dmaengine fixes from Vinod Koul:
"Here are hopefully last set of fixes for 4.1. This time we have:
- fixing pause capability reporting on both dmaengine pause & resume
support by Krzysztof
- locking fix fir at_xdmac by Ludovic
- slave configuration fix for at_xdmac by Ludovic"
* 'fixes' of git://git.infradead.org/users/vkoul/slave-dma:
dmaengine: Fix choppy sound because of unimplemented resume
dmaengine: at_xdmac: rework slave configuration part
dmaengine: at_xdmac: lock fixes
Linus Torvalds [Mon, 15 Jun 2015 01:46:43 +0000 (15:46 -1000)]
Merge tag 'ntb-4.1' of git://github.com/jonmason/ntb
Pull NTB fixes from Jon Mason:
"I apologize for the tardiness of this request. Here are a couple of
last minute NTB bug fixes for v4.1:
NTB bug fixes to address issues in unmapping the MW reg base and
vbase, and an uninitialized variable on Atom platforms"
* tag 'ntb-4.1' of git://github.com/jonmason/ntb:
ntb: initialize max_mw for Atom before using it
ntb: iounmap MW reg and vbase in error path
Linus Torvalds [Mon, 15 Jun 2015 01:38:57 +0000 (15:38 -1000)]
Merge branch 'upstream' of git://git.linux-mips.org/ralf/upstream-linus
Pull more MIPS fixes from Ralf Baechle:
"Another round of 4.1 MIPS fixes, one fix to a MIPS-specific #if
condition in lib/mpi, one fix to the MIPS GIC irqchip driver and one
SSB fix.
Details:
- fix handling of clock in chipco SSB driver.
- fix two MIPS-specific #if conditions to correctly work for GCC 5.1.
- fix damage to R6 pgtable bits done by XPA support.
- fix possible crash due to unloading modules that contain statically
defined platform devices.
- fix disabling of the MSA ASE on context switch to also work
correctly when a new thread/process has the CPU for the very first
time.
This is part of linux-next and has been beaten to death on
Imagination's test farm.
While things are not looking too grim this pull request also means the
rate of fixes for 4.1 remains nearly constant so I'd not be unhappy if
you'd delay the release"
* 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus:
MPI: MIPS: Fix compilation error with GCC 5.1
IRQCHIP: mips-gic: Don't nest calls to do_IRQ()
MIPS: MSA: bugfix - disable MSA correctly for new threads/processes.
MIPS: Loongson: Do not register 8250 platform device from module.
MIPS: Cobalt: Do not build MTD platform device registration code as module.
SSB: Fix handling of ssb_pmu_get_alp_clock()
MIPS: pgtable-bits: Fix XPA damage to R6 definitions.
Linus Torvalds [Mon, 15 Jun 2015 00:53:02 +0000 (14:53 -1000)]
Merge branch 'irq-urgent-for-linus' of git://git./linux/kernel/git/tip/tip
Pull irqchip fix from Thomas Gleixner:
"A single fix for an off by one bug in the sunxi irqchip driver"
* 'irq-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip:
irqchip: sunxi-nmi: Fix off-by-one error in irq iterator
Linus Torvalds [Mon, 15 Jun 2015 00:03:11 +0000 (14:03 -1000)]
Merge branch 'locking-urgent-for-linus' of git://git./linux/kernel/git/tip/tip
Pull lockdep fix from Ingo Molnar:
"A lockdep/modules unload race fix that can oops"
* 'locking-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip:
lockdep: Fix a race between /proc/lock_stat and module unload
Linus Torvalds [Mon, 15 Jun 2015 00:00:13 +0000 (14:00 -1000)]
Merge branch 'perf-urgent-for-linus' of git://git./linux/kernel/git/tip/tip
Pull perf fixes from Ingo Molnar:
"A regression fix for a crash, and a Intel HSW uncore PMU driver fix"
* 'perf-urgent-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip:
Revert "perf/x86/intel/uncore: Move uncore_box_init() out of driver initialization"
perf/x86/intel/uncore: Fix CBOX bit wide and UBOX reg on Haswell-EP
Linus Torvalds [Sun, 14 Jun 2015 23:55:24 +0000 (13:55 -1000)]
Merge tag 'sound-4.1-rc8' of git://git./linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Most of commits are regression fixes for HD-audio: a few corner case
fixes for regmap transition, and i915 binding issues.
In addition, a quirk for another USB-audio device supporting DSD"
* tag 'sound-4.1-rc8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Abort the probe without i915 binding for HSW/BDW
ALSA: hda - Re-add the lost fake mute support
ALSA: hda - Continue probing even if i915 binding fails
ALSA: hda - Don't actually write registers for caps overwrites
ALSA: hda - fix number of devices query on hotplug
ALSA: usb-audio: add native DSD support for JLsounds I2SoverUSB