This patch fixes a few obvious bugs in DSP loader stuff:
- Fix possible memory leaks in the error path
- Avoid double-free calls in dma_reset()
- Properly set/unset WC bits for DMA buffers
- Add missing error status checks
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Base the DSP firmware transfer and communication timeouts on jiffy values.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the HD Audio Device IDs for the Intel Wellsburg PCH
Signed-off-by: James Ralston <james.d.ralston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A Haswell test machine showed that the invalid connection list, but
this time it has only a single pin on the codec, thus the former fixup
code doesn't work as it assumes the three pins blindly.
This patch splits the former fixup code to two parts:
- Enable eDP 1.2 for Haswell codec
- Fix the connection list of pins on Haswell codec;
the converter list is recorded dynamically in hdmi_add_cvt(), and
applied in hdmi_add_pin()
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Haswell machines support more than one display outputs (HDMI or DP),
but its BIOS may not enable the codec's 2nd and 3rd pin and output cvt widgets.
This patch implements a board-specific fixup for Intel Haswell Machines:
If the hidden pins are not enabled by BIOS, the driver will enable them
and call common code to update the codec tree.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A codec may allow software to hide some unused pin/cvt widgets.
Sometimes BIOS does not enable the hidden widgets properly although they are
needed for the board. Thus the driver need to enable them as a board-specific
fixup and the whole tree will change.
This patch implements a common code for rereading codec widgets. So the fixup
code can call it after enabling the hidden widgets.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.
The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.
Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Currently we set the max number of connections to be 32, but there
seems codec that gives longer connection lists like AD1988, and we see
errors in proc output and else. (Though, in the case of AD1988, it's
a list of all codecs connected to a single vendor widget, so this must
be something fishy, but it's still valid from the h/w design POV.)
This patch tries to remove this restriction. For efficiency, we still
use the fixed size array in the parser, but takes a dynamic array when
the size is reported to be greater than that.
Now the fixed array size is found only in patch_hdmi.c, but it should
be fine, as the codec itself can't support so many pins.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise capture activity on a compressed DAI would mute any playback
on the same DAI.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Fix these two compile errors on s390 which does not have HAS_IOPORT
nor GENERIC_HARDIRQS:
sound/pci/lx6464es/lx6464es.c: In function ‘snd_lx6464es_free’:
sound/pci/lx6464es/lx6464es.c:565:2: error: implicit declaration of function ‘ioport_unmap’
sound/soc/codecs/wm8903.c: In function ‘wm8903_set_pdata_irq_trigger’:
sound/soc/codecs/wm8903.c:1954:9: error: implicit declaration of function ‘irq_get_irq_data’
Signed-off-by: Heiko Carstens <heiko.carstens@de.ibm.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch completes the replacement of the existing max98090 driver,
by installing a more complete driver.
Signed-off-by: Jerry Wong <jerry.wong@maximintegrated.com>
Tested-by: Matthew Mowdy <matthew.mowdy@maximintegrated.com>
Reviewed-by: Ralph Birt <ralph.birt@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes the existing max98090 driver prior to installing a more
complete one.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/pci/hda/patch_ca0132.c: In function ‘ca0132_is_vnode_effective’:
sound/pci/hda/patch_ca0132.c:3331:15: warning: ‘nid’ may be used uninitialized in this function [-Wmaybe-uninitialized]
sound/pci/hda/patch_ca0132.c:4345:13: warning: ‘ca0132_download_dsp’ defined but not used [-Wunused-function]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loopback list is referred by the VIA codec driver no matter
whether CONFIG_PM is set or not, thus we need to enable it always.
Otherwise it gets compile errors.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The wm->regs[] array has WM8766_REG_COUNT (16) elements not
WM8766_REG_RESET (31).
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's mostly harmless to apply it for new models even if they have no
mic mute LED (just toggling an unused GPIO pin).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tegra already supports the common clock framework, but had issues:
1) The clock driver was located in arch/arm/mach-tegra/ rather than
drivers/clk/.
2) A single "Tegra clock" type was implemented, rather than separate
clock types for PLL, mux, divider, ... type in HW.
3) Clock lookups by device drivers were still driven by device name
and connection ID, rather than through device tree.
This pull request solves all three issues. This required some DT changes
to add clocks properties, and driver changes to request clocks more
"correctly". Finally, this rework allows all AUXDATA to be removed from
Tegra board files, and various duplicate clock lookup entries to be
removed from the driver.
This pull request is based on the previous pull request, with tag
tegra-for-3.9-cleanup.
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Merge tag 'tegra-for-3.9-soc-ccf' of git://git.kernel.org/pub/scm/linux/kernel/git/swarren/linux-tegra into next/soc
From Stephen Warren:
ARM: tegra: Common Clock Framework rework
Tegra already supports the common clock framework, but had issues:
1) The clock driver was located in arch/arm/mach-tegra/ rather than
drivers/clk/.
2) A single "Tegra clock" type was implemented, rather than separate
clock types for PLL, mux, divider, ... type in HW.
3) Clock lookups by device drivers were still driven by device name
and connection ID, rather than through device tree.
This pull request solves all three issues. This required some DT changes
to add clocks properties, and driver changes to request clocks more
"correctly". Finally, this rework allows all AUXDATA to be removed from
Tegra board files, and various duplicate clock lookup entries to be
removed from the driver.
This pull request is based on the previous pull request, with tag
tegra-for-3.9-cleanup.
* tag 'tegra-for-3.9-soc-ccf' of git://git.kernel.org/pub/scm/linux/kernel/git/swarren/linux-tegra: (31 commits)
clk: tegra30: remove unused TEGRA_CLK_DUPLICATE()s
clk: tegra20: remove unused TEGRA_CLK_DUPLICATE()s
ARM: tegra30: remove auxdata
ARM: tegra20: remove auxdata
ASoC: tegra: remove auxdata
staging: nvec: remove use of clk_get_sys
ARM: tegra: paz00: add clock information to DT
ARM: tegra: add clock properties to Tegra30 DT
ARM: tegra: add clock properties to Tegra20 DT
spi: tegra: do not use clock name to get clock
ARM: tegra: remove legacy clock code
ARM: tegra: migrate to new clock code
clk: tegra: add clock support for Tegra30
clk: tegra: add clock support for Tegra20
clk: tegra: add Tegra specific clocks
ARM: tegra: define Tegra30 CAR binding
ARM: tegra: define Tegra20 CAR binding
ARM: tegra: move tegra_cpu_car.h to linux/clk/tegra.h
ARM: tegra: add function to read chipid
ARM: tegra: fix compile error when disable CPU_IDLE
...
Signed-off-by: Olof Johansson <olof@lixom.net>
Conflicts:
arch/arm/mach-tegra/board-dt-tegra20.c
arch/arm/mach-tegra/board-dt-tegra30.c
arch/arm/mach-tegra/common.c
arch/arm/mach-tegra/platsmp.c
drivers/clocksource/Makefile
The ASoC compressed API did not implement the copy callback in its
compressed ops which is required for DSPs that are not memory mapped.
This patch creates a local copy of the compress ops for each runtime and
modifies them with a copy callback as appropriate.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The same constraint is going to be set in the snd_dmaengine_pcm_open()
function, so there is no need to set it here as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These days, GUIs such as Gnome sound settings want to be able to
show the correct jack status even when no streams are currently
running. I doubt this gives any measurable difference in power,
but if it does, the "Jack Detect" control can still be used to
turn polling off.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The VT1708 has no unsol event capability, and polling is set using
the "Jack Detect" alsamixer control. In order not to create
phantom Jack controls, temporary enable jackpoll during build_controls.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
From Kukjin Kim:
AS I commented, this makes <mach/*.h> local so that they could be removed.
* 'next/cleanup-header' of git://git.kernel.org/pub/scm/linux/kernel/git/kgene/linux-samsung: (26 commits)
ARM: S3C64XX: Fix missing header error with CONFIG_CPU_IDLE enabled
ARM: S3C64XX: make regs-syscon-power.h local
ARM: S3C64XX: make regs-sys.h local
ARM: S3C64XX: make regs-srom.h local
ARM: S3C64XX: make regs-modem.h local
ARM: S3C64XX: make regs-gpio-memport.h local
ARM: S3C64XX: make crag6410.h local
ARM: S3C24XX: remove dsc.c and make regs-dsc.h local
ARM: S3C24XX: remove idle.h
ARM: S3C2412: cleanup regs-s3c2412.h
ARM: S3C2416: remove regs-s3c2416-mem.h and regs-s3c2416.h
ARM: S3C24XX: make vr1000-cpld.h, vr1000-irq.h and vr1000-map.h local
ARM: S3C24XX: make otom-map.h local
ARM: S3C24XX: make osiris-cpld.h and osiris-map.h local
ARM: S3C24XX: make h1940.h and h1940-latch.h local
ARM: S3C24XX: make gta02.h local
ARM: S3C24XX: make bast-cpld.h, bast-irq.h and bast-map.h local
ARM: S3C24XX: make anubis-cpld, anubis-irq and anubis-map local
ARM: SAMSUNG: cleanup mach/gpio-fns.h gpio-track.h and gpio-nrs.h
ARM: SAMSUNG: cleanup mach/regs-audss.h file
...
Signed-off-by: Olof Johansson <olof@lixom.net>
Allow the regmap API to use asynchronous I/O where supported to minimise
the delay between transfers, reducing firmware download times.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Convert MicBias widgets to supply widget.
On tlv320aic3x, Mic bias power on/off shares the same register bits
with output mic bias voltage. So, when power on mic bias, we need
reclaim it to voltage value.
Provide a new platform data so that the micbias voltage can be sent
according to board requirement. Now since tlv320aic3x codec driver
is DT aware, update dt files and functions to handle this new
"micbias-vg" platform data.
Because of sharing of bits, when enabling the micbias, voltage also
needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD
macro to create an event to handle this.
Since micbias is converted to supply widget, updated machine drivers as
well.
This change is runtime tested on da850-evm with audio loopback
(arecord|aplay) for confirmation.
Signed-off-by: Hebbar Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
disable clock and correct sequence when unload
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
correct the sequence when unload this module
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ssc private data has been set in ssc driver, this cause the error
private data set to ssc, remove it
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When build as module, it reports following error, using this patch
fix it
sound/soc/atmel/atmel-pcm-pdc.c:387: error: redefinition of 'atmel_pcm_pdc_platform_register'
sound/soc/atmel/atmel-pcm.h:95: note: previous definition of 'atmel_pcm_pdc_platform_register' was here
sound/soc/atmel/atmel-pcm-pdc.c:393: error: redefinition of 'atmel_pcm_pdc_platform_unregister'
sound/soc/atmel/atmel-pcm.h:99: note: previous definition of 'atmel_pcm_pdc_platform_unregister' was here
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
Cc: stable@vger.kernel.org
... to be less confusing for the update path.
This new kconfig will choose CONFIG_SND_HDA_DSP_LOADER, which is
basically a device-independent feature in hda_intel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit d45e6889ee ("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>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new PCI ID 0x0a0c for Haswell ULT platform.
Signed-off-by: Wang Xingchao <xingchao.wang@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cc: 2.6.37+ <stable@vger.kernel.org>
Some compilers complain when building sound for MOP500:
sound/soc/ux500/mop500.c:27:27:
fatal error: mop500_ab8500.h: No such file or directory
That file is actually in the local directory rather than include.
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds runtime suspend to resume support for I2S.
I2S clk is disabled at suspend and enabled at resume.
Signed-off-by: R. Chandrasekar <rcsekar@samsung.com>
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit a08485d8 ("ASoC: Samsung: Do not register samsung audio dma
device as pdev") introduced compilation error.
Combination of CONFIG_CPU_S3C2440 and CONFIG_SND_S3C24XX_I2S
caused undefined symbols 's3c_i2sv2_register_dai' and 's3c2412_i2s_dai'
in sound/soc/samsung/s3c24xx-i2s.c.
This patch fixes the problem and makes S3C2440 I2S usable again. It does
not affect S3C2412 (aka I2S-v2).
Signed-off-by: Alexey Galakhov <agalakhov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC clock gate settings are continuous/gated only.
This patch decides it as bool, then, gated clock will be default.
Special thanks to Stephen
Cc: Stephen Warren <swarren@wwwdotorg.org>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Playing a mono stream on the TLV320AIC3x results in too fast playback rate.
Remove mono support so that mono streams can be played correctly on this codec.
Tested with imx-ssi (i.MX25) and TLV320AIC3104.
Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau@advansee.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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>
It looks like MODULE_SUPPORTED_DEVICES() is not implemented yet, but
still, having the entries in the list consistently separated by commas
and with balanced parenthesis won't hurt.
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Update Kconfig file to enable I2S,PCM audio for wm8994 and spdif on all
samsung platforms.
Signed-off-by: Sangsu Park <sangsu4u.park@samsung.com>
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the basic device tree based lookup.
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for device based discovery.
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Configlink clock information is added to device tree. Get the clocks
using device node. Remove AUXDATA.
Signed-off-by: Prashant Gaikwad <pgaikwad@nvidia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Migrate Tegra clock support to drivers/clk/tegra, this involves
moving:
1. definition of tegra_cpu_car_ops to clk.c
2. definition of reset functions to clk-peripheral.c
3. change parent of cpu clock.
4. Remove legacy clock initialization.
5. Initialize clocks using DT.
6. Remove all instance of mach/clk.h
Signed-off-by: Prashant Gaikwad <pgaikwad@nvidia.com>
[swarren: use to_clk_periph_gate().]
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Maximise robustness for the widest range of firmwares by ensuring the DSP
is in a consistent state when halted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes a regression of the external mic not working on
HP Probook 4520s.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
This reverts commit 25b8d31488.
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>
Commit 23caaf19b1 (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>
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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Merge tag 'asoc-3.8-rc4' of 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.
u32 rotate = (32 - word_length) / 4;
This implementation is wrong, but it works only for 16, or 32 bit audio data.
(rotation for 16 or 32 bit is same as in code I present) Mcasp rotated data in
4 bits (max value 0x7)and then masks them . That data are sended to i2s bus.
For 24 bit or 20 bit or other data formats, this code rotates data badly and
you hear somethink like noise. You need to use
u32 rotate = (word_length / 4) & 0x7;
to proper data rotation.
Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds snd_soc_of_parse_daifmt() and supports below style on DT.
[prefix]format = "i2c";
[prefix]clock-gating = "continuous";
[prefix]bitclock-inversion;
[prefix]bitclock-master;
[prefix]frame-master;
Each driver can use specific [prefix]
(ex simple-card,cpu,dai,format = xxx;)
This sample will be
SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CONT |
SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There were missing break statements so everything used
TWO_CHANNEL_SUPPORT.
Also I added a return statement to silence a GCC warning:
sound/soc/dwc/designware_i2s.c: In function ‘dw_i2s_hw_params’:
sound/soc/dwc/designware_i2s.c:236:32: warning: ‘ch_reg’ may be
used uninitialized in this function
[-Wmaybe-uninitialized]
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I2S module need to be reset after S2R. Keeping the S/W rst
control part in resume didn't help in playing audio after resume.
So this patch adds S/W RST control part in startup function which
gets triggered for every new audio stream playback.
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: R. Chandrasekar <rcsekar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]
For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros. This should have the same effect but shut up warnings like
above.
But since we had already put ifdefs, changing to inline functions
would trigger compile errors. So, such ifdefs is removed in this
patch.
In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too. For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.
Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a better power filter hook for powering down unused
widgets in the generic parser.
The feature is enabled by setting hda_gen_spec.power_down_unused
flag.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put the power state synchronization at the end of the parsing of
codec. This is necessary when the power filter is changed during the
codec probe. Since the first power-up sequence is performed without
the special filter, all widgets are supposed to be ON at this point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a hook to struct hda_codec for filtering the target power state of
each widget when powering up/down. The current hackish EAPD check is
implemented as the default hook pointer, too.
This allows codec drivers to implement own power filter. In the
upcoming changes, the generic parser will have the better power filter
based on the active paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AC_VERB_GET_POWER_STATE returns the combined bits of the actual state
and the target state. Thus, comparing the obtained value directly
with the target value can't work. The value has to be shifted and
masked properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The arguments to call is_active_nid() in activate_amp() were swapped,
and this resulted in the muted amp on some SPDIF output pins.
Also, the index to be passed to is_active_nid() must be idx_to_check.
Otherwise it checks the wrong connection in the case of implicit aamix
connection paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'tegra30_i2s_startup' and 'tegra30_i2s_shutdown' are used only in this file and
hence made static. Fixes the following sparse warnings:
sound/soc/tegra/tegra30_i2s.c:74:5: warning:
symbol 'tegra30_i2s_startup' was not declared. Should it be static?
sound/soc/tegra/tegra30_i2s.c:101:6: warning:
symbol 'tegra30_i2s_shutdown' was not declared. Should it be static?
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixes the following sparse warnings:
sound/soc/tegra/tegra30_ahub.c:583:16: warning:
Using plain integer as NULL pointer
sound/soc/tegra/tegra30_ahub.c:600:16: warning:
Using plain integer as NULL pointer
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
__devinitconst has been removed from the kernel and gives
the following build errors:
sound/soc/tegra/tegra20_ac97.c:460:58: error: Expected ; at end of declaration
sound/soc/tegra/tegra20_ac97.c:460:58: error: got __devinitconst
Cc: Lucas Stach <dev@lynxeye.de>
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This has been removed from the kernel recently and gives following build errors:
sound/soc/tegra/tegra_wm9712.c:155:58: error:
expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘__devinitconst’
sound/soc/tegra/tegra_wm9712.c:165:21: error:
‘tegra_wm9712_of_match’ undeclared here (not in a function)
Cc: Lucas Stach <dev@lynxeye.de>
Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Delayed work was scheduled but not initialised, this patch adds the
actual work and initialises it.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the pcm_mutex to serialise the compressed ops.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All Samsung SoCs has max 3 i2s controllers. So the i2s secondary fifo
interface device id was named as samsung-i2s.4. Renaming this to
"samsung-i2s-sec" to support device tree in i2s driver.
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allows ADSP control code to set the dsp clock rate to match the
sys clock rate.
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using the new chained_before flag, we can correct the headphone jack
detection capability easily over the existing ALC880 6stack model
(which disables the jack detection intentionally for compatibility
reason).
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=901846
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Sometimes we want to call a fixup after applying other existing
fixups, but currently the fixup chain mechanism allows only the call
the others after the target fixup. This patch adds a new flag,
chained_before, to struct hda_fixup, for allowing the chained call
before the current execution.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD1988 family and AD1882 codecs have another mixer widget (0x21)
between the analog-loopback mixer widget (0x20) and the actual
outputs. Due to this hole, the analog-loopbacks aren't sent properly
to the output pins.
As a band-aid fix, introduce another fields holding the aamix merge
path, and activate it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The fixup function is called multiple times before parsing the pins,
so snd_BUG_ON() hits when loaded. Move it to the proper place in the
if block.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a merge of really big changes: the generic parser is heavily
enhanced for handling all cases, based on the former Realtek codec
driver code. And all codec drivers except for a few ones (CA0132,
HDMI and modem) have been converted to use the new generic driver.
Conflicts:
sound/pci/hda/patch_realtek.c
Now all AD codecs have the proper BIOS auto-parser, and we can make
it for default, finally. (AD1988 already did it because it had the
auto-parser.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As done for patch_conexant.c, put ifdef ENABLE_AD_STATIC_QUIRKS for
preparing t odrop the static quirk codes in patch_analog.c.
The whole static quirk code can be omitted by commenting out
ENABLE_AD_STATIC_QUIRKS define now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert all uses of devm_request_and_ioremap() to the newly introduced
devm_ioremap_resource() which provides more consistent error handling.
devm_ioremap_resource() provides its own error messages so all explicit
error messages can be removed from the failure code paths.
Signed-off-by: Thierry Reding <thierry.reding@avionic-design.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@ti.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
AD codecs have strange implementations for choosing the SPDIF-output
mux source: the digital audio out widget may take the sources from
multiple connections, where 0x01 indicates it's a PCM while others
point ADCs. It's obviously invalid in the HD-audio spec POV, but it's
somehow convincing, too. And, to make things more complex, AD1988A
and AD1882 have deeper connection routes that aren't expressed
correctly.
In this patch, the SPDIF mux control is implemented in two ways:
- For easier one like AD1981, AD1983, AD1884 and AD1984, where the
SPDIF audio out widget takes just two or three sources, we can
simply implement via the normal input_mux and connection verb
calls.
- For the complex routes like AD1988A (but not AD1988B) or AD1882, we
prepare "faked" paths represented statically, and switch the paths
using these static ones, instead of parsing the routes from the
widget tree.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since both snd_hda_codec_flush_amp_cache() and
snd_hda_codec_flush_cmd_cache() are called usually at the same time,
we can simply combine them to a single function,
snd_hda_codec_flush_cache().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture volume put callback may call the node selection change,
and its actual call won't be triggered unless flushed. In general,
we always need to call both snd_hda_codec_flush_amp_cache() and
snd_hda_codec_flush_cmd_cache() at the same place...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both the HP auto-mute and the independent HP mode conflict with each
other. Make HP auto-mute disabled (only for the affected HP jack)
during the driver is in HP independent mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It'd be better to give another name to the secondary (alt) analog PCM
stream, which is dedicated for the independent HP out and extra
inputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The aamix NIDs are also missing for AD codecs. All AD codecs seem to
have a (more or less) working aamix widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
IDT codecs have analog-loopback mixer widgets, but we haven't cared
about it, so far. Let's set them. This will avoid also possible
wrong routes for the input paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch eventually fixes two issues:
- Handle the case where the primary output is a headphone and can have
independent HP mode;
so far we checked only the case where the headphone is the secondary
output.
- Fix the conflict of HP independent mode and aamix mode;
when switched to aamix mode, the DAC might be also switched to
another widget shared with other outputs. Then even if we disable
the DAC for the original output, it doesn't change -- because the
active route is from another (shared) DAC to HP pin through aamix.
So, in such a case, we have to prohibit the switch to aamix for HP
routes.
This fixes issues appearing on VT codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many codecs provide routes to multiple output pins through an aamix
widget, but most of them do it only from a single DAC. However, the
current generic parser checks only the aamix paths from the original
(directly bound) DACs through aamix NID, and miss the path:
primary DAC -> aamix -> target out pin
This patch adds a more check for the routes like the above.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a patch couldn't be resolved in try_assign_dacs() although the
target DAC is expected, we forgot to add a proper badness value but
continued.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since fill_and_eval_dacs() may be called repeatedly with different
configurations, setting pinctls at each time there isn't optimal.
We can set it better only once after deciding the output configuration
in parse_output_paths().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Print the information of outputs in a bit more details and concisely
in a single place instead of printing the path at each time when
detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Place a cap on the number of channels clocks are generated for. This is
intended for use with systems which have the WM5102 master an I2S bus with
multiple data lines.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems may wish to support switching between telephony and CD audio
clock rates but this is restricted by enforcement of constraints on the
current DAI clock. Support setting clocks to zero and don't enforce any
constraints in that case in order to facilitate this use case.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix up all callers as they were before, with make one change: an
unsigned module taints the kernel, but doesn't turn off lockdep.
Signed-off-by: Rusty Russell <rusty@rustcorp.com.au>
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>
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>
The option allows you to remove TTY and compile without errors. This
saves space on systems that won't support TTY interfaces anyway.
bloat-o-meter output is below.
The bulk of this patch consists of Kconfig changes adding "depends on
TTY" to various serial devices and similar drivers that require the TTY
layer. Ideally, these dependencies would occur on a common intermediate
symbol such as SERIO, but most drivers "select SERIO" rather than
"depends on SERIO", and "select" does not respect dependencies.
bloat-o-meter output comparing our previous minimal to new minimal by
removing TTY. The list is filtered to not show removed entries with awk
'$3 != "-"' as the list was very long.
add/remove: 0/226 grow/shrink: 2/14 up/down: 6/-35356 (-35350)
function old new delta
chr_dev_init 166 170 +4
allow_signal 80 82 +2
static.__warned 143 142 -1
disallow_signal 63 62 -1
__set_special_pids 95 94 -1
unregister_console 126 121 -5
start_kernel 546 541 -5
register_console 593 588 -5
copy_from_user 45 40 -5
sys_setsid 128 120 -8
sys_vhangup 32 19 -13
do_exit 1543 1526 -17
bitmap_zero 60 40 -20
arch_local_irq_save 137 117 -20
release_task 674 652 -22
static.spin_unlock_irqrestore 308 260 -48
Signed-off-by: Joe Millenbach <jmillenbach@gmail.com>
Reviewed-by: Jamey Sharp <jamey@minilop.net>
Reviewed-by: Josh Triplett <josh@joshtriplett.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Conexant CX20551 codec has a mixer in NID 0x19 and a few outputs have
to take the input through this widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Looking through the whole definitions, some fields have inappropriate
array sizes, especially about the capture. The array assigned to each
input (pin) should have HDA_MAX_NUM_INPUTS entries while the array
assigned to each ADC should have AUTO_CFG_MAX_INS entries.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch "ALSA: hda - fix wrong adc_idx in generic parser" fixed the
adc_idx for the capture volume and capture switch controls. But also
modified the adc_idx retrieval for the capture source controls
wrongly. As multiple capture source controls are created in a single
shot with counts > 1, the id.index doesn't contain the real value.
The real index has to be taken via snd_ctl_get_ioffidx() as in the
original code.
This patch reverts the fixes partially to recover from the
regression.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If there's one each of HDMI and SPDIF, we should not add an index
on the one that comes second.
[slight code refactoring by tiwai]
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just stumbled over this one while reading the code.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I found a codec configuration which had six inputs, so the max of
five was not appropriate.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some BIOS version of FSC Lifebook S7110 laptop seems to give a wrong
default pin config for NID 0x15, which confuses the parser. Give a
fixup to correct the value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Although I commented that boost volumes would be added only for
line-in and mic pins in the source code, the actual code excludes but
for mic-in. Fix it to accept the line-ins, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two hooks in hda_gen_spec, cap_sync_hook and capture_switch_hook, play
very similar roles. The only differences are that the former is
called more often (e.g. at init or switching capsrc) while the latter
can take an on/off argument.
As a more generic implementation, consolidate these two hooks, and
pass snd_ctl_elem_value pointer as the second argument. If the
secondary argument is non-NULL, it can take the on/off value, so the
caller handles it like the former capture_switch_hook. If it's NULL,
it's called in the init or capsrc switch case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a standard capture switch without multiple binding is used, the
call for capture_switch_hook isn't called properly. Replace the put
ops to add the hook call in that case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
In the current generic parser code, we look for the (mic) boost
controls only on input pins. But many codecs assign the boost volume
to a widget connected to each input pin instead of the input amp of
the pin itself.
In this patch, the parser tries to look through more widgets connected
to the pin and find a boost amp.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an amp in the activation path is associated with mixer controls,
activate_amp() tries to skip the initialization. It's good, but only
if the mixer really initializes both mute and volume. Otherwise,
either the mute of the volume is left uninitialized.
This patch adds this missing check and properly initialize the
partially controlled amps in an activation path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are a few places creating the labels and indices of kctls for
each input pin in the current generic parser code. This is redundant
and makes harder to maintain. Let's create the labels and indices at
once and keep them in hda_gen_spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the imux table entries can be a subset of autocfg.input table,
the indices of these aren't always same. For passing the proper index
value of autocfg.input at creating input ctl labels (via
snd_hda_autocfg_input_label()), keep the corresponding autocfg.input
idx value in the index field of each imux item, which isn't used in
the generic driver.
Also, this makes easier to check the invalid imux pin for stereo mix.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sets the ADSP1 clock rate to match the system clock
rate. To support this the codec driver provides
details of register containing the system clock
control bits.
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implement support for a new revision of the coefficeint file format for
ADSP cores.
Since coefficient file format 0 has not been widely deployed and is very
unlikely to ever be used with this driver code support for it has been
removed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Finally we reached here. All codecs driver (except for CA0132, which
has really device-specific requirements) have been converted to use
the generic parser.
This patch appears bigger than others since it also involves with the
code shuffling, but mostly the cut-off of parser codes and adapt to
the generic parser flags. Most of fixup codecs haven't been changed
but just removed a few unnecessary codes.
The only missing stuff is the SPDIF mux control. It'll be added again
later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* test/hda-gen-parser:
ALSA: hda - Improve naming rule for primary output
ALSA: hda - Add PCM capture hook to hda_gen_spec
ALSA: hda - Record all detected ADCs in hda_gen_spec
ALSA: hda - Move vmaster TLV parsing to snd_hda_gen_parse_auto_config()
ALSA: hda - Add input jack mode enum controls to generic parser
ALSA: hda - Give more comments to hda_gen_spec flags
ALSA: hda - Add suppress_auto_mute flag to hda_gen_spec
ALSA: hda - Record the current speaker / LO mute status in hda_gen_spec
ALSA: hda - Properly call automute/switch hooks at init
When the volume or mute control of the primary output is shared with
other (headphone or speaker) outputs, we shouldn't name it as a
specific output type but rather name it with the channel name or a
generic name like "PCM".
Also, this check should be performed individually for the volume and
the mute controls because some codecs may have shared volumes but
separate mute controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since the generic parser reduces the ADC list, copy the list of the
all detected ADCs and keep it.
This list can be later referred by the codec driver for finer power
controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add vmaster_tlv[] to hda_gen_spec and store the suggested TLV data
in snd_hda_gen_parse_auto_config(). This allows the codec driver to
correct the TLV data (e.g. mute capability) before actually creating
vmaster instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just like the jack mode enum ctls for output jacks, add the support
for similar enum ctls for input pins to control the bias Vref.
The new controls will be added when spec->add_in_jack_modes is set
either by the codec driver or by a hint string.
Note that ground and 100% vrefs are excluded from the list for
simplicity, currently. We may add a new flag to allow them, too.
But I guess it's easier to put a value override in the pinfix in such
a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- spec->hp_detect has to be overridden in HDA_FIXUP_ACT_PARSE, not in
PRE_PARSE.
- Remove err == 0 check but return directly -EINVAL from
stac92xx_parse_auto_config()
- Set spec->default_polarity for 92HD71bxx
- Some code shuffles
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a very simple machine driver using the Wolfson wm9712 AC97
codec.
Signed-off-by: Lucas Stach <dev@lynxeye.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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
A new flag to skip the auto-mute handling in the generic parser, just
like suppress_auto_mic flag. It has to be set before calling
snd_hda_gen_parse_auto_config().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* test/hda-gen-parser:
ALSA: hda - Make sure fill_all_dac_nids is called for digital only codecs
ALSA: hda - force different capture controls if amp caps differ
ALSA: hda - do not add non-existing Mic boost controls
ALSA: hda - initialize channel counts correctly
ALSA: hda - fix wrong adc_idx in generic parser
ALSA: hda - Check array bounds in get_input_path
ALSA: hda - Add prefer_hp_amp flag to hda_gen_spec
ALSA: hda - fix OOPS in hda_mark_cmd_cache_dirty
ALSA: hda - Check pincap while parsing the configuration
Otherwise no PCM will be built for codecs without analog I/O.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise setting the capture volume for amps will be weird and
inconsistent (it will try to set values outside the range of the
second amp based on capabilities of the first amp).
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the input node does not have any volume capable input amp,
don't add such a control.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Chris Rattray <crattray@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
__devinitconst and friends have recently been removed and must not be
used anymore.
Signed-off-by: Thierry Reding <thierry.reding@avionic-design.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Even a single DAC can output two channels, so the channel count
is twice the number of DACs.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We use knew->index for adc_idx when we create "Capture Volume" and
"Capture Switch", so use the same to retrieve adc_idx.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We only log the result and since the interrupt triggers on loss of lock
during shutdown this may lead to spurious interrupts during shutdown
delaying the process.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
sound/pci/hda/patch_ca0132.c:387:19: sparse: symbol 'ca0132_voicefx' was not declared. Should it be static?
Reported-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With HDSP_TOGGLE_SETTING in place, these functions are no longer
required. Removing them makes the code DRY and considerably shorter.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDSP_TOGGLE_SETTING and its corresponding functions allow to change
settings in the control register. Instead of using the specialised
functions, use the generic code to make the code DRY.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver contains multiple similar functions that change only a single
bit in the control register, only the bit position varies.
This patch implements a generic function to toggle a certain bit
position that will be used to replace the old code.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current iobox detection code reportedly fails for various users, so
simply do what the Win32 driver does instead.
Patch originally by Karl Grill <kgrill@chello.at> and then modified to
comply with kernel coding guidelines + current HEAD.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag to indicate whether HP amp is turned on as default for
speaker or line-outs, and enable this for ALC260 codec, as many
machines with this codec require the HP amp even for speakers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c: In function ‘ca0132_effects_set’:
sound/pci/hda/patch_ca0132.c:3391:2: warning: too many arguments for
format [-Wformat-extra-args]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Spotted by smatch,
sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: potential
null dereference 'dma_engine'. (kzalloc returns null)
sound/pci/hda/patch_ca0132.c:1950 dspxfr_image() error: we
previously assumed 'dma_engine' could be null (see line 1857)
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/hda/patch_ca0132.c:1781 dspxfr_one_seg() info: why not
propagate 'status' from dsp_dma_stop() instead of (-5)?
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent update of ca0132 driver replaced the pinctl setup to the
direct write via snd_hda_codec_write() again. This should be covered
by snd_hda_set_pin_ctl() to be safer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit c3b4eea262.
Since the recent firmware loader code supports caching at S3/S4 by
itself, we don't have to handle f/w caching in the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Handle a potential dma_engine alloc error and fix the possible use of an
uninitialized status variable in dspxfr_one_seg(). Also correct the initial
sampling rate for Mic 1.
Update the module description.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the controls used for tuning the DSP effects.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the unsolicited response handler for incoming DSP responses and
jack detection reporting, and routines for reading the incoming DSP response.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the playback PCM open callback.
PCM stream setup and cleanup functions are added for use by PCM callbacks.
Delay stream cleanup if effects are on, to allow time for any effects tail to
finish.
Add the analog capture PCM callbacks.
Change the max channels of analog playback to 6.
Add two new PCMs: AMic2 and What-U-Hear.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the kcontrols for the DSP effects, playback and recording
source selection.
ca0132_is_vnode_effective() checks whether virtual node settings have
taken effect.
The control change helpers ca0132_pe_switch_set(), ca0132_voicefx_set()
and ca0132_cvoice_switch_set() are added to toggle playback / capture
DSP effects, ca0132_voicefx_info(), _get() and _put() are added for
input path DSP effect value access. The volume helpers are updated to
volume_info(), _get() and _set() to use the virtual nodes.
The redundant headphone and speaker switches and ct_extension function
are removed.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the framework to set effect parameters: ca0132_effects_set()
and ca0132_setup_defaults() are general functions for parameter setting and
initializing to default values. dspio_set_param() and dspio_set_uint_param()
are lower-level fns to simplify setting individual DSP parameters via an
SCP buffer transfer to the firmware.
The CA0132 chip parameter init code is added in ca0132_init_params().
In chipio_[write,read]_data(), the current chip address is auto-incremented
if no error has occurred.
ca0132_select_out() selects the current output. If autodetect is enabled,
use headphones (if jack detected) or speakers (if no jack).
ca0132_select_mic() selects the current mic in. If autodetect is enabled,
use exterior mic (if jack detected) or built-in mic (if no jack).
Init digital mic and switch between dmic and amic with ca0132_init_dmic(),
ca0132_set_dmic(). amic2 is initialized in ca0132_init_analog_mic2().
Finally, add verb tables for configuring DSP firmware.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds definitions and structs used for configuring DSP effects,
virtual nodes, effect tuning controls, and mixer control helpers.
The effect structs are also initialized.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Instead of checking the codec SSID in find_mute_led_cfg() for HP Mini
110, set the proper spec->default_polairty in the fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI vendor ID check in find_mute_led_cfg() is now superfluous
because the function is called in the fixup table entries of HP
machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Finally all codecs in patch_sigmatel.c have been converted to use the
standard fixup helpers. This change also includes trivial cleanups
like the call of common setup for GPIO LED or the removal of unused
function.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This one is rather a simple conversion. The fixups for Dell machines
are implemented by fixup functions in the end.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This time, the only intrusive changes are for HP machines.
As the mute LED fixup and the bass speaker switch are required only
for HP machines, we can move these checks into the fixup entries; the
former is applied generically to all HP machines while the latter for
only certain models.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes (or rather often) BIOS sets the pin default configurations
obviously wrongly. Looking through these failures, one common pattern
is to enable some dead pins that are usually marked as speaker pins.
In such a case, we can skip them if the pins don't have the output
capability.
In this patch, add a check for the valid pin cap bit for each parsed
pin, and filter out when it's invalid.
The fix was originally suggested by Raymond Yau.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This conversion is a bit tricky. Since STAC927x may take two
different volume-knob initialization values depending on the model, a
new flag, spec->volknob_init, is introduced to indicate whether it's
the standard volume-knob initialization or not.
Also, Dell BIOS model is now directly mapped onto the fixup table
instead of parsing in the function. This resulted in a new model ref,
STAC_927X_DELL_BIOS_SPDIF, which is a chained entry.
Also, for reducing the fixups, virtual entries like
STAC_927X_DELL_DMIC and STAC_D965_VERBS are introduced.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rather straightforward conversion, except for ones for Intel Mac.
As Intel Mac have only unique codec SSIDs, we need to remap the fixup
again for the codec SSID and call the new fixup there.
Also, we can reduce model enums like STAC_MACMINI, which are model
aliases for backward compatibility, since they can be pointed directly
via hda_model_fixup table.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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>
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>
Mark Brown