In slave mode, the card can only detect the base frequency (32..48kHz)
on the MADI link (exception: 96k frames), so the real external sample
rate is this base frequency multiplied by 1, 2 or 4 depending on the
speed mode.
This patch enables 64..192kHz sample rates in clock slave mode, which
failed before due to an alleged sample rate mismatch between the MADI
link (e.g., 48kHz) and the application in DS/QS mode (e.g., 96kHz,
192kHz).
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When running in slave mode (no clock master), there is no way to
determine the real wirespeed on the MADI link (single/double/quad
speed). Like physical gear, simply provide the user with a tristate
switch to select the appropriate format.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert array index from the loop bound to the loop index.
A simplified version of the semantic patch that fixes this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e1,e2,ar;
@@
for(e1 = 0; e1 < e2; e1++) { <...
ar[
- e2
+ e1
]
...> }
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
sound: oss: rename local change_bits to avoid powerpc bitsops.h definition
ALSA: hda - Fix duplicated DAC assignments for Realtek
ALSA: asihpi - off by one in asihpi_hpi_ioctl()
ALSA: hda - Fix Oops with Realtek quirks with NULL adc_nids
ALSA: asihpi - bug fix pa use before init.
ALSA: hda - Add support for vref-out based mute LED control on IDT codecs
The driver only supports playback firstly.
For recording, as we have to use two saif instances to implement full
duplex (playback & recording) due to hardware limitation, we need to
figure out a good design to fit in ASoC.
Signed-off-by: Dong Aisheng <b29396@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove empty and useless g_input and s_input ioctls.
This fixes one fail of v4l2-compliance test.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Change locking to allow tea575x-radio device to be opened multiple times.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Convert radio-sf16fmr2 to use generic TEA575x implementation. Most of the
driver code goes away as SF16-FMR2 is basically just a TEA5757 tuner
connected to ISA bus.
The card can optionally be equipped with PT2254A volume control (equivalent
of TC9154AP) - the volume setting is completely reworked (with balance control
added) and tested.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Convert tea575x-tuner to use the new V4L2 control framework. Also add
ext_init() callback that can be used by a card driver for additional
initialization right before registering the video device (for SF16-FMR2).
Also embed struct video_device to struct snd_tea575x to simplify the code.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Hans Verkuil <hverkuil@xs4all.nl>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
PCM_TX/RX are the same as SNDRV_PCM_STREAM_PLAYBACK/CAPTURE. Use
them directly.
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The use of the "soc-audio" platform device is no longer en vogue,
update the code to the newer, simpler way of doing things.
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PSC audio drivers (psc-ac97/psc-i2s) register the DMA platform_device
on their own. This is frowned upon, from now on board code must
register a simple pcm dma platform device for each PSC with sound duties.
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that an ASoC variant is available, tell users that this
driver is now living on borrowed time...
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a machine driver suitable for the AC97 part on the DB1000/DB1500/DB1100
boards.
Run-tested on DB1500.
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This collides with powerpc exported functions from bitops.h. Rename the
local copy in the oss soundblaster mixer and ad1848 driver.
Signed-off-by: Andy Whitcroft <apw@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Copying hp_pins and speaker_pins from line_out_pins may confuse the
parser, and it can lead to duplicated initializations for the same pin
with a wrong DAC assignment. The problem appears in 3.0 kernel code.
Cc: <stable@kernel.org> (for 3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"adapter" is used as an array index in the adapters[] array so
the off by one would make us read past the end.
1c073b6797 "ALSA: asihpi - Remove spurious adapter index check"
reverted Dan Rosenberg's check that would have prevented the
overflow here.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Somce quirk models don't set adc_nids but let the parser filling it.
But the recent code has unnecessary NULL-checks of spec->input_mux,
and it resulted in NULL dereferences.
This patch fixes that regression.
Reported-and-tested-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes bug introduced by 1c073b67.
Also declare pa local to block in which it is used.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch also registers all necessary callbacks to support mute LED
only when such control is enabled. And it keeps codec AFG in D0 or D1
state all the time when aggressive power managemnt is enabled for vref-out
control (and mute LED) work correctly.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs-2.6:
merge fchmod() and fchmodat() guts, kill ancient broken kludge
xfs: fix misspelled S_IS...()
xfs: get rid of open-coded S_ISREG(), etc.
vfs: document locking requirements for d_move, __d_move and d_materialise_unique
omfs: fix (mode & S_IFDIR) abuse
btrfs: S_ISREG(mode) is not mode & S_IFREG...
ima: fmode_t misspelled as mode_t...
pci-label.c: size_t misspelled as mode_t
jffs2: S_ISLNK(mode & S_IFMT) is pointless
snd_msnd ->mode is fmode_t, not mode_t
v9fs_iop_get_acl: get rid of unused variable
vfs: dont chain pipe/anon/socket on superblock s_inodes list
Documentation: Exporting: update description of d_splice_alias
fs: add missing unlock in default_llseek()
* 'next/devel2' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/linux-arm-soc: (47 commits)
OMAP: Add debugfs node to show the summary of all clocks
OMAP2+: hwmod: Follow the recommended PRCM module enable sequence
OMAP2+: clock: allow per-SoC clock init code to prevent clockdomain calls from clock code
OMAP2+: clockdomain: Add per clkdm lock to prevent concurrent state programming
OMAP2+: PM: idle clkdms only if already in idle
OMAP2+: clockdomain: add clkdm_in_hwsup()
OMAP2+: clockdomain: Add 2 APIs to control clockdomain from hwmod framework
OMAP: clockdomain: Remove redundant call to pwrdm_wait_transition()
OMAP4: hwmod: Introduce the module control in hwmod control
OMAP4: cm: Add two new APIs for modulemode control
OMAP4: hwmod data: Add modulemode entry in omap_hwmod structure
OMAP4: hwmod data: Add PRM context register offset
OMAP4: prm: Remove deprecated functions
OMAP4: prm: Replace warm reset API with the offset based version
OMAP4: hwmod: Replace RSTCTRL absolute address with offset macros
OMAP: hwmod: Wait the idle status to be disabled
OMAP4: hwmod: Replace CLKCTRL absolute address with offset macros
OMAP2+: hwmod: Init clkdm field at boot time
OMAP4: hwmod data: Add clock domain attribute
OMAP4: clock data: Add missing divider selection for auxclks
...
This allows us to move duplicated code in <asm/atomic.h>
(atomic_inc_not_zero() for now) to <linux/atomic.h>
Signed-off-by: Arun Sharma <asharma@fb.com>
Reviewed-by: Eric Dumazet <eric.dumazet@gmail.com>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: David Miller <davem@davemloft.net>
Cc: Eric Dumazet <eric.dumazet@gmail.com>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This update includes the changes necessary for supporting the
CS421x family of codecs. Previously this file only supported
the CS420x family of codecs.
This file also contains init verbs to correct several issues in
the CS421x hardware.
Behavior between the CS421x and CS420x codec families is similar,
so several functions have been reused with "if" statements to
determine which codec family (CS421x or CS420x) is present.
Also, this file will be updated sometime in the near future in
order to add support for a system using CS421x that requires
mono mix on the speaker output only.
[Fix const usages and adaption for new APIs by tiwai]
Signed-off-by: Tim Howe <tim.howe@cirrus.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The procedure for codec D-state change may have exceptional cases
depending on the codec chip, such as a longer delay or suppressing D3.
This patch adds a new codec ops, set_power_state() to override the system
default function. For ease of porting, snd_hda_codec_set_power_to_all()
helper function is extracted from the default set_power_state() function.
As an example, the Conexant codec-specific delay is removed from the
default routine but moved to patch_conexant.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new ops, post_suspend(), which is called after suspend() ops is
performed. This is called only in the case of the real PM suspend, and
the codec driver can use this for further changing of D-state or
clearing the LED, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It makes little sense to enable power-saving without PM.
This removes SND_HDA_NEEDS_RESUME define so that we can use CONFIG_PM
in all places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds ASoC support for the AC97 and I2S controllers
on the old Au1000/Au1500/Au1100 chips,
AC97 Tested on a Db1500. I2S untested since none of the boards
actually have an I2S codec wired up (just test pins).
Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds checking of mute state on all outputs besides just
speakers to calculate the master mute state for mute led support.
It also renames and splits the function that does it for better code
clarity.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Codec state is not restored immediately on resume but on the first
access when power-save is enabled. That leads to an invalid mute led
state after resume until either sound is played or some control is
changed. This patch adds a possibility for a vendor specific patch to
restore codec state immediately after resume if required. And it adds
code to restore IDT codecs state immediately on resume on HP systems
with mute led support.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stream event debug can be noisy on larger audio devices so improve the
debug SNR by changing it to the verbose level.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for Dynamic PCM (AKA DSP) support.
This adds a callback function to be called at the completion of a DAPM stream
event.
This can be used by DSP components to perform calculations based on DAPM graphs
after completion of stream events.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (43 commits)
fs: Merge split strings
treewide: fix potentially dangerous trailing ';' in #defined values/expressions
uwb: Fix misspelling of neighbourhood in comment
net, netfilter: Remove redundant goto in ebt_ulog_packet
trivial: don't touch files that are removed in the staging tree
lib/vsprintf: replace link to Draft by final RFC number
doc: Kconfig: `to be' -> `be'
doc: Kconfig: Typo: square -> squared
doc: Konfig: Documentation/power/{pm => apm-acpi}.txt
drivers/net: static should be at beginning of declaration
drivers/media: static should be at beginning of declaration
drivers/i2c: static should be at beginning of declaration
XTENSA: static should be at beginning of declaration
SH: static should be at beginning of declaration
MIPS: static should be at beginning of declaration
ARM: static should be at beginning of declaration
rcu: treewide: Do not use rcu_read_lock_held when calling rcu_dereference_check
Update my e-mail address
PCIe ASPM: forcedly -> forcibly
gma500: push through device driver tree
...
Fix up trivial conflicts:
- arch/arm/mach-ep93xx/dma-m2p.c (deleted)
- drivers/gpio/gpio-ep93xx.c (renamed and context nearby)
- drivers/net/r8169.c (just context changes)
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs-2.6:
fs: take the ACL checks to common code
bury posix_acl_..._masq() variants
kill boilerplates around posix_acl_create_masq()
generic_acl: no need to clone acl just to push it to set_cached_acl()
kill boilerplate around posix_acl_chmod_masq()
reiserfs: cache negative ACLs for v1 stat format
xfs: cache negative ACLs if there is no attribute fork
9p: do no return 0 from ->check_acl without actually checking
vfs: move ACL cache lookup into generic code
CIFS: Fix oops while mounting with prefixpath
xfs: Fix wrong return value of xfs_file_aio_write
fix devtmpfs race
caam: don't pass bogus S_IFCHR to debugfs_create_...()
get rid of create_proc_entry() abuses - proc_mkdir() is there for purpose
asus-wmi: ->is_visible() can't return negative
fix jffs2 ACLs on big-endian with 16bit mode_t
9p: close ACL leaks
ocfs2_init_acl(): fix a leak
VFS : mount lock scalability for internal mounts
* 'next/fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/linux-arm-soc: (35 commits)
ARM: msm: platsmp: determine number of CPU cores at boot time
ARM: Tegra: Seaboard: Fix I2C bus numbering for ADT7461
ARM: Tegra: Trimslice: Tri-state DAP3 pinmux
ARM: orion5x: fixup 5181 MPP mask check
ARM: mxs-dma: include <linux/dmaengine.h>
ARM: i.MX53: consistently use MX53_UART_PAD_CTRL for uart txd/rxd/rts/cts
ARM: i.MX53: UARTn_CTS pin should not change RTS input select
ARM: i.MX53: UARTn_TXD pin should not change RXD input select
ARM: mx25: Fix typo on CAN1_RX pad setting
iomux-mx53: add missing 'IOMUX_CONFIG_SION' for some I2C pad definitions
ARM: NUC93X: add UL suffix to VMALLOC_END to ensure it is properly typed
ARM: LPC32XXX: add UL suffix to VMALLOC_END to ensure it is properly typed
ARM: CNS3XXX: add UL suffix to VMALLOC_END to ensure it is properly typed
ARM: i.MX53: Fix IOMUX type o's
ARM i.MX dma: Fix burstsize settings
mach-mx5: fix the I2C clock parents
ARM: mxs/tx28: according to the TX28's datasheet D4-D7 are not used for MMC0
ARM i.MX23/28: platform-mxsfb: Add missing include of linux/dma-mapping.h
ARM: mx53: Fix some interrupts marked as reserved.
MXC: iomux-v3: correct NO_PAD_CTRL definition
...
Fix up trivial conflict in arch/arm/mach-imx/mach-mx31_3ds.c
This reverts commit d7c3e9525a as it does
not currently build due to missing dependencies in the Samsung tree.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Formatting a PCM name is useful for module debug too.
Add snd_prefix when making function public.
[minor coding-style fixes by tiwai]
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (297 commits)
ALSA: asihpi - Replace with snd_ctl_boolean_mono_info()
ALSA: asihpi - HPI version 4.08
ALSA: asihpi - Add volume mute controls
ALSA: asihpi - Control name updates
ALSA: asihpi - Use size_t for sizeof result
ALSA: asihpi - Explicitly include mutex.h
ALSA: asihpi - Add new node and message defines
ALSA: asihpi - Make local function static
ALSA: asihpi - Fix minor typos and spelling
ALSA: asihpi - Remove unused structures, macros and functions
ALSA: asihpi - Remove spurious adapter index check
ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro
ALSA: asihpi - DSP code loader API now independent of OS
ALSA: asihpi - Remove controlex structs and associated special data transfer code
ALSA: asihpi - Increase request and response buffer sizes
ALSA: asihpi - Give more meaningful name to hpi request message type
ALSA: usb-audio - Add quirk for Roland / BOSS BR-800
ALSA: hda - Remove a superfluous argument of via_auto_init_output()
ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
ALSA: hda - Add documentation for codec-specific mixer controls
...
Fix a regression in the DAC filling code in patch_realtek.c. The already
filled DACs in multiout.dac_nids[] were ignored because of num_dacs=0,
thus always pointed to the first DAC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'timers-cleanup-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
mips: Fix i8253 clockevent fallout
i8253: Cleanup outb/inb magic
arm: Footbridge: Use common i8253 clockevent
mips: Use common i8253 clockevent
x86: Use common i8253 clockevent
i8253: Create common clockevent implementation
i8253: Export i8253_lock unconditionally
pcpskr: MIPS: Make config dependencies finer grained
pcspkr: Cleanup Kconfig dependencies
i8253: Move remaining content and delete asm/i8253.h
i8253: Consolidate definitions of PIT_LATCH
x86: i8253: Consolidate definitions of global_clock_event
i8253: Alpha, PowerPC: Remove unused asm/8253pit.h
alpha: i8253: Cleanup remaining users of i8253pit.h
i8253: Remove I8253_LOCK config
i8253: Make pcsp sound driver use the shared i8253_lock
i8253: Make pcspkr input driver use the shared i8253_lock
i8253: Consolidate all kernel definitions of i8253_lock
i8253: Unify all kernel declarations of i8253_lock
i8253: Create linux/i8253.h and use it in all 8253 related files
* 'spi/next' of git://git.secretlab.ca/git/linux-2.6: (34 commits)
spi/imx: add device tree probe support
spi/imx: copy gpio number passed by platform data into driver private data
spi/imx: use soc name in spi device type naming scheme
spi/imx: merge type SPI_IMX_VER_0_7 into SPI_IMX_VER_0_4
spi/imx: do not use spi_imx2_3 to name SPI_IMX_VER_2_3 function and macro
spi/imx: use mx21 to name SPI_IMX_VER_0_0 function and macro
spi/imx: do not make copy of spi_imx_devtype_data
spi/dw: Add spi number into spi irq desc
spi/tegra: Use engineering names in DT compatible property
spi/fsl_spi: fix CPM spi driver
mach-s3c2410: remove unused spi-gpio.h file
spi: remove obsolete spi-s3c24xx-gpio driver
mach-gta2: remove unused spi-gpio.h include
mach-qt2410: convert to spi_gpio
mach-jive: convert to spi_gpio
spi/pxa2xx: Remove unavailable ssp_type from documentation
spi/bfin_spi: uninline fat queue funcs
spi/bfin_spi: constify pin array
spi/bfin_spi: use structs for accessing hardware regs
spi/topcliff-pch: Support new device ML7223 IOH
...
Fix up trivial conflict in arch/arm/mach-ep93xx/Makefile
HPI Version is used to check for firmware compatibility.
This version will accept 4.08.xx released firmware,
and will also accept 4.09.xx beta firmware
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add names corresponding to new HPI node types.
Shorten some names so that constructed names don't overflow the
maximum name length.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Because mutex is used in adapter struct defined here.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Subsystem requests don't have or need a valid adapter index.
The adapter index is already checked further on, before it is used to index
the adapters array. (Reverts 4a122c10f)
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Work towards moving the function into alsa common header.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loader API has been revised so that OS specific data is kept
local to hpidspcd.c, and the public API is unchanged across OSes.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some cobranet control data would not fit in an original HPI message.
Now that HPI is able to transfer larger messages, this special handling
is no longer required.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow for up to 256 bytes of extra data on top of standard hpi
request and response sizes.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Having a 'request message' makes more sense than a 'message message'
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for Roland/BOSS BR-800 (0582:011e) to snd-usb-audio driver.
This allows playback and recording, which has been tested and found to
work. The third interface should be MIDI (MTC/SMPTE?) for DAW interface
and is set as per ME-25, but this has not been tested. SDHC card access
is already supported by usb-storage for Backup/Rhythm Editor/Wave
Convertor mode which should not conflict with this.
Signed-off-by: David G Turner <dgturner@iee.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes non-working indep-HP control on VT1708* codecs.
The problems are that via_independent_hp_put() wasn't fixed to follow
the recent change of three HP paths, and hp_indep_path didn't contain
the amp nids of mixer elements.
Together with the fixes, a few code clean-ups are done.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for ASoC Dynamic PCM (AKA DSP) support.
Provide convenience methods to retrieve the soc_card or snd_card from a
DAPM context.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I2S in Exynos4 and S5PC110(S5PV210) has a internal dma.
It can be used low power audio mode and 2nd channel transfer.
This patch can support idma.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Previously, I2S driver only can support system dma.
In this patch, i2s driver can support internal dma too.
IDMA h/w configuration is initialized on idma.c
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In davinci_vcif_trigger() function, a break() statement was missing
causing the davinci_vcif_stop() function to be called as a fallback
after calling davinci_vcif_start().
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
According to DM365 voice codec data sheet at [1], before starting
recording or playback, ADC/DAC modules should follow a reset and
enable cycle. Writing a 1 to the ADC/DAC bit in the register resets
the module and clearing the bit to 0 will enable the module. But the
driver seems to be doing the reverse of it.
[1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf
Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
This closes the small race between a status being read in response to an
interrupt and clearing the interrupt, meaning that if the status changes
between those periods we might not get a reassertion of the interrupt.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Print a hint when the user has a setup where CONFIG_REGULATOR is really
needed to make the driver work.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code for registering the internal ldo was present twice. Turn it
into a function instead. Also, inform the user if LDO is used now.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In one comment, cpu_dai was mentioned although codec_dai was used in the
code. Also, fix the name for the card dai list which has no seperation
into card_dai and codec_dai.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since quite a few drivers are not managing to flag the cache as needing
to be resynced after suspend and it's a reasonable thing to do flag the
cache as needing sync automatically when suspending.
The expectation is that systems will mainly only keep the CODEC powered
when doing audio through the CODEC so we won't actually suspend the
device anyway; drivers which want to can override this behaviour when
they resume.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
This patch changes the behavior of independent-HP enum switch. Now
instead of returning a busy error, the driver switches dynamically the
stream of the HP (and shared) DACs according to the current mode.
The logic is similar like the dual-mic ADC switch, but a bit more
complicated because of the presence of shared DAC.
Together with the change, a mutex is introduced to protect against the
possible races for the indep-HP mode setting.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the dynamic control of analog-loopback for VIA codecs.
When the loopback is enabled, the inputs from line-ins and mics are
mixed with the front DAC, and sent to the front outputs. The very same
input is routed to the headhpones and speakers in loopback mode.
However, since the loopback mix can't take other than the front DAC,
there is no longer individual volume controls for headphones and
speakers. Once when the loopback control is off, these volumes take
effect.
Since the individual volumes are more desired in general use caess, the
loopback mode is set to off as default for now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit dd203fa97b (ALSA: virtuoso: remove non-working controls on
Essence ST Deluxe) made it impossible to adjust the volume after the
driver initialized it to muted.
Ensure that those DACs that can be accessed with I2C are initialized
to the same volume that is the reset default of the DAC without I2C.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.38+ <stable@kernel.org>
The WM8994 and WM8958 series of devices have two MICBIAS supplies rather
than one, the current widget actually manages the microphone detection
control register bit (which is managed separately by the relevant API).
Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2
widgets.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Makes life a little easier if you want to add subsequences to an existing
driver as you can use -1 to put things at the start of sequences.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If DAIs are idle but their clocks are in use for some reason (eg, as
SYSCLK or for accessory detect) then set the clock dividers to the maximum
to reduce slightly the power consumption of the unclocked circuits.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Not only fixes error handling but also some uninitialized variable
warnings.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Try the completion before we start the FLL so that if an interrupt was
delayed long enough for us to miss it we don't wait for the completion
it signalled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8983 is a low power, high quality stereo CODEC
designed for portable multimedia applications. Highly flexible
analogue mixing functions enable new application features,
combining hi-fi quality audio with voice communication.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://bugs.launchpad.net/bugs/774895
The original reporter states that his volume keys do not change the
desired Master and PCM mixer elements together, so apply the hp+mute led
quirk for his PCI SSID.
Reported-by: Jeffrey Finkelstein
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the rewrite, the check of spec->need_dac_fix and the corresponding
num_dacs change was dropped from the channel-mode control.
This patch re-adds it, and also enables need_dac_fix for ALC880 as default,
as this feature was originally introduced to fix h/w bugs of this chip.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We should spot them anyway on state changes but logging them gives us
better time information about when the misconfiguration happened.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using 256fs or 512fs will result in distortion of 24-bit
audio samples. This is because the lrclk generated is not
proper. Using 384 fs generates proper output.
Signed-off-by: Giridhar Maruthy <giridhar.maruthy@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we have interrupts then wait for the FLL lock interrupt rather than
using dead reckoning when waiting for the FLL to start.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The individual devices should set the flag dcs_done_irq in the hubs
shared data structure to indicate that they will flag the interrupt
by calling wm_hubs_dcs_done().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This ensures appropriate clocking for bypass paths to speaker and
headphone and direct voice paths on affected revisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Chip documentation explicitly requires that the reset values
of reserved register bits are left untouched. It is possible
there are differences between STA326 and STA328 or future
chip revisions in these bits, and clobbering them might
cause malfunction.
Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The STA32x has a number of preset EQ settings, but also
allows full user control of the biquad filter coeffcients
(when "Automode EQ" is set to "User").
Each biquad has five signed, 24bit, fixed-point coefficients
representing the range -1...1. The five biquad coefficients
can be uploaded in one atomic operation into on-chip
coefficient RAM.
There are also a few prescale, postscale and mixing
coefficients, in the same numeric format and range
(a negative coefficient inverts phase).
These coefficients are made available as SNDRV_CTL_ELEM_TYPE_BYTES
mixer controls.
Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This commit is a fix up for commit acfa634f.
commit acfa634f7e
Author: Takashi Iwai <tiwai@suse.de>
Date: Tue Jul 12 17:27:46 2011 +0200
ALSA: hda - Add Kconfig for the default buffer size
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch gives M-Audio Fast Track Pro and M-Audio Quattro quirks and
endpoints to boot and setup those devices with special options (digital
inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are
just adapted to match the new global M-Audio parameters.
Special configurations can be then loaded through a modprobe conf file.
For example, to set the 24 bits mode on the Fast Track Pro add
/etc/modprobe.d/fast_track_pro.conf :
options snd_usb_audio vid=0x763 pid=0x2012 device_setup=0x08
Here is a list of the possibilities in this example :
http://files.parisson.com/debian/fast-track-pro.conf
Signed-off-by: Guillaume Pellerin <yomguy@parisson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a Kconfig entry to specify the default buffer size.
Distros using PulseAudio can choose a larger value here.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1718S and co have a secret connection from DAC to AA-mix, which
doesn't appear in the connection list obtained from the h/w.
Currently the driver fixes the connection index locally at init, but
now we can expose it statically via snd_hda_override_connections()
so that this conection can be checked better by the parser in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the codec proc outputs, read the raw connections instead of the
cached connection list, i.e. proc files contain only raw values.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a function to add/modify the connection-list cache entry.
It'll be useful to fix a buggy hardware result.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some machines seem to use EAPD control of the unused pin for controlling
the overall EAPD. Since the driver currently doesn't check the EAPD of
unused pins, the EAPD isn't enabled. For avoiding such a problem, turn
all extra EAPDs on as default.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For smart51 pins, we need to preserve the input pin-control bits at
auto-mute controls instead of overwriting zero or pin-out-only.
Otherwise the VREF won't be set properly when smart51 is disabled
again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When Independent-HP mode is changed for VIA, the driver needs to
re-issue the auto-mute check so that the line-out pins are set properly
without influence of HP pin state.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the line-jack is plugged/unplugged, the driver must check also
the headphone jack state in addition to the line-out jack. Currently
it checks only the line-out state and ignores the headphone.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of checking the model quirk, use a fixup table for workaround
of 44kHz-fixed PCM for Lenovo IdeaPad with ALC269.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
it shouldn't contain space letters and
special letters like parentheses.
aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's harmless but annyoing.
sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now all alc*_parse_auto_config() do almost same thing except for the
NID list to ignore and the PINs for SSID-check, we can merge all these
to a single function. A good amount of code reduction.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One more code reduction. This codec has less DACs, thus the wiring
to DAC can't be filled uniquely for all output pins, i.e. some outputs
share the same volume control.
Except for that, all seems working fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge more auto-parser code in patch_realtek.c, now for ALC861.
The topology of this codec is pretty simple, and can be parsed well
by the current starndard parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
query_amp_caps() may return non-zero if the amp cap isn't supported
by the codec. Thus one needs to check widget-caps first, then check
the corresponding amp-caps.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A regression fix from commit 21268961d3
ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs
The auto-mic wasn't detected properly when no ADC-switch is needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, to create Independent HP
control.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, there're only two DACs. So smart51
control shouldn't be created.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT2002P, VT1802 and VT1812 codecs, the original activate_output_path()
function can't initialize output and hp path correctly, since mixers connected to
output pin widgets are not considered. So modify the activate_output_path()
function to satisify this kind of codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unused variable 'dai' to eliminate below warning.
CC sound/soc/pxa/pxa2xx-pcm.o
sound/soc/pxa/pxa2xx-pcm.c: In function 'pxa2xx_soc_pcm_new':
sound/soc/pxa/pxa2xx-pcm.c:91: warning: unused variable 'dai'
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Put the all static quirk codes out of patch_realtek.c, split into the
file for each codec model. For controlling the build of quirk codes,
a new Kconfig, CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS is introduced.
By setting this off, all quirk codes won't be built, thus you can save
lots of memory.
The codes in patch_realtek.c are also shuffled and more comments are
given, but the contents aren't changed. This is just a refactoring.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes the auto-parser and the auto-mic handling codes to
allow more flexible dynamic ADC-switching with Realtek codecs.
In the new code, the following strategy is taken:
- When a cap-src can't handle all input-sources, either skip it, or
switch to the ADC-switching mode. In ADC-switching mode, like the
former dual-ADC mode for ALC275, it changes ADC on the fly according
to the current input source.
- When auto-mic is possible, always assign imux. If the mic pins are
set statically via a quirk, rebuild imux according to the pins.
In the auto-mic mode, the driver always changes the imux (although
the imux isn't exposed as a mixer element).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The legacy mode has been accidentaly removed by commit:
ASoC: twl6040: add all ABE DAIs
Add back the twl6040-hifi dai.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the PLL handling has been simplified, and
rebased on 0, there is no longer need for converting
the PLL ID.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Avoid configuring the PLL several times during audio startup.
We can configure the PLL at prepare time with parameters collected
earlier hw_param, and set_dai_sysclk calls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We can manage the sample rate constraints without the need
to maintain a variable and a pointer.
This simplifies the handling of the constraint, and makes it
more robust.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is better if the selection between the Low power,
and High performance PLL is handled within the codec
driver, not in machine driver(s) to avoid duplicated
code, and also to have consistent tracking of the selected
PLL, and the resulting differences in supported sample
rates.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the variable names to be neutral (not refering to HS).
This will ease up the introduction of PLL selection, which
going to use the same enum strings.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The twl6040_request_irq/free_irq inline functions are going
to be removed, so replace them with direct calls.
The irq number is provided by the core driver via resource.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Felipe Balbi <balbi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dmaengine expects the maxburst parameter in words, not bytes.
The imxdma driver and its users do this wrong. Fix this.
As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work
with imx-dma. This broke the driver with imx-sdma support which
correctly takes the maxburst parameter in words. This patch
puts the sdma based sound back to work.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Instead of assigning each default hda_pcm_stream pointers, do NULL-checks
and assign default values in alc_build_pcms().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The only different implmentation was alc880_auto_init_input_src(),
and now it covers this variant, and we can use the single function
for all codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now with the new code for looking for ADCs and MUXs, we can replace
the whole ADC assignment with the parsed results.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All alc*_auto_init_analog_input() calls are identical, so let's use
the same function more clearly without aliases.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Parse ADCs and cap-srcs in alc_auto_create_input_ctls() by itself
instead of passing explicitly from the caller. By this change, all
alc*_auto_create_input_ctls() can be unified to the same calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the same common code for auto-parsing the output paths and their
initializations, based on the existing ALC662 code, which is smarter
than the old ALC880/2 code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When multiple inputs are present on the mixer widget (typically a DAC
and a loopback), mute/unmute both inputs with the corresponding mixer
element.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In alc662_auto_fill_dac_nids(), the HP and speaker DACs aren't parsed
when the corresponding pins aren't fixed with single DACs.
Now check these DACs even for non-fixed pins.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the dual-adc switching mode is active in Realtek auto-parser,
we need to couple all ADCs as a single capture-volume. Currently, the
volume control changes only the first ADC, thus others may remain silent.
This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of ALC269VC codec.
Also delete the unnecessary codec_variant type enum list:
now only three variants (ALC269VA ALC269VB ALC269VC) are needed.
In addition, added some aliases:
- Add ALC269VB alias name ALC277
- Add ALC269VC alias name ALC259 ALC281X
- Add ALC269VC for Lenovo device 0x21f3 name ALC3202
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a minimal driver for the Tegra SPDIF controller.
In hardware, the SPDIF output signal is always routed to any active HDMI
display controllers, and may also be routed to external pins on Tegra
using the pinmux.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform probe to register platform kcontrols and DAPM just like
the CODEC probe().
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform driver widgets to perform any IO required for DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In preparation for Dynamic PCM (AKA DSP) support.
Allow platform drivers to register kcontrols.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Now we have supply widgets there's no need to open code the handling of
the ACTIVE bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The snd_card->driver field contains a driver name string, and in
general it shouldn't contain space or special letters. The commit
2b39535b9e changed the string copy from
card->name, but the long name string may contain such letters, thus
it may still lead to a segfault.
A temporary fix is not to copy the long name string but just keep it
empty as the earlier version did.
Reported-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for ASoC Dynamic PCM (AKA DSP) support.
Allow platform driver to perform IO. Intended for platform DAPM.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit af46800 ("ASoC: Implement mux control sharing") revealed that
"Left Line1[L | R] Mux" and "Right Line1[L | R] Mux" widgets were pointing
to the same kcontrols and codec registers and thus soc-core falsely detected
them as shared controls. This is actually wrong since there are separate
registers in hardware that configure Line1L to RADC and Line1R to LADC cross
connects so these muxes should not be shared.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Enable ramp down/up step to be configured based on
platform.
Signed-off-by: Axel Castaneda Gonzalez <x0055901@ti.com>
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Set default sysclk constraints to high performance mode.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove dependency between pll (hppll, lppll) and headset power
mode (low-power, high-performance), as headset power mode can
be used with any pll.
A new control is created to allow headset power mode configuration
from userspace. Changing headset power mode during earpiece related
usecases is not propagated down to the codec as earpiece requires
HS DAC in HP mode.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add other supported sample rates to LP and HP modes.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add all DAIs to fully support OMAP4 ABE.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Convert TWL6040 CODEC driver into a TWL6040 MFD child, it implies
that MFD-level operations like register accesses, clock setting
and power management are done through MFD APIs, not directly by
CODEC driver anymore. To avoid conflicts with the other MFD child,
vibrator registers are skipped in CODEC driver.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This delay is very conservative.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
The clock needed by the I2S driver is associated with the I2S device name
in the standard fashion. Hence, use clk_get(dev) instead of clk_get_sys(clk_name).
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The I2S controller needs a clock to respond to register writes. Without
this, register writes will at worst hang the CPU. In practice, I've only
observed writes being dropped.
Luckily, the dropped register writes historically had no effect:
TEGRA_I2S_TIMING: The value we wrote was the reset default.
TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data
when one slot was empty. The requested value was for the FIFOs to request
when four slots were empty. The DMA controller in the mainline kernel is
configured to burst a single entry at a time into the FIFO, hence there
was no issue. The only negative effect was on bus efficiency losses due
to an increased number of arbitration attempts.
However, in various non-upstream changes, the DMA controller now bursts
four entries at a time into the FIFO. If there is only space for one
entry, the data is simply dropped. In practice, this resulted in 3/4 of
samples being dropped, and playback at 4x the expected rate and pitch.
By fixing the clocking issue, this is solved.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allign the platform data names for twl4030 audio submodule:
twl4030_audio_data: for the core MFD driver
twl4030_codec_data: for ASoC codec driver
twl4030_vibra_data: for the input/ForceFeedback driver
To avoid breakage, change all depending drivers, files
to use the new types.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Rename the driver, and header file from twl4030-codec to
twl4030-audio.
To avoid breakage change depending drivers at the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
CC: Misael Lopez Cruz <misael.lopez@ti.com>
Acked-by: Samuel Ortiz <sameo@linux.intel.com>
Since we now route the front DAC via aa-mix widget, adding the aa-mix
to surrounds will result in a mix-up of both front and surround PCM
signals. For avoiding this, the aa-mix routes have to be disabled
for surround paths.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the individual DAC is available for the headphone output, the driver
should create the DAC for its volume control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, the multi-channel path should be like following:
DAC 0-->Mixer 9(index 5)-->Mixer 0(index 1)-->Front Pin;
DAC 1-->Mixer 1(index 0)-->Surround Pin;
DAC 2-->C/LFE Pin;
DAC 3-->Mixer 2(index 0)-->Side Pin;
But current code built Surround and Side path through index 1 of
Mixer 1 and 2. So Adjusting Surround and Side channel amplifier is
invalid. This patch fixes the issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For VT1718S, Mixer 9 doesn't expose the connection to DAC 0. So when
building up a 'PCM Playback' amplifier control, it will fail since
getting DAC 0 index of Mixer 9 returned -1. So I added a dac_mixer_idx
to indicated the actual index of DAC 0 to Mixer 9. Following is the
patch and next mail is another.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for ASoC dynamic PCM support (AKA ASoC DSP)
Platform will also support DAPM so separate out the probe function
to simplify the code (just like the codec probe).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unmute DAC on front speaker path when Independent HP is enabled.
When to enable Independent HP, the front speaker won't output any sound
for VT1708, VT1708B, VT1708S and VT1702.
I find the via_independent_hp_put() routine will mute DAC 0 path in Mixer 0.
For these codecs, when using Independent HP, there could have two
independent streams, one is from DAC0-->Mixer0-->Front Pin, the other is
from DAC3-->GainSW3-->Side Pin.
So I added a check for DAC-->Mixer path in activate_output_path().
If current path is DAC-->Mixer, no need to mute DAC index in Mixer.
In fact, to change connection of Headphone pin or Mux connected with HP
is enough.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the re-implementation of Independent-HP mode in the
case where the DAC is shared between HP and side-channel streams.
Now the driver tries to parse the output-path using the pre-parsed
side-channel DAC for the independent HP output, too.
When a playback PCM stream is opened with this shared mode, the
Independent-HP mixer switch can't be changed for avoiding the conflict,
thus it returns -EBUSY error.
One remaining unintuitive issue is that the DAC volume is still
controlled as "Side" volume although it's shared by both independent-HP
and side streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The char can be unsigned on some architectures. Since the code checks
the negative values, they should be declared as signed char explicitly.
sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type
sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the wrongly converted short values:
sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type
sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More for neatness than any actual performance improvement.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
It was consistently wrong for everything except WM8993 so should be no
functional change.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.comm>
The Blackfin DMA controller can report one frame beyond the end of the
buffer in the wraparound case but ALSA requires that the pointer always
be in the buffer. Do the wraparound to handle this. A similar bug is
likely to apply to the other Blackfin PCM drivers but the code is less
obvious to inspection and I don't have a user to test.
Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
I noticed that the last character of the ELD monitor name is lost,
this fixes the issue.
This fix should be confirming to the HDA spec, and works together with
the DRM part of the ELD patch.
The HDA spec does not mention that Monitor_Name_String is an '\0'
ending string, and it allows NML to be 1, which is only valid when MNL
does not count the possible ending '\0'.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move codec init verb which is only applicatable for VT1708.
I've found the root cause that jack plugged in can't be detected.
The verb in vt1708_init_verbs is used to power down jack detect circuit.
This verb is only applicable to VT1708. vt1708 didn't implement jack
detect function in hardware, so we should shut down this function to
avoid noise. But for other codecs, hardware implement jack detect
function. If sending this verb during initialization, jack detect will
be invalid. So I move this verb from via_parse_auto_config() to
patch_vt1708().
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create snd_hda_get_conn_index() helper function for obtaining the
connection index of the widget. Replaced the similar codes used in
several codec-drivers with this common helper.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to enable unsolicited response to line-out pins.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a machine driver to support the EVAL-ADAV801 and EVAL-ADAV803 boards
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch updates the email address of the sound drivers supported by me to an
email account I will use on a more regular basis in the future.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we're not using the new auto parser as a fallback yet,
add it manually as a quirk.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conexant ID 506c was found on Acer Aspire 3830TG. As users report
no playback, sending to stable should be safe.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/783582
Reported-by: andROOM
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use static inline for dummy function to fix the warnings like below
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_init’:
sound/pci/hda/patch_sigmatel.c:4387:3: warning: statement with no effect
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_resume’:
sound/pci/hda/patch_sigmatel.c:4927:3: warning: statement with no effect
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More similar fixes like previous commits: handle the exceptional case
like ALC267 where no volume amp is found in ADC widget but in the
capsrc widget instead.
Also minor checks for avoiding possible erros: no connection-select
when the pin has a single selection, and add beep verbs only when the
0x1d is used for beep.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC259 seems to provide an invalid widget capability for the input-src
selector widget. The widget shows the input-amp while it's a selector,
and this confuses the current ALC882 initialization code that is used
for ALC259, too. For fixing this, check the amp capability and handle
the connection selection individually.
Also, ALC259 has no mute bit in DAC volume, so we need to initialize
it as ZERO instead of MUTE.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 and compatible codecs have the output volume in DACs, thus we
can't use the ALC880's code as is. Fixed by checking the amp caps and
picking up the right widget for initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Look through the known NIDs that may have EAPD capabilities and turn
on/off them appropriately instead of checking the individual vendor ids.
This will also avoid the forgotten entries of newly added codec ids
in future.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the regression introduced by the commit
1f0f4b8036
ALSA: hda - Reduce static init verbs for Realtek auto-parsers
The input amps of mixer widgets should be unmuted as default (as
usually they have no assigned mixer switches).
More fixes in this commit are, however, for ALC260: ALC260 codec can
have multiple output mixers connnected to a single DAC althouh the
driver didn't pick up them properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Realtek codecs have the output pins hardwired with certain DACs.
These DACs have to be assigned at first and assign the rest for
multi-DAC pins so that all DACs can be assigned properly.
Without such an optimization, speaker outputs may be assigned to the
same DAC as the headphone or others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In alc_auto_add_multi_channel_mode(), when the primary HP workaround
is enabled, it re-initializes the DAC list but calls alc662's function
in a fixed way. This isn't pretty suitable for other codecs, of course.
Now we call it with fill_dac function pointer so that the proper
function can be called at that point.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using fixed init verbs, initialize DACs, ADCs and mixers
more dynamically for Realtek auto-parsers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add idma related register definitions to support idma.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I2S registers can be used for control idma.
Previously, register is defined in i2s.c.
For sharing the registers, It is moved to i2s-regs.h
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add WM8994 PCM machine driver to support PCM audio
on SMDKV310, SMDKC210 boards.
Playback and Capture supports 8kHz sampling rates.
and It is tested on SMDKV310, SMDKC210.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jaswinder.singh@linaro.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a new helper function snd_hda_get_conn_list().
Unlike snd_hda_get_connections(), this function doesn't copy the
connection-list but gives the raw pointer for the cached list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
each device has a unique mac address, which can be used to distinguish
multiple devices in the same machine. we therefore include the full mac
address in the device shortname and the last 6 bytes in the device id.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the sound/ directory there are two files (flagged by 'make
versioncheck'); sound/pci/asihpi/asihpi.c and
sound/soc/codecs/wm8991.c that include linux/version.h although they
don't need it. This patch removes the unneeded includes.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change alc_get_line_out_pfx() in patch_realtek.c to provide the channel
specific name and assign the index so that each caller doesn't have to
set the channel name by itself.
Also, check the multi-io case with the primary hp-out; for the multi-io
channels, assign the channel name instead of "Headphone" with indices.
This makes the mixer names more intuitive and reduces confusion.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some BIOS set up the pin config wrongly as line-out although it's
supposed to be a speaker out. In most cases, though, we can judge
the validity by checking the connection type -- when it's FIXED,
mostly it's an invalid line-out but a speaker.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Even if the machine has no line-out but only HP-out, try to detect the
multi-io. It'll allow more possibilities for 5.1 outputs on laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
SND_MXC_SOC_SSI looks to be unused, so kill it.
Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8918 is register compatible with the WM8904 with a subset of the
functionality. Add the device ID, a subsequent patch will ensure that only
the relevant functionality is exported to userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Some devices do not support manual updates of the DC servo.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
BIOS lists the internal speaker as an internal line-out. Change to
internal speaker + model=auto for better auto-mute capabilities.
BugLink: http://bugs.launchpad.net/bugs/754964
Reported-by: Marc Legris <marc.legris@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure]
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some VIA codecs like VT1702 provide the input-route only to specific
ADCs such as digital-mic inputs. These routes aren't covered by the
normal primary ADC, and for now, user had to open the capture stream
assigned to that special ADC manually for using such inputs.
This patch implements a way to switch the current ADC dynamically per
the input-source selection in such a case. When this workaround is
activated, the driver provides only one capture stream and one input-
source control but with the full possible inputs. The driver switches
the ADC to be used (or being used) according to the input-source on the
fly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The base hardware revision of the Maxim 98095 part is 0x40; the code
which outputs the revision of the hardware has been updated to
properly use uppercase alphabetic values for the revision numbers.
Also, the use of a constant for the length 'max98095_dai' has been
replaced with ARRAY_SIZE().
Signed-off-by: Taylor Hutt <thutt@chromium.org>
Acked-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When smart51 mode is enabled, auto-mute these surround outputs
as well as the primary line-out. Also this patch includes minor
clean-ups.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unify the VT1709 10ch and 6ch parsers, as well as VT1708B 8ch and 4ch
parsers. They have no difference now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codecs like VT1708 needs more complicated routing using the mixer
widget rather than the simple selector widgets.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The surround/CLFE/side DACs on VT1708B and co have no amp but the
connected selector widgets have the amp instead. Fix the parser to
check these selector widgets for the possible mixer controls as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the check of the multiple loopback-mixer, which gave sometimes
a wrong index assigned to an element even for different names, e.g.
Mic and Front Mic. Now check the label properly for avoid duplication.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input jacks assigned as the smart51 outputs must be in the same
stack, either rear, front or other. Also, prefer line-in as the surround
to mic-in.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a issue to create playback volume control if pin has amplifier capability
but not DAC.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the order of the output-path list in a way from the DAC to the
target pin. Also now the list include the target pin, too.
Together with this format change, simplify the arguments of
parse_output_path() function, and fix the initialization in
via_auto_init_output().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop "Capture" prefix from the mic-boost names.
Otherwise some control names can overflow the max name length.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create patch_ca0132.c, to add support for devices featuring the
Creative CA0132 HD-audio codec.
This driver implements :-
* 1 playback subdevice to headphone and speaker
* 2 capture subdevices:
i - Mic-in
ii- Line-in
* mixer device
Advanced DSP features are not yet included.
Developed and maintained by Creative Labs, Inc.
Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create a master volume and mute control of playback for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When switch HP independent mode, mute/unmute connctions of mixer which is
connected to headphone for VT1718S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some invalid initial verbs and correct some wrong initial verbs
for VT1718S codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER.
This keeps headphone automute and microphone input from operating
on at least one laptop from Opti Systems.
Without the override, the BIOS parser does a fine job setting the
card up and everything works.
Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com>
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"ret" is supposed to be signed here. The current code will only
return -EIO on error, instead of a more appropriate error code such
as -EAGAIN etc.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The reporter, who is running kernel 2.6.38, reports that
he needs to set model=auto for the headphone output to work
correctly.
BugLink: http://bugs.launchpad.net/bugs/761022
Cc: stable@kernel.org (v2.6.38+)
Reported-by: Jo
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the existing aa-loop list for simplifying the check for analog
low-current mode. Also fix the stream count test for playback streams.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Issue the init verbs of unsolicited events dynamically from the parsed
results for VIA codecs. Also, consolidate the unsol handlers for HP
and line-out mutes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly like the previous commit, initialize the input-paths dynamically
from the parsed results instead of the fixed array for VIA codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of fixed array for each codec type, initialize the output path
dynamically from the parsed results.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adrian Knoth