Commit Graph

10512 Commits

Author SHA1 Message Date
Linus Torvalds d3ec4844d4 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
* '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)
2011-07-25 13:56:39 -07:00
Linus Torvalds 0003230e82 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs-2.6
* '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
2011-07-25 12:53:15 -07:00
Linus Torvalds dd58ecba48 Merge branch 'next/fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/linux-arm-soc
* '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
2011-07-25 11:53:35 -07:00
Mark Brown a0c27ab242 ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver"
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>
2011-07-24 22:47:39 +01:00
Al Viro e55d92b92d get rid of create_proc_entry() abuses - proc_mkdir() is there for purpose
Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
2011-07-24 10:12:33 -04:00
Vitaliy Kulikov 0c27c18052 ALSA: hda - Add support of the 4 internal speakers on certain HP laptops
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-24 13:36:24 +02:00
Eliot Blennerhassett acb03d440b ALSA: Make snd_pcm_debug_name usable outside pcm_lib
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>
2011-07-24 13:34:32 +02:00
Linus Torvalds e498037105 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* '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
  ...
2011-07-23 10:59:37 -07:00
Takashi Iwai 8f398ae72f ALSA: hda - Fix DAC filling for multi-connection pins in Realtek parser
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>
2011-07-23 18:57:11 +02:00
Linus Torvalds a99a7d1436 Merge branch 'timers-cleanup-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip
* '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
2011-07-22 16:51:56 -07:00
Linus Torvalds 7235dd74a4 Merge branch 'spi/next' of git://git.secretlab.ca/git/linux-2.6
* '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
2011-07-22 14:52:44 -07:00
Takashi Iwai 76531d4166 Merge branch 'topic/hda' into for-linus 2011-07-22 08:43:27 +02:00
Takashi Iwai 7d339ae997 Merge branch 'topic/misc' into for-linus 2011-07-22 08:43:24 +02:00
Takashi Iwai 13b137ef03 Merge branch 'topic/asoc' into for-linus 2011-07-22 08:43:19 +02:00
Takashi Iwai 000477a0fe ALSA: asihpi - Replace with snd_ctl_boolean_mono_info()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:57:44 +02:00
Eliot Blennerhassett 509a714744 ALSA: asihpi - HPI version 4.08
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>
2011-07-22 07:55:02 +02:00
Eliot Blennerhassett fe0aa88eec ALSA: asihpi - Add volume mute controls
Mute functionality was recently added to the DSP firmware

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:54:20 +02:00
Eliot Blennerhassett c830613574 ALSA: asihpi - Control name updates
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>
2011-07-22 07:53:45 +02:00
Eliot Blennerhassett 3d0591eee4 ALSA: asihpi - Use size_t for sizeof result
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:53:07 +02:00
Eliot Blennerhassett 5ddc5bef5c ALSA: asihpi - Explicitly include mutex.h
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>
2011-07-22 07:52:31 +02:00
Eliot Blennerhassett b7f12482ca ALSA: asihpi - Add new node and message defines
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:52:15 +02:00
Eliot Blennerhassett 33162d2dfa ALSA: asihpi - Make local function static
Fixes a sparse warning.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:52:02 +02:00
Eliot Blennerhassett 938c565a82 ALSA: asihpi - Fix minor typos and spelling
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:51:41 +02:00
Eliot Blennerhassett 4bf8cff05a ALSA: asihpi - Remove unused structures, macros and functions
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-22 07:50:57 +02:00
Eliot Blennerhassett 1c073b6797 ALSA: asihpi - Remove spurious adapter index check
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>
2011-07-22 07:50:44 +02:00
Eliot Blennerhassett 0a17e99307 ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro
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>
2011-07-22 07:50:03 +02:00
Eliot Blennerhassett 95a4c6e785 ALSA: asihpi - DSP code loader API now independent of OS
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>
2011-07-22 07:49:23 +02:00
Eliot Blennerhassett 58fbf77ff5 ALSA: asihpi - Remove controlex structs and associated special data transfer code
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>
2011-07-22 07:46:14 +02:00
Eliot Blennerhassett c6c2c9aba1 ALSA: asihpi - Increase request and response buffer sizes
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>
2011-07-22 07:45:26 +02:00
Eliot Blennerhassett 82b5774fe0 ALSA: asihpi - Give more meaningful name to hpi request message type
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>
2011-07-22 07:45:06 +02:00
David G Turner 6a6d822e12 ALSA: usb-audio - Add quirk for Roland / BOSS BR-800
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>
2011-07-21 19:00:57 +02:00
Takashi Iwai a353fbb179 ALSA: hda - Remove a superfluous argument of via_auto_init_output()
"force" argument is always true, so let's strip it off.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21 14:24:25 +02:00
Takashi Iwai 020066d1ec ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
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>
2011-07-21 13:55:10 +02:00
Liam Girdwood 4805608ac1 ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
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>
2011-07-20 21:15:51 +01:00
Sangbeom Kim d7c3e9525a ASoC: SAMSUNG: Add I2S0 internal dma driver
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>
2011-07-20 20:58:12 +01:00
Sangbeom Kim 61100f405d ASoC: SAMSUNG: Modify I2S driver to support idma
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>
2011-07-20 20:58:07 +01:00
Rajashekhara, Sudhakar 82d1d52103 ASoC: davinci: add missing break statement
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
2011-07-20 20:51:35 +01:00
Rajashekhara, Sudhakar 3012f43eaf ASoC: davinci: fix codec start and stop functions
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
2011-07-20 20:51:23 +01:00
Mark Brown 3198b9eb51 ASoC: Acknowledge WM8962 interrupts before acting on them
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>
2011-07-20 20:50:08 +01:00
Wolfram Sang 09bddc8eb2 ASoC: sgtl5000: guide user when regulator support is needed
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>
2011-07-19 16:19:30 +01:00
Wolfram Sang e94a4062c8 ASoC: sgtl5000: refactor registering internal ldo
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>
2011-07-19 16:19:22 +01:00
Wolfram Sang 1c8371d61e ASoC: core: make comments fit the code
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>
2011-07-19 16:19:15 +01:00
Mark Brown 7be4ba24a3 ASoC: Mark cache as dirty when suspending
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
2011-07-19 16:16:00 +01:00
Linus Torvalds 524196d2ad Merge branch 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ASoC: Correct WM8994 MICBIAS supply widget hookup
  ASoC: Fix shift in WM8958 accessory detection default implementation
  ASoC: sh: fsi-hdmi: fixup snd_soc_card name
  ASoC: sh: fsi-da7210: fixup snd_soc_card name
  ASoC: sh: fsi-ak4642: fixup snd_soc_card name
2011-07-18 09:05:59 -07:00
Takashi Iwai 3b607e3d3a ALSA: hda - Switch HP DAC dynamically with indep-HP switch for VIA
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>
2011-07-18 16:54:40 +02:00
Takashi Iwai 3214b9665c ALSA: hda - Implement dynamic loopback control for VIA codecs
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>
2011-07-18 16:47:33 +02:00
Clemens Ladisch c81c6b356b ALSA: virtuoso: fix silent analog output on Xonar Essence ST Deluxe
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>
2011-07-18 09:39:50 +02:00
Arnd Bergmann bc574e190d Merge branches 'omap/prcm' and 'omap/mfd' of git+ssh://master.kernel.org/pub/scm/linux/kernel/git/arm/linux-arm-soc into next/devel-2 2011-07-17 21:48:22 +02:00
Mark Brown 4400855986 Merge branch 'for-3.0' into for-3.1 2011-07-17 18:25:58 +09:00
Mark Brown b793eb60a0 ASoC: Correct WM8994 MICBIAS supply widget hookup
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
2011-07-17 18:24:29 +09:00
Mark Brown b0b3e6f861 ASoC: Don't use -1 to boostrap subseq so it can be used by drivers
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>
2011-07-17 18:23:31 +09:00
Mark Brown 7d02173cd1 ASoC: Reduce power consumption for idle DAIs in WM8994
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>
2011-07-17 17:51:10 +09:00
Mark Brown ca1004bab9 ASoC: Report an error for unknown adav80x formats
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>
2011-07-17 13:15:20 +09:00
Mark Brown f0f5039c3d ASoC: Handle failed WM8994 FLL lock waits
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>
2011-07-17 13:15:00 +09:00
Arnd Bergmann 6277839602 Merge branch 'fixes-for-arnd' of git://git.pengutronix.de/git/imx/linux-2.6 into imx/fixes 2011-07-15 21:56:37 +02:00
Mark Brown 1479c3fb5f ASoC: Handle spurious wm_hubs DC servo done interrupts
Don't assume the first fire indicates that we're done.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-15 23:48:05 +09:00
Dimitris Papastamos 6b3860b0a2 ASoC: WM8983: Initial driver
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>
2011-07-15 23:39:53 +09:00
Mark Brown 47d90a03eb Merge branch 'for-3.0' into for-3.1 2011-07-15 22:43:07 +09:00
Mark Brown b35e160a11 ASoC: Fix shift in WM8958 accessory detection default implementation
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-15 22:41:38 +09:00
Daniel T Chen f21169aa87 ALSA: intel8x0: Apply headphones+mute LED quirk for Dell Inspiron 9300
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>
2011-07-15 07:54:26 +02:00
Takashi Iwai 00ef9610ac ALSA: hda - Fix krealloc() replacement in hda_codec.c
It was obviously wrong, grr....

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-14 15:58:02 +02:00
Takashi Iwai 7b1655f5f2 ALSA: hda - Re-add need_dac_fix check for multi-io jacks of Realtek codecs
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>
2011-07-14 15:33:59 +02:00
Axel Lin 58499906c8 ASoC: wm8900: fix a memory leak if wm8900_set_fll fails
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 20:04:26 +09:00
Mark Brown 3b1af3f8c8 ASoC: Log WM8994 FIFO errors from the interrupt
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>
2011-07-14 17:17:23 +09:00
Giridhar Maruthy b3d7615f2a ASoC: SAMSUNG: 24-bit audio playback on Exynos4210
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>
2011-07-14 08:25:42 +09:00
Mark Brown f05bdb8bb6 ASoC: Don't warn on low WM8994/58 AIFnCLKs
We can have valid but very low clocks in accessory detection modes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:30 +09:00
Mark Brown c7ebf932e5 ASoC: Use WM8994 FLL lock interrupt
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>
2011-07-14 00:38:22 +09:00
Mark Brown b30ead5f39 ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:14 +09:00
Mark Brown d96ca3cd0b ASoC: Implement DC servo completion IRQ handling for wm_hubs devices
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>
2011-07-14 00:38:04 +09:00
Mark Brown b70a51bab9 ASoC: Use late enable handling for direct voice, speaker and headphone
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>
2011-07-14 00:37:52 +09:00
Johannes Stezenbach 889ebae537 ASoC: STA32x: Preserve reserved register bits
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>
2011-07-14 00:24:32 +09:00
Johannes Stezenbach 7968843915 ASoC: STA32x: Add mixer controls for biquad coefficients
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>
2011-07-14 00:24:31 +09:00
Paul Menzel cf01b73e26 ALSA: hda - fix up typos in Kconfig help for default buffer size introduced in acfa634f
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>
2011-07-12 20:17:23 +02:00
Guillaume Pellerin 0f5733b0c8 ALSA: usb-audio - Add quirks for M-Audio Fast Track Pro and Quattro
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>
2011-07-12 18:15:45 +02:00
Takashi Iwai acfa634f7e ALSA: hda - Add Kconfig for the default buffer size
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>
2011-07-12 17:31:46 +02:00
Takashi Iwai 3101ba035c ALSA: Use krealloc() in possible places
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 08:05:16 +02:00
Takashi Iwai 30b4503378 ALSA: hda - Expose secret DAC-AA connection of some VIA codecs
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>
2011-07-12 07:45:02 +02:00
Takashi Iwai 9e7717c9eb ALSA: hda - Always read raw connections for proc output
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>
2011-07-12 07:45:01 +02:00
Takashi Iwai b2f934a0df ALSA: hda - Add snd_hda_override_conn_list() helper function
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>
2011-07-12 07:44:46 +02:00
Takashi Iwai 19110595c8 ALSA: hda - Turn on extra EAPDs on Conexant codecs
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>
2011-07-11 14:46:44 +02:00
Jiri Kosina b7e9c223be Merge branch 'master' into for-next
Sync with Linus' tree to be able to apply pending patches that
are based on newer code already present upstream.
2011-07-11 14:15:55 +02:00
Takashi Iwai 9499473463 ALSA: hda - Preserve input pin-ctl bits in HP-automute for VIA codec
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>
2011-07-11 11:36:44 +02:00
Takashi Iwai 6e969d9155 ALSA: hda - Set line-out pin-ctls properly when indep-HP mode changes
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>
2011-07-11 11:28:13 +02:00
Takashi Iwai 21ce0b6527 ALSA: hda - Via Fix speaker-mute checks in VIA driver
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>
2011-07-11 10:33:47 +02:00
Mark Brown 5b7396709e ASoC: Conditionalize the enable of WM8994 ADC TDM mode
Future devices will not benefit from this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-07-09 23:16:48 +09:00
Mark Brown 3db1bbfd4a Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-3.1 2011-07-09 23:16:12 +09:00
Takashi Iwai 017f2a104c ALSA: hda - Implement 44kHz workaround for IdeadPad as fixup
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>
2011-07-09 14:42:25 +02:00
Mark Brown 3f9c42ed6b Merge branch 'for-3.0' into for-3.1 2011-07-09 19:06:33 +09:00
Kuninori Morimoto 2c7beb9285 ASoC: sh: fsi-hdmi: fixup snd_soc_card name
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>
2011-07-09 19:06:16 +09:00
Kuninori Morimoto f15c941331 ASoC: sh: fsi-da7210: fixup snd_soc_card name
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>
2011-07-09 19:06:05 +09:00
Kuninori Morimoto 505b04e0f8 ASoC: sh: fsi-ak4642: fixup snd_soc_card name
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>
2011-07-09 19:05:55 +09:00
Takashi Iwai e8fd86efaa Merge branch 'fix/asoc' into for-linus 2011-07-09 11:56:43 +02:00
Takashi Iwai abaead6ac5 ALSA: hda - Fix a copmile warning
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>
2011-07-09 11:55:28 +02:00
Takashi Iwai e320bc42be Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2011-07-09 11:43:04 +02:00
Mark Brown 71ae391d45 Merge branch 'for-3.0' into for-3.1 2011-07-09 18:20:36 +09:00
Takashi Iwai 18361bbe31 Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc 2011-07-09 09:44:09 +02:00
Takashi Iwai 3e6179b844 ALSA: hda - Merge alc*_parse_auto_config() functions in patch_realtek.c
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>
2011-07-08 16:55:13 +02:00
Takashi Iwai 8452a982fb ALSA: hda - Merge ALC260 auto-parser code
Finally the last one.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 16:19:48 +02:00
Takashi Iwai 4c11398edc ALSA: hda - Merge ALC269 parser code
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>
2011-07-08 16:12:05 +02:00
Takashi Iwai be9bc37bcc ALSA: hda - Merge ALC268/269 auto-parser codes
Now coming to ALC268/269 parser codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 16:01:47 +02:00
Takashi Iwai 72dcd8e76b ALSA: hda - Merge ALC861 auto-parser code
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>
2011-07-08 15:16:55 +02:00
Takashi Iwai 44c0240052 ALSA: hda - Fix amp-cap checks in patch_realtek.c
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>
2011-07-08 15:14:19 +02:00
Takashi Iwai a1f649d547 ALSA: hda - Merge ALC861-VD auto-parse to the standard parser
The existing standard auto-parser can work well with this codec, too.
Let's merge.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 14:39:03 +02:00
Takashi Iwai 268ff6fbe7 ALSA: hda - Fix auto-mic detection in Realtek codec-parser
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>
2011-07-08 14:37:35 +02:00
Lydia Wang 28dc10a5f1 ALSA: hda - Fix output-path of VT1812 codec
For VT1812, add dac_mixer_idx for initialization.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 12:37:19 +02:00
Takashi Iwai 21d45d2ba9 ALSA: hda - Fix Oops in smart51 parsing in VIA codec
Typical off-by-one thinko.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 11:35:11 +02:00
Takashi Iwai e477062958 ALSA: hda - Provide the standard auto_init for Realtek codecs
Remove redundant definitions.  Ideally, all init functions should be
identical in future.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 11:12:09 +02:00
Takashi Iwai afcd551508 ALSA: hda - Merge ALC680 auto-parser to the standard parser
Improved the standard Realtek auto-parser to support the codec topology
like ALC680.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 11:07:59 +02:00
Takashi Iwai e59ea3ed9f ALSA: hda - Add a fix-up for HP RP5800
The BIOS provides bogus pin configs, and also invalid SSID.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 10:20:18 +02:00
Takashi Iwai 08ef79490d ALSA: pcmcia - Use pcmcia_request_irq()
The drivers don't require the exclusive irqs.  Let's fix the deprecated
warnings.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 10:11:35 +02:00
Pavel Roskin 81b85b6bd9 ALSA: usb-audio: replace "void *" with more specific pointers
Signed-off-by: Pavel Roskin <proski@gnu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 10:10:25 +02:00
Lydia Wang a2a870c827 ALSA: hda - Fix Independent-HP detection on VT2002P/1802/1812 codecs
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>
2011-07-08 08:20:05 +02:00
Lydia Wang 5c9a5615de ALSA: hda - Fix DAC checks for VT2002P/1802/1812 codecs
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>
2011-07-08 08:19:28 +02:00
Lydia Wang d69607b3c3 ALSA: hda - Fix VIA output-path init for VT2002P/1802/1812
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>
2011-07-08 08:18:29 +02:00
Mark Brown b5d5f59be2 Merge branch 'for-3.0' into for-3.1 2011-07-07 09:54:19 -07:00
Axel Lin e12c28a98f ASoC: pxa2xx-pcm: remove unused variable 'dai'
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>
2011-07-07 09:54:09 -07:00
Kuninori Morimoto bd7fdbcaa2 ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2
mask didn't cover update-data

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-07 09:46:06 -07:00
Takashi Iwai 1d045db96a ALSA: hda - Split quirk codes from patch_realtek.c
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>
2011-07-07 18:27:29 +02:00
Takashi Iwai 0e4a73ae58 ALSA: hda - Use common paser for digital I/O for ALC260
Avoid open-codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 18:03:12 +02:00
Takashi Iwai 21268961d3 ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs
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>
2011-07-07 18:02:43 +02:00
Takashi Iwai a926757f04 ALSA: hda - Fix warning with ALC882 digital-out detection
The digital out pin on ALC882 may have multiple connections.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 16:08:13 +02:00
Peter Ujfalusi 21385eeb02 ASoC: twl6040: Add back support for legacy mode
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>
2011-07-07 14:23:47 +03:00
Peter Ujfalusi ff593ca1a4 ASoC: twl6040: No need to convert the PLL ID
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>
2011-07-07 14:23:46 +03:00
Peter Ujfalusi 753621c215 ASoC: twl6040: Configure PLL only once
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>
2011-07-07 14:23:45 +03:00
Peter Ujfalusi f53c346c08 ASoC: twl6040: Simplify sample rate constraint handling
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>
2011-07-07 14:23:44 +03:00
Peter Ujfalusi af958c72af ASoC: twl6040: Move PLL selection to codec driver
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>
2011-07-07 14:23:44 +03:00
Peter Ujfalusi 7cca606794 ASoC: twl6040: Use neutral name for power mode text/enum
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>
2011-07-07 14:23:43 +03:00
Peter Ujfalusi 2a433b9daf ASoC: twl6040: Do not use wrapper for irq request
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>
2011-07-07 14:23:22 +03:00
Sascha Hauer 6584cb8825 ARM i.MX dma: Fix burstsize settings
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>
2011-07-07 09:55:50 +02:00
Takashi Iwai c2d986b0d2 ALSA: hda - Clean-up PCM assignments in patch_realtek.c
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>
2011-07-07 09:35:17 +02:00
Takashi Iwai f970de2555 ALSA: hda - Unify alc*_auto_init_input_src() in patch_realtek.c
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>
2011-07-07 09:35:14 +02:00
Takashi Iwai d6cc9fabd5 ALSA: hda - Parse ADCs and CAPSRCs dynamically for Realtek auto-parser
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>
2011-07-07 09:34:46 +02:00
Takashi Iwai 0a7f532090 ALSA: hda - Unify alc_auto_init_analog_input() calls
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>
2011-07-07 09:31:25 +02:00
Takashi Iwai b78217096b ALSA: hda - Parse ADCs in alc_auto_create_input_ctls()
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>
2011-07-07 09:31:21 +02:00
Takashi Iwai 343a04be37 ALSA: hda - Code consolidation for ALC88x and ALC662 auto-parsers
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>
2011-07-07 09:31:18 +02:00
Takashi Iwai 97aaab7b49 ALSA: hda - Create bind-mutes appropriately for ALC662 auto-parser
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>
2011-07-07 09:31:15 +02:00
Takashi Iwai cd51155676 ALSA: hda - Initialize DACs in ALC662 auto-parser mode
The initialization of DACs was missing in ALC662 parser code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:31:13 +02:00
Takashi Iwai bb8bf4d40c ALSA: hda - Parse HP and speaker DACs even for multi connections for ALC662
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>
2011-07-07 09:31:09 +02:00
Takashi Iwai 8e89995c58 Merge branch 'fix/hda' into topic/hda 2011-07-07 09:28:47 +02:00
Takashi Iwai 9c7a083d94 ALSA: hda - Change all ADCs for dual-adc switching mode for Realtek
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>
2011-07-07 09:25:54 +02:00
Kailang Yang b68785714b ALSA: hda - Add Realtek ALC269VC codec support
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>
2011-07-06 09:53:28 +02:00
Stephen Warren 774fec338b ASoC: Tegra: Implement SPDIF CPU DAI
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>
2011-07-05 12:20:56 -07:00
Liam Girdwood a82ce2ae0d ASoC: core - Add platform IO tracing
Trace platform IO just like CODEC IO.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-05 11:08:10 -07:00
Liam Girdwood cb2cf612fb ASoC: core - Add convenience register for platform kcontrol and DAPM
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>
2011-07-05 11:07:41 -07:00
Liam Girdwood b795064137 ASoC: core - Add platform widget IO
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>
2011-07-05 11:07:39 -07:00
Liam Girdwood a491a5c84f ASoC: core - Add API call to register platform kcontrols.
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>
2011-07-05 11:07:34 -07:00
Mark Brown 8a27bd9a33 ASoC: Manage WM8731 ACTIVE bit as a supply widget
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>
2011-07-05 11:07:33 -07:00
Mark Brown 4c7c5374ce ASoC: Manage WM8731 ACTIVE bit as a supply widget
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>
2011-07-05 11:00:21 -07:00
Takashi Iwai 873bd4cb4f ASoC: Don't set invalid name string to snd_card->driver field
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>
2011-07-05 14:39:27 +02:00
Takashi Iwai f187700c2d Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2011-07-05 08:20:19 +02:00
Takashi Iwai 8d9afa08fe Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc 2011-07-05 08:20:00 +02:00
Takashi Iwai 56aa533910 Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2011-07-05 07:33:23 +02:00
Takashi Iwai 63bc975016 Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc 2011-07-05 07:33:06 +02:00
Liam Girdwood f1442bc1e9 ASoC: core - Add platform read and write.
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>
2011-07-04 12:41:07 -07:00
Jarkko Nikula 404b566569 ASoC: tlv320aic3x: Add correct hw registers to Line1 cross connect muxes
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>
2011-07-04 19:54:38 +01:00
Axel Castaneda Gonzalez 1fbe99529d ASoC: twl6040: Configure ramp step based on platform
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>
2011-07-04 19:36:29 +03:00
Liam Girdwood f7026c9996 ASoC: twl6040: set default constraints.
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>
2011-07-04 19:36:28 +03:00
Misael Lopez Cruz 6bba63b68d ASoC: twl6040: Remove pll and headset mode dependency
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>
2011-07-04 19:36:27 +03:00
Liam Girdwood e17e4ab801 ASoC: twl6040: Support other sample rates in constraints.
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>
2011-07-04 19:36:26 +03:00
Liam Girdwood 6510bdc3f4 ASoC: twl6040: add all ABE DAIs
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>
2011-07-04 19:36:25 +03:00
Misael Lopez Cruz fb34d3d505 ASoC: twl6040: Convert into TWL6040 MFD child
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>
2011-07-04 19:35:06 +03:00
Mark Brown 469bb638dc Merge branch 'for-3.0' into for-3.1 2011-07-04 08:54:40 -07:00
Mark Brown 8e9ddf811b ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting
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
2011-07-04 08:51:44 -07:00
Stephen Warren b5f9cfed12 ASoC: Tegra: I2S: s/clk_get_sys/clk_get/
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>
2011-07-04 08:49:24 -07:00
Stephen Warren 713d136978 ASoC: Tegra: I2S: Ensure clock is enabled when writing regs
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>
2011-07-04 08:49:05 -07:00
Peter Ujfalusi 4ae6df5e10 MFD: twl4030-audio: Rename platform data
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>
2011-07-04 18:44:02 +03:00
Peter Ujfalusi 57fe7251f5 MFD: twl4030-codec -> twl4030-audio: Rename the driver
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>
2011-07-04 18:43:56 +03:00
Takashi Iwai bac4b92cf7 ALSA: hda - Don't add aa-mix for VIA surrounds
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>
2011-07-04 17:37:57 +02:00
Takashi Iwai 18bd2c44b9 ALSA: hda - Create HP-vol control properly for VIA codecs
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>
2011-07-04 15:55:44 +02:00
Takashi Iwai de6c74f3e3 ALSA: hda - Define some constants in patch_via.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-04 14:53:30 +02:00
Lydia Wang b89596a160 ALSA: hda - Fix invalid multi-channel amplifiers for VT1718S
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>
2011-07-04 14:53:25 +02:00
Lydia Wang c4394f5b80 ALSA: hda - Fix issue that front can't output sound for VT1718S
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>
2011-07-04 14:33:23 +02:00
Liam Girdwood 956245e9cd ASoC: core - Make platform probe more like codec probe.
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>
2011-07-02 11:50:16 -07:00
Lydia Wang e5e1468140 ALSA: hda - Fix the silent front with independent-HP for VIA codecs
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>
2011-07-01 08:33:06 +02:00
Mark Brown 67d0c479d9 ASoC: Improve error reporting in Speyside WM8962 driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-30 13:17:49 -07:00
Takashi Iwai 350434ee53 ALSA: hda - Fix missing initialization in alc662 auto-parser
A missing initialization resulted in wrong DAC assignments in
ALC662 (and other) auto-parsers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-30 21:29:12 +02:00
Takashi Iwai 2525050518 ALSA: hda - Re-implementation of VIA Independent-HP sharing with side stream
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>
2011-06-30 17:24:47 +02:00
Takashi Iwai 286bed0f0c ALSA: hdspm - Fix compile warnings with PPC
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>
2011-06-30 12:45:36 +02:00
Takashi Iwai 71276410e1 ALSA: cs5535 - Fix invalid big-endian conversions
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>
2011-06-30 12:35:45 +02:00
Mark Brown 57cc2432e1 Merge branch 'for-3.0' into for-3.1 2011-06-29 09:49:04 -07:00
Mark Brown 4e8e78e37c ASoC: Change WM9081 speaker output enable to _OUT_DRV
More for neatness than any actual performance improvement.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-29 09:48:55 -07:00
Mark Brown d5b040c92d ASoC: Correct left/right swap in wm_hubs DC offset correction
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>
2011-06-29 09:48:36 -07:00
Mark Brown e999dc5040 ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values
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
2011-06-29 09:47:53 -07:00
Wu Fengguang f5b2d0ef63 ALSA: HDMI - fix ELD monitor name length
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>
2011-06-29 08:48:24 +02:00
Lydia Wang e322a36d39 ALSA: hda - Fix jack-detection on non-VT1708 VIA codecs
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>
2011-06-29 08:02:46 +02:00
Takashi Iwai 94230c11da ALSA: hda - Fix unused variable warning
sound/pci/hda/patch_cmedia.c: In function ‘cmi9880_fill_multi_init’:
sound/pci/hda/patch_cmedia.c:401:15: warning: unused variable ‘len’

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-29 08:02:33 +02:00
Takashi Iwai c82693db52 ALSA: hda - Enable auto-parser as default for Conexant codecs
Let's use auto-parser as default now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-29 08:02:21 +02:00
Takashi Iwai c2549312d2 Merge branch 'fix/hda' into topic/hda 2011-06-29 08:02:09 +02:00
Takashi Iwai 8d087c7600 ALSA: hda - Create snd_hda_get_conn_index() helper function
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>
2011-06-29 08:01:46 +02:00
Lydia Wang 63f10d2ca7 ALSA: hda - Fix unsol event initializations for VIA codecs
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>
2011-06-29 08:01:23 +02:00
Lars-Peter Clausen aef05294df ASoC: Blackfin: Add machine driver for EVAL-ADAV80X boards
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>
2011-06-28 17:12:44 +01:00
Lars-Peter Clausen cc52688a08 ASoC: Add ADAV80x codec driver
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>
2011-06-28 17:11:20 +01:00
Hans-Christian Egtvedt 0cfae7c937 ALSA: atmel - update author email for ABDAC, AC97C and AT73C213
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>
2011-06-28 16:56:07 +02:00
David Henningsson 9966db22ca ALSA: HDA: Add model=auto quirk for Acer Aspire 3830TG
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>
2011-06-28 14:09:57 +02:00
David Henningsson f0ca89b031 ALSA: HDA: Add a new Conexant codec ID (506c)
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>
2011-06-28 14:09:41 +02:00
Takashi Iwai ff2b7e2a3f ALSA: hda - Fix warnings with CONFIG_SND_POWER_SAVE=n
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>
2011-06-28 08:59:30 +02:00
Stephen Rothwell 880a050f4a ALSA: hda - remove SND_HDA_POWER_SAVE protection of struct hda_loopback_check
to fix build problems when it is disabled.

Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-28 08:55:56 +02:00
Takashi Iwai 4f574b7b1a ALSA: hda - More volume-init fixes for ALC267 codec
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>
2011-06-27 16:17:07 +02:00
Takashi Iwai 7ec9c6ccc6 ALSA: hda - Fix volume-init for ALC259 with invalid widget caps
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>
2011-06-27 15:53:38 +02:00
Takashi Iwai 050ea75317 ALSA: hda - Fix volume-init of ALC299 & co
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>
2011-06-27 15:48:17 +02:00