Commit Graph

6592 Commits

Author SHA1 Message Date
Takashi Iwai 69b5655a85 ALSA: hda - Fix Dell S14 pin setup
The pin setup for Dell S14 quirk is rather wrong for the latest driver.
Fixed pin 0x0a, 0x0b, 0x0d and 0x0f.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-15 12:37:42 +02:00
Takashi Iwai 44da531e95 ALSA: hda - Fix IDT92HD83* codec setup
Remove unnecessary (and buggy) init sequences left for IDT92HD83*
codecs in the previous fixes.  The DACs are now dynamically connected,
thus shouldn't be set statically in init verbs.  Also, the mono_nid
is detected dynamically, thus shouldn't be set staticaly, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-15 12:35:56 +02:00
Linus Torvalds 2ca7d674d7 Merge branch 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (257 commits)
  [ARM] Update mach-types
  ARM: 5636/1: Move vendor enum to AMBA include
  ARM: Fix pfn_valid() for sparse memory
  [ARM] orion5x: Add LaCie NAS 2Big Network support
  [ARM] pxa/sharpsl_pm: zaurus c3000 aka spitz: fix resume
  ARM: 5686/1: at91: Correct AC97 reset line in at91sam9263ek board
  ARM: 5640/1: This patch modifies the support of AC97 on the at91sam9263 ek board
  ARM: 5689/1: Update default config of HP Jornada 700-series machines
  ARM: 5691/1: fix cache aliasing issues between kmap() and kmap_atomic() with highmem
  ARM: 5688/1: ks8695_serial: disable_irq() lockup
  ARM: 5687/1: fix an oops with highmem
  ARM: 5684/1: Add nuc960 platform to w90x900
  ARM: 5683/1: Add nuc950 platform to w90x900
  ARM: 5682/1: Add cpu.c and dev.c and modify some files of w90p910 platform
  ARM: 5626/1: add suspend/resume functions to amba-pl011 serial driver
  ARM: 5625/1: fix hard coded 4K resource size in amba bus detection
  MMC: MMCI: convert realview MMC to use gpiolib
  ARM: 5685/1: Make MMCI driver compile without gpiolib
  ARM: implement highpte
  ARM: Show FIQ in /proc/interrupts on CONFIG_FIQ
  ...

Fix up trivial conflict in arch/arm/kernel/signal.c.

It was due to the TIF_NOTIFY_RESUME addition in commit d0420c83f ("KEYS:
Extend TIF_NOTIFY_RESUME to (almost) all architectures") and follow-ups.
2009-09-14 17:48:14 -07:00
Mark Brown 3eef08ba52 ASoC: Fix display of stream name in DAPM debugfs
Also display streams all the time while we're here.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-14 16:56:25 +01:00
Takashi Iwai 6e34c03321 ALSA: hda - Add support for HP dv6
Add the quirk entry for HP dv6.  Also add a workaround for the headphone
detection by setting hp_detect=1 beforehand.  Without this, the driver
won't do auto-muting because BIOS doesn't give any HP pin but only a
line-out pin.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-14 15:42:18 +02:00
Takashi Iwai 5f380eb1ef ALSA: hda - Fix HP/line-out initialization with IDT/STAC codecs
It's possible that hp_detect is set even though no headphone pin is
detected.  The driver issues, however, an unsol event only to hp_pins[0],
which can be invalid.

This patch adds the check of the valid pin to send an unsol event
at initialization and resume callbacks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-14 15:36:14 +02:00
Takashi Iwai fc64b26cfa ALSA: hda - Set default GPIO for IDT92HD71bxx
A smiliar fix for IDT 92HD71Bxx codecs like the previous commit for
other IDT/STAC codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-14 15:33:01 +02:00
Takashi Iwai af6ee30202 ALSA: hda - Set default GPIO for STAC/IDT codecs
IDT92HD73xx and STAC927x codecs use GPIO0 bit as EAPD on many machines.
However, currently we don't set it unless the model is specified just
for safety reason.  But, most machines do need this bit, so this safety
handling is rather annoying.

This patch enables GPIO0 setup as default for them.  Many HP / Dell
laptops should work even without model override with this change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-14 15:03:12 +02:00
Barry Song 472df3cbae ASoC: Provide API for reordering channels
The patch adds an interface to set the relationship between audio
channel number and slot number. The interface should be really useful
because audio channel n doesn't always use slot n in all platforms. And
for some devices, the relationship even can change with sound mode
switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-13 12:37:53 +01:00
Julia Lawall 33d7f77850 ASoC: Clean up error handling in MPC5200 DMA setup
Error handling code following a kzalloc should free the allocated data.
Error handling code following an ioremap should iounmap the allocated data.

The semantic match that finds the first problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,f1,l;
position p1,p2;
expression *ptr != NULL;
@@

x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
<... when != x
     when != if (...) { <+...x...+> }
(
x->f1 = E
|
 (x->f1 == NULL || ...)
|
 f(...,x->f1,...)
)
...>
(
 return \(0\|<+...x...+>\|ptr\);
|
 return@p2 ...;
)

@script:python@
p1 << r.p1;
p2 << r.p2;
@@

print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-12 13:41:50 +01:00
Russell King 87d721ad7a Merge branch 'master' into devel 2009-09-12 12:04:37 +01:00
Russell King ddd559b13f Merge branch 'devel-stable' into devel
Conflicts:
	MAINTAINERS
	arch/arm/mm/fault.c
2009-09-12 12:02:26 +01:00
Russell King cf7a2b4fb6 Merge branches 'arm', 'at91', 'bcmring', 'ep93xx', 'mach-types', 'misc' and 'w90x900' into devel 2009-09-12 12:01:34 +01:00
Takashi Iwai 3d3792cb45 ALSA: hda - Add missing model=auto entry for ALC269
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-11 07:50:47 +02:00
Takashi Iwai 1110afbe72 Merge branch 'topic/ymfpci' into for-linus
* topic/ymfpci:
  sound: ymfpci: increase timer resolution to 96 kHz
2009-09-10 15:33:09 +02:00
Takashi Iwai fd30afa454 Merge branch 'topic/usb-audio' into for-linus
* topic/usb-audio:
  ALSA: usb-audio - Fix types taken in min()
  sound: usb-audio: do not make URBs longer than sync packet interval
  sound: usb-audio: add MIDI drain callback
  sound: usb-audio: use multiple output URBs
  sound: usb-audio: use multiple input URBs
  sound: usb-audio: Xonar U1 digital output support
2009-09-10 15:33:07 +02:00
Takashi Iwai b34c866394 Merge branch 'topic/tlv-minmax' into for-linus
* topic/tlv-minmax:
  ALSA: usb-audio - Correct bogus volume dB information
  ALSA: usb-audio - Use the new TLV_DB_MINMAX type
  ALSA: Add new TLV types for dBwith min/max
2009-09-10 15:33:06 +02:00
Takashi Iwai 3827119e20 Merge branch 'topic/soundcore-preclaim' into for-linus
* topic/soundcore-preclaim:
  sound: make OSS device number claiming optional and schedule its removal
  sound: request char-major-* module aliases for missing OSS devices
  chrdev: implement __[un]register_chrdev()
2009-09-10 15:33:04 +02:00
Takashi Iwai 9d416811f8 Merge branch 'topic/snd-printk' into for-linus
* topic/snd-printk:
  ALSA: Fixed a typo of printk()
  ALSA: Add debug module option
  ALSA: core - strip too long file names in snd_print*()
2009-09-10 15:33:03 +02:00
Takashi Iwai df9200dd04 Merge branch 'topic/pcm-estrpipe-in-pm' into for-linus
* topic/pcm-estrpipe-in-pm:
  ALSA: pcm - Tell user that stream to be rewound is suspended
2009-09-10 15:33:02 +02:00
Takashi Iwai 2c0d19a78d Merge branch 'topic/pcm-drain-nonblock' into for-linus
* topic/pcm-drain-nonblock:
  ALSA: pcm - Increase protocol version
  ALSA: pcm - Fix drain behavior in non-blocking mode
2009-09-10 15:33:00 +02:00
Takashi Iwai 05a33e3d6f Merge branch 'topic/oxygen' into for-linus
* topic/oxygen:
  sound: oxygen: work around MCE when changing volume
2009-09-10 15:32:59 +02:00
Takashi Iwai fa28519002 Merge branch 'topic/oss' into for-linus
* topic/oss:
  ALSA: allocation may fail in	snd_pcm_oss_change_params()
  sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
  sound: fix OSS MIDI output data loss
2009-09-10 15:32:58 +02:00
Takashi Iwai 9cd9f42767 Merge branch 'topic/misc' into for-linus
* topic/misc:
  ALSA: Remove unneeded ifdef from sound/core.h
  ALSA: Remove struct snd_monitor_file from public sound/core.h
  ALSA: Release v1.0.21
2009-09-10 15:32:57 +02:00
Takashi Iwai 0f23c5cc50 Merge branch 'topic/midi' into for-linus
* topic/midi:
  sound: rawmidi: disable active-sensing-on-close by default
  sound: seq_oss_midi: remove magic numbers
  sound: seq_midi: do not send MIDI reset when closing
  seq-midi: always log message on output overrun
2009-09-10 15:32:56 +02:00
Takashi Iwai 8a3351bbb9 Merge branch 'topic/ice1724-pm' into for-linus
* topic/ice1724-pm:
  ALSA: ice1724 - Fix section mismatch
  ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD2
2009-09-10 15:32:55 +02:00
Takashi Iwai dcb37d509a Merge branch 'topic/hdsp' into for-linus
* topic/hdsp:
  ALSA: hdsp - allow proc reporting with disconnected io box
2009-09-10 15:32:54 +02:00
Takashi Iwai 2d4ff66ad7 Merge branch 'topic/hda' into for-linus
* topic/hda: (92 commits)
  ALSA: hda - Use auto model for HP laptops with ALC268 codec
  ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
  ALSA: hda - Add support of Alienware M17x laptop
  ALSA: hda - Remove dead codes from patch_sigmatel.c
  ALSA: hda - Fix input source selection of IDT92HD73xx
  ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT
  ALSA: hda - Unmute docking line-out as default with AD1984A codec
  ALSA: hda - Add another entry for Nvidia HDMI device
  ALSA: hda - Add missing GPIO initialization for AD1984A laptop model
  ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop model
  ALSA: hda - Fix ALC268/ALC269 headphone pin routing
  ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT codecs
  ALSA: hda - Add more quirk for HP laptops with AD1984A
  ALSA: hda - Add / fix model entries for HD-audio driver
  ALSA: hda - Add full audio support on Acer Aspire 7730G notebook
  ALSA: hda - Improve auto-cfg mixer name for ALC662
  ALSA: hda - Improve auto-cfg mixer name for ALC861-VD
  ALSA: hda - Improve auto-cfg mixer name for ALC262
  ALSA: hda - Improve auto-cfg mixer name for ALC260
  ALSA: hda - Improve auto-cfg mixer name for ALC880
  ...
2009-09-10 15:32:52 +02:00
Takashi Iwai 6a0f402146 Merge branch 'topic/dummy' into for-linus
* topic/dummy:
  ALSA: dummy - Increase MAX_PCM_SUBSTREAMS to 128
  ALSA: dummy - Add debug proc file
  ALSA: Add const prefix to proc helper functions
  ALSA: Re-export snd_pcm_format_name() function
  ALSA: dummy - Fake buffer allocations
  ALSA: dummy - Fix the timer calculation in systimer mode
  ALSA: dummy - Add more description
  ALSA: dummy - Better jiffies handling
  ALSA: dummy - Support high-res timer mode
2009-09-10 15:32:51 +02:00
Takashi Iwai f9892a52e2 Merge branch 'topic/dma-sgbuf' into for-linus
* topic/dma-sgbuf:
  ALSA: Fix SG-buffer DMA with non-coherent architectures
2009-09-10 15:32:50 +02:00
Takashi Iwai 6c5cb93b1e Merge branch 'topic/ctxfi' into for-linus
* topic/ctxfi:
  ALSA: ctxfi - Simple code clean up
  ALSA: ctxfi - Native timer support for emu20k2
2009-09-10 15:32:48 +02:00
Takashi Iwai f604529d0c Merge branch 'topic/ctl-add-remove-fixes' into for-linus
* topic/ctl-add-remove-fixes:
  sound: snd_ctl_remove_user_ctl: prevent removal of kernel controls
  sound: snd_ctl_remove_unlocked_id: simplify user control counting
  sound: snd_ctl_remove_unlocked_id: simplify error paths
  sound: snd_ctl_elem_add: fix value count check
2009-09-10 15:32:47 +02:00
Takashi Iwai 124e39b34d Merge branch 'topic/cs46xx' into for-linus
* topic/cs46xx:
  ALSA: cs46xx - Fix minimum period size
2009-09-10 15:32:46 +02:00
Takashi Iwai 9d2743f84d Merge branch 'topic/cmi8330' into for-linus
* topic/cmi8330:
  ALSA: cmi8330: Allow MPU-401-less operation
  ALSA: cmi8330: find OPL3 port automatically
  cmi8330: Add basic CMI8329 support
  ALSA: cmi8330: revert comments about AD1848 back
2009-09-10 15:32:45 +02:00
Takashi Iwai d0064a1b22 Merge branch 'topic/cleanup' into for-linus
* topic/cleanup:
  ALSA: info - Use krealloc()
2009-09-10 15:32:43 +02:00
Takashi Iwai b81e5ab34d Merge branch 'topic/azt3328' into for-linus
* topic/azt3328:
  ALSA: azt3328: fix previous breakage, improve suspend, cleanups
  ALSA: azt3328: large codec cleanup, add I2S port etc.
  ALSA: azt3328: fix Kconfig entry
2009-09-10 15:32:41 +02:00
Takashi Iwai e0b3032bcd Merge branch 'topic/asoc' into for-linus
* topic/asoc: (226 commits)
  ASoC: au1x: PSC-AC97 bugfixes
  ASoC: Fix WM835x Out4 capture enumeration
  ASoC: Remove unuused hw_read_t
  ASoC: fix pxa2xx-ac97.c breakage
  ASoC: Fully specify DC servo bits to update in wm_hubs
  ASoC: Debugged improper setting of PLL fields in WM8580 driver
  ASoC: new board driver to connect bfin-5xx with ad1836 codec
  ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI
  ASoC: davinci: i2c device creation moved into board files
  ASoC: Don't reconfigure WM8350 FLL if not needed
  ASoC: Fix s3c-i2s-v2 build
  ASoC: Make platform data optional for TLV320AIC3x
  ASoC: Add S3C24xx dependencies for Simtec machines
  ASoC: SDP3430: Fix TWL GPIO6 pin mux request
  ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time
  ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop
  ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI
  OMAP: McBSP: Use textual values in DMA operating mode sysfs files
  ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
  ASoC: Select core DMA when building for S3C64xx
  ...
2009-09-10 15:32:40 +02:00
Takashi Iwai 45fae5c78d Merge branch 'topic/ali5451-cleanup' into for-linus
* topic/ali5451-cleanup:
  ALSA: ali5451: remove dead code
2009-09-10 15:32:38 +02:00
Mike Rapoport 2ba9fd0d15 [ARM] pxa: update pxa2xx-ac97.c to use 'struct dev_pm_ops'
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
2009-09-10 19:15:37 +08:00
Joonyoung Shim 2312fd8f6b ASoC: AK4671: add ak4671 codec driver
The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier,
Receiver-Amplifier and Headphone-Amplifier.

The datasheet for the ak4671 can find at the following url:
http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-10 00:27:57 +01:00
Mark Brown 215edda3ad ASoC: Allow per-route connectedness checks for supplies
Some chips with complex internal supply (particularly clocking)
arragements may have multiple options for some of the supply
connections. Since these don't affect user-visible audio routing
the expectation would be that they would be managed automatically
by one of the drivers.

Support these users by allowing routes to have a connected function
which is queried before the connectedness of the path is checked as
normal. Currently this is only done for supplies, other widgets
could be supported but are not currently since the expectation for
them is that audio routing will be under the control of userspace.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-08 19:24:56 +01:00
Manuel Lauss cdc65fbe18 ASoC: au1x: PSC-AC97 bugfixes
This patch fixes the following bugs:

- only reprogram bitdepth if it has changed since last call to hw_params.
- add locking inside ac97_read/write functions:
  When reprogramming sample depth, the ac97 unit has to be disabled,
  which should not be done in the middle of codec register accesses.

- retry timed-out codec register accesses.

- wait for status bits to set/clear when starting/stopping various
  functional blocks; very important after reenabling AC97 unit else
  sound may be distorted (e.g. high-pitch noise in 1kHz sine wave).

- clear fifos before/after starting/stopping RX/TX.

- longer timeouts waiting for PSC/AC97 ready after cold reset
  with certain codecs this can take ridiculous amounts of time.

Run-tested on various Au1200 platforms with various codecs.

Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-08 19:21:27 +01:00
Takashi Iwai b888d1ce82 ALSA: dummy - Increase MAX_PCM_SUBSTREAMS to 128
Increase the limit of PCM substreams to 128.  The default value is
unchanged; only the max accept value is increased.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-08 18:15:17 +02:00
Takashi Iwai 9b151fec13 ALSA: dummy - Add debug proc file
Added the debug proc file to see or change the snd_pcm_hardware fields
to emulate.  The parameters can be changed by writing to a proc file like:

    # echo periods_min 4 > /proc/asound/card1/dummy_pcm

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-08 14:46:49 +02:00
Takashi Iwai 4f7454a997 ALSA: Add const prefix to proc helper functions
Add appropriate const prefix to char * arguments in proc helper functions.
Also fixed the caller side to be proper const pointers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-08 14:45:06 +02:00
Takashi Iwai 6e5265ec34 ALSA: Re-export snd_pcm_format_name() function
Re-export snd_pcm_format_name() function to be used outside the PCM core.
As a first example, usbaudio is changed to use it now again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-08 14:26:51 +02:00
Takashi Iwai 33d7867458 ALSA: hda - Use auto model for HP laptops with ALC268 codec
The HP laptops with ALC268 codec seem working better with model=auto
than model=toshiba; e.g. the auto model fixes missing digital outputs.
Let's fix quirk entry to choose auto model explicitly.

Tested-by: Jens Jorgensen <jbj1@ultraemail.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-08 11:07:56 +02:00
Sophie Hamilton 6148b130eb ALSA: cs46xx - Fix minimum period size
Fix minimum period size for cs46xx cards. This fixes a problem in the
case where neither a period size nor a buffer size is passed to ALSA;
this is the case in Audacious, OpenAL, and others.

Signed-off-by: Sophie Hamilton <kernel@theblob.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-08 10:59:49 +02:00
Mark Brown 87831cb660 ASoC: Fix WM835x Out4 capture enumeration
It's the 8th enum of a zero indexed array. This is why I don't let
new drivers use these arrays of enums...

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2009-09-07 18:56:24 +01:00
Takashi Iwai 82a783f4bc ALSA: Remove struct snd_monitor_file from public sound/core.h
The struct snd_monitor_file is used locally only in sound/core/init.c,
thus it should be moved there from the public sound/core.h.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-07 15:50:18 +02:00
Clemens Ladisch f1bc07af9a sound: oxygen: work around MCE when changing volume
When the volume is changed continuously (e.g., when the user drags a
volume slider with the mouse), the driver does lots of I2C writes.
Apparently, the sound chip can get confused when we poll the I2C status
register too much, and fails to complete a read from it.  On the PCI-E
models, the PCI-E/PCI bridge gets upset by this and generates a machine
check exception.

To avoid this, this patch replaces the polling with an unconditional
wait that is guaranteed to be long enough.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Johann Messner <johann.messner at jku.at>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-07 12:15:43 +02:00
Joonyoung Shim 341c9b84bc ASoC: Factor out I2C 8 bit address 8 bit data I/O
This patch is for the AK4671 codec driver using this format.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-07 11:14:12 +01:00
Takashi Iwai a68c4d1133 ALSA: dummy - Fake buffer allocations
Instead of allocating the real buffers, use a fake buffer and ignore
read/write in the dummy driver so that we can save the resources.
For mmap, a single page (unique to the direction, though) is reused
to all buffers.

When the app requires to read/write the real buffers, pass fake_buffer=0
module option at loading time.  This will get back to the old behavior.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-07 09:01:10 +02:00
ddiaz@cenditel.gob.ve a65cc60f63 ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
The model clevo-m540r was created with 6-channel and digital support. All
functions verified except spdif. Tested with a VIT D2000 laptop which has:

[lspci extract]
 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio
Controller [8086:284b] (rev 03)
        Subsystem: CLEVO/KAPOK Computer Device [1558:5409]

[/proc/asound/card0/codec\#0 header]
Codec: Realtek ALC883
Address: 0
Function Id: 0x1
Vendor Id: 0x10ec0883
Subsystem Id: 0x15585409
Revision Id: 0x100002

[Added a comment about HP mute and the model description by tiwai]

Signed-off-by: Dhionel Diaz <ddiaz@cenditel.gob.ve>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-07 07:32:33 +02:00
Linus Torvalds b71b7dc09a Merge branch 'fix/oxygen' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/oxygen' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  sound: oxygen: handle cards with missing EEPROM
  sound: oxygen: fix MCLK rate for 192 kHz playback
2009-09-05 14:55:30 -07:00
Mark Brown 85488037bb ASoC: Add source argument to PLL configuration
More and more devices feature PLLs and FLLs with the ability to select
between multiple input clocks. In order to better support these devices
a new argument, source, has been added to the set_pll() configuration
API. Using set_clkdiv() is often difficult due to the need to stop the
PLL/FLL before any reconfiguration can be done.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-05 18:52:16 +01:00
Robert Schwebel 367da1527a ASoC: fix pxa2xx-ac97.c breakage
Today's linux-next fails to build with

  sound/arm/pxa2xx-ac97.c: In function 'pxa2xx_ac97_probe':
  sound/arm/pxa2xx-ac97.c:211: error: 'pxa2xx_audio_ops_t' has no member named 'codec_data'
  make[2]: *** [sound/arm/pxa2xx-ac97.o] Error 1

It looks like commit e2365bf313 has
introduced this; patch below.

Signed-off-by: Robert Schwebel <r.schwebel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-04 20:19:56 +01:00
Takashi Iwai b5d1078173 ALSA: dummy - Fix the timer calculation in systimer mode
Fix the expire-time calculation in the systimer mode when the buffer
size isn't aligned to the period size.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-04 08:45:11 +02:00
Takashi Iwai b142037b4c ALSA: dummy - Better jiffies handling
In the system-timer mode, snd-dummy driver issues each tick to update
the position.  This is highly inefficient and even inaccurate if the
timer can't be triggered at each tick.

Now rewritten to wake up only at the period boundary.  The position
is calculated from the current jiffies.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-03 16:01:06 +02:00
Takashi Iwai c631d03c68 ALSA: dummy - Support high-res timer mode
Allow snd-dummy driver to use high-res timer as its timing source
instead of the system timer.  The new module option "hrtimer" is added
to turn on/off the high-res timer support.  It can be switched even
dynamically via sysfs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-03 15:59:26 +02:00
Clemens Ladisch 92653453c3 sound: oxygen: handle cards with missing EEPROM
The card model detection code introduced in 2.6.30 that tries to work
around partially broken EEPROM contents by reading the EEPROM directly
does not handle cards where the EEPROM has been omitted.  In this case,
we have to use the default ID to allow the driver to load.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Ozan Çağlayan <ozan@pardus.org.tr>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-03 07:38:06 +02:00
Mark Brown 2eff31e809 ASoC: Fully specify DC servo bits to update in wm_hubs
Avoids potential issues if we read back unexpected values during
a read/modify/write cycle.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-02 19:36:22 +01:00
Takashi Iwai 842ae63800 ALSA: hda - Add support of Alienware M17x laptop
Added the quirk for Alienware M17x with IDT 92HD73* codec chip.
It has two HP and one line-out jack, one mic jack, a built-in
speaker and a built-in mic.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-02 07:43:08 +02:00
Takashi Iwai 4a9678909b ALSA: hda - Remove dead codes from patch_sigmatel.c
Due to the previous fix of input source for IDT92HD73xx, the amp mux
and amp vol stuff became unused.  Let's rip off dead codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-02 01:09:54 +02:00
Takashi Iwai e2aec17100 ALSA: hda - Fix input source selection of IDT92HD73xx
Fix the mux_nids to select directly the input source instead of mux
mixers so that it works with the current mux enum handler for IDT
92HD73xx codecs.

Also, clean up useless / unnecessary mixer controls and init verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-02 01:00:05 +02:00
Takashi Iwai d94ff6b7ca ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT
Fix the old dead CONFIG_SND_DEBUG_DETECT to CONFIG_SND_DEBUG_VERBOSE.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-02 00:20:21 +02:00
jassi brar 5c0d38c947 ASoC: Debugged improper setting of PLL fields in WM8580 driver
Bug was caught while trying to use WM8580 as I2S master on SMDK.
Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved
by referring to WM8580A manual and setting mask value correctly and
making the code to not touch 'reserved' bits of PLL4 register.

Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-01 11:37:41 +01:00
Barry Song dce944dbb2 ASoC: new board driver to connect bfin-5xx with ad1836 codec
As discussed, the patch uses the original TDM order without rewriting.
For the match between TDM slot number and audio channel number, a new
API need be added.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-01 11:36:13 +01:00
Takashi Iwai 2ad81ba014 ALSA: hda - Unmute docking line-out as default with AD1984A codec
Unmute the docking-station line-out as default on machines with
AD1984A codec chip.  It can be still muted via "Dock" mixer switch.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-01 09:09:26 +02:00
Takashi Iwai f8ff035e38 ALSA: hda - Add another entry for Nvidia HDMI device
Added another entry for Nvidia HDMI device (10de:0003).

Reference: kernel bug#14097
	http://bugzilla.kernel.org/show_bug.cgi?id=14097

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-01 08:53:19 +02:00
Clemens Ladisch b91ab72b83 sound: oxygen: fix MCLK rate for 192 kHz playback
Do not forget to program the MCLK ratio for the I2S output.
Otherwise, the master clock frequency can be too high for
the DACs at sample frequencies above 96 kHz.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-01 08:45:40 +02:00
Linus Torvalds cda9856f1c Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix MacBookPro 3,1/4,1 quirk with ALC889A
  ALSA: hda - Add missing mux check for VT1708
2009-08-31 17:36:10 -10:00
Roel Kluin cbbb05703d ALSA: allocation may fail in snd_pcm_oss_change_params()
Allocation may fail, show if it did.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
[Additional fix for invalid runtime->oss.prepare flag set by tiwai]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-31 16:33:23 +02:00
Takashi Iwai fe7e56814c ALSA: hda - Add missing GPIO initialization for AD1984A laptop model
A similar initialization of GPIO1 pin like mobile model is needed
for laptop model, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-31 08:37:46 +02:00
Takashi Iwai 17bbaa6f60 ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop model
Add the support of automatic mute and mic-switching of the docking
station HP and mic plugs for AD1984A laptop model for some HP machines.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-31 08:32:27 +02:00
Takashi Iwai be0ae923a4 Merge branch 'fix/hda' into topic/hda
Conflicts:
	sound/pci/hda/patch_realtek.c
2009-08-31 08:27:10 +02:00
Takashi Iwai e9af4f365f ALSA: hda - Fix ALC268/ALC269 headphone pin routing
Fix the headphone pin routing of ALC268/ALC269 codecs.  Using alc882
routine doesn't work because alc268/alc269 parser assumes the
independent DACs for both HP and speaker outputs.  Need to assign the
DAC depending on the pin.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-31 08:25:58 +02:00
Takashi Iwai a3f730af7e ALSA: hda - Fix MacBookPro 3,1/4,1 quirk with ALC889A
This patch fixes the wrong headphone output routing for MacBookPro 3,1/4,1
quirk with ALC889A codec, which caused the silent headphone output.
Also, this gives the individual Headphone and Speaker volume controls.

Reference: kernel bug#14078
	http://bugzilla.kernel.org/show_bug.cgi?id=14078

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-08-31 08:23:13 +02:00
Takashi Iwai 0f67a61162 ALSA: hda - Add missing mux check for VT1708
In patch_vt1708(), the check of MUX nids is missing and this results in
the -EINVAL error in accessing Input Source mixer element.  Simpliy
adding the call of get_mux_nids() fixes the problem.

Reference: Novell bnc#534904
	https://bugzilla.novell.com/show_bug.cgi?id=534904

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-31 08:12:29 +02:00
Takashi Iwai 96f845de89 ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT codecs
So far, the digital mic capture volume wasn't created.  This is because
IDT codecs have output amps for digital mics, not input amps, while
input amps should be used for other analog pins.  Thus the automatic
capture volume creation should check both directions for digital mics.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-29 00:49:36 +02:00
Jarkko Nikula d2c0bdaa93 ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI
The McBSP1 port in OMAP3 processors (I believe OMAP2 too but I don't have
specifications to check it) have additional CLKR and FSR pins for McBSP1
receiver. Reset default is that receiver is using bit clock and frame
sync signal from those pins but it is possible to configure to use
also CLKX and FSX pins as well. In fact, other McBSP ports are doing that
internally that transmitter and receiver share the CLKX and FSX.

Add functionaly that machine drivers can set the CLKR and FSR sources by
using the snd_soc_dai_set_sysclk.

Thanks to "Aggarwal, Anuj" <anuj.aggarwal@ti.com> for reporting the issue.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-28 18:36:43 +01:00
Chaithrika U S f4890b5c04 ASoC: davinci: i2c device creation moved into board files
Also, the codec setup data structure has to remain for successful
probe.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-28 10:33:10 +01:00
Takashi Iwai 36ce99c1dc ALSA: Add debug module option
Add debug module option to snd core.
This controls the debug print level.  When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value.  debug=0 means no debug messsages.
As default, it's set to the verbose level 2.

Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-27 17:42:08 +02:00
Takashi Iwai 286f5875ca ALSA: hda - Add more quirk for HP laptops with AD1984A
More entries for HP laptops to get them working properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-27 14:37:51 +02:00
Takashi Iwai 1b0053a0f0 ALSA: core - strip too long file names in snd_print*()
When modules are built with M= option, they pass long file paths to
__FILE__.  This results in ugly outputs of snd_print*() when
CONFIG_SND_VERBOSE_PRINTK is set.

This patch adds a check of the path and strips the leading path dirs
if the file name is an absolute path to improve the readability of logs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-27 12:39:35 +02:00
Mark Brown f1e887de2d ASoC: Don't reconfigure WM8350 FLL if not needed
If the requested FLL configuration is the one we're currently running
in it's at best pointless to reconfigure the FLL.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-26 15:27:57 +01:00
Mark Brown 5dc0748182 ASoC: Fix s3c-i2s-v2 build
We now need the PCM header to kick the DMA.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-26 15:27:57 +01:00
Mark Brown 977d49e00d ASoC: Make platform data optional for TLV320AIC3x
Now that we don't need the I2C address for the device the platform data
is redundant so allow it to be omitted.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Chaithrika U S <chaithrika@ti.com>
2009-08-26 15:27:56 +01:00
Mark Brown bc36681fdc ASoC: Add S3C24xx dependencies for Simtec machines
No point in building them for S3C64xx, certainly no sense in running
into build issues there.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-26 15:27:56 +01:00
Roel Kluin f1d269bac2 sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
Since !LI_CCFG_* evaluates to 0, this did not change anything to
cfgval and ctlval.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-26 12:42:43 +02:00
Sudhakar Rajashekhara 60902a2cb1 davinci: EDMA: multiple CCs, channel mapping and API changes
- restructure to support multiple channel controllers by using
  additional struct resources for each CC

- interface changes visible to EDMA clients

  Introduce macros to build IDs from controller and channel number,
  and to extract them. Modify the edma_alloc_slot function to take an
  extra argument for the controller.

  Also update ASoC drivers to use API.  ASoC changes
  Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

- Move queue related mappings to dm<soc>.c

  EDMA in DM355 and DM644x has two transfer controllers while DM646x
  has four transfer controllers. Moving the queue to tc mapping and
  queue priority mapping to dm<soc>.c will be helpful to probe these
  mappings from platform device so that the machine_is_* testing will
  be avoided.

- add channel mapping logic

  Channel mapping logic is introduced in dm646x EDMA. This implies
  that there is no fixed association for a channel number to a
  parameter entry number. In other words, using the DMA channel
  mapping registers (DCHMAPn), a PaRAM entry can be mapped to any
  channel. While in the case of dm644x and dm355 there is a fixed
  mapping between the EDMA channel and Param entry number.

Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Reviewed-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
2009-08-26 10:56:56 +03:00
Candelaria Villareal, Jorge 30cd0c4ad5 ASoC: SDP3430: Fix TWL GPIO6 pin mux request
Fix the write to PMBR1 register through I2C. Also, the constant which
holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This
name is based on TRM to avoid confusion.

Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-25 19:30:32 +01:00
Linus Torvalds a206e9417f Merge branch 'fix/misc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/misc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  sound: pcm_lib: fix unsorted list constraint handling
  sound: vx222: fix input level control range check
  ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready()
2009-08-25 09:47:06 -07:00
Denis Kuplyakov fc86f95415 ALSA: hda - Add full audio support on Acer Aspire 7730G notebook
1) Added support of internal subwoofer (it sounds!!!)
2) Auto muting front speakers and internal subwoofer on headphones plug.
3) Internal mic works.
4) 3 channel mods (jack maps):
       black  pink         blue
2ch: front   ext mic     line in
4ch: front   ext mic     surround
6ch: front   CLFE        surround
  Can be changed in mixer.
5) Sound can be recorded from:
 Internal mic
 Ext mic
 Cd
 Line in
6) 2 separate capture channels.

Signed-off-by: Denis Kuplyakov <dener.kup@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-25 18:16:55 +02:00
Takashi Iwai 0d884cb936 ALSA: hda - Improve auto-cfg mixer name for ALC662
The last patch in this series is for ALC662; pretty similar as the
previous patch for ALC861-VD.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-25 16:14:35 +02:00
Takashi Iwai a4fcd49109 ALSA: hda - Improve auto-cfg mixer name for ALC861-VD
One more patch to give a better name for the primary output controls,
this time for ALC861-VD codec.  The change is simple, just checking the
pin connection whether it's a speaker-out.  When both speaker and HP
are assigned, we name the volume as "PCM" as this influences on both
outputs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-25 16:12:15 +02:00
Takashi Iwai c3fc1f502a ALSA: hda - Improve auto-cfg mixer name for ALC262
Similar improvements for ALC262 codec like previous two commits:
assign a better name, either Master or Speaker, for the primary output
controls.

However, in the case of ALC262 codec, the necessary changes are larger
than others because we need to check the possibility of different mixer
amps depending on the pins.  The pin 0x16 is mono, and bound with the
dedicated mixer 0x0e while other pins are bound with 0x0c.  Thus, there
are two possible volumes.

When only one of them is used, we can name it as "Master".  OTOH, when
both are used at the same time, they have to be named uniquely.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-25 16:08:47 +02:00
Takashi Iwai 23112d6d2d ALSA: hda - Improve auto-cfg mixer name for ALC260
Instead of fixed "Front" mixer name, try to assign a better name, e.g.
"Master" or "Speaker" fot the primary output volume controls of ALC260
codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-25 16:07:08 +02:00
Takashi Iwai cb162b6bf2 ALSA: hda - Improve auto-cfg mixer name for ALC880
When there is only one DAC is used for ALC880, try to assign a better
name, either Speaker or Front, depending on the output pin type.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-25 16:05:03 +02:00
Shine Liu faf907c7ba ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time
s3c24xx dma has the auto reload feature, when the the trnasfer is done,
CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA
ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So
the transmission is repeated.

IRQ is issued while auto reload occurs. We change the DISRC and
DCON[19:0] in the ISR, but at this time, the auto reload has been
performed already. The first block is being re-transmitted by the DMA.

So we need rewrite the DISRC and DCON[19:0] for the next block
immediatly after the this block has been started to be transported.

The function s3c2410_dma_started() is for this perpose, which is called
in the form of "s3c2410_dma_ctrl(prtd->params->channel,
S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger().

But it is not correct. DMA transmission won't start until DMA REQ signal
arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1)
is called in s3c24xx_i2s_trigger().

In the current framework, s3c24xx_pcm_trigger() is always called before
s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in
s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or
s3c24xx_snd_rxctrl(1) is called in this function.

However, s3c2410_dma_started() is dma related, to call this function we
should provide the channel number, which is given by
substream->runtime->private_data->params->channel. The private_data
points to a struct s3c24xx_runtime_data object, which is define in
s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c

Fix this by moving the call to signal the DMA started to the DAI
drivers.

Signed-off-by: Shine Liu <liuxian@redflag-linux.com>
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-25 13:09:05 +01:00
Takashi Iwai 05f5f47708 ALSA: hda - Generalize input pin parsing in patch_realtek.c
Provide a standard parser for input pins to create the input mixer
and input source controls instead of having a difference one for each
Realtek codec.  The new helper parses the codec connections dynamically
isntead of fixed indicies.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-25 13:10:18 +02:00
Jarkko Nikula d09a2afc93 ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop
Functionality of functions omap_mcbsp_xmit_enable and omap_mcbsp_recv_enable
can be merged into omap_mcbsp_start and omap_mcbsp_stop since API of
those omap_mcbsp_start and omap_mcbsp_stop was changed recently allowing
to start and stop individually the transmitter and receiver.

This cleans up the code in arch/arm/plat-omap/mcbsp.c and in
sound/soc/omap/omap-mcbsp.c which was the only user for those removed
functions.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-25 10:20:48 +01:00
Jarkko Nikula 32080af7a6 ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI
Commit ca6e2ce086 is setting up few XCCR and
RCCR bits for I2S and DPS_A formats. Part of the bits are already set
for all formats and I believe that XDISABLE and RDISABLE bits are
format independent.

As XCCR and RCCR are found only from OMAP2430 and OMAP34xx, I move setup
of XDISABLE and RDISABLE to where those cpu's are tested and remove format
dependent part for simplicity.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-25 10:20:48 +01:00
Clemens Ladisch b1ddaf681e sound: pcm_lib: fix unsorted list constraint handling
snd_interval_list() expected a sorted list but did not document this, so
there are drivers that give it an unsorted list.  To fix this, change
the algorithm to work with any list.

This fixes the "Slave PCM not usable" error with USB devices that have
multiple alternate settings with sample rates in decreasing order, such
as the Philips Askey VC010 WebCam.

http://bugzilla.kernel.org/show_bug.cgi?id=14028

Reported-and-tested-by: Andrzej <adkadk@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-25 08:52:34 +02:00
Mark Brown e4aa8dd5ca Merge branch 'topic/digital-mixing' into for-2.6.32 2009-08-24 20:44:41 +01:00
Mark Brown 239a22aaa9 ASoC: Select core DMA when building for S3C64xx
Ensure that the core DMA support is available when building for
S3C64xx.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-24 20:42:48 +01:00
Takashi Iwai 9d0b71b1cf ALSA: hda - Reuse ALC268 parser for ALC269
Reuse a part of the code of ALC268 parser for ALC269.
This will change the default output volume either to Front or Speaker
depending on the pin configuration.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-24 14:10:30 +02:00
Clemens Ladisch edd1365e90 sound: vx222: fix input level control range check
Fix a logic error in the range check of the input level control that
would prevent setting any volume less than the maximum.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-24 09:46:08 +02:00
Wu Fengguang fd72d00846 ALSA: hda: move open coded tricks into get_wcaps_channels()
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-24 09:42:48 +02:00
Takashi Iwai c6ea2af76a ASoC: Remove unneeded inclusion of linux/regulator/consumer.h
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-24 09:41:32 +02:00
Takashi Iwai 20496ff378 ASoC: add missing inclusion of debugfs.h
To fix compile errors.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-24 09:41:05 +02:00
Marek Vasut e2365bf313 ASoC: Pass correct platform data from pxa2xx-ac97
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-23 18:18:01 +01:00
Bartlomiej Zolnierkiewicz 848bffef28 ALSA: ali5451: remove dead code
Remove code covered by #if/endif 0 and #ifdef/endif CODEC_RESET
(CODEC_RESET is never defined).

Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-23 18:59:14 +02:00
Bartlomiej Zolnierkiewicz 70bdbd3d1a ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready()
Modify loops in such way that the register value is checked also after
the timeout condition, just in case the heavy interrupt load etc. caused
the thread to sleep for the time period exceeding the timeout value.

While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready().

Reported-by: Jack Byer <ojbyer@usa.net>
Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-23 18:58:07 +02:00
Roel Kluin 821ebc86ef ASoC: free socdev if init_card() fails in wm9705_soc_probe()
Free socdev if snd_soc_init_card() fails.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-23 10:41:06 +01:00
Mark Brown 79fb9387f8 ASoC: Add DAPM widget power decision debugfs files
Currently when built with DEBUG DAPM will dump information about
the power state decisions it is taking for each widget to dmesg.
This isn't an ideal way of getting the information - it requires
a kernel build to turn it on and off and for large hub CODECs the
volume of information is so large as to be illegible. When the
output goes to the console it can also cause a noticable impact
on performance simply to print it out.

Improve the situation by adding a dapm directory to our debugfs
tree containing a file per widget with the same information in
it. This still requires a decision to build with debugfs support
but is easier to navigate and much less intrusive.

In addition to the previously displayed information active streams
are also shown in these files.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-21 17:17:59 +01:00
Kuninori Morimoto b8e583f601 ASoC: Add FSI-AK4642 sound support for SuperH
This patch is tested by ms7724se

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-21 11:02:03 +01:00
Kuninori Morimoto a3a83d9a7c ASoC: Add ak4642/ak4643 codec support
This is very simple driver for ALSA
It supprt headphone output and stereo input only
This patch is tested by ms7724se

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-21 10:54:02 +01:00
Ben Dooks b2ec22e263 ASoC: S3C24XX: Support for Simtec Hermes boards
Add support for the tlv320aic3x CODEC on the Simtec Hermes board.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-21 10:53:06 +01:00
Ben Dooks aa6b904e66 ASoC: tlv320aic3x: fixup board device changes
Fixup the device changes by modifying the files that we just removed the
explicit device creation from with i2c_register_board_info() until this
can be moved into the relevant board files.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-21 10:52:57 +01:00
Ben Dooks cb3826f524 ASoC: tlv320aic3x: Change to use device model
The tlv320aic3x driver managed its own i2c device, instead of an extant
one created by the board support code. Change the code to make it so that
the driver binds to an extant (in this case i2c) device.

Add explict tlv320aic33 as well as tlv320aic3x to the supported device
table and remove the old driver bindings from the users of this code.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-21 10:52:49 +01:00
Ben Dooks 14412acde5 ASoC: S3C24XX: Add audio core and tlv320aic23 for Simtec boards
Add core support for the range of S3C24XX Simtec boards with TLV320AIC23
CODECs on them. Since there are also boards with similar IIS routing the
AMP and the configuration code is placed in a core file for re-use with
other CODEC bindings.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-21 10:52:42 +01:00
Eduardo Valentin a0a499c579 ASoC: OMAP: Use DMA operating mode of McBSP
Configures DMA sync mode depending on McBSP operating mode value.
The value is configurable by McBSP instance. So, depending
on McBSP operating mode, the DMA sync mode is passed from
omap-mcbsp to omap-pcm. Besides that, it also configures
McBSP threshold value depending on which McBSP mode is activated.

Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-20 20:10:29 +01:00
Eduardo Valentin caebc0cb3b ASoC: OMAP: Use McBSP threshold to playback and capture
This patch changes the way DMA is done in omap-pcm.c
in order to reduce power consumption. There is no need
to have so much SW control in order to have DMA in idle
state during audio streaming. Configuring McBSP threshold value
and DMA to FRAME_SYNC are sufficient.

Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-20 20:10:29 +01:00
Eero Nurkkala ca6e2ce086 ASoC: Always syncronize audio transfers on frames
All these steps are required for ASoC to behave correctly.
rccr and xccr are format dependent, for example TDM audio
has different values than I2S or DSP_A. Also the
omap_mcbsp_xmit_enable and/or omap_mcbsp_recv_enable must
be called right after the DMA has started.
This provides no longer L and R channels switching at random.

Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-20 20:10:29 +01:00
Eero Nurkkala c721bbdad7 ASoC: Add runtime check for RFIG and XFIG
This is, no RFIG or XFIG (Not defined in 34xx), correct
initiliazation of rccr and xccr.

Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-20 20:10:28 +01:00
Eduardo Valentin a152ff24b9 ASoC: OMAP: Make DMA 64 aligned
Align DMA address to DMA burst transaction sizes.

Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-20 20:10:28 +01:00
Eduardo Valentin 9599d485cb ASoC: OMAP: Enable DMA burst mode
Improve DMA transfers by enabling Burst transaction.

Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-20 20:10:28 +01:00
Kuninori Morimoto a4d7d550a9 ASoC: Add SuperH FSI driver support for ALSA
This driver is very simple.
It support playback only now.
This patch is tested by ms7724se board.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-20 20:01:42 +01:00
Shine Liu f61c890ec6 ASoC: S3C24XX : Align the peroid size to the buffer size
> Then it's a driver bug.  If unaligned period size is allowed, it means
> that the irq is really generated in that period, not at the buffer
> boundary.  Otherwise, it must have a proper hw-constraint to align the
> period size to the buffer size.

This patch will fix the bug metioned in the above mail. Force the peroid
size to be aligned with the buffer size.

Based and tested on linux-2.6.31-rc6.

Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-20 19:42:40 +01:00
Linus Torvalds a1d1251115 Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix probe of Toshiba laptops with ALC268 codec
  ALSA: hda: add model for Intel DG45ID/DG45FC boards
  ALSA: hda: enable speaker output for Compaq 6530s/6531s
2009-08-20 10:19:39 -07:00
Takashi Iwai 4cdc115fd3 ALSA: pcm - Fix drain behavior in non-blocking mode
The current PCM core has the following problems regarding PCM draining
in non-blocking mode:

- the current f_flags isn't checked in snd_pcm_drain(), thus changing
  the mode dynamically via snd_pcm_nonblock() after open doesn't work.
- calling drain in non-blocking mode just return -EAGAIN error, but
  doesn't provide any way to sync with draining.

This patch fixes these issues.
- check file->f_flags in snd_pcm_drain() properly
- when O_NONBLOCK is set, PCM core sets the stream(s) to DRAIN state
  but quits ioctl immediately without waiting the whole drain; the
  caller can sync the drain manually via poll()

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-20 16:40:16 +02:00
Mark Brown f8bae4caaa ALSA: Restore support for DMAless DAIs on PXA
Used for applications such as direct bluetooth connections on
smartphones which don't go via the CPU. This used to be supported
before the refactoring to share code but this check was removed
during that move.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-19 20:30:14 +01:00
Takashi Iwai 454e134d0e Merge branch 'fix/hda' into topic/hda 2009-08-19 20:10:24 +02:00
Takashi Iwai 3abf2f3639 ALSA: hda - Fix probe of Toshiba laptops with ALC268 codec
There are many variants of Toshiba laptops with ALC268 codec, and
it seems that a few of them don't work with model=toshiba preset
since they have the secondary ALC268 codec just for HDMI output.
This is a regression due to the previous clean-up work to merge all
Toshiba quirk entries into a single check.

This patch adds the identification of such laptops to apply the
standard BIOS-probing method.  Unfortunately, Toshiba laptops have
all the same PCI SSID, so we need to check the codec SSID to identify
each device.

Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-19 20:05:02 +02:00
Mark Brown 474e09ca01 ASoC: Provide default set_bias_level() implementation
If the CODEC does not provide a set_bias_level() then update the
bias_level variable for it since other parts of the system expect
that to be maintained.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-19 14:18:53 +01:00
Takashi Iwai 1c11ce8118 Merge branch 'fix/hda' into topic/hda 2009-08-19 12:11:06 +02:00
Wu Fengguang ae709440ed ALSA: hda: add model for Intel DG45ID/DG45FC boards
The BIOS pin configs are in fact correct and shall not be overwritten.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-19 12:10:25 +02:00
Wu Fengguang 150fe14c1a ALSA: hda: enable speaker output for Compaq 6530s/6531s
HP Compaq 6530s and 6531s internal speaker is silence or becomes silence
within 1 minute after fresh boot. It is found that pin 0x1c must be set to
PIN_OUT mode to make the speaker work. This is weird - line-in pin 0x1c and
speaker pin 0x16 seem to be unrelated.

The codec differences before/after patch are:

@@ Node 0x17 [Pin Complex] wcaps 0x40020b:
   Pin Default 0x41a6e130: [N/A] Mic at Ext Rear
     Conn = Digital, Color = White
     DefAssociation = 0x3, Sequence = 0x0
     Misc = NO_PRESENCE
-  Pin-ctls: 0x24: IN
+  Pin-ctls: 0x40: OUT
@@ Node 0x1c [Pin Complex] wcaps 0x40018d:
   Pin Default 0x41813021: [N/A] Line In at Ext Rear
     Conn = 1/8, Color = Blue
     DefAssociation = 0x2, Sequence = 0x1
-  Pin-ctls: 0x24: IN VREF_80
+  Pin-ctls: 0x40: OUT VREF_HIZ
   Unsolicited: tag=00, enabled=0
   Connection: 1
      0x24

Tests show that it won't impact (external) Mic recording.

Reported-by: "Lin, Ming M" <ming.m.lin@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-19 12:07:27 +02:00
Takashi Iwai fdbc66266c ALSA: hda - Fix invalid capture mixers with some ALC268 models
The auto-mic clean-up patches caused regressions on some ALC268 models
that have no proper input_mux but with "Input Source" mixer elements.
Such a combination results in Oops when accessed.

[A reason why set_capture_mixer() isn't used in patch_alc268() is that
ALC268 codec have HDA_OUTPUT direction for capture volumes unlike other
codecs.  Thus it needs own definitions of capture elements.]

This patch fixes the issues:
- Add a capture mixer definition without input-source
- Use the new capture mixer appropriately

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-19 00:22:17 +02:00
Mark Brown b5ab887e6d ASoC: Add TLV information to WM8711
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 21:29:31 +01:00
Mark Brown 431f777177 ASoC: WM8711 minor cleanups
Coding style changes only.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 21:17:34 +01:00
Mark Brown 08aff8cd7a ASoC: Add SPI support to WM8711
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 21:15:14 +01:00
Mark Brown d97d2e35b9 ASoC: Factor out WM8711 cache I/O
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 21:12:30 +01:00
Mark Brown f72222c74b Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into upstream/wm8711 2009-08-18 20:59:01 +01:00
Mark Brown 318b0b8d90 ASoC: Update WM8711 to driver model registration method
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 20:57:33 +01:00
Mike Arthur bd6d417743 ASoC: Add WM8711 CODEC driver
The WM8711 or WM8711L (WM8711/L) is a low power stereo DAC with an
integrated headphone driver. The WM8711/L is designed specifically for
portable MP3 audio and speech players. The WM8711/L is also ideal for
MD, CD machines and DAT players.

Signed-off-by: Mike Arthur <Mike.Arthur@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 20:37:49 +01:00
Mark Brown 59ae07a580 ASoC: WM8993 digital mixing support
The WM8993 provides digital sidetone paths and also allows each
channel on the audio interface to be routed separtately to the
DACs and ADCs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 16:06:13 +01:00
Mark Brown 010ff26226 ASoC: Add input and output AIF widgets
Currently DAPM interfaces with the audio streams to and from the
processor at the DAC and ADC widgets. As the digital capabilities
of parts increases this is becoming a less and less able to meet
the needs of parts.

To meet the needs of these devices create new widgets interfacing
with the TDM bus but not integrated into any other functionality.
Audio can then be routed to and from these widgets using existing
routing widgets.

A slot number is provided in the definition but this is currently
not used yet. This is intended to support devices which can use
more than one TDM slot on a single interface.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 16:06:08 +01:00
Mark Brown d1a5e44b89 ASoC: Remove duplicate ADC/DAC widgets from wm_hubs.c
These need to be in the CODEC since the DAIs supported by the CODECs
aren't static.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 16:04:24 +01:00
Mark Brown b2472b1d4c ASoC: Reenable S3C64xx I2S support
Joonyoung Shim reports that S3C64xx I2S is working on the NCP boards so
allow it to be selected in Kconfig.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmciro.com>
2009-08-18 16:02:59 +01:00
Joonyoung Shim 0914b93f4f ASoC: Fix data format configuration for S3C64XX IISv2
The data format configuration for S3C64xx IISv2 was hardcoded for IISMOD
register. This patch changes to the defined values it.

And instead of bits 9 and 10 of IISMOD we should clear bits 13 and 14.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 16:02:36 +01:00
Mark Brown d3c9e9a139 ASoC: Implement TDM configuration for WM8993
Note that the number of slots used internally is specified in terms
of stereo slots while the external API works with mono slots.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 18:53:50 +01:00
Mark Brown 0182dcc52c ASoC: Fix WM8993 MCLK configuration for high frequency MCLKs
When used without the PLL we were accidentally clearing the MCLK/2
divider, resulting in a double rate SYSCLK when the divider should
have been used.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 18:53:44 +01:00
Russell King 29c08460d4 Merge branch 'next-s3c' of git://aeryn.fluff.org.uk/bjdooks/linux into devel-stable 2009-08-17 18:16:28 +01:00
Mark Brown 1ca04065c3 ASoC: Power speakers and headphones simultaneously
Speaker and headphone outputs do not need to be handled separately
since they can't be part of the same path.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 16:26:59 +01:00
Mark Brown b14b76a56e ASoC: Fix handling of bias levels for non-DAPM codecs
If the system doesn't have any DAPM widgets then we can't use their
state to check if the bias level for the codec should be up.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 12:57:59 +01:00
Shine Liu 0c093fb542 ASoC: UDA134X: Fix mistaken mute/unmute code
There is a mistake in current uda134x_mute function: mute_reg has been
changed in line 162 or line 164, so uda134x_write should write
"mute_reg" but not "mute_reg & ~(1<<2)" to
UDA134X_DATA010.

Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 12:56:57 +01:00
Clemens Ladisch 18dd0aa5af sound: snd_ctl_remove_user_ctl: prevent removal of kernel controls
Ensure that userspace can remove only user controls.  Controls created
by kernel drivers must not be removed because they might be referenced
in calls to snd_ctl_notify().

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:21 +02:00
Clemens Ladisch f217ac59b6 sound: snd_ctl_remove_unlocked_id: simplify user control counting
Move the decrementing of the user controls counter from
snd_ctl_elem_remove to snd_ctl_remove_unlocked_id; this saves the
separate locking of the controls semaphore, and therefore removes
a harmless race.

Since the purpose of the function is to operate on user controls (the
control being unlocked is just a prerequisite), rename it to
snd_ctl_remove_user_ctl.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:15 +02:00
Clemens Ladisch 317b80817f sound: snd_ctl_remove_unlocked_id: simplify error paths
Use a common exit path to release the mutex and to return a possible
error.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:06 +02:00
Clemens Ladisch 2a031aedf7 sound: snd_ctl_elem_add: fix value count check
Make sure that no user element that has no values can be added.

The check for count>1024 is not needed because the count is checked
later for the individual control types.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:00 +02:00
Janusz Krzysztofik 471e3dec3a ASoC: OMAP: Enhance OMAP1510 DMA progress software counter
Enhance period_index accuracy, particularly just before buffer rewind, by
making use of DMA interrupt status flags in addition to simply counting up
interrupts.

Created against linux-2.6.31-rc5.

Tested on Amstrad Delta.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 11:00:34 +01:00
Janusz Krzysztofik 64844a6ac8 ASoC: OMAP: Make use of DMA channel self linking on OMAP1510
Use newly implemented DMA channel self linking on OMAP1510 like on other OMAP
models. Remove unnecessary DMA transfer restart from interrupt handler
routine.

The interrupt routine used to maintain a period index, originally needed for
counting up periods up to a full buffer in order to restart the DMA transfer.
For some time, this counter is also used as a replacement for hardware DMA
progress counter that has been found unusable on OMAP1510 in case of playback.
Thus, the period index calculation cannot be omitted completely. However, the
accuracy of this counter can still suffer from missing DMA interrupts.

In order to work correctly, it requires patch 1 from this series also applied:
[RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510

Created against linux-2.6.31-rc5.

Tested on Amstrad Delta.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 10:59:59 +01:00
Mark Brown 1e97f50b70 ASoC: Factor out cache I/O from WM8974
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 12:15:10 +01:00
Mark Brown 37cfa1950e Merge branch 'wm8974-upstream' into for-2.6.32 2009-08-15 11:52:43 +01:00
Mark Brown 29e02cb3ff ASoC: Hook i.MX into build
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 11:37:30 +01:00
Mark Brown d555a552ae ASoC: Staticise unexported variables
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 11:36:49 +01:00
Mark Brown a2d512a978 ASoC: Remove unneeded i.MX dependency on SND
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 11:36:20 +01:00
Mark Brown 08229de4b4 Merge branch 'for-2.6.32' into mxc
Conflicts:
	sound/soc/Makefile
2009-08-15 11:20:44 +01:00
Takashi Iwai 7570ef1834 ALSA: hda - Add missing num_adc_nids definition for IDT92HD8xxx
The previous fix removed the definition of num_adc_nids wrongly, and
this resulted in the missing input-source control.  Now readded again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-15 11:57:53 +02:00
Barry Song 2a708137fd ASoC: delete -spi suffix in ad1938 and free private data while registers fail
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 17:53:02 +01:00
Peter Ujfalusi 9028935d75 ASoC: TWL4030: Fix for capture mixer strings
Change the strings related to capture in order to be
interpreted correctly by alsamixer and possible other
UI based mixer applications.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 17:52:59 +01:00
Mark Brown d91e9a7ab9 ARM: S3C24XX: Add platform device for AC97 controller
Move the definition of the "generic" IRQ in the process.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
2009-08-14 01:13:29 +01:00
Marek Vasut 4ac0478f2a ALSA: Allow passing platform_data for pxa2xx-ac97
This patch adds support for passing platform data to ac97 bus devices
from PXA2xx-AC97 driver..

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:37 +01:00
Chaithrika U S 30230f4cd7 ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM
There is one instance of McASP on DA850/OMAP-L138 SoC. This is
connected to TLV320AIC3106 codec for audio playback and capture.
This patch adds audio support on this platform. Some of the
structure prefix names which are common for DA830/OMAP-L137 EVM and
DA850/OMAP-L138 EVM have been renamed to da8xx from da830.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:36 +01:00
Chaithrika U S 517ee6cf69 ASoC: DaVinci: Add a DAI format to McASP driver
The patch adds a DAI format: Codec bit clock master and frame sync slave,
to the driver.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:35 +01:00
Chaithrika U S 6a99fb5fb8 ASoC: DaVinci: McASP driver enhacements
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO
support. This FIFO provides additional data buffering. It also provides
tolerance to variation in host/DMA controller response times.
The read and write FIFO sizes are 256 bytes each. If FIFO is enabled,
the DMA events from McASP are sent to the FIFO which in turn sends DMA requests
to the host CPU according to the thresholds programmed.
More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=
sprufm1&fileType=pdf

This patch adds support for FIFO configuration. The platform data has a
version field which differentiates the McASP on different SoCs.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:35 +01:00
Mark Brown a2342ae325 ASoC: Factor out shared code from WM8993
The WM8993 analogue control is shared with other devices in the same
product line.  Since this is a very substantial proportion of the
driver move the definitions of these controls into a new wm_hubs module
which allows them to be shared between the two.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:31 +01:00
Takashi Iwai 667067d898 ALSA: hda - Fix / clean up IDT92HD83xxx codec parser
A few improvements for IDT 92HD83xxx codec pareser:
- Remove unused / deprecated mixer-amp controls
- Handle d-mics as normal inputs since this codec has no separate
  MUXes for analog and digital
- Don't create duplicated controls for capture volumes with Mux
  capture volumes

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-13 18:14:42 +02:00
Takashi Iwai a6cd7a71fd Merge branch 'topic/hda-dmic-fix' into topic/hda 2009-08-13 18:14:02 +02:00
Mark Brown e9ade7f933 ASoC: Minor cleanups to AD1938 driver
- Build in SND_SOC_ALL_CODECS.
- Remove null suspend/resume stuff.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 15:19:42 +01:00
Barry Song 7eaae41ea5 new ad1836 codec driver based on asoc
There has been an ad1836 driver in sound/blackfin based on traditional alsa.
The new driver is based on asoc. The architecture of ad1836 codec driver is
very much like ad1938.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 15:18:53 +01:00
Peter Ujfalusi 9008adf9a9 ASoC: TWL4030: Introduce PGAs for outputs
Dynamically control and control only the needed output amplifier
muting/un-muting.

The original code was muting and un-muting the following output
amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time
regardless which pin is actually in use at the given moment.

Move these as separate PGA so only the needed amplifier will be touched.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 14:56:13 +01:00
Barry Song c4ff357ada ASoC: add output/input widgets in ad1938 to make dac/adc dynamic PM work
According to the function dapm_dac_check_power() in
sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any
output widget as sink. And according to dapm_adc_check_power(), adc
power can't be on/off stand-alone without any input widget as source. So
we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC
to hope their power can be managed dynamically.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 10:47:22 +01:00
Takashi Iwai 1c4bdf9be0 ALSA: hda - Enable line-out detection only with speakers
Enable line-out detection for IDT/STAC codecs only when speaker pins
exist.  In some cases, the speaker itself is identified as line-out,
and this confuses the situation.  Only the extra line-outs should do
auto-muting.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-13 08:27:38 +02:00
Tim Blechmann c18bc9b927 ALSA: hdsp - allow proc reporting with disconnected io box
the hdsp driver refuses to report any information via the proc
interface, if the io box is not connected. with this patch, the
content of the control and status registers is printed before the
iobox check.

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-12 18:21:30 +02:00
Mark Brown d2a382143b ASoC: Update AD1938 for new TDM slot API
It's only actually paying attention to the slot count anyway.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-12 14:30:33 +01:00
Takashi Iwai 8884be98bc Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Don't override ADC definitions for ALC codecs
  ALSA: hda - Add missing vmaster initialization for ALC269
2009-08-12 08:05:20 +02:00
Takashi Iwai 909a2607a5 Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: Add missing DRV_NAME definitions for fsl/* drivers
2009-08-12 08:05:19 +02:00
Herton Ronaldo Krzesinski 5908589f31 ALSA: hda - fix noise issue when recording from digital mic with alc268
With auto config model of alc268 realtek codec, it allows to select any
of possible available digital microphone inputs when only one is
available. For example, when only digital mic in nid 0x12 is available,
on second input source it will allow you to select unavailable digital
mic in nid 0x13. The problem is that selecting unavailable digital mic
creates a source of noise when recording (I'm not sure if this happens
on all machines with alc268 and only one digital mic input, but testing
on a quanta uw1 netbook a lot of noise is introduced in recording from
digital mic 0x12/first input source, when you select the unavailable
digital mic 0x13 for capture source 0x24 in the second input source in
mixer).

Then to avoid noise when recording from digital mic with auto model in
this case, prevent a digital mic input source to be selected if
microphone is not available.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-12 07:35:02 +02:00
Takashi Iwai 4f5d170620 ALSA: hda - Clean up init and setup hooks for Realtek codecs
Move static codes to setup from init_hook for each model.

Also, use the common auto-mic selection helper for devices that support
auto-mic selection.  They just need to set up ext_mic, int_mic and
auto_mic flag in the setup section.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 18:17:46 +02:00
Mark Brown e0026beac0 ASoC: Update WM9081 for tdm_slot() API change
Store the TDM slot width then if it's set use that rather than the
sample size to calculate BCLK. Leave imposing constraints to the
core (which should do this but doesn't yet) or machine driver.

Also allow 0 TDM slots to be configure (for use when disabling TDM).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-11 16:29:21 +01:00
Takashi Iwai e9c364c04f ALSA: hda - Add setup hook to ALC preset struct
Added setup hook to ALC preset struct to be called at in the parser
but not at each init callback.
This can be used for setting up the static pins, etc, while the
init hook should be used for updating the status again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 17:16:13 +02:00
Takashi Iwai 4d8e22e0f6 ALSA: hda - Add a white-list for MSI option
Created a white-list to enable MSI since some devices require MSI
explicitly due to BIOS/ACPI problems.  Simply using a quirk list.
As the first case, take HP Compaq CQ40.

Reference: Novell bnc#529971
	https://bugzilla.novell.com/show_bug.cgi?id=529971

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 14:25:46 +02:00
Mark Brown 1921bab217 Merge commit 'a5479e389e989acfeca9c32eeb0083d086202280' into for-2.6.32 2009-08-11 13:09:27 +01:00
Randy Dunlap 17244c24f9 ASoC: fix I2C build errors
Fix soc build errors when I2C is built as a loadable module:

(.text+0x5d26b): undefined reference to `i2c_master_send'
soc-cache.c:(.text+0x5d32d): undefined reference to `i2c_transfer'

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-11 10:47:25 +01:00
Takashi Iwai b59bdf3b0c ALSA: hda - Check connectivity for auto-mic of Realtek codecs
Some Realtek codecs don't provide the full connections for certain pins
from each ADC; e.g. ACL662/ALC272 gives only one of two digital-mic pins
for each ADC.  Thus, depending on the digital mic pin, the ADC/MUX to be
used has to be chosen properly.

This patch adds the check of the connectivity of pins at auto-mic mode.
If no proper connectivity is found, auto_mic flag is turned off to be
sure.

Also the mux_idx is determined during this check so it won't be checked
in the unsol event any more.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 09:47:30 +02:00
Takashi Iwai 52b5deefbb Merge branch 'fix/hda' into topic/hda 2009-08-11 08:47:38 +02:00
Takashi Iwai dd704698f5 ALSA: hda - Don't override ADC definitions for ALC codecs
ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition.  This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).

This patch avoids the overriding by adding the proper checks.

Reference: Novell bnc#529467
	https://bugzilla.novell.com/show_bug.cgi?id=529467

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 08:45:11 +02:00
Takashi Iwai f1e6d3c5cf ALSA: usb-audio - Fix types taken in min()
Fix the compile warning due to different integer types used in min():
  sound/usb/usbaudio.c: In function 'init_substream_urbs':
  sound/usb/usbaudio.c:1087: warning: comparison of distinct pointer types lacks a cast

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 08:16:15 +02:00
Takashi Iwai 2a22d3f812 ALSA: hda - Use only one capture stream for auto-mic
When the auto-mic feature is enabled, we should support only one
capture stream.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 18:56:05 +02:00
Takashi Iwai 6c81949227 ALSA: hda - Add auto-mic support for Realtek codecs
Added the support for automatic mic selection via plugging for
Realtek codecs (in auto-probing mode).  The auto-mic mode is enabled
only when one internal mic and one external mic are present.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 18:47:44 +02:00
Tejun Heo 93fe4483e6 sound: make OSS device number claiming optional and schedule its removal
If any OSS support is enabled, regardless of built-in or module,
sound_core claims full OSS major number (that is, the old 0-255
region) to trap open attempts and request sound modules using custom
module aliases.  This feature is redundant as chrdev already has such
mechanism.  This preemptive claiming prevents alternative OSS
implementation.

The custom module aliases are scheduled to be removed and the previous
patch made soundcore emit the standard chrdev aliases too to help
transition.

This patch schedule the feature for removal in a year and makes it
optional so that developers and distros can try new things in the
meantime without rebuilding the kernel.  The pre-claiming can be
turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel parameter
soundcore.preclaim_oss.

As this allows sound minors to be individually grabbed by other users,
this patch updates sound_insert_unit() such that if registering
individual device region fails, it tries the next available slot.

For details on removal plan, please read the entry added by this patch
in feature-removal-schedule.txt .

Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:59:36 +02:00
Mark Brown e0c48a18f7 ASoC: Drop unneeded declaration of removed wm8731 SPI write function
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-10 12:59:29 +01:00
Tejun Heo 0a848680a8 sound: request char-major-* module aliases for missing OSS devices
Till now missing OSS devices emitted sound-slot/service-* module
alises instead of the standard char-major-* if a missing device number
is opened if soundcore is loaded.  The custom module aliases don't
have any inherent benefit than backward compatibility.

sound-slot/service-* module aliases is scheduled to be removed and to
help the transition this patch makes soundcore emit the standard
module alises along with the custom ones.

Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:59:26 +02:00
Clemens Ladisch 5e8e7c3853 sound: fix OSS MIDI output data loss
In the 2.1.6 kernel, the output loop in midi_poll() was changed to
enable interrupts during the outputc() call.  Unfortunately, the check
whether the device has accepted the current byte ("ok") was moved behind
the code that removes the byte from the output queue, so one byte would
be lost every time the hardware FIFO is full.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:15:43 +02:00
Clemens Ladisch 6e2efaacb3 sound: ymfpci: increase timer resolution to 96 kHz
Allow the interval timer to be programmed with its full 96 kHz
precision.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:14:46 +02:00
Clemens Ladisch 765e8db078 sound: usb-audio: do not make URBs longer than sync packet interval
Using more packets in one URB do avoid interrupts does not make sense
when we have a sync pipe whose packets generate interrupts more often.
Therefore, limit the URB size to the synchronization packet interval.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:13:56 +02:00
Takashi Iwai d5c9c8912a Merge branch 'fix/hda' into topic/hda 2009-08-10 11:58:09 +02:00
Takashi Iwai 100d5eb36b ALSA: hda - Add missing vmaster initialization for ALC269
Without the initialization of vmaster NID, the dB information got
confused for ALC269 codec.

Reference: Novell bnc#527361
	https://bugzilla.novell.com/show_bug.cgi?id=527361

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-08-10 11:57:05 +02:00
Takashi Iwai da2a2aaa8e ALSA: hda - Fix Oops due to STAC/IDT auto-mic changes
The previous auto-mic patch for STAC/IDT codecs causes the Oops on
machines without digital mic pins.  This patch fixes the problem.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 07:44:09 +02:00
Mark Brown 35b1207b34 ASoC: Convert WM8776 to use factored out register cache code
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-08 10:37:33 +01:00
Chaithrika U S 7ae5945f0c ASoC: DaVinci: Support Audio on DA830 EVM
Add support for audio on DA830 EVM- here McASP1 is interfaced to
TLV320AIC3106 codec.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-08 09:12:54 +01:00
Uwe Kleine-König dbe9ea6e79 ASoC: s3c2443-ac97: convert semaphore to mutex
This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig):

	  CC [M]  sound/soc/s3c24xx/s3c2443-ac97.o
	sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX'
	sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration
	sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read':
	sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function)
	sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once
	sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.)
	sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write':
	sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function)

Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-08 08:50:13 +01:00
Takashi Iwai afc5e65245 ASoC: Add missing DRV_NAME definitions for fsl/* drivers
Module builds are broken due to missing DRV_NAME for
efika-audio-fabric and pcm030-audio-fabric.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-07 16:33:53 +02:00
Janusz Krzysztofik b7b8f9bf0c TTY/ASoC: Rename N_AMSDELTA line discipline to N_V253
The patch changes the line discipline name registered in include/linux/tty.h
and updates the ams-delta machine driver to use it.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-07 11:48:02 +01:00
Mark Brown 06cddefc1f Merge branch 'reg-cache' into for-2.6.32 2009-08-07 11:43:58 +01:00
Mark Brown b9b5cc26d0 Merge branch 'for-2.6.31' into for-2.6.32 2009-08-07 11:42:01 +01:00
Troy Kisky 6a90d536fe ASoC: DaVinci: pcm, constrain buffer size to multiple of period
The dma setup code assumes that the buffer size is a multiple
of the period size.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-07 11:38:29 +01:00
Troy Kisky 9bb7415056 ASoC: DaVinci: i2s: don't bounce through rtd to get dai
dai is a parameter to the functions, so use it instead of
looking it up.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-07 11:38:29 +01:00
Jarkko Nikula c12abc012e ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/stop
Simultaneous audio playback and capture on OMAP1510 can cause that second
stream is stalled if there is enough delay between startup of the audio
streams.

Current implementation of the omap_mcbsp_start is starting both transmitter
and receiver at the same time and it is called only for firstly started
audio stream from the OMAP McBSP based ASoC DAI driver.

Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is
missed if there is no DMA transfer set up at that time when the first word
after McBSP startup is transmitted. The problem hasn't noted before since
later OMAPs are using level sensitive DMA request lines.

Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by
allowing to start and stop individually McBSP transmitter and receiver
logics. Then call those functions individually for both audio playback
and capture streams. This ensures that DMA transfer is setup before
transmitter or receiver is started.

Thanks to Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> for detailed problem
analysis and Peter Ujfalusi <peter.ujfalusi@nokia.com> for info about DMA
request line behavior differences between the OMAP generations.

Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-07 10:57:42 +01:00
Daniel Ribeiro a5479e389e ASoC: change set_tdm_slot api to allow slot_width override.
Extend set_tdm_slot to allow the user to arbitrarily set the frame width
and active TX/RX slots.

Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c
still doesn't handle the slot_width override.

While being there, correct an incorrect use of SlotsPerFrm(7) use in
bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ).

(this series is meant for Mark's for-2.6.32 branch)

Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-06 15:52:24 +01:00
Janusz Krzysztofik 9029bb316b ASoC: CX20442: simplify codec controller usage
This patch is a workaround for the problem of several subsequent control
statements not being applied correctly to the codec controller (modem).

In order to follow the hook switch state change from handset to handsfree
while
in full duplex mode, two consecutive +VLS control commands were sent to the
modem. The first one was M1 (microphone only), the seconds one was M1S1 (both
microphone and speaker). As there was no real modem handshaking procedure
implemented, neither in the codec nor in the machine driver part of the line
discipline, the modem was having the second command missed.

Since a possibility to switch to microphone only mode (and speaker only mode
as well) seams of no value, I have modified the code to issue single M1S1
command only for any of those cases.

Tested on my Amstrad Delta.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-06 11:36:46 +01:00
Janusz Krzysztofik 4977b03e3d ASoC: CX20442: add some debugging
This patch adds debugging statement that can help in tracing
how the driver is trying to control the codec device.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-06 11:36:45 +01:00
Mark Brown 924914ee95 ASoC: Add WM8776 CODEC driver
The WM8776 is a high performance, stereo audio CODEC with five channel
input selector. The WM8776 is ideal for surround sound processing
applications for home hi-fi, DVD-RW and other audio visual equipment.

This driver implements support for most WM8776 features - currently the
ADC automatic level control/limiter functionality is omitted.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-06 11:36:45 +01:00
Takashi Iwai 1972d02504 ALSA: hda - Add quirks for some HP laptops
The new HP laptops have PCI SSID 103c:701x and requires model=hp-dv5.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-06 08:44:43 +02:00
javier Martin fbb474deda ASoC: Fix review issues in i.MX2x PCM driver
Signed-off-by: javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:31:55 +01:00
javier Martin 2ccafed43a ASoC: add machine driver for i.mx27_visstrim_m10 board
This adds support for i.mx27_visstrim_sm10 board machine driver which
uses an i.mx27 processor plus a wm8974 codec.

It has been tested on a visstrim_sm10 board.

Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:31:54 +01:00
javier Martin 9d8bc2968c ASoC: add DAI platform ssi driver for MXC
This adds support for DAI platform for the SSI present in MXC platforms.

It currently does not support i.MX3, the only thing necessary to do
this is to export DMA data for i.MX3 interface which I haven't done
because I don't have a i.MX3 based board available.

It has been tested on i.MX27 board.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:31:54 +01:00
javier Martin fd6a6394d7 ASoC: add DMA platform driver for MX1x and MX2x
This adds support for DMA platform valid for i.MX1 and i.MX2 platforms.

This is not valid for i.MX3 since it doesn't share the same DMA
interface than i.MX1 and i.MX2.

It has been tested on i.MX27 board.

Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:31:54 +01:00
Daniel Mack 15b5bdaeeb ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
Power management for the cs4270 codec is currently implemented as part
of the i2c_driver struct. The disadvantage of doing it this way is that
the callbacks registered in the snd_soc_card struct are called _before_
the codec's callbacks.

That doesn't work, because the snd_soc_card callbacks will most likely
switch down the codec's power domains or pull the reset GPIOs, and
hence make the i2c communication bail out.

Fix this by binding the suspend and resume code to the
snd_soc_codec_device driver model and let the I2C functions only call
the SoC core function for resume and suspend, which do nothing currently
but will do later.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:13:49 +01:00
John Bonesio b0a2712ffd ASoC: MPC5200: Support for buffer wrap around
The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that
s->runtime->control->appl_ptr can wrap around to the beginning of the
buffer. This change fixes this problem.

Signed-off-by: John Bonesio <bones@secretlab.ca>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 22:10:14 +01:00
Mark Brown 4bc4c9a5f5 ASoC: Existing S3C24xx AC97 drivers should depend on S3C24xx
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-05 17:15:04 +01:00
Linus Torvalds 6ce90c430b Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Read buffer overflow
  ALSA: hda: Correct EAPD for Dell Inspiron 1525
  ALSA: hda: warn on spurious response
  ALSA: hda: remember last command for each codec
  ALSA: hda: read CORBWP inside reg_lock
  ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
  ALSA: hda: take cmd_mutex in probe_codec()
  ALSA: hda: track CIRB/CORB command/response states for each codec
  ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
2009-08-04 15:39:55 -07:00
Takashi Iwai 16ffe32c77 ALSA: hda - Fix line-out jack handling with STAC/IDT codec
When the line-out jack is plugged/unplugged, the driver needs to check
the headphone plug, not only the line-out jack itself.  Otherwise the
headphone or the speaker may be wrongly muted/unmuted.

As a result, both STAC_HP_EVENT and STAC_LO_EVENT need to call the
same function, stac92xx_hp_detect().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-04 13:40:54 +02:00
Mark Brown 27ded041f0 ASoC: Factor out 7 bit register 9 bit data SPI write
This converts all the Wolfson drivers using this format (the only devices
that do) except WM8753 to use it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:16 +01:00
Mark Brown 8d50e447d1 ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:16 +01:00
Mark Brown afa2f1066e ASoC: Factor out I2C 8 bit address 16 bit data I/O
As part of this refactoring the type of the CODEC hw_read operation
is changed to match the regular read operation.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:15 +01:00
Mark Brown 7084a42b96 ASoC: Add I/O control bus information to factored out cache setup
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.

Initially just use this to factor out hw_write_t for I2C.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-03 16:59:09 +01:00
Takashi Iwai 15cfa2b3db ALSA: hda - Fix line-out jack detection
The commit fefd67f31e
    ALSA: hda - Add line-out jack detection on IDT/STAC codecs
enabled wrong pins for jack detections.  Fixed to the correct ones.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 14:23:33 +02:00
Lubomir Rintel 51840409b6 ALSA: pcm - Tell user that stream to be rewound is suspended
Return STRPIPE instead of EBADF when userspace attempts to rewind
of forward a stream that was suspended in meanwhile, so that it
can be recovered by snd_pcm_recover().

This was causing Pulseaudio to unload the ALSA sink module under a race
condition when it attempted to rewind the stream right after resume from
suspend, before writing to the stream which would cause it to revive the
stream otherwise. Tested to work with Pulseaudio patched to attempt to
snd_pcm_recover() upon receiving an error from snd_pcm_rewind().

Signed-off-by: Lubomir Rintel <lkundrak@v3.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:57:00 +02:00
Takashi Iwai e44d4e4cee Merge branch 'fix/hda' into topic/hda 2009-08-03 08:37:17 +02:00
Wu Fengguang 559059b27f ALSA: hda: add IbexPeak/Clarkdale HDMI model with static cvt/pin number
The new IbexPeak HDMI codec has 3 pin nodes and 2 converter nodes.
Here we assume only the first ones will be used.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:35:05 +02:00
Roel Kluin 4b35d2ca23 ALSA: hda - Read buffer overflow
Check whether index is within bounds before testing the element.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:34:06 +02:00
Chengu Wang 84d3dc200f ALSA: hda: Correct EAPD for Dell Inspiron 1525
The commit 24918b61b5 statically changes
the model from dell-bios to dell-3stack to solve the sound decreasing
regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another
problem that the 2nd headphone jack doesn't work
(https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think
the commit 249**2dc is just a workaround. I would like to give a true solution
here.

The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and
the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as
GPIO2. This patch changes EAPD to GPIO0 to solve the problem.

Signed-off-by: Chengu Wang <wangchengu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:30:56 +02:00
Wu Fengguang e310bb0646 ALSA: hda: warn on spurious response
To help disclose hardware bugs.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:27:53 +02:00
Wu Fengguang feb273404f ALSA: hda: remember last command for each codec
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:27:39 +02:00
Wu Fengguang c32649feb4 ALSA: hda: read CORBWP inside reg_lock
This converts the last CORBWP access outside of reg_lock.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:26:55 +02:00
Wu Fengguang cdb1fbf231 ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
Just for safety.  azx_init_cmd_io() and azx_free_cmd_io() may be
called when switching to single command mode.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:26:42 +02:00
Wu Fengguang a678cdee25 ALSA: hda: take cmd_mutex in probe_codec()
Now that each codec will have its own module, it is possible
for the user to load one codec while another one is running.

So cmd_mutex would be a safe addition to probe_codec().

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:26:23 +02:00
Wu Fengguang deadff1665 ALSA: hda: track CIRB/CORB command/response states for each codec
Recently we hit a bug in our dev board, whose HDMI codec#3 may emit
redundant/spurious responses, which were then taken as responses to
command for another onboard Realtek codec#2, and mess up both codecs.

Extend the azx_rb.cmds and azx_rb.res to array and track each codec's
commands/responses separately. This helps keep good codec safe from
broken ones.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:26:13 +02:00
Takashi Iwai ce577e8cf5 ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.

Reference: Novell bnc#526325
	https://bugzilla.novell.com/show_bug.cgi?id=526325

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:23:52 +02:00
Janusz Krzysztofik 6d7f68a1ea ASoC: add support for Amstrad E3 (Delta) machine
This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC.

Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.

Depends on:
1) latest version of the CX20442 codec driver that exposes v253_ops
   structure[1],
2) patch 2/3 form this series: TTY: Add definition of a new line
   discipline required by Amstrad E3 (Delta) ASoC driver[2].

CPU DAI parameters best matching the codec DAI has been selected out
empirically for best user experience.

Board specific audio function control (with related DAPM widgets) has been
modeled after empirically discovered codec capabilities.

Unlike other ASoC machine drivers, this one makes use of a codec provided line
discipline that is required for talking to a modem chip that can control the
codec behavoiur. As the line discipline operations must call board specific
bits as well, the machine driver registers its own line discipline ops, not
the codec provided, and then calls those codec provided from inside its own
callbacks.
If some kind of a glue, like a bus over a tty, exsited that could help in
runtime detection of a modem (bus adapter) over a more generic line discipline
(bus driver)[3], the line discipline code could be probably designed in a
more generic way.

In order to work at all, this driver requires a working McBSP1. On OMAP1510
based machines (not sure if other OMAP1 variants as well), where McBSP1 is a
DSP public peripheral, that means the kernel must provide basic DSP support,
ie. omap_dsp_init(), in order to power up the DSP. This used to be included in
linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c.
Without that, the driver would not work, ie. not shift in/out any bits over
the CPU DAI[4]. This limitation is not board, but CPU specific, and may apply
to other code that makes use of McBSP1/McBSP3 on affected machines. I provide
an extra patch (4/3) as a temporary solution.

To work correctly in playback mode, this driver requires my prevoiusly
submitted patch that corrects pcm pointer calculation for OMAP1510 based
machines[5] (already included in linux-2.6.31-rc3).

To support codec controls, this driver requires my previously submitted patch
that adds support for modem found on Amstrad Delta[6].

[1] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019780.html
[2] http://www.spinics.net/lists/linux-serial/msg01862.html
[3] http://www.spinics.net/lists/linux-serial/msg01856.html
[4] http://www.spinics.net/lists/linux-omap/msg15114.html
[5] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.html
[6] http://www.spinics.net/lists/linux-omap/msg15432.html

Credits to:
Mark Underwood - for his initial, omap-alsa based sound driver for
this machine,
Mark Brown - for his help, patience and excellent subsytem maintainer support.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-31 22:38:44 +01:00
Janusz Krzysztofik ad120dae12 ASoC: CX20442: push down machine independent line discipline bits
This corrected patch adds machine independent line discipline code, prevoiusly
exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX20442
codec driver. The code can be used as a standalone line discipline, or as a
set of codec specific functions called from machine's line discipline
callbacks. Anyway, the line discipline itself must be registered by a machine
driver.

Applies on top of the followup to my initial driver version:
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.html

Suggested by ASoC manintainer Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-31 22:38:44 +01:00
Lars-Peter Clausen b8e22c1fe3 ASoC: jack: Fix race in snd_soc_jack_add_gpios
The irq can fire as soon as it has been requested, thus all fields accessed
from within the irq handler must be initialized prior to requesting the irq.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-31 22:38:43 +01:00
Mark Brown 77ee09c67e ASoC: Allow CODECs to flag invalid registers
This helps CODECs with sparse register maps work better with the
register cache display interface.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-31 18:54:48 +01:00
Takashi Iwai ec86fe5209 Merge branch 'fix/oss' into for-linus
* fix/oss:
  sound: mpu401.c: Buffer overflow
  sound: aedsp16: Buffer overflow
2009-07-31 10:17:45 +02:00
Takashi Iwai d62e345f14 Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: sound/aoa: Add kmalloc NULL tests
2009-07-31 10:17:44 +02:00
Takashi Iwai 6280b61af5 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Increase PCM stream name buf in patch_realtek.c
  ALSA: hda: fix out-of-bound hdmi_eld.sad[] write
  ALSA: hda - Add quirk for Dell Studio 1555
2009-07-31 10:17:42 +02:00
Julia Lawall f065fabc86 ALSA: sound/aoa: Add kmalloc NULL tests
Check that the result of kzalloc is not NULL before a dereference.

The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@@
expression *x;
identifier f;
constant char *C;
@@

x = \(kmalloc\|kcalloc\|kzalloc\)(...);
... when != x == NULL
    when != x != NULL
    when != (x || ...)
(
kfree(x)
|
f(...,C,...,x,...)
|
*f(...,x,...)
|
*x->f
)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-31 10:14:58 +02:00
Takashi Iwai aa563af763 ALSA: hda - Increase PCM stream name buf in patch_realtek.c
The name buf with size 16 is too short for some codec names, e.g.
truncated like "ALC861-VD Analo".  Now the size is doubled.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-31 10:06:34 +02:00
Takashi Iwai 03cb2dafcb Merge branch 'topic/hda-cirrus' into topic/hda 2009-07-30 18:09:04 +02:00
Takashi Iwai d195658bd7 Merge branch 'fix/hda' into topic/hda 2009-07-30 18:08:54 +02:00
Takashi Iwai fefd67f31e ALSA: hda - Add line-out jack detection on IDT/STAC codecs
Add the automatic mute of speakers via line-out jack plugging on
STAC/IDT codecs.  The feature is enabled when the HP detect is present.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 18:08:46 +02:00
Stelian Pop 3a38516750 ALSA: hda - Enable HP output with Macbook Pro 5, 5
The patch below, to be applied on the latest sound-unstable-2.6.git,
enables headphones output on my MacBookPro 5,5, together with the
automuting feature.

Here is the exact soundcard id:
	Vendor Id: 0x10134206
	Subsystem Id: 0x106b4d00
	Revision Id: 0x100301

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 14:57:25 +02:00
Mark Brown a1daf67d72 Merge branch 'gta02-audio' into for-2.6.32 2009-07-30 13:21:38 +01:00
Takashi Iwai 5207e10ed4 ALSA: hda - Integrate Digital Input Source to Input Source
STAC/IDT codecs provide both "Input Source" and "Digital Input Source"
controls to choose the analog input source and the digital input source.
But this is far user-unfriendly.

This patch merges the input source selections into one "Input Source"
control.  To have separate digital and analog input source controls,
you can pass "separate_dmux = 1 " hint string.

At the same time, this patch gets rid of analog mixer stuff that was
already disabled in previous patches.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 13:09:08 +02:00
Takashi Iwai bf677bd8fb ALSA: hda - Fix typos of Capture controls.
The commit 6479c63188
    ALSA: hda - Create Capture controls dynamically
introduced typos of "Capture".  Fixed now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 09:24:29 +02:00
Wu Fengguang 6732bd0d15 ALSA: hda: add HP automute support to Intel ALC889/ALC889A models
It auto mutes all 8-channel outputs at rear panel when
the front panel headphone is connected.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 09:19:14 +02:00
Wu Fengguang dd7714c922 ALSA: hda: add 2-channel mode to Intel ALC889/ALC889A models
This 2-channel mode is useful in that it will broadcast
a 2-channel audio stream to all front/side/... ports.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-30 08:53:56 +02:00
Barry Song 3a39f832a5 ASoC: Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm dai
1. fix "line over 80 characters" checkpatch warnings
2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instead
3. fix typos

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-29 21:31:53 +01:00
Lars-Peter Clausen 82c4362ee3 ASoC: neo1973_gta02_wm8753: Replace deprecated s3c_gpio calls with gpiolib
With the s3c platform has implementing gpiolib support the s3c_gpio api has been
deprecated.
This patch gets rid of all s3c_gpio calls and replaces them by using gpiolib.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-29 21:29:33 +01:00
Lars-Peter Clausen 69331fbdee ASoC: neo1973_gta02_wm8753: Replace snd_soc_cnew with snd_soc_add_controls.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-29 21:29:33 +01:00
Takashi Iwai 71443b0b74 ALSA: hda - No analog mix input source as default for IDT92HD71bxx
The analog mix is disabled now as default (unless "analog_mixer" hint
is given), so it shoudn't appear in the digital input source as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 18:41:29 +02:00
Takashi Iwai 15b4f296fc ALSA: hda - Add missing DMUX initialization for auto-mic with STAC/IDT
Added the missing initialization of DMUX connection (to analog input)
for auto-mic mode with STAC/IDT codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 16:32:55 +02:00
Takashi Iwai 26a2798053 ALSA: hda - Remove static connection in IDT 92HD71bxx
We don't need any more static connection to the port F (which is often
used for docking stations) since its connection is done dynamically via
DAC assignment now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 16:30:14 +02:00
Roel Kluin a987004fbc sound: mpu401.c: Buffer overflow
mpu_synth_info[m].name is a char[30], and the minimum length of the data
written by sprintf is 31 bytes including terminating null.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:41:24 +02:00
Roel Kluin c45ec06c74 sound: aedsp16: Buffer overflow
DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes
including terminating null.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:37:12 +02:00
Roel Kluin 78735cffc2 ALSA: hda: fix out-of-bound hdmi_eld.sad[] write
e->sad[] is declared with size ELD_MAX_SAD=16, but the guard
allows range 0-31.

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:35:20 +02:00
Takashi Iwai 3d21d3f7e7 ALSA: hda - Support auto-mic switching with IDT/STAC codec
Support the automatic mic-switching with some devices with IDT/STAC
codecs.  The condition is that the device has only two inputs, one
for an external mic and one for an internal mic.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:32:56 +02:00
Takashi Iwai 62558ce157 ALSA: hda - Avoid overwrite of jack events with STAC/IDT
Since only one event can be associated to a (pin) widget, it's safer
to avoid the multiple mapping.  This patch fixes the behavior of the
STAC/IDT codec driver.

Now stac_get_event() doesn't take the type argument but simply returns
the first hit element.  Then enable_pin_detect() checks the validity
of the type, and returns non-zero only if a valid entry.  The caller
can call stac_issue_unsol_event() after checking the return value.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-29 14:28:37 +02:00
Barry Song c8489c3ed3 ASoC: board driver to connect bf5xx with ad1938
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-28 22:16:02 +01:00
Barry Song 01e2ab207c ASoC: blackfin I2S(TDM mode) CPU DAI driver
The I2S DAI driver for blackfin SPORT, but works in TDM mode.
I2S is not a special case of TDM with only left and right two slots for
SPORT interface. I2S coordinates with TDM in SPORT, but not a part of
TDM. TDM require different hardware configuration with I2S, not only
different slot number.  One is "Stereo Serial Operation" mode of SPORT,
the other one is "Multichannel Operation" mode. They are incompatible
at the same time.
Hardware and DMA description and data transfer flow are much different
for I2S and TDM. Merging them as a whole will be very ugly and difficult
to maintain.
So we don't define a new DAI type, but give two DAI instances for standard
I2S and TDM, both in I2S-family DAI type. The TDM instance still uses the
I2S-family DAI type.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-28 22:15:27 +01:00
Janusz Krzysztofik b84eab08a6 ASoC: CX20442: fix issues pointed out by subsystem maintainer
The patch fixes some checkpatch identified issues and adds a comment about
line discipline interaction to my driver code, as requested by Mark on my
inital submission (thank you Mark for applying my imperfect patch anyway).
It also fixes MODULE_ALIAS mismatch as used in my machine driver.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-28 22:15:23 +01:00
Takashi Iwai 50c62f068e ALSA: hda - Don't create analog mixer for IDT92HD71bxx
The analog mixer unit on IDT 92HD71Bxx codecs is almost useless
since we use only the direct connections from DAC to pin.

Remove the controls to avoid unneeded confusion as default now.
This can be still back via "analog_mixer = 1" hint.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 18:28:27 +02:00
Takashi Iwai 6479c63188 ALSA: hda - Create Capture controls dynamically
Instead of static snd_kcontrol_new arrays, create "Capture Volume"
and "Capture Switch" controls dynamically based on the mixer attr
values (made via HDA_COMPOSE_AMP_VAL()).
This reduces the code size and gives more flexibility to change
the number of controls later.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 18:25:03 +02:00
Takashi Iwai 4417932315 ALSA: hda - Don't create unneeded digital input source for IDT 92HD71x
The current driver creates always the digital input source mixer
elements for IDT 92HD71x codecs no matter whether digital mics are
present.  This patch adds the proper check to avoid the creation of
these controls if unnecessary.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 17:03:49 +02:00
Takashi Iwai 9a11f1aa8e ALSA: hda - Reword information messages for BIOS auto-probing mode
The sentense "Unknown model for xxx, ..." makes people too nervous
and drives them to a direction to a wrong "fix" by giving any
mismatching model option.

Let's rephrase the messages to be more nice and easy (at least that
won't make people suspect conspiracies).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 16:06:56 +02:00
Takashi Iwai 626f5cefc6 ALSA: hda - Add quirk for Dell Studio 1555
Added a quirk entry for Dell Studio 1555.

Reference: Novell bnc#525244
	https://bugzilla.novell.com/show_bug.cgi?id=525244

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-28 00:54:39 +02:00
Takashi Iwai 1ba7a7c650 ALSA: hda - Add exception for volume-knob in snd_hda_get_connections()
Volume-knob widgets may have connections even if they have no CONN_LIST
cap bit.  Allow the query exceptionally in snd_hda_get_connections().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-27 12:56:26 +02:00
Takashi Iwai a22d543a95 ALSA: hda - Introduce get_wcaps_type() macro
Add a helper macro to retrieve the widget type from wiget cap bits.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-27 12:54:26 +02:00
Takashi Iwai 57e4a5c4f8 Merge branch 'fix/usb-audio' into for-linus
* fix/usb-audio:
  ALSA: usb-audio - Volume control quirk for QuickCam E 3500
2009-07-26 11:07:08 +02:00
Takashi Iwai b88158846f Merge branch 'fix/pcm-hwptr' into for-linus
* fix/pcm-hwptr:
  ALSA: pcm - Fix hwptr buffer-size overlap bug
  ALSA: pcm - Fix warnings in debug loggings
  ALSA: pcm - Add logging of hwptr updates and interrupt updates
  ALSA: pcm - Fix regressions with VMware
2009-07-26 11:07:07 +02:00
Takashi Iwai de5d674c02 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Fix mute control with some ALC262 models
  ALSA: hda - Restore GPIO1 properly at resume with AD1984A
  ALSA: hda - Use snprintf() to be safer
2009-07-26 11:07:06 +02:00
Takashi Iwai f35e2965b2 Merge branch 'fix/ctxfi' into for-linus
* fix/ctxfi:
  ALSA: ctxfi - Fix uninitialized error checks
2009-07-26 11:07:05 +02:00
Takashi Iwai 29769d533b Merge branch 'fix/caiaq' into for-linus
* fix/caiaq:
  ALSA: snd_usb_caiaq: add support for Audio2DJ
2009-07-26 11:07:04 +02:00
Takashi Iwai 7679d5c65b Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: tlv320aic3x: Enable PLL when not bypassed
2009-07-26 11:07:03 +02:00
Takashi Iwai a3daf68931 Merge branch 'fix/hda' into topic/hda 2009-07-24 16:54:59 +02:00
Takashi Iwai 8de56b7deb ALSA: hda - Fix mute control with some ALC262 models
The master mute switch is wrongly implemented as checking the pointer
instead of its value, thus it can be never muted.  This patch fixes
the issue.

Reference: Novell bnc#404873
	https://bugzilla.novell.com/show_bug.cgi?id=404873

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-07-24 16:54:31 +02:00
Marek Vasut 4ce2f2fe61 ASoC: Switch palm27x-asoc to jack detection api
This patch removes the old method of jack detection from palm27x-asoc
driver and adds jack detection api. It also removes some other (now)
useless stuff from the driver and corrects pin configuration for the
codec.

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-24 11:29:03 +01:00
Janusz Krzysztofik 178b699c25 ASoC: Jack handling enhancements as suggested by subsystem maintainer
The patch adds a few small enhancements to the ASoC jack handling, as
suggested by Mark in his comments to my Amstrad Delta driver, and a few fixes
for related bugs found while learning Mark's code and testing results.

Enhancements:
1. Update status of an ASoC jack while associating it with new gpios.
2. Really update DAPM pins while associating them with an ASoC jack.
3. Export ASoC jack gpios over gpiolib sysfs for diagnostic purposes.

Fixes:
1. Apply mask on jack status report before using it, just for case.
2. While updating jack associated DAPM pins, use full resulting jack status,
   not the status report passed as an argument.

Created and tested on linux-2.6.31-rc3

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-24 11:28:59 +01:00
Daniel Mack b30c494773 ALSA: snd_usb_caiaq: add support for Audio2DJ
This adds support for Native Instrument's freshly announced Audio2DJ
sound device hardware. Version number bumped to 1.3.19.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 16:31:58 +02:00
Takashi Iwai 947ca210f1 ALSA: pcm - Fix hwptr buffer-size overlap bug
The fix 79452f0a28 introduced another
bug due to the missing offset for the overlapped hwptr.
When the hwptr goes back to zero, the delta value has to be corrected
with the buffer size.  Otherwise this causes looping sounds.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 16:21:08 +02:00
Takashi Iwai 8935064043 ALSA: pcm - Fix warnings in debug loggings
Add proper cast.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 14:28:37 +02:00
Marek Vasut 474828a40f ALSA: Allow passing platform_data to devices attached to AC97 bus
This patch allows passing platform_data to devices attached to AC97 bus
(like touchscreens, battery measurement chips ...).

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 11:30:56 +01:00
Joonyoung Shim a7569afa8b ASoC: MAX9877: fix write operation for register
The MAX9877 needs an address of start register when we write values to
registers through i2c_master_send(), but the code for this was missed in
max9877_write_regs().

If the value of control is 0 in the max9877_set_out_mode(), the value is
not increased to 1, but actually the value to write to the register
should be 1.
And the register bits for out_mode and osc_mode should be cleared before
writing.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 11:08:30 +01:00
Janusz Krzysztofik 459dc35233 ASoC: Add support for Conexant CX20442-11 voice modem codec
This patch adds support for Conexant CX20442-11 voice modem codec, suitable
for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related
sound card driver will follow.

This codec is an optional part of the Conexant SmartV three chip modem design.
As such, documentation for its proprietary digital audio interface is not
available. However, on Amstrad Delta board, thanks to Mark Underwood who
created an initial, omap-alsa based sound driver a few years ago[1], the codec
has been discovered to be accessible not only from the modem side, but also
over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound
card that can access the codec DAI directly. The DAI configuration parameters
(sample rate and format, number of channels) has been selected out empirically
for best user experience.

The codec analogue interface consists of two pairs of analogue I/O pins:
speakerphone interface or telephone handset/headset interface. Furthermore, it
seams to provide two operation modes for speakerphone I/O: standard and
advanced, with automatic gain control and echo cancelation. Even if the codec
control interface is unknown and not available, all those interfaces and modes
can be selected over the modem chip using V.253 commands. The driver is able
to issue necessary commands over a suitable hw_write function if provided by a
sound card driver. Otherwise, the codec can be controlled over the modem from
userspace while inactive.

Even if nothig is known about the codec internal power management
capabilities, DAPM widgets has been used to model the codec audio map.
Automatically performed powering up/down of those virtual widgets results in
corresponding V.253 commands being issued.

Some driver features/oddities may be board specific, but I have no way to
verify that with any board other than Amstrad Delta.

[1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html

Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 11:08:29 +01:00
Jaroslav Kysela 87a8c3702e ALSA: hda - Add better Intel IbexPeak platform support
Here are the new sound enabling patches for IbexPeak.

Summary of tested features:

  - playback
    - Front Headphone: OK
    - 8 channel audio: Front/Rear/CLFE/Side all OK

  - recording
    - Front Mic/Rear Mic: both OK
      (front/rear/line mics are selectable in the "Input source" alsamixer control)
    - Line In: not working
      (in 6ch mode, its amp/mute, direction and route all looks fine,
       so I'm a little puzzled)
      (hopefully no one will care this feature)

  - digital SPDIF input/output: not tested (no equipment)

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 11:25:42 +02:00
Takashi Iwai cedb8118e8 ALSA: pcm - Add logging of hwptr updates and interrupt updates
Added the logging functionality to xrun_debug to record the hwptr
updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt(),
corresponding to 16 and 8, respectively.

For example,
	# echo 9 > /proc/asound/card0/pcm0p/xrun_debug
will record the position and other parameters at each period interrupt
together with the normal XRUN debugging.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 11:09:03 +02:00
Mark Brown c30853df98 Merge branch 'for-2.6.31' into for-2.6.32 2009-07-23 08:22:58 +01:00
Lopez Cruz, Misael d756b27748 ASoC: OMAP: Staticise pcm creation function of omap-pcm
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 08:22:16 +01:00
Chaithrika U S 06c71282a9 ASoC: tlv320aic3x: Enable PLL when not bypassed
PLL was not being enabled when it was not bypassed. This patch
enables the PLL when it is used. Additionally, it disables the PLL
when it is bypassed.

Without this patch, the audio on TI DM646x EVM and DM355 EVM
does not work properly. The bit clocks and the frame sync signals
from the codec are not correct and hence the playback/record are faster
than usual for most sample rates. The reason for this was that the PLL
was not enabled when it was not bypassed.

Tested on DM6467 EVM, playback tested on DM355 EVM.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-23 08:14:29 +01:00
Takashi Iwai 4012ade933 ALSA: hda - Restore GPIO1 properly at resume with AD1984A
The commit 099db17e66 introduced a
regression at suspend/resume where the GPIO1 bit isn't properly
restored, thus the speaker output gets muted initially after resume.

The fix is simple, use the cached write for storing GPIO data.

Reference: Novell bnc#522764
	https://bugzilla.novell.com/show_bug.cgi?id=522764

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 18:15:10 +02:00
Takashi Iwai 35ebf6e721 ALSA: ctxfi - Simple code clean up
- replace NULL == xxx with !xxx
- replace NULL != xxx with xxx
- similar trivial cleanups

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 17:12:34 +02:00
Takashi Iwai 89e1b9511f Merge branch 'fix/ctxfi' into topic/ctxfi 2009-07-22 17:06:53 +02:00
Takashi Iwai 68110661e8 ALSA: ctxfi - Fix uninitialized error checks
Fix a few uninitialized error checks that were introduced recently
mistakenlly during the clean-up:
  sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’:
  sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this function
  sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’:
  sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this function
  sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’:
  sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 17:05:15 +02:00
Jaroslav Kysela 16a433d8b3 ALSA: hda-intel: Cleanups for widget connection list handling
This patch adds a check to snd_hda_get_connections() routine for
presence of AC_WCAP_CONN_LIST. Also, make sure that negative error
codes from noted route are handled on all places as errors.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 16:30:03 +02:00
Takashi Iwai 86de741660 ALSA: hda - Use snprintf() to be safer
Use snprint() for creating the jack name string instead of sprintf()
in patch_sigmatel.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 16:02:46 +02:00
Alexey Fisher 2cf313ee75 ALSA: usb-audio - Volume control quirk for QuickCam E 3500
- E3500 report cval->max more than it actually can handel, so if you
set 95% capture level it will be silently muted.
- Betwen cval->min and cval-max(real) is 2940 control units,
but real are only 7 with cval->res = 384.
- Alsa can't handel less than 10 controls, so make it more
and set cval->res = 192.

Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 15:52:15 +02:00
Jaroslav Kysela 254da007f9 ALSA: hda_generic: use AC_WCAP_CONN_LIST check for widget connections
Previous patch used widget type, but the presence flag of the connection
list is in the widget capabilities.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 15:39:56 +02:00
Jaroslav Kysela 0529604838 ALSA: hda_generic: do not read connections for widged with an unknown type
Reading node connections for an unknown widget can confuse HDA codec bus.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 15:39:44 +02:00
Takashi Iwai 1c20930a41 ALSA: hda - Fix ALC861 auto-mode parser
Fix the logic of ALC861 auto-mode parser for the outputs.
Instead of assuming the fixed DAC list, parse the conection and assign
the DAC dynamically.

Also, unmute the unused output connections to avoid noises on inputs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 15:36:25 +02:00
Takashi Iwai 79452f0a28 ALSA: pcm - Fix regressions with VMware
VMware tends to report PCM positions and period updates at utterly
wrong timing.  This screws up the recent PCM core code that tries
to correct the position based on the irq timing.

Now, when a backward irq position is detected, skip the update
instead of rebasing.  (This is almost the old behavior before
2.6.30.)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 12:55:56 +02:00
Takashi Iwai 05ff7e11b7 ALSA: hda - Reduce click noise at power-saving
Add some tricks to reduce the click noise at powering down to D3
in the power saving mode on STAC/IDT codecs.
The key seems to be to reset PINs before the power-down, and some
delay before entering D3.  The needed delay is significantly long,
but I don't know why.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-22 12:39:24 +02:00
Joonyoung Shim e458a48f87 ASoC: MAX9877: separate callback functions
The callback function to control register was used by whole controls in
MAX9877 driver, but this causes using many if statement for double
register control or invert.
So, the callback function for double register control is separate
differently, and the code for invert is added in the callback function.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-22 11:02:49 +01:00
javier Martin 25cbf46520 ASoC: Correct a bug with "ADC Inversion Switch" in wm8974 codec.
This corrects a bug with ADC Inversion Switch in wm8974 codec.

Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-22 00:13:27 +01:00
John Bonesio ed0f19b237 ASoC: MPC5200: Increase the delay time between resets
Reset was failing with the original udelay(50) between the code in
psc_ac97_cold_reset() and the call to psc_ac97_warm_reset(). Through testing
it was found that a delay of 1ms was necessary for the cold_reset code to
consistently complete successfully.

Signed-off-by: John Bonesio <bones@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-22 00:07:51 +01:00
Takashi Iwai 44f167d376 Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: ca0106 - Fix the max capture buffer size
  ALSA: OSS sequencer should be initialized after snd_seq_system_client_init
  ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
2009-07-21 19:03:22 +02:00
Takashi Iwai a9d90c81b5 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs
  ALSA: hda - Add quirk for Gateway T6834c laptop
  ALSA: hda_codec: Check for invalid zero connections
2009-07-21 19:03:20 +02:00
Takashi Iwai 36766835ed Merge branch 'fix/ctxfi' into for-linus
* fix/ctxfi:
  ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k2
2009-07-21 19:03:19 +02:00
Takashi Iwai 9c6c529a21 Merge branch 'fix/ctxfi' into topic/ctxfi 2009-07-20 17:08:01 +02:00
Frank Roth 55fe27f7e2 ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k2
On Soundblaster X-FI Titanium with emu20k2 the SIDE and SURROUND
channels were swapped and wrong. 
I double checked it with connector colors and creative soundblaster
windows drivers.

So I swapped them to the true order.
Now "speaker-test -c6" and "speaker-test -c8" are working fine.

Signed-off-by: Frank Roth <frashman@freenet.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-20 17:07:10 +02:00
Takashi Iwai 34fdeb2d07 ALSA: ca0106 - Fix the max capture buffer size
The capture buffer size with 64kB seems broken with CA0106.
At least, either the update timing or the DMA position is wrong,
and this screws up pulseaudio badly.

This patch restricts the max buffer size less than that to make life
a bit easier.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-07-20 15:49:46 +02:00
Takashi Iwai b04add9566 ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs
The recent rewrite of the codec parser for STAC9872 caused a regression
for some Sony VAIO models that don't give proper pin default configs
by BIOS.  Even using model=vaio doesn't work because the pin definitions
are set after the pin overrides.

This patch fixes the pin definitions in patch_stac9872() to be put
in the right place before the pin overrides.  Also the patch adds the
new quirk entry for VAIO F/S to have the correct pin default configs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-07-20 15:12:41 +02:00
Takashi Iwai bc5304b6fb ALSA: ctxfi - Native timer support for emu20k2
Added the native timer support for emu20k2, which gives much more
accurate update timing than the system timer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-20 13:41:35 +02:00
Hao Song 42b95f0c6b ALSA: hda - Add quirk for Gateway T6834c laptop
Gateway T6834c laptops need EAPD always on while the default behavior
for the STAC9205 reference board is to turn it off upon every HP plug.
By using the special "eapd" model, which is first introduced for Gateway
T1616 laptops for this same reason, this peculiarity can be properly
handled.

Signed-off-by: Hao Song <baritono.tux@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-20 09:05:53 +02:00
Jaswinder Singh Rajput f96e080821 ALSA: OSS sequencer should be initialized after snd_seq_system_client_init
When build SND_SEQUENCER in kernel then OSS sequencer(alsa_seq_oss_init)
is initialized before System (snd_seq_system_client_init) which leads to
memory leak :

unreferenced object 0xf6b0e680 (size 256):
  comm "swapper", pid 1, jiffies 4294670753
  backtrace:
    [<c108ac5c>] create_object+0x135/0x204
    [<c108adfe>] kmemleak_alloc+0x26/0x4c
    [<c1087de2>] kmem_cache_alloc+0x72/0xff
    [<c126d2ac>] seq_create_client1+0x22/0x160
    [<c126e3b6>] snd_seq_create_kernel_client+0x72/0xef
    [<c1485a05>] snd_seq_oss_create_client+0x86/0x142
    [<c1485920>] alsa_seq_oss_init+0xf6/0x155
    [<c1001059>] do_one_initcall+0x4f/0x111
    [<c14655be>] kernel_init+0x115/0x166
    [<c10032af>] kernel_thread_helper+0x7/0x10
    [<ffffffff>] 0xffffffff
unreferenced object 0xf688a580 (size 64):
  comm "swapper", pid 1, jiffies 4294670753
  backtrace:
    [<c108ac5c>] create_object+0x135/0x204
    [<c108adfe>] kmemleak_alloc+0x26/0x4c
    [<c1087de2>] kmem_cache_alloc+0x72/0xff
    [<c126f964>] snd_seq_pool_new+0x1c/0xb8
    [<c126d311>] seq_create_client1+0x87/0x160
    [<c126e3b6>] snd_seq_create_kernel_client+0x72/0xef
    [<c1485a05>] snd_seq_oss_create_client+0x86/0x142
    [<c1485920>] alsa_seq_oss_init+0xf6/0x155
    [<c1001059>] do_one_initcall+0x4f/0x111
    [<c14655be>] kernel_init+0x115/0x166
    [<c10032af>] kernel_thread_helper+0x7/0x10
    [<ffffffff>] 0xffffffff
unreferenced object 0xf6b0e480 (size 256):
  comm "swapper", pid 1, jiffies 4294670754
  backtrace:
    [<c108ac5c>] create_object+0x135/0x204
    [<c108adfe>] kmemleak_alloc+0x26/0x4c
    [<c1087de2>] kmem_cache_alloc+0x72/0xff
    [<c12725a0>] snd_seq_create_port+0x51/0x21c
    [<c126de50>] snd_seq_ioctl_create_port+0x57/0x13c
    [<c126d07a>] snd_seq_do_ioctl+0x4a/0x69
    [<c126d0de>] snd_seq_kernel_client_ctl+0x33/0x49
    [<c1485a74>] snd_seq_oss_create_client+0xf5/0x142
    [<c1485920>] alsa_seq_oss_init+0xf6/0x155
    [<c1001059>] do_one_initcall+0x4f/0x111
    [<c14655be>] kernel_init+0x115/0x166
    [<c10032af>] kernel_thread_helper+0x7/0x10
    [<ffffffff>] 0xffffffff

The correct order should be :

System (snd_seq_system_client_init) should be initialized before
OSS sequencer(alsa_seq_oss_init) which is equivalent to :

1. insmod sound/core/seq/snd-seq-device.ko
2. insmod sound/core/seq/snd-seq.ko
3. insmod sound/core/seq/snd-seq-midi-event.ko
4. insmod sound/core/seq/oss/snd-seq-oss.ko

Including sound/core/seq/oss/Makefile after other seq modules
fixes the ordering and memory leak.

Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-19 19:10:01 +02:00
Julia Lawall fcb2954b96 ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
If spin_lock_irqsave is called twice in a row with the same second
argument, the interrupt state at the point of the second call overwrites
the value saved by the first call.  Indeed, the second call does not need
to save the interrupt state, so it is changed to a simple spin_lock.

The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)

// <smpl>
@@
expression lock1,lock2;
expression flags;
@@

*spin_lock_irqsave(lock1,flags)
... when != flags
*spin_lock_irqsave(lock2,flags)
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-19 14:02:29 +02:00
Takashi Iwai 05e870d29a Merge branch 'fix/hda' into topic/hda 2009-07-19 13:52:31 +02:00
Jaroslav Kysela 2e9bf24706 ALSA: hda_codec: Check for invalid zero connections
To prevent "Too many connections" message and the error path for some HDMI
codecs (which makes onboard audio unusable), check for invalid zero
connections for CONNECT_LIST verb.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-19 13:51:45 +02:00
Mark Brown bca146578c ASoC: Fix checkpatch issues in AD1938
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-18 11:09:42 +01:00
Mark Brown 0c11f65555 ASoC: Fix FLL reference clock division setup in WM8993
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-17 22:13:01 +01:00
Mark Brown 8aa2df5308 ASoC: Bodge around GCC 4.4.0 flow analysis bug in GCC 4.4.0
GCC 4.4.0 doesn't appear to be able to spot that we don't apply any FLL
configuration if the output frequency is zero.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-17 21:53:49 +01:00
Takashi Iwai 3f3b7c1aed ALSA: hda - Fix ALC268 parser for mono speaker
- Parse the mono output pin 0x16 correctly even as the primary output
- Create "Speaker" volume control if the primary output is a speaker
- Fix the wrong direction of (optional) "Mono" switch

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 14:36:59 +02:00
Takashi Iwai 82e1b804b0 ALSA: hda - Fix the previous sanity check in make_codec_cmd()
The newly added sanity-check for a codec verb can be better written
with logical ORs.  Also, the parameter can be more than 8bit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 12:47:34 +02:00
Candelaria Villareal, Jorge c5910a7038 ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6
Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6
line, controlled by register INTBR_PMBR1. Machine driver takes care
of enabling gpio line through i2c and codec driver manipulates the
line during headset ramp up/down sequence.

Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-17 10:32:39 +01:00
Wu Fengguang 6430aeeb30 ALSA: hda - add bounds checking for the codec command fields
A recent bug involves passing auto detected >0x7f NID to codec command,
creating an invalid codec addr field, and finally lead to cmd timeout
and fall back into single command mode. Jaroslav fixed that bug in
alc880_parse_auto_config().

It would be safer to further check the bounds of all cmd fields.

Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 11:18:21 +02:00
Andiry Brienza 9176b672c2 ALSA: hda - Add support for new AMD HD audio devices
Add support for new AMD HD audio devices. Use generic driver to detect HD audio
devices with Vendor ID AMD.

Signed-off-by: Andiry Xu <andiry.xu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 11:16:51 +02:00
Takashi Iwai 416c8fe3cd ASoC: Kill direct accesses to driver_data
Replaced with dev_{get|set}_drvdata().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-17 07:48:03 +02:00
Takashi Iwai 15c2ac051c Merge branch 'fix/usb-audio' into for-linus
* fix/usb-audio:
  sound: usb-audio: add workaround for Blue Microphones devices
2009-07-16 16:35:50 +02:00
Takashi Iwai 9d79b13691 Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: riptide -  proper handling of pci_register_driver for joystick
2009-07-16 16:35:48 +02:00
Takashi Iwai 26887793b6 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checking
2009-07-16 16:35:47 +02:00
Takashi Iwai 9d5b28d530 Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free
2009-07-16 16:35:46 +02:00
Daniel Drake 0fb67e982a ALSA: hda - Add CX20582 and OLPC XO-1.5 support
This adds support for the Conexant CX20582 codec, based on code from
http://www.linuxant.com/alsa-driver/alsa-driver-linuxant-1.0.19ppch12-1.noarch.rpm.zip

This is the codec to be shipped in the OLPC XO-1.5, so this patch also
includes an XO-specific profile. Resultant configuration:
http://dev.laptop.org/~dsd/20090713/codec0.txt
http://dev.laptop.org/~dsd/20090713/codec0.svg

As the Linuxant code is structured differently to the other codecs,
I was unable to cleanly reimplement everything in the generic and Dell
profiles as more info is needed (e.g. codec graphs). I simplified those
profiles so that hopefully it will not break anyone's audio. If it does,
it may be worth returning -ENODEV from patch_cx5066 on non-OLPC systems,
and then fixing snd_hda_codec_configure() to fall back on the generic
parser, at least until support for other systems is figured out.

Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-16 16:03:08 +02:00
Barry Song 1274738d85 ASoC: new ad1938 codec driver based on asoc
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-16 11:14:39 +01:00
Kevin Hilman 3e46a44739 ASoC: davinci: don't use clock names
clock name strings are no longer passed on platform_data.  Instead,
we rely entirely on struct device and clkdev to find the right clock.

Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-16 10:59:52 +01:00
Joonyoung Shim 9db9ed977d ASoC: MAX9877: add MAX9877 amp driver
The MAX9877 combines a high-efficiency Class D audio power amplifier
with a stereo Class AB capacitor-less DirectDrive headphone amplifier.

The max9877_add_controls() is called to register the MAX9877 specific
controls on machine specific init() of the machine driver.

The datasheet for the MAX9877 can find at the following url:
http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf

[Slight edit to sort the ALL_CODECS entries -- broonie.]

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-15 16:59:31 +01:00
Jaswinder Singh Rajput cb65c8732a ALSA: riptide - proper handling of pci_register_driver for joystick
We need to check returning error for pci_register_driver(&joystick_driver)

On failure, we should unregister formerly registered audio drivers

This also fixed the compiler warning :

  CC [M]  sound/pci/riptide/riptide.o
 sound/pci/riptide/riptide.c: In function ‘alsa_card_riptide_init’:
 sound/pci/riptide/riptide.c:2200: warning: ignoring return value of ‘__pci_register_driver’, declared with attribute warn_unused_result

Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 14:00:40 +02:00
Mark Brown 4b75e94767 ASoC: Error out if we can't determine a suitable WM9081 sysclk
Due to the flexibility of the WM9081 FLL this should never happen
in a real system.

Reported-by: Jaswinder Singh Rajput <jaswinder@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-15 11:03:51 +01:00
Andreas Mohr 78df617acf ALSA: azt3328: fix previous breakage, improve suspend, cleanups
- fix my previous codec activity breakage (_non-warned_ variable assignment
  issue)
- convert suspend/resume to 32bit I/O access (I/O is painful; to improve
  suspend/resume performance)
- change DEBUG_PLAY_REC to DEBUG_CODEC for consistency
- printk cleanup
- some logging improvements
- minor cleanup/improvements

The variable assignment issue above was a conditional assignment to the
call_function variable (this ended with the non-preinitialized variable
not getting assigned in some cases, thus a dangling stack value, yet gcc 4.3.3
unbelievably did _NOT_ warn about it in this case!!),
needed to change this into _always_ assigning the check result.
Practical result of this bug was that when shutting down
_either_ playback or capture, _both_ streams dropped dead :P

Tested, working (plus resume) and checkpatch.pl:ed on 2.6.30-rc5,
applies cleanly to 2.6.30 proper with my previous (committed)
patches applied.

Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 12:03:26 +02:00
Clemens Ladisch 2d4b842014 sound: rawmidi: disable active-sensing-on-close by default
Sending an Active Sensing message when closing a port can interfere with
the following data if the port is reopened and a note-on is sent before
the device's timeout has elapsed.  Therefore, it is better to disable
this setting by default.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 11:57:20 +02:00
Clemens Ladisch 08d033405a sound: seq_oss_midi: remove magic numbers
Instead of using magic numbers for the controlles sent when resetting
a port, use the symbols from asoundef.h.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 11:57:08 +02:00
Clemens Ladisch b86c87288c sound: seq_midi: do not send MIDI reset when closing
Sending a MIDI reset message when closing a port is wrong because we
only want to shut the device up, not to reset all settings.
Furthermore, many devices ignore this message.

Fortunately, the RawMIDI layer already shuts the device up, so we can
ignore this matter here.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 11:56:58 +02:00
Clemens Ladisch a65dd997b3 sound: usb-audio: add MIDI drain callback
When draining, instead of waiting for fifty milliseconds, just wait for
the currently active URBs to complete.  This cuts the usual waiting time
down to one USB frame, or zero in the common case when there is no URB.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 11:56:41 +02:00
Clemens Ladisch ed4affa532 sound: usb-audio: use multiple output URBs
Some newer USB MIDI interfaces use rather small packet sizes, so to get
enough bandwidth, we have to be able to send multiple packets in one USB
frame, so we have to use multiple URBs.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 11:56:30 +02:00
Clemens Ladisch 4773d1fb8f sound: usb-audio: use multiple input URBs
Some newer USB MIDI interfaces use rather small packet sizes, so to get
enough bandwidth, we have to be able to receive multiple packets in one
USB frame, so we have to use multiple URBs.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 11:56:19 +02:00
Clemens Ladisch f907ed94f9 seq-midi: always log message on output overrun
It turns out that the main cause of output buffer overruns is not slow
drivers but applications that generate too many messages.  Therefore, it
makes more sense to make that error message always visible, and to
rate-limit it.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 11:56:06 +02:00
Clemens Ladisch 468b8fde24 sound: usb-audio: Xonar U1 digital output support
Add support for the Asus Xonar U1.  This device is mostly class compliant, but
the digital output requires a vendor-specific request.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 11:55:25 +02:00
Clemens Ladisch 8886f33f25 sound: usb-audio: add workaround for Blue Microphones devices
Blue Microphones USB devices have an alternate setting that sends two
channels of data to the computer.  Unfortunately, the descriptors of
that altsetting have a wrong channel setting, which means that any
recorded data from such a device has twice the sample rate from what
would be expected.

This patch adds a workaround to ignore that altsetting.  Since these
devices have only one actual channel, no data is lost.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-15 11:55:00 +02:00
Mark Brown e465d544fa ASoC: Fix sample rate lookup in WM8993
We need to use the best value we picked, not the last value we
looked at.

Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-15 10:01:30 +01:00
Cliff Cai 82d76f4d9f ASoC: Blackfin I2S: fix resume handling
There is no need to manually start playback/capture ourselves as the PCM
driver will handle things for us.

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-14 19:44:52 +01:00
Cliff Cai 18d02bc32c ASoC: Blackfin AC97: fix resume handling
There is no need to manually start playback/capture ourselves as the PCM
driver will handle things for us.

Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-14 19:44:52 +01:00
Mark Brown ba3b64b976 Merge branch 'for-2.6.31' into for-2.6.32 2009-07-13 23:05:51 +01:00
Kevin Hilman 0a0cf58d93 ASoC: spdif: set module licence to GPL
Without MODULE_LICENCE("GPL"), when built as a module it will fail
to load because it uses other GPL symbols from kernel.

Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-13 23:01:30 +01:00
Kevin Hilman a27e304b5c ASoC: spdif codec: enable use by modules
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-13 23:01:30 +01:00
Rongrong Cao 087d53ab11 ASoC: fix checking for external widgets bug
In SOC DAPM layer of SOUND subsystem, when add signal route (in the
function snd_soc_dapm_add_route() ), the original code has wrong logic
when dapm layer check each widget whether an external one.

Signed-off-by: Rongrong Cao <rrcao@ambarella.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-13 23:01:29 +01:00
Roel Kluin 33e319fba7 ASoC: Keep index within stac9766_reg[]
Keep index within stac9766_reg[]

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-13 10:32:18 +01:00
Greg Kroah-Hartman 864e1e8db4 Sound: remove direct access of driver_data
This is the last in-kernel direct usage of driver_data, replace it with
the proper dev_get/set_drvdata() calls.

Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2009-07-12 13:02:10 -07:00
Linus Torvalds f00caa7629 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:
  ALSA: hda - targa and targa-2ch fix
  ALSA: hda - fix beep tone calculation for IDT/STAC codecs
  ALSA: hda - Missing volume controls for Intel HDA (ALC269/EeePC)
  ALSA: hda - Disable AMD SB600 64bit address support only
  ALSA: hda - Check widget types while parsing capture source in patch_via.c
  ALSA: hda - Fix capture source selection in patch_via.c
  ALSA: hda - Add missing EAPD initialization for VIA codecs
  ALSA: hda - Clean up VT170x dig-in initialization code
  ALSA: hda - Fix error path in the sanity check in azx_pcm_open()
  ALSA: hda - move 8086:fb30 quirk (stac9205) to the proper section
  ASoC: Fix wm8753 register cache size and initialization
  ASoC: add locking to mpc5200-psc-ac97 driver
  ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared
  ASoC: Fix register cache initialisation for WM8753
2009-07-10 19:19:09 -07:00
Mark Brown 030c819e79 Merge branch 'tlv320aic3x' into reg-cache 2009-07-10 21:06:33 +01:00
Takashi Iwai 3c6aae4489 ALSA: hda - Check codec errors in snd_hda_get_connections()
The codec read errors in snd_hda_get_connections() are ignored so far,
and it causes a problem like the bug in the commit
    9d30937acc
    ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checking

Better to check errors in the function and returns a proper error code
rather than passing bogus NID values.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-10 12:56:41 +02:00
Jaroslav Kysela 9d30937acc ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checking
On some IbexPeak systems with ALC889A errors like "azx_get_response
timeout, switching to polling mode: last cmd=0xaf9f000b" are produced,
because non-existent codec #10 is wrongly accessed.

The problem is that snd_hda_get_connections() returns out-of-range result
for NID 0x1c (something like 0xf8f9 or 0xffff).

This patch adds a check to alc880_parse_auto_config() to avoid using
of this out-of-range NIDs. A better fix maybe to improve
snd_hda_get_connections() routine to check for valid NID ranges if
NIDs are expected as result.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-10 12:55:49 +02:00
Takashi Iwai 31909b83ea ALSA: hda - Fix the merge error
Fix the merge error at the commit 305355aad8,
an addition of the missing alc880_gpio3_init_verbs to ALC882_TARGA model.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-10 12:33:48 +02:00
Takashi Iwai 3ae3079666 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - targa and targa-2ch fix
  ALSA: hda - fix beep tone calculation for IDT/STAC codecs
  ALSA: hda - Missing volume controls for Intel HDA (ALC269/EeePC)
  ALSA: hda - Disable AMD SB600 64bit address support only
  ALSA: hda - Check widget types while parsing capture source in patch_via.c
  ALSA: hda - Fix capture source selection in patch_via.c
  ALSA: hda - Add missing EAPD initialization for VIA codecs
  ALSA: hda - Clean up VT170x dig-in initialization code
  ALSA: hda - Fix error path in the sanity check in azx_pcm_open()
  ALSA: hda - move 8086:fb30 quirk (stac9205) to the proper section
2009-07-10 11:17:12 +02:00
Takashi Iwai f371f12f3e Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: Fix wm8753 register cache size and initialization
  ASoC: add locking to mpc5200-psc-ac97 driver
  ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared
  ASoC: Fix register cache initialisation for WM8753
2009-07-10 11:17:11 +02:00
Takashi Iwai 305355aad8 Merge branch 'fix/hda' into topic/hda
Conflicts:
	sound/pci/hda/patch_realtek.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-09 18:48:38 +02:00
David Heidelberger 005b10769c ALSA: hda - targa and targa-2ch fix
Simplify ALC882_TARGA and return gpio3 to ALC883_TARGA_DIG and
ALC883_TARGA_2ch_DIG, which I accidentally removed in commit id
64a8be7435

Signed-off-by: David Heidelberger <d.okias@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-09 18:45:46 +02:00
Mark Brown d7dbf6ea40 [ARM] 5596/1: at91sam9g20-ek: Register WM8731 in board file
The WM8731 driver has been updated to allow registration via normal
device model methods rather than from within the ASoC driver probe
so update the AT91SAM9G20-EK to make use of this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Andrew Victor <linux@maxim.org.za>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
2009-07-09 17:15:23 +01:00
Mark Brown cc369cf504 ASoC: WM8510 has a single frame clock so needs symmetric rates
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-09 11:28:07 +01:00
Takashi Iwai 63b2413b2f ALSA: hda - don't build digital output controls if not exist
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-09 11:45:59 +02:00
Daniel Mack b7d4de7ff0 ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_free
Check for rtd->params->drcmr != NULL before accessing it.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-09 10:36:19 +01:00
Paul Vojta 369693dc93 ALSA: hda - fix beep tone calculation for IDT/STAC codecs
In the beep tone calculation for IDT/STAC codecs, lower numbers correspond
to higher frequencies and vice versa.  The current code has this backwards,
resulting in beep frequencies which are way too high (and sound bad on
tinny laptop speakers, resulting in complaints).

[Also added hz <= 0 check by tiwai]

Signed-off-by: Paul Vojta <vojta@math.berkeley.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-09 09:14:29 +02:00
Mark Brown cb507e7e79 ASoC: Add pop delay debug at end of DAPM sequencing
Provide an interval after the end of DAPM sequencing so that we
can distinguish between a pop in the final step of the sequence
and a pop generated from some other source outside DAPM.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 18:54:57 +01:00
Mark Brown 96fd6d471b ASoC: Configure WM8731 SYSCLK at startup on AT91SAM9G20-EK
The system clock is currently fixed by the driver and this avoids
the need for us to handle errors with enabling and disabling MCLK
(which was incorrect previously so this fixes bugs in error
handling).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 18:41:05 +01:00
Joe Perches ad361c9884 Remove multiple KERN_ prefixes from printk formats
Commit 5fd29d6ccb ("printk: clean up
handling of log-levels and newlines") changed printk semantics.  printk
lines with multiple KERN_<level> prefixes are no longer emitted as
before the patch.

<level> is now included in the output on each additional use.

Remove all uses of multiple KERN_<level>s in formats.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2009-07-08 10:30:03 -07:00
Mark Brown 22df8eb4fe ASoC: Disable microphone input for AT91SAM9G20-EK by default
As shipped the board does not have inputs but it is relatively
straightforward to modify the board to hook them up so support
is provided in the driver. When these modifications have not
been made enabling the microphone stage can cause problems.

Add an ifdef to disable this by default. Don't put it into
Kconfig since users will have to get their soldering irons
out to change things.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 18:18:19 +01:00
Mark Brown 2a01e5f358 ASoC: Use CODEC as clock master on AT91SAM9G20-EK
This simplifies the driver by removing the need to manually
configure dividers within the CPU and improve audio performance
by ensuring that the optimal phase relationships between the
clocks in the system are maintained.

Note that currently this means that for playback to work the
Output Mixer HiFi switch must be enabled since otherwise CODEC
will not generate the DAC clock.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 18:05:51 +01:00
Mark Brown 4934482d93 ASoC: Limit WM8731 to symmetric rates
While the hardware is capable of some limited asynmmetric modes the
driver does not currently support those modes so tell applications
that only symmetric rates are available.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 17:48:12 +01:00
Mark Brown 942c435ba7 ASoC: Add WM8993 CODEC driver
The WM8993 is a highly integrated ultra-low power hi-fi CODEC designed
for portable devices such as multimedia phones.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 17:20:20 +01:00
Mark Brown ff7d04b130 ASoC: DaVinci I2S needs mach/asp.h
Reported-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 17:18:30 +01:00
Mark Brown ef38ed888e ASoC: Correct WM8731 Mic Capture Switch control name
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 17:18:28 +01:00
Mark Brown d00efa648d ASoC: Add TLV information for WM8731
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 17:08:40 +01:00
Troy Kisky 6e5414750a ASoC: DaVinci: pcm, don't play 1st sound period twice
Update the dma link with correct data as soon as
the master channel has copied it. Otherwise, the
1st period will play twice.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-08 16:00:20 +01:00
Darren Salt 508f711090 ALSA: hda - Missing volume controls for Intel HDA (ALC269/EeePC)
There is a regression, introduced in aa202455ee
(in alsa-kernel) which I noticed when trying to use the headphone socket on
my EeeCPC 901: the output was *very* quiet, practically silent.

This patch corrects the control types to that which was obviously intended in
the referenced commit.

Signed-off-by: Darren Salt <linux@youmustbejoking.demon.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-08 16:37:45 +02:00
Takashi Iwai cc6a8acdee ALSA: Fix SG-buffer DMA with non-coherent architectures
Using SG-buffers with dma_alloc_coherent() is often very inefficient
on non-coherent architectures because a tracking record could be
allocated in addition for each dma_alloc_coherent() call.
Instead, simply disable SG-buffers but just allocate normal continuous
buffers on non-supported (currently all but x86) architectures.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-08 14:20:20 +02:00
William Weston b1e4422f96 ALSA: hda - Add quirks for RTL888 & RV630/M76 based MSI GX710
Signed-off-by: William Weston <weston@sysex.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-08 14:18:49 +02:00
Takashi Iwai 277a57c710 ALSA: hda - Fix compile warnings in patch_cirrus.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-08 12:42:08 +02:00
Andiry Brienza dc4c2e6bde ALSA: hda - Disable AMD SB600 64bit address support only
HDA driver disabled HD audio 64bit address support for all AMD
SB600/SB700/SB800 platforms with commit
09240cf429 due to one SB600 issue
reported by community, but we do not see the similar issue on
SB700/SB800 platforms.
This patch is to refine the workaround for SB600 only.

Signed-off-by: Andiry Xu <andiry.xu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-08 08:01:47 +02:00
Takashi Iwai 1c55d521f4 ALSA: hda - Check widget types while parsing capture source in patch_via.c
Check the widget type and don't take invalid widgets while parsing
the capture source in patch_via.c.

Also, fixed some compile warnings introduced in the previous commit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-08 07:45:46 +02:00
Mark Brown 3f405b46a9 Merge branch 'davinci' into for-2.6.32
Conflicts:
	sound/soc/davinci/davinci-i2s.c
2009-07-07 19:18:46 +01:00
Ondrej Zary 72b43cf140 ALSA: cmi8330: Allow MPU-401-less operation
Adding MPU-401 support to cmi8330 driver could cause a regression (non-working
sound) on a system where there is no free IRQ for the MPU-401 device (which
is not very uncommon as this card requires two separate IRQs plus a third one
for MPU-401).

When MPU-401 PnP configuration fails (mostly because of unavailable IRQ), just
ignore MPU-401 and continue without it.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-07 18:24:53 +02:00
Takashi Iwai 337b9d02b4 ALSA: hda - Fix capture source selection in patch_via.c
The fixed widget NIDs in patch_via.c seem wrong for some codecs,
and it resulted in the invalid capture source selection.

This patch adds the code to parse the topology instead of using
fixed numbers in order to get the right MUX widget id corresponding
to the ADCs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-07 18:20:23 +02:00
Takashi Iwai d3a11e601a ALSA: hda - Add missing EAPD initialization for VIA codecs
If the output pin is used and EAPD capability is present, turn on
the EAPD bit.  This fixes the silent output problem on ASUS laptops
with VT1708S codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-07 13:44:29 +02:00
Takashi Iwai 55d1d6c1ef ALSA: hda - Clean up VT170x dig-in initialization code
Minor clean up for initializing the digital-in pin.
No functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-07 13:39:03 +02:00
Takashi Iwai b4dabfc452 ALSA: hda - Fix the speaker volume control name
Increase the name string buffer size so that "Surround Speaker Playback
Volume" won't be truncated.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-07 09:05:07 +02:00
Takashi Iwai ed208255e7 ALSA: hda - Add GPIO setup for MacBook pro 5,5 with CS420x
GPIO3 seems corresponding to EAPD that is required for the speaker
output.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-07 09:04:26 +02:00
Ondrej Zary 0b95916723 ALSA: cmi8330: find OPL3 port automatically
My CMI8329 had OPL3 port specified in SB16 resources. But now I found out that
it was my modification of the card's PnP EEPROM a couple of years ago (can be
done using C9SETROM.EXE utility). I did it because the OPL3 port was
completely missing from PnP data. It seems to be hardwired to 0x388 on
CMI8329.

Find OPL3 port automatically by searching in WSS and SB16 resources. If not
found, assume that it's hardwired to 0x388.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-07 08:02:57 +02:00
Mark Brown 4ec5c9693b Merge branch 'for-2.6.31' into for-2.6.32 2009-07-06 21:49:35 +01:00
Takashi Iwai a6bae20559 ALSA: hda - Add quirk for MacBook Pro 5,5 with CS4206
Add the default pin configs for MBP55.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-06 15:18:46 +02:00
Takashi Iwai 60e53882ac ALSA: hda - Fix double creation of SPDIF input controls
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-06 15:01:09 +02:00
Takashi Iwai 9983aa62c3 ALSA: info - Use krealloc()
Use krealloc() to resize the buffer in sound/core/info.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-06 14:31:59 +02:00
Takashi Iwai 40c20fa05a ALSA: hda - Add CS420x-specific coef setup
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-06 13:02:44 +02:00
Takashi Iwai ea35929b88 ALSA: hda - Force to initialize input mixer setup for CS420x
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-06 13:02:42 +02:00
Takashi Iwai 21a4dc43ac ALSA: hda - Fix cirrus codec parsing
The parser wasn't called in the proper order.
Split now the parser to be called in patch_cirrus(), and the rest
are just for building PCMs and controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-06 13:02:20 +02:00
Andreas Mohr dfbf951115 ALSA: azt3328: large codec cleanup, add I2S port etc.
- fully separate codec I/O port handling, enabling the use of a single
  function each for all codecs (playback, capture, I2S out)
- add a new separate pcm for I2S out port (UNTESTED, no I2S DAC
  available yet)
- switch gameport to low frequency while idle, to try to reduce noise/power
- improve snd_azf3328_codec_setdmaa() calculation
- minor variable type cleanup (u16, bool etc.)
- add some doc updates (help those lost Windows users, debug help, ...)

Note that due to the large cleanup aspect of the codec I/O change,
I was able to fit everything including all improvements into the
same binary size!! (a measly 10 bytes more or so)

This should now be the almost last patch to this driver
(minus some possible kernel clocksource patch and x86_64 fixes or so).
I just felt like taking a break from the usual stuff and wanted to
get this driver's structure finished, and it's rather clean now...

Tested, working and checkpatch.pl:ed on 2.6.30-rc5,
applies cleanly to 2.6.30 proper.

Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-06 08:24:47 +02:00
Andreas Mohr 3eff895830 ALSA: azt3328: fix Kconfig entry
This driver is about as far from being experimental as it can ever get
for an undocumented card, thus create this patch (interestingly it was the only
EXPERIMENTAL remaining in the entire Kconfig file).

Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-06 08:24:34 +02:00
Mark Brown f6f1eb1033 ASoC: Factor out WM8580 register cache code
Note the slightly tricky cache usage in the volume update function due
to the requirement for a separate write for the VU bit.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 17:57:57 +01:00
Mark Brown 5f345346dd ASoC: Remove use of hw_read from TLV320AIC3x driver
The TLV320AIC3x driver is currently the only user of the CODEC hw_read
operation and is jumping through some hoops in order to do so.  In order
to support future refactoring to make the hw_read operation more usable
unwrap the usage in this driver to avoid its use.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 17:35:28 +01:00
Mark Brown 1e30a5828e ASoC: Remove unused AK4535 hardware read functionality
Nothing uses it and the existing hw_read operation needs to be
refectored so it's easier to remove it rather than work with it.
Support can be re-added if the code requires volatile registers.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 17:28:41 +01:00
Mark Brown 17a52fd60a ASoC: Begin to factor out register cache I/O functions
A lot of CODECs share the same register data formats and therefore
replicate the code to manage access to and caching of the register
map. In order to reduce code duplication centralised versions of
this code will be introduced with drivers able to configure the use
of the common code by calling the new snd_soc_codec_set_cache_io()
API call during startup.

As an initial user the 7 bit address/9 bit data format used by many
Wolfson devices is supported for write only CODECs and the drivers
with straightforward register cache implementations are converted to
use it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 17:24:50 +01:00
Mark Brown 5420f30723 ASoC: Fix leaks in WM8988 registration error handling
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 16:29:39 +01:00
Mark Brown 1a01417e85 ASoC: Fix WM8960 leaks on probe failure
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 15:47:03 +01:00
Mark Brown fe5422fc4a ASoC: Fix leaks in WM8731 probe error handling
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 15:18:01 +01:00
Mark Brown 096e49d5e6 ASoC: Add CODEC volatile register operation
Add a volatile_register() operation to the CODEC structure providing a
standard operation to query if a register is volatile. This will be used
to factor out the register cache I/O operations for the CODECs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 15:12:22 +01:00
Troy Kisky af0adf3e81 ASoC: DaVinci: i2s, add davinci_i2s_prepare and shutdown
If the codec is master then prepare should call
mcbsp_start, not trigger.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 13:14:35 +01:00
Troy Kisky f5cfa954e6 ASoC: DaVinci: i2s, fix mcbsp_word_length update
Code previously just "ors" in this field without clearing
first. Fix, by never reading this register.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 12:59:08 +01:00
Troy Kisky 9333b594a0 ASoC: DaVinci: i2s, minor cleanup of davinci_i2s_startup
Save a few lines of code.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 12:59:08 +01:00
Troy Kisky 1bef449989 ASoC: DaVinci: i2s, only start sample generator if needed
Only start sample generator if needed, and more
cleanup on davinci_mcbsp_start.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 12:59:07 +01:00
Troy Kisky eba575c30b ASoC: DaVinci: i2s cleanup
Move variable declaration closer to use.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 12:59:07 +01:00
Troy Kisky f9af37cc63 ASoc: DaVinci: i2s, minor cleanup
Add davinci_mcbsp_dev as argument to davinci_mcbsp_start
and davinci_mcbsp_stop.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 12:59:07 +01:00
Troy Kisky c392bec716 ASoC: DaVinci: i2s toggle clock to complete reset
Add toggle_clock function to complete i2s reset earlier.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 12:59:06 +01:00
Troy Kisky 35cf63583d ASoC: DaVinci: i2s, remove MOD_REG_BIT macro
No functional changes. Rename variable w to something
more meaningful. Remove code obfuscating macro MOD_REG_BIT.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-05 12:59:06 +01:00
Ondrej Zary 69eb88825a cmi8330: Add basic CMI8329 support
Add basic support for CMI8329 cards. Makes PCM and OPL3 work.
Does not break CMI8330 (tested).

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-05 11:47:24 +02:00
Takashi Iwai aba6653617 ALSA: hda - Fix error path in the sanity check in azx_pcm_open()
Release resources cleanly after errors in the sanity check in
azx_pcm_open().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-05 11:44:46 +02:00
Takashi Iwai 1475ef0f03 Merge branch 'fix/hda' into topic/hda 2009-07-04 12:20:25 +02:00
Herton Ronaldo Krzesinski 02358fcfa5 ALSA: hda - move 8086:fb30 quirk (stac9205) to the proper section
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-04 12:20:16 +02:00
Takashi Iwai 7ce1695c40 Merge branch 'fix/soundcore' into for-linus
* fix/soundcore:
  sound: do not set DEVNAME for OSS devices
2009-07-03 23:50:47 +02:00
Takashi Iwai 854ace9c40 Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Add sanity check in PCM open callback
  ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callback
  ALSA: hda - Avoid invalid formats and rates with shared SPDIF
  ALSA: hda - Improve ASUS eeePC 1000 mixer
  ALSA: hda - Add GPIO1 control at muting with HP laptops
2009-07-03 23:50:45 +02:00
Kay Sievers 954a973cab sound: do not set DEVNAME for OSS devices
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-03 23:36:13 +02:00
Takashi Iwai b0bb3aa623 ALSA: hda - Don't override maxbps for FLOAT sharing with linear formats
When FLOAT PCM format is available but together with other linear
PCM formats, don't override maxbps value.  For FLOAT format, it's always
32, thus it can be better checked in snd_hda_calc_stream_format().
Otherwise the maxbps 32 might be used wrongly even if the linear PCM
doesn't support it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-03 23:25:37 +02:00
Takashi Iwai c470331e69 ALSA: hda - Add sanity check in PCM open callback
Add some sanity checks of struct snd_pcm_hardware fields in the PCM
open callback of hda driver.  This makes a bit easier to debug any PCM
setup errors in the codec side.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-03 23:10:23 +02:00
Takashi Iwai 70d321e638 ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callback
The PCM rates bit field may have been changed by the codec open callback.
In that case, we need to reset rate_min and rate_max.  So, simply call
snd_pcm_lib_hw_rates() again after the codec open callback.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-03 23:06:45 +02:00
Takashi Iwai 022b466fc3 ALSA: hda - Avoid invalid formats and rates with shared SPDIF
Check whether formats and rates don't result in zero due to the
restriction of SPDIF sharing.  If any of them can be zero, disable
the SPDIF sharing mode instead.  Otherwise it will lead to a PCM
configuration error.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-03 23:03:30 +02:00
Lars-Peter Clausen 637a935aab ASoC: Fix wm8753 register cache size and initialization
Register cache space was not being allocated for the final register,
causing bugs when it was used.  Allocate space for it.

Also ensure that the final register is displayed in sysfs.

[Commit message rewritten to document actual issue. -- broonie]

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-03 14:35:49 +01:00
Takashi Iwai e488fca79f Merge branch 'fix/hda' into topic/hda 2009-07-03 15:03:35 +02:00
Takashi Iwai aa202455ee ALSA: hda - Improve ASUS eeePC 1000 mixer
The mixer elements created for ASUS eeePC 1000 with ALC269 aren't
standard but strange words like "LineOut".  Rename the element names
to follow the standard one like "Headphone" and "Speaker".
Also, split the volumes to each so that the virtual master can control
them.

The alc269_fujitsu_mixer is removed because it's now identical with
the new eeepc mixer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-03 15:00:54 +02:00
Takashi Iwai 099db17e66 ALSA: hda - Add GPIO1 control at muting with HP laptops
HP laptops with AD1984A codecs (at least mobile models) need to set
GPIO1 appropriately to indicate the mute state.  The BIOS checks this
bit to judge whether the mute on or off is sent via F8 key.
Without changing this bit, the BIOS can be confused and may toggle
the mute wrongly.

Reference: Novell bnc#515266
	https://bugzilla.novell.com/show_bug.cgi?id=515266

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-03 14:58:59 +02:00
Lopez Cruz, Misael 6814044324 ASoC: TWL4030: Add tristate callbacks for HiFi and Voice
Add "set_tristate" callbacks for HiFi and Voice DAIs.
Machine drivers can enable and disable tristate for each
DAI with "snd_soc_dai_set_tristate" function.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-03 11:01:05 +01:00
Candelaria Villareal, Jorge 1b6ebdd7bb ASoC: Zoom2: Update twl4030_setup_data parameters
Add support for EXTMUTE in Zoom2 machine driver. This is necessary
to further reduce pop noise problem. Signal EXTMUTE is connected to
signal GPIO 153 in Zoom2 board.

In addition, change ramp delay value to 3 (218/161/109 ms). With
previous ramp delay value, pop noise was louder. With a longer value
the beep tone can be observed.

Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-03 11:01:05 +01:00
Grant Likely 0827d6ba0b ASoC: add locking to mpc5200-psc-ac97 driver
AC97 bus register read/write hooks need to provide locking, but the
mpc5200-psc-ac97 driver does not.  This patch adds a mutex around
the register access routines.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-03 10:41:47 +01:00
Grant Likely 07573534b0 ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared
When doing register reads, it is possible for there to be a stale
data ready bit set which will cause subsequent reads to return
prematurely with incorrect data.  This patch fixes the issues by
ensuring stale data is cleared before starting another transaction.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-03 10:41:46 +01:00
Mark Brown 1df892cba4 ASoC: Fix register cache initialisation for WM8753
The wrong register cache variable was being used to provide the size for
the memcpy(), resulting in a copy of only a void * of data.

Reported-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2009-07-03 10:41:45 +01:00
Takashi Iwai 8ab9e0af6d ALSA: hda - Manually expand alc882_init_verbs
Instead of expanding alc882_init_verbs to two elements via a macro,
manually expand to each entry.  This makes clear that some have already
the full slot for init_verbs array (currently 5).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-03 10:58:12 +02:00
Takashi Iwai cb6381225a ALSA: hda - Add missing mixer amp initialization for ALC882
After merting patch_alc882() and patch_alc883(), the initialization of
mixer amp 0x0b was missing in alc882_base_init_verbs[].
This is usually no critical problem, but it can disable the power-saving
as the default state, so better to put to mute these channels.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-03 10:57:10 +02:00
Takashi Iwai dbe45d0ce3 Merge branch 'fix/oxygen' into for-linus
* fix/oxygen:
  sound: virtuoso: fix Xonar D1/DX silence after resume
2009-07-02 17:48:34 +02:00
Takashi Iwai 4413c8d2be Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Add quirk for HP 6930p
  ALSA: hda - Add missing static to patch_ca0110()
2009-07-02 17:48:33 +02:00
Takashi Iwai cfbc78003e Merge branch 'fix/caiaq' into for-linus
* fix/caiaq:
  ALSA: usx2y - reparent sound device
  ALSA: snd_usb_caiaq: reparent sound device
2009-07-02 17:48:32 +02:00
Takashi Iwai dc8a4f3fc3 Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: Only disable pxa2xx-i2s clocks if we enabled them
  ASoC: OMAP: fix OMAP1510 broken PCM pointer callback
  ASoC: remove BROKEN from Efika and pcm030 fabric drivers
  ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig
2009-07-02 17:48:31 +02:00
Candelaria Villareal, Jorge 4e49ffd10f ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effect
According to TRM, an external FET controlled by a 1.8V output signal
can be used to reduce the pop-noise heard when the audio amplifier is
switched on. It is suggested that GPIO6 of TWL4030 be used, but any
other gpio can be used instead. This is indicated in machine driver
with the following twl4030_setup_data members:

-hs_extmute. Set to 1 if board has support for EXTMUTE.

-set_hs_extmute. Set to a callback funcion to control an external gpio
line. Set to NULL if MUTE[GPIO6] pin is used.

Codec driver takes care of enabling and disabling this output during
the headset pop attenuation sequence.

Also add a delay to let VMID settle in ramp up sequence.

Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-02 11:06:26 +01:00
Takashi Iwai 3f5d3465be ALSA: usx2y - reparent sound device
Fix the parent device to be the USB interface, not the USB device.
A similiar commit like 563c2bf59d.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-02 11:53:12 +02:00
Daniel Mack 563c2bf59d ALSA: snd_usb_caiaq: reparent sound device
The sound device instance needs to be a child of the USB interface, not
the USB device. Newer udev versions pay attention to that.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Lennart Poettering <lennart@poettering.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-02 10:58:33 +02:00
Clemens Ladisch 826390796d sound: virtuoso: fix Xonar D1/DX silence after resume
When resuming, we better take the DACs out of the reset state before
trying to use them.

Reference: kernel bug #13599
	http://bugzilla.kernel.org/show_bug.cgi?id=13599

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-02 10:35:35 +02:00
Takashi Iwai cd77538724 ALSA: ice1724 - Fix section mismatch
Now snd_vt1724_chip_reset() is used in the resume callback, thus
it cannot be __devinit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-02 10:30:50 +02:00
Mark Brown 2115d2c173 Merge branch 'for-2.6.31' into for-2.6.32 2009-07-01 21:45:47 +01:00
Mark Brown 1dcf98ff8e ASoC: Add WM8523 CODEC driver
The WM8523 is a high performance stereo DAC with integral charge
pump providing 2Vrms line driver outputs using a single 3.3V power
supply rail.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-01 18:47:45 +01:00
Mark Brown da9ff1f796 ASoC: Only disable pxa2xx-i2s clocks if we enabled them
The clock API can't cope with unbalanced enables and disables and
we only enable in hw_params() but try to disable in shutdown.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-01 18:31:32 +01:00
Takashi Iwai e5f1424807 ALSA: hda - Add Cirrus Logic CS420x support
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-01 18:11:44 +02:00
Takashi Iwai c785786933 Merge branch 'fix/hda' into topic/hda 2009-07-01 18:09:15 +02:00
Takashi Iwai ff84847171 ALSA: hda - Add quirk for HP 6930p
Added a quirk model=laptop for HP 6930p (103c:30dc) with AD1984A codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-01 18:08:01 +02:00
Takashi Iwai b5025c50b5 ALSA: hda - Allow FLOAT PCM format
So far, the FLOAT PCM format is used only exclusivley set.  But
this can be a combination with other formats.

This patch changes the parser to allow the FLOAT format in addition
to other PCM formats.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-01 18:05:27 +02:00
Takashi Iwai 1e1689536f ALSA: hda - Add missing static to patch_ca0110()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-01 08:34:32 +02:00
Mark Brown 8a123ee2a4 ASoC: WM8974 DAPM cleanups
Also implement AUX mode control.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 21:30:42 +01:00
Mark Brown 8b83a19367 ASoC: WM8974 cosmetic cleanups
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 21:30:35 +01:00
Mark Brown cb11d39ead ASoC: Use symmetric rates for WM8974
The chip has a single LRCLK.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 21:30:25 +01:00
Mark Brown a5f8d2f1b8 ASoC: Add WM8974 TLV information
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 19:34:32 +01:00
Mark Brown 91d0c3ecba ASoC: Refresh WM8974 PLL configuration
Move away from a fixed table to runtime calculation.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 19:34:23 +01:00
Mark Brown 33d81af4d1 ASoC: Declare 2 channels for WM8974
The device is a mono device but it can read two channel data and
many I2S controllers only understand 2 channels.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 19:34:16 +01:00
Mark Brown df1ef7a38d ASoC: Refresh WM8974 bias configuration
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 19:34:09 +01:00
Mark Brown 416356fcfa ASoC: Convert to dev_pm_ops
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 19:32:57 +01:00
Takashi Iwai 0d971c9fcf ALSA: hda - Fix input pinctl for ALC882 auto mode
alc882_auto_init_analog_input() sets the input pins to VREF-80 regardless
of the input pin types although it shouldn't be for line-in pins.

This patch fixes the behavior to follow other codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-30 16:11:11 +02:00
Takashi Iwai 4953550a6c ALSA: hda - Merge patch_alc882() and patch_alc883()
Merge patch_alc882() and patch_alc883() to the former one since both
codecs have fairly similar connections but just a slight difference.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-30 16:10:34 +02:00
Takashi Iwai b9791ca013 Merge branch 'topic/hda-patch' into topic/hda 2009-06-30 15:29:38 +02:00
Takashi Iwai 546861f1d3 Merge branch 'fix/hda' into topic/hda 2009-06-30 15:12:11 +02:00
Janusz Krzysztofik 1bdd741991 ASoC: OMAP: fix OMAP1510 broken PCM pointer callback
This patch tries to work around the problem of broken OMAP1510 PCM playback
pointer calculation by replacing DMA function call that incorrectly tries to
read the value form DMA hardware with a value computed locally from an
already maintained variable omap_runtime_data.period_index.

Tested on OMAP5910 based Amstrad Delta (E3) using work in progress ASoC
driver.

Based on linux-2.6-asoc.git v2.6.31-rc1.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 10:39:12 +01:00
Grant Likely 40d9ec14e7 ASoC: remove BROKEN from Efika and pcm030 fabric drivers
The needed spin_event_timeout() macro is now merged in from the
powerpc tree, so these drivers are no longer broken.  This reverts
commit 0c0e09e21a (ASoC: Mark MPC5200
AC97 as BROKEN until PowerPC merge issues are resolved)

Tested against 2.6.31-rc1.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 10:23:41 +01:00
Grant Likely 6a84c234da ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig
ALSA SoC drivers should be specify SND_SOC_AC97_BUS instead, not AC97_BUS.
Without SND_SOC_AC97_BUS defined, an AC97 device will not get correctly
registered on the AC97 bus, which prevents thinks like the WM9712
touchscreen driver from getting probed.

Tested against 2.6.31-rc1.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-30 10:23:40 +01:00
Jiri Slaby 9ea5ca75a2 sound: OSS: mpu401, fix deadlock
mpu401_chk_version is called with a spin lock already held. Don't take it
again.

Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-29 18:05:52 +02:00
Takashi Iwai 580c9d373c Merge branch 'fix/pci-vdevice' into for-linus
* fix/pci-vdevice:
  sound: Use PCI_VDEVICE for CREATIVE and ECTIVA
  sound: Use PCI_VDEVICE
2009-06-29 18:05:09 +02:00
Takashi Iwai e2cb0d95ba Merge branch 'fix/oxygen' into for-linus
* fix/oxygen:
  sound: oxygen: make mic volume control mono
2009-06-29 18:05:07 +02:00
Takashi Iwai 7d489c460e Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: cmi8330: fix MPU-401 PnP init copy&paste bug
2009-06-29 18:05:06 +02:00
Takashi Iwai 0baddbc895 Merge branch 'fix/lx6464es' into for-linus
* fix/lx6464es:
  ALSA: lx6464es - configure ethersound io channels
2009-06-29 18:05:05 +02:00
Takashi Iwai cdae219cec Merge branch 'fix/hda-samsung-p50' into for-linus
* fix/hda-samsung-p50:
  ALSA: hda - Fix support for Samsung P50 with AD1986A codec
  ALSA: hda - Generalize the pin-detect quirk for Lenovo N100
  ALSA: hda - Simplify AD1986A mixer definitions
2009-06-29 18:05:04 +02:00
Mark Brown 9ea21ebca1 ASoC: Fix WM8961 suspend function type
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-29 16:34:57 +01:00
Mark Brown 372a14a4a4 ASoC: Remove unreferenced wm8974_add_controls()
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-29 16:14:19 +01:00
Takashi Iwai 1d955ebd4d ALSA: hda - Add missing initializations for ALC268 and ALC269
During the changes to clean up / fix the realtek codec initialization
routines in commit 4a79ba34ca,
I forgot to add the check for ALC268 and ALC269.
This resulted in the missing EAPD and COEF setup for these codecs.

This patch adds the missing checks for these codecs.

Reference: bko#13633
	http://bugzilla.kernel.org/show_bug.cgi?id=13633

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-29 11:33:53 +02:00
Takashi Iwai a983b9e500 Merge branch 'fix/hda-samsung-p50' into topic/hda 2009-06-29 08:51:09 +02:00
Takashi Iwai b7b51141b4 ALSA: hda - Check "beep" hint
Check the hint "beep" in snd_hda_attach_beep_device() to avoid the beep
device creation if user doesn't want.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-29 08:37:00 +02:00
Ondrej Zary 74a0094cd9 ALSA: cmi8330: revert comments about AD1848 back
In ALSA 1.0.20, the comments were changed to say CMI8330 instead of AD1848.
The CMI8330 chip includes two codecs - AD1848 and SB16, so the comments were
correct and are misleading now. Revert them back.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-29 08:06:57 +02:00
Ondrej Zary c2a30d7118 ALSA: cmi8330: fix MPU-401 PnP init copy&paste bug
Fix copy&paste bug in PnP MPU-401 initialization.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-29 08:06:10 +02:00
Tony Vroon 684a88429c ALSA: hda - Line In for Acer Inspire 6530G model
The Line In connector is set up as PIN_IN by default, using
VREF_HIZ. It is connected to both ADCs, so add it to both
input selectors.
Also add the ability to use the input mix (on a SoundBlaster
one would call this "What You Hear").

Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-29 07:41:08 +02:00
Mark Brown c2caa4da46 ASoC: Fix widget powerdown on shutdown
We need to set the widget power state we want to implement.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-26 15:36:56 +01:00
Lopez Cruz, Misael e3c7dbb07c ASoC: Remove word "Switch" from Handsfree switch name
SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls,
removing word "Switch" from HandsfreeL/HandsfreeR widget name
for avoiding to duplicate it.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-26 10:39:52 +01:00
Atsushi Nemoto 647613e97f ASoC: txx9aclc: dynamically allocate dmaengine devname
Use kasprintf to allocate temporary devname string instead of a
fixed size string.
This fixes "FIXME" introduced on removal of BUS_ID_SIZE.

Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-25 21:06:30 +01:00
Clemens Ladisch 14744d7da2 sound: oxygen: make mic volume control mono
The microphone input and its volume register have only one channel, so
we have to make the corresponding mixer control a mono control.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-25 15:39:02 +02:00
Takashi Iwai 62b1653e29 Merge branch 'for-2.6.32' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2009-06-25 15:28:39 +02:00
Mark Brown d5fc3b5fe3 Merge branch 'for-2.6.31' into for-2.6.32 2009-06-25 13:58:37 +01:00
Mark Brown e2f551dacb ASoC: Add core suspend and resume callbacks to WM8961
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-25 13:57:59 +01:00
Tim Blechmann 7e895cfaad ALSA: lx6464es - configure ethersound io channels
as long as the io channel number is not set by the driver, the card
is not visible from the ethersound network

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-25 09:41:53 +02:00
Igor Chernyshev b40e953812 ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD2
I've built a small HTPC and had to add suspend/resume support in ice1724
driver. There seem to be 3 existing bugs related to that:

https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3748
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2314

Due to hardware (un)availability, I only enabled the fix for Audiotrak
Prodigy HD2 card, which is installed in my HTPC. However, most of my code
should be reusable in the future on other ice1724-based cards as well (as
long as people add card-specific peices of code). The fix is currently based
on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel).

Signed-off-by: Igor Chernyshev <igor.ch75+alsa at gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-25 09:31:07 +02:00
Joe Perches 0d7392e544 sound: Use PCI_VDEVICE for CREATIVE and ECTIVA
Here's a patch on top of the others to use CREATIVE and ECTIVA

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-25 08:53:57 +02:00
Joe Perches 28d27aae94 sound: Use PCI_VDEVICE
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-25 08:52:49 +02:00
Takashi Iwai dde6535686 ALSA: hda - Use model=acer-aspire-6530g for Acer Aspire 6930G
For Acer Aspire 6930G (1025:015e), acre-aspire-6530g model matches
obviously better.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-25 08:26:38 +02:00
Emilio López 320d592001 ALSA: hda - Fix acer-aspire-6530g model quirk
Fix the following bugs of acer-aspire-6530g model with ALC888:
- HP jack to mute all speaker outputs including LFE
- Make digital built-in mic working

Signed-off-by: Emilio López <buhitoescolar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-25 08:25:22 +02:00
Takashi Iwai 261c240740 ALSA: hda - Add pin-sense trigger when needed for Realtek codecs
Realtek codecs require the pin-sense trigger call before actually
reading the pin-sense.  Without this, the pin-detection might not be
done accurately.

This patch adds the pin-capability check and issues the trigger call
if required.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-25 08:15:03 +02:00
Takashi Iwai c912e7a580 ALSA: hda - Fix support for Samsung P50 with AD1986A codec
Samsung P50 requires the HP auto-muting unlike other Samsung models.
Added a new model=samsung-p50 to support this.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-24 14:14:34 +02:00
Takashi Iwai 03c405ad31 ALSA: hda - Generalize the pin-detect quirk for Lenovo N100
Add a new flag to ad_spec struct so that the same hack can be used for
any other models (if any).  This also allows other models to reuse the
auto-mute functions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-24 14:10:15 +02:00
Takashi Iwai 16d11a829e ALSA: hda - Simplify AD1986A mixer definitions
Split mixer element arrays of AD1986A models to several pieces so that
each model can share the same mixer arrays.
This removes lots of duplicated data.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-24 14:07:53 +02:00
Takashi Iwai 4ea6fbc8eb ALSA: hda - Add patch module option
Added the patch module option to apply a "patch" as a firmware to
modify pin configurations or give additional hints to the driver
before actually initializing and configuring the codec.

This can be used as a workaround when the BIOS doesn't give sufficient
information or give wrong information that doesn't match with the real
hardware setup, until it's fixed statically in the driver via a quirk.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-24 11:53:43 +02:00
Takashi Iwai a1e21c9078 ALSA: hda - Don't call snd_hda_codec_configure in snd_hda_codec_new()
The codec setup call via snd_hda_codec_configure() isn't necessarily
called in snd_hda_codec_new().  For the later added feature, it's better
to change the code flow like:
 - create all codec instances
 - configure each codec

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-24 11:52:59 +02:00
Mark Brown 517374704d ASoC: Add a shutdown callback
Ensure that the audio subsystem is powered down cleanly when the system
shuts down by providing a shutdown operation. This ensures that all the
components have been returned to an off state cleanly which should avoid
audio issues from partially charged capacitors or noise on digital inputs
if the system is restarted quickly.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Ben Dooks <ben-linux@fluff.org>
2009-06-23 23:48:53 +01:00
Linus Torvalds c488eef815 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:
  ALSA: hda - Make jack-plug notification selectable
  ALSA: ctxfi - Add PM support
  sound: seq_midi_event: fix decoding of (N)RPN events
  ALSA: hda - Add digital-mic support to ALC262 auto model
  ALSA: hda - Fix check of input source type for realtek codecs
  ALSA: hda - Add quirk for Sony VAIO Z21MN
  ALSA: hda - Get back Input Source for ALC262 toshiba-s06 model
  ALSA: hda - Fix unsigned comparison in patch_sigmatel.c
  ALSA: via82xx: add option to disable 500ms delay in snd_via82xx_codec_wait
  sound: fix check for return value in snd_pcm_hw_refine
  ALSA: ctxfi - Allow unknown PCI SSIDs
  ASoC: Blackfin: update the bf5xx_i2s_resume parameters
  ASoC: Blackfin: keep better track of SPORT configuration state
2009-06-23 10:45:39 -07:00
Lopez Cruz, Misael 30808ca751 ASoC: TWL4030: Correct bypass event for voice sidetone
Event for voice sidetone was being interpreted as an
analog HiFi bypass event because VSTPGA register offset
is less than ARXR2_APGA_CTL offset. Reordering the
register checks allows to handle voice digital bypass
event properly.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-23 10:57:38 +01:00
Lopez Cruz, Misael 328d0a138e ASoC: TWL4030: Add AVADC Clock Priority
AVDAC clk priority allows to determine the path ADC must
be connected when the codec is in option2 and both HiFi
and Voice paths are enabled.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-23 10:57:37 +01:00
Linus Torvalds 9e268beb92 Merge branch 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (49 commits)
  [ARM] idle: clean up pm_idle calling, obey hlt_counter
  [ARM] S3C: Fix gpio-config off-by-one bug
  [ARM] S3C64XX: add to_irq() support for EINT() GPIO
  [ARM] S3C64XX: clock.c: fix typo in usb-host clock ctrlbit
  [ARM] S3C64XX: fix HCLK gate defines
  [ARM] Update mach-types
  [ARM] wire up rt_tgsigqueueinfo and perf_counter_open
  OMAP2 clock/powerdomain: off by 1 error in loop timeout comparisons
  OMAP3 SDRC: set FIXEDDELAY when disabling SDRC DLL
  OMAP3: Add support for DPLL3 divisor values higher than 2
  OMAP3 SRAM: convert SRAM code to use macros rather than magic numbers
  OMAP3 SRAM: add more comments on the SRAM code
  OMAP3 clock/SDRC: program SDRC_MR register during SDRC clock change
  OMAP3 clock: add a short delay when lowering CORE clk rate
  OMAP3 clock: initialize SDRC timings at kernel start
  OMAP3 clock: remove wait for DPLL3 M2 clock to stabilize
  [ARM] Add old Feroceon support to compressed/head.S
  [ARM] 5559/1: Limit the stack unwinding caused by a kthread exit
  [ARM] 5558/1: Add extra checks to ARM unwinder to avoid tracing corrupt stacks
  [ARM] 5557/1: Discard some ARM.ex*.*exit.text sections when !HOTPLUG or !HOTPLUG_CPU
  ...
2009-06-22 14:56:13 -07:00
Takashi Iwai 3f67f6693c Merge branch 'topic/seq-midi-fix' into for-linus
* topic/seq-midi-fix:
  sound: seq_midi_event: fix decoding of (N)RPN events
2009-06-22 17:03:56 +02:00
Takashi Iwai 12bc738596 Merge branch 'topic/pcm-jiffies-check' into for-linus
* topic/pcm-jiffies-check:
  sound: fix check for return value in snd_pcm_hw_refine
2009-06-22 17:03:54 +02:00
Takashi Iwai b61eaa924c Merge branch 'topic/misc' into for-linus
* topic/misc:
  ALSA: via82xx: add option to disable 500ms delay in snd_via82xx_codec_wait
2009-06-22 17:03:52 +02:00
Takashi Iwai 11a93fd940 Merge branch 'topic/hda' into for-linus
* topic/hda:
  ALSA: hda - Make jack-plug notification selectable
  ALSA: hda - Add digital-mic support to ALC262 auto model
  ALSA: hda - Fix check of input source type for realtek codecs
  ALSA: hda - Add quirk for Sony VAIO Z21MN
  ALSA: hda - Get back Input Source for ALC262 toshiba-s06 model
  ALSA: hda - Fix unsigned comparison in patch_sigmatel.c
2009-06-22 17:03:48 +02:00
Takashi Iwai 366890df46 Merge branch 'topic/ctxfi' into for-linus
* topic/ctxfi:
  ALSA: ctxfi - Add PM support
  ALSA: ctxfi - Allow unknown PCI SSIDs
2009-06-22 17:03:44 +02:00
Takashi Iwai 70052413e7 Merge branch 'topic/asoc' into for-linus
* topic/asoc:
  ASoC: Blackfin: update the bf5xx_i2s_resume parameters
  ASoC: Blackfin: keep better track of SPORT configuration state
2009-06-22 17:03:39 +02:00
Takashi Iwai 8c8145b873 ALSA: hda - Make jack-plug notification selectable
Make the jack-plug notification via input layer selectable via Kconfig.
This is often unnecessary, and the similr function will be provided
using the ALSA control API in near future anyway.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-22 17:02:03 +02:00
Wai Yew CHAY 29959a09cc ALSA: ctxfi - Add PM support
Added the suspend/resume support to ctxfi driver.

The team tested on the following seems ok:
  AMD Athlon 64 3500+ / ASUS A8N-E / 512MB DDR ATI / Radeon X1300
  20k1 & 20k2 cards

Signed-off-by: Wai Yew CHAY <wychay@ctl.creative.com>
Singed-off-by: Ryan RICHARDS <ryan_richards@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-22 14:53:51 +02:00
Lopez Cruz, Misael c264301c77 ASoC: TWL4030: Fix voice interface clock masters
Voice interface of twl4030 codec supports: CBM_CFM and
CBS_CFS. It doesn't support CBS_CFM.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-By: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-22 11:24:04 +01:00
Mark Brown e370f45aeb Merge branch 'wm8961' into for-2.6.32 2009-06-22 11:08:26 +01:00
Clemens Ladisch 6423f9ea80 sound: seq_midi_event: fix decoding of (N)RPN events
When decoding (N)RPN sequencer events into raw MIDI commands, the
extra_decode_xrpn() function had accidentally swapped the MSB and LSB
controller values of both the parameter number and the data value.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-22 11:11:49 +02:00
Takashi Iwai 8c927b4acf ALSA: hda - Add digital-mic support to ALC262 auto model
Add the digital-mic support with ALC262 auto model.
The new ALC262 models have the digital mic at NID 0x12.  This widget
isn't checked in the current alc262_auto_create_analog_input_ctls()
since it's under 0x18.  So, just reuse the routine for alc269 to fix
the behavior.

But, it doesn't suffice: the digital mic is supported only with the
ADC0, we have to exclude other ADCs when d-mic is detected.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-22 11:06:18 +02:00
Takashi Iwai 0169b6b33b ALSA: hda - Fix check of input source type for realtek codecs
Fix the check of the input-source type by checking the widget type of
each capture-source item.  Since some codecs can have both the mixer
and selector types depending on the ADC, alc_mux_enum_put() needs to
check each widget.

With this change, spec->capture_style gets unneeded, so it's removed,
too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-22 11:06:05 +02:00
Takashi Iwai 376b508ffd ALSA: hda - Add quirk for Sony VAIO Z21MN
It needs model=toshiba-s06 to work with the digital-mic.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-06-22 11:05:56 +02:00
Takashi Iwai ae14ef68e8 ALSA: hda - Get back Input Source for ALC262 toshiba-s06 model
The commit f9e336f65b
    ALSA: hda - Unify capture mixer creation in realtek codes
removed the "Input Source" mixer element creation for toshiba-s06 model
because it contains a digital-mic input.

This patch take the control back.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2009-06-22 10:13:36 +02:00