Commit Graph

7000 Commits

Author SHA1 Message Date
Tony Lindgren d702d12167 Merge with mainline to remove plat-omap/Kconfig conflict
Conflicts:
	arch/arm/plat-omap/Kconfig
2010-03-01 14:19:05 -08:00
Linus Torvalds 524df55725 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: (252 commits)
  ASoC: Check progress when reporting periods from i.MX FIQ handler
  ASoC: Remove a unused variables from i.MX FIQ runtime data
  ALSA: hda - Add/fix ALC269 FSC and Quanta models
  ALSA: hda - Add ALC670 codec support
  OMAP4: PMIC: Add support for twl6030 codec
  ALSA: hda - remove unnecessary msleep on power state transitions
  usb/gadget/{f_audio,gmidi}.c: follow recent changes in audio.h
  ASoC: fsi: Modify over/under run error settlement
  ASoC: OMAP4: Add McPDM platform driver
  ASoC: OMAP4: Add support for McPDM
  ASoC: OMAP: data_type and sync_mode configurable in audio dma
  ALSA: hda - Add missing description in HD-Audio-Models.txt
  ALSA: add support for Macbook Air 2,1 internal speaker
  ALSA: usbaudio: consolidate header files
  ALSA: usbmixer: bail out early when parsing audio class v2 descriptors
  ALSA: usbaudio: implement basic set of class v2.0 parser
  ALSA: usbaudio: introduce new types for audio class v2
  ALSA: usbaudio: parse USB descriptors with structs
  ALSA: hda - enable snoop for Intel Cougar Point
  ALSA: hda - Remove identical definitions for macmini3 model
  ...
2010-03-01 08:58:44 -08:00
Takashi Iwai 6679ee1870 Merge branch 'topic/asoc' into for-linus 2010-03-01 12:38:59 +01:00
Takashi Iwai a91a4aa1ee Merge branch 'topic/hda' into for-linus 2010-03-01 12:38:54 +01:00
Takashi Iwai 12c2a682b5 Merge branch 'topic/misc' into for-linus 2010-03-01 12:38:49 +01:00
Takashi Iwai a86ba28583 Merge branch 'fix/misc' into for-linus 2010-03-01 12:38:39 +01:00
Manuel Lauss 05ae323180 MIPS/SOUND: Alchemy: DB1200 AC97+I2S audio support.
Machine driver for DB1200 AC97 and I2S audio systems, intended as a proper
reference asoc machine for Alchemy-based systems.  AC97/I2S can be selected
at boot time by setting switch S6.7.

Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Cc: Linux-MIPS <linux-mips@linux-mips.org>
Cc: alsa-devel@alsa-project.org
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
2010-02-27 12:53:01 +01:00
Manuel Lauss 963accbc82 MIPS: Alchemy: change dbdma to accept physical memory addresses
DMA can only be done from physical addresses; move the "virt_to_phys"
source/destination buffer address translation from the dbdma queueing
functions (since the hardware can only DMA to/from physical addresses)
to their respective users.

Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
2010-02-27 12:52:55 +01:00
Manuel Lauss ea071cc705 MIPS: Alchemy: remove dbdma compat macros
Remove dbdma compat macros, move remaining users over to default
queueing functions and -flags.

(Queueing function signature has changed in order to give
 a build failure instead of silent functional changes due
 to the no longer implicitly specified DDMA_FLAGS_IE flag)

Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
2010-02-27 12:52:54 +01:00
Linus Torvalds 6ebdc661b6 Merge branch 'next-devicetree' of git://git.secretlab.ca/git/linux-2.6
* 'next-devicetree' of git://git.secretlab.ca/git/linux-2.6: (41 commits)
  of: remove undefined request_OF_resource & release_OF_resource
  of/sparc: Remove sparc-local declaration of allnodes and devtree_lock
  of: move definition of of_chosen into common code.
  of: remove unused extern reference to devtree_lock
  of: put default string compare and #a/s-cell values into common header
  of/flattree: Don't assume HAVE_LMB
  of: protect linux/of.h with CONFIG_OF
  proc_devtree: fix THIS_MODULE without module.h
  of: Remove old and misplaced function declarations
  of/flattree: Make the kernel accept ePAPR style phandle information
  of/flattree: endian-convert members of boot_param_header
  of: assume big-endian properties, adding conversions where necessary
  of: use __be32 for cell value accessors
  of/flattree: use OF_ROOT_NODE_{SIZE,ADDR}_CELLS DEFAULT for fdt parsing
  of/flattree: use callback to setup initrd from /chosen
  proc_devtree: include linux/of.h
  of: make set_node_proc_entry private to proc_devtree.c
  of: include linux/proc_fs.h
  of/flattree: merge early_init_dt_scan_memory() common code
  of: add 'of_' prefix to machine_is_compatible()
  ...
2010-02-25 15:38:37 -08:00
Takashi Iwai a0b62329bb Merge branch 'for-2.6.34' of git://opensource.wolfsonmicro.com/linux-2.6-asoc into topic/asoc 2010-02-25 19:44:00 +01:00
Mark Brown b4e82b5b78 ASoC: Check progress when reporting periods from i.MX FIQ handler
Currently the i.MX FIQ handler is reporting periods as elapsed based
purely on a timer running in the CPU. This means that any clock
mismatch between the CPU and the audio subsystem can result in the
status reported to applications drifting away from the actual status
of the hardware. This is particularly likely at present since the
SSI driver is only capable of operating in slave mode so it's very
likely that the interface will be clocked from a different source.

Instead check the offset reported by the FIQ and only notify when we
have transferred at least one period, re-firing the timer if we didn't
do so. Also factor out the calculation of the timer expiry time for
make it a bit easier to experiment with.

Note that this only improves the situation, problems can still be
triggered.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-25 15:25:07 +00:00
Mark Brown 9e4a10d27e ASoC: Remove a unused variables from i.MX FIQ runtime data
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-25 15:25:07 +00:00
Kailang Yang 61c2d2b5e7 ALSA: hda - Add/fix ALC269 FSC and Quanta models
Specify proper quirk models for FSC and Quanta machines with ALC269 codec.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-25 08:49:06 +01:00
Kailang Yang 6227cdced0 ALSA: hda - Add ALC670 codec support
- Fixed alc_subsystem_id( ) typo and add new function.
   - !(ass & 0x100000)) ==> Delete this check. It is unnecessary check.
   - Add porti
- ALC670 support

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-25 08:48:44 +01:00
Zhang, Rui dd2b4a7abf ALSA: hda - remove unnecessary msleep on power state transitions
This will save ~15ms boot time.

The first 10ms sleep was introduced in commit d2595d86e5 for (buggy)
Cxt codecs, so better to limit the sleep to the problem hardware.

For the second 10ms sleep, the HDA spec says:

Power State[1:0]:
00: Node Power state (D0) is fully on.
01: Node Power state (D1) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms, excepting analog pass through circuits (e.g., CD analog
playback) which must remain fully on.
10: Node Power state (D2) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms. For modems, this is the "wake on ring" power state.
11: Node Power state (D3) allows for (does not require) lowest possible power consuming state under software
control. Note that any low power state set by software must retain sufficient operational capability to properly
respond to subsequent software Power State command.

So 10ms is actually the max wait time. It should be safe to
remove/reduce it and rely on the loop of 1ms-sleeps.

CC: Marc Boucher <marc@linuxant.com>
CC: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Zhang Rui <rui.zhang@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-24 09:12:57 +01:00
Ilkka Koskinen 83905c1345 ASoC: OMAP-McBSP: ASoC interface for McBSP sidetone
Add ASoC interface for OMAP McBSP2 and McBSP3 sidetones.

Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Tony Lindgren <tony@atomide.com>
2010-02-23 10:57:39 -08:00
Kuninori Morimoto 47fc9a0a80 ASoC: fsi: Modify over/under run error settlement
In current FSI driver, playback function cares only overrun,
and capture function cares only underrun.

But playback function should had cared about underrun,
and capture function should had cared about overrun too.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23 10:42:07 +00:00
Misael Lopez Cruz db72c2f897 ASoC: OMAP4: Add McPDM platform driver
McPDM platform driver is configured to use sDMA in order to transfer
to/from memory. Support for interfacing with ABE will be added later.

McPDM dai currently supports up to 4 downlink channels and 2 uplink
channels simultaneously, as well as 88.2 and 96 KHz, and a sample
size of 32 bits.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23 10:41:05 +00:00
Candelaria Villareal, Jorge b3b0b4580b ASoC: OMAP4: Add support for McPDM
McPDM is the interface between Phoenix audio codec
and the OMAP4430 processor. It enables data to be transfered
to/from Phoenix at sample rates of 88.4 or 96 KHz.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23 10:39:48 +00:00
Misael Lopez Cruz e17dd32f34 ASoC: OMAP: data_type and sync_mode configurable in audio dma
Allow client drivers to set the data_type (16, 32) and the
sync_mode (element, packet, etc) of the audio dma transferences.

McBSP dai driver configures it for a data type of 16 bits and
element sync mode.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-23 10:38:52 +00:00
Reimundo Heluani 76e6f5a9ef ALSA: add support for Macbook Air 2,1 internal speaker
Add support for Macbook Air 2,1 (late 2008) internal speaker and
headphones. Create a "mba21" model for snd-hda-intel.

Signed-off-by: Reimundo Heluani <rheluani@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 10:55:03 +01:00
Daniel Mack de48c7bc6f ALSA: usbaudio: consolidate header files
Use the definitions from linux/usb/audio.h all over the ALSA USB audio
driver and add some missing definitions there as well.

Use the endpoint attribute macros from linux/usb/ch9 and remove the own
things from sound/usb/usbaudio.h.

Now things are also nicely prefixed which makes understanding the code
easier.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:51:56 +01:00
Daniel Mack 7b8a043f26 ALSA: usbmixer: bail out early when parsing audio class v2 descriptors
This is just a quick hack that needs to be removed once the new units
defined by the audio class v2.0 standard are supported.

However, it allows using these devices for now, without mixer support.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:26 +01:00
Daniel Mack 53ee98fe8a ALSA: usbaudio: implement basic set of class v2.0 parser
This adds a number of parsers for audio class v2.0. In particular, the
following internals are different and now handled by the code:

* the number of streaming interfaces is now reported by an interface
  association descriptor. The old approach using a proprietary
  descriptor is deprecated.

* The number of channels per interface is now stored in the AS_GENERAL
  descriptor (used to be part of the FORMAT_TYPE descriptor).

* The list of supported sample rates is no longer stored in a variable
  length appendix of the format_type descriptor but is retrieved from
  the device using a class specific GET_RANGE command.

* Supported sample formats are now reported as 32bit bitmap rather than
  a fixed value. For now, this is worked around by choosing just one of
  them.

* A devices needs to have at least one CLOCK_SOURCE descriptor which
  denotes a clockID that is needed im the class request command.

* Many descriptors (format_type, ...) have changed their layout. Handle
  this by casting the descriptors to the appropriate structs.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:24 +01:00
Daniel Mack 8fee4aff8c ALSA: usbaudio: introduce new types for audio class v2
This patch adds some definitions for audio class v2.

Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have
different numerical representations in both standards, so there is need
for a _V1 add-on now. usbmixer.c is changed accordingly.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:20 +01:00
Daniel Mack 28e1b77308 ALSA: usbaudio: parse USB descriptors with structs
In preparation of support for v2.0 audio class, use the structs from
linux/usb/audio.h and add some new ones to describe the fields that are
actually parsed by the descriptor decoders.

Also, factor out code from usb_create_streams(). This makes it easier to
adopt the new iteration logic needed for v2.0.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:40:12 +01:00
Seth Heasley 32679f95ca ALSA: hda - enable snoop for Intel Cougar Point
This patch enables snoop, eliminating static during playback.
This patch supersedes the previous Cougar Point audio patch.

Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:15:37 +01:00
Takashi Iwai d01aecdf90 ALSA: hda - Remove identical definitions for macmini3 model
The channel mode definitions for macmini3 model are identical with mb5.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-23 08:07:15 +01:00
Takashi Iwai ad6cfc2ac7 Merge remote branch 'alsa/fixes' into fix/misc 2010-02-22 18:45:34 +01:00
Peter Ujfalusi b9dd94a87e ASoC: core: On resume also check the soc device state
Check the card->codec on soc_resume to detect if the soc
device is properly initialized.
If the card->codec is NULL, than do not continue the resume
operation, since the device is not initialized properly.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-22 14:39:42 +00:00
Clemens Ladisch bf30a4309d ALSA: via82xx: add quirk for D1289 motherboard
Add a headphones-only quirk for the Fujitsu Siemens D1289.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Marc Haber <mh+alsa201002@zugschlus.de>
Cc: <stable@kernel.org>

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-02-22 11:15:11 +01:00
Chris J Arges 40717382e0 ALSA: usbaudio Mbox support, output only
Signed-off-by: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 09:56:26 +01:00
Paul Menzel 0708cc582f ALSA: hda-intel: Add position_fix quirk for ASUS M2V-MX SE.
With PulseAudio and an application accessing an input device like `gnome-volume-manager` both have high CPU load as reported in [1].

Loading `snd-hda-intel` with `position_fix=1` fixes this issue. Therefore add a quirk for ASUS M2V-MX SE.

The only downside is, when now exiting for example MPlayer when it is playing an audio file a high pitched sound is outputted by the speaker.

$ lspci -vvnn | grep -A10 Audio
20:01.0 Audio device [0403]: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10)
	Subsystem: ASUSTeK Computer Inc. Device [1043:8290]
	Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx-
	Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx-
	Latency: 0, Cache Line Size: 64 bytes
	Interrupt: pin A routed to IRQ 17
	Region 0: Memory at fbffc000 (64-bit, non-prefetchable) [size=16K]
	Capabilities: <access denied>
	Kernel driver in use: HDA Intel

[1] http://sourceforge.net/mailarchive/forum.php?thread_name=1265550675.4642.24.camel%40mattotaupa&forum_name=alsa-user

Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:37:15 +01:00
Paul Menzel 2448158ed2 ALSA: Typo. s/distrubs/disturbs/
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:36:56 +01:00
Takashi Iwai 9d54f08bc7 ALSA: hda - Clean up Intel Mac unsol codes
Use the standard unsol_event callback with each setup callback for
IntelMac models with Realtek ALC885 codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:34:40 +01:00
Luke Yelavich e458b1fadf ALSA: hda - Add Macmini 3,1 support
BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989

Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The
pinout is almost identical to the mb5 quirk, except for no microphone and
the line-in mixer controls being on a different index. Everything works in
2ch mode, but as I am not sure what needs to be changed for 6ch mode, or
whether the Mac Mini's chip supports 6ch mode, I have simply duplicated
the code from the mb5 quirk for the mac mini chmode management. The new
model parameter for this quirk is "macmini3".

Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:27:57 +01:00
Daniel T Chen ba579eb7b3 ALSA: hda: Use 3stack quirk for Toshiba Satellite L40-10Q
BugLink: https://bugs.launchpad.net/bugs/524948

The OR has verified that the existing model=laptop-eapd quirk does not
function correctly but instead needs model=3stack.  Make this change
so that manual corrections to module-init-tools file(s) are not
required.

Reported-by: Lasse Havelund <lasse@havelund.org>
CC: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-22 08:15:21 +01:00
Florian Zumbiehl 04510a74bf ALSA: cs46xx - fix some typos
Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-18 08:12:30 +01:00
Florian Zumbiehl 7fb2d723e6 ALSA: cs46xx - Do test writes to register AC97_REC_GAIN in
snd_cs46xx_codec_reset() bypassing the register cache, so as to not
clobber the cached register value during resume.

Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-18 08:10:54 +01:00
Tony Lindgren 80c20d543d Merge branch 'omap-fixes-for-linus' into omap-for-linus 2010-02-17 14:08:58 -08:00
Mark Brown 6c5f1fed49 ASoC: Make pmdown_time a long
Fixes a warning.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-17 14:37:20 +00:00
Peter Ujfalusi e47c796d58 ASoC: TWL4030: Use codec defaults for Headset initial configuration
Disable the amplifiers for the headset outputs, and do not select
routings by default to the headset outputs.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-17 14:37:20 +00:00
Takashi Iwai 7fb3a069bc Merge branch 'fix/misc' into topic/misc
Conflicts:
	sound/pci/hda/patch_realtek.c
2010-02-17 14:24:46 +01:00
Takashi Iwai 9d3415a8cc Merge remote branch 'alsa/fixes' into fix/misc 2010-02-17 14:22:21 +01:00
Giuliano Pochini b721e68bdc ALSA: Echoaudio, fix Guru Meditation #00000005.48454C50
This patch fixes a division by zero error in the irq handler.

There is a small window between the hw_params() callback and when
runtime->frame_bits is set by ALSA middle layer. When another substream is
already running, if an interrupt is delivered during that window the irq
handler calls pcm_pointer() which does a division by zero. The patch below
makes the irq handler skip substreams that are initialized but not started
yet. Cc to Clemens Ladisch because he proposed an alternate fix.

For more information, please read the original thread in the linux-kernel
mailing list: http://lkml.org/lkml/2010/2/2/187

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-02-17 13:02:29 +01:00
Peter Ujfalusi 7833ae0edf ASoC: tlv320dac33: Correct the OSCSET calculation
OSCSET calculation was not correct in case of 44.1KHz
sampling rate.
With small adjustment both 48 and 44.1 KHz calculation
now gives the correct value.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-16 19:14:53 +00:00
Peter Ujfalusi e5e878c1c3 ASoC: tlv320dac33: Clearing FIFOFLUSH flag before playback
In repeated playback the FIFOFLUSH bit remained set, and
never has been cleared.
Clear it during the setup phase.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-16 19:14:52 +00:00
Mark Brown dbe21408b1 ASoC: Make pmdown_time runtime configurable
Provide a sysfs file allowing userspace to inspect and change the
pmdown_time setting at runtime.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-16 19:14:52 +00:00
Mark Brown 96dd362284 ASoC: Make pmdown_time a per-card setting
Make the pmdown_time a per-card setting rather than a global one,
initialised before the card initialisation runs. This allows cards
to override the default setting if it makes sense to do so (for
example, due to an unavoidable pop).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-16 19:14:52 +00:00