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Commit Graph

6340 Commits

Author SHA1 Message Date
peer chen
db32f99816 ALSA: hda_intel: Add the Linux device ID for NVIDIA HDA controller
Add the generic device ID for NVIDIA HDA controller.

Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-30 11:59:12 +01:00
Stas Sergeev
b71207e9dc ALSA: pcsp - Fix nforce workaround
The attached patch fixes the problems introduced in this commit:
http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdiff;h=eea0579fc85e64e9f05361d5aacf496fe7a151aa

- Fix nForce workaround by honouring the pointer_update var
- Revert "ns" to u64, as per the hrtimer API
- Revert to the zero-delay timer startup, since I can't reproduce any
  problem with it (please, give me the hint!)

Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-30 11:55:22 +01:00
Mark Brown
98078bf904 Merge branch 'for-2.6.32' into for-2.6.33 2009-10-30 10:36:23 +00:00
Kuninori Morimoto
07102f3cef ASoC: sh: FSI: Add capture support
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-30 10:35:30 +00:00
Kuninori Morimoto
9ddc9aa910 ASoC: sh: FSI: Remove DMA support
SuperH FSI device have the hardware limitation to use DMA.
If DMA is used, LCD output will be broken.
Maybe there are some solution. But I don't know how to do it now.
This patch remove DMA support and use soft transfer.

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-30 10:35:30 +00:00
Wu Zhangjin
97609458ce ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency
SND_CS5535AUDIO is available on Loongson(MIPS compatible) family
machines, and checked it with ARCH=x86_64, no relative compiling
warnings & errors, so, remove the platform dependency directly.

Reported-by: rixed@happyleptic.org
Acked-by: Andres Salomon <dilinger@collabora.co.uk>
Signed-off-by: Wu Zhangjin <wuzhangjin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-30 11:31:33 +01:00
Anuj Aggarwal
67e646cd7b ASoC: Modifying Kconfig/Makefile for AM3517 EVM
Modifying the Kconfig and Makefile in sound/soc/omap folder
to add support for OMAP3517 / AM3517 EVM in Alsa SoC.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 22:28:18 +00:00
Anuj Aggarwal
89e9abe781 ASoC: Adding OMAP3517 / AM3517 EVM support in ASOC
Adding support for OMAP3517 / AM3517 EVM in Alsa SoC.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 22:28:18 +00:00
Anuj Aggarwal
ed146aeb68 ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal
The pop-removal specific values are configured for TWL4030 codec
for OMAP3EVM through this patch.

Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 12:58:22 +00:00
Peter Ujfalusi
1c3d200271 ASoC: TWL4030: Add APLL supply for the capture path
Capture path also need the APLL enabled, adding DAPM_SUPPLY
for the Virtual ADCs.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 12:58:22 +00:00
Peter Ujfalusi
7729cf7493 ASoC: TWL4030: Change APLL powering sequence
It seams that certain part of the twl4030 codec needs the APLL
enabled before they are enabled.
Paths which has any digital processing needs need the APLL
enabled before they can function.
For example the vibra output will have some random data after
it is enabled and before the APLL also enabled.

If only analog components are in use (analog bypass), than it
seams, that the APLL does not need to be enabled. This lowers
the power consumption with around ~0.005A.

Adding DAPM_SUPPLY to the Digital playback route and also
to the capture route to enable and disable the APLL.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 12:58:21 +00:00
Jari Vanhala
86139a13ce ASoC: TWL4030: Vibra motor stop fix when it is driven with audio
This patch fixes vibrator playing incoherently, when driven
with audio. There is something wrong in switch 3 at
H-bridge and VIBRA_SET still affects PWM generator.
Slowest value fixes things.

Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 12:58:21 +00:00
Daniel Mack
7e1aa1dcd0 ASoC: CS4270: export de-emphasis filter as ALSA control
The CS4270 codec features an de-emphasis filter for compensation of
audio material filtered by an 50/15 uS algorithm. Not sure whether we
should always enable it for 44100Hz sampling frequency, but it should at
least be configurable by the user.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 12:58:13 +00:00
Mark Brown
26d95b6e30 ASoC: Minor SMDK64xx WM8580 cleanups
Fix up some comments, remove all enable_pin() calls (edge widgets
are all enabled by default) and mark the microphone as disabled by
default since it requires a resistor fit to connect it.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-29 12:55:56 +00:00
Ben Dooks
e3d8024891 ARM: S3C: Add info for supporting circular DMA buffers
The S3C64XX DMA implementation will work a lot better with the ability
to enqueue circular buffers as the hardware can do it's own linked-list
management.

Add a function s3c_dma_has_circular() to show that the system can do this
and a flag for the channel.

Update the s3c24xx/s3c64xx I2S DMA code to deal with this.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Acked-by: Mark Brown <broonie@@opensource.wolfsonmicro.com>
2009-10-28 18:22:57 +00:00
Peter Ujfalusi
2845fa13e5 ASoC: TWL4030: Change codec_muted to apll_enabled
codec_muted is missleading, change it to apll_enabled,
which is what it is doing: enabing and disabling the APLL.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-28 10:04:21 +00:00
Peter Ujfalusi
78e08e2f20 ASoC: TWL4030: Remove bypass tracking
Since ASoC core now handling the codec bias differently
there is no need to do the tracking of bypass switch states
anymore.

Handling of the common bit for analog loopbacks is done with
DAPM_SUPPLY for the bypass paths.

Now this bit is only enabled when there is a complete analog
bypass path, compared to the previous implementation, when the
global switch was enabled if there were any of the analog
bypass switch was on (regardless if there were complete path or
not)

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-28 10:04:21 +00:00
Kumar Gala
f8a3ae6c84 powerpc: Minor cleanup to sound/ppc/Kconfig
We can replace PPC32 || PPC64 as a dependancy with just PPC as all
powerpc platforms (32-bit and 64-bit) define PPC now.

Signed-off-by: Kumar Gala <galak@kernel.crashing.org>
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
2009-10-27 16:42:42 +11:00
Mark Brown
7dea7c01da ASoC: Add regulator support for WM8731
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-26 15:37:37 +00:00
Peter Ujfalusi
7a1fecf57f ASoC: TWL4030: Driver registration via twl4030_codec MFD
Change the way how the twl4030 soc codec driver is
loaded/probed.
Use the device probing via tlw4030_codec MFD device.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-25 17:15:10 +00:00
Peter Ujfalusi
1f0f9b67f9 ASoC: TWL4030: use the twl4030-codec.h for register descriptions
Remove the register descriptions from the twl4030.h file and use
the linux/mfd/twl4030-codec.h instead, which has the codec
related register descriptions also.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-25 17:15:10 +00:00
Janusz Krzysztofik
b214f11fb9 ASoC: Amstrad Delta: add info about the line discipline requirement to Kconfig help text
I thought it could be usefull to add some information on how to get the device
fully supported by loading a line discipline on the modem line.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-25 17:10:59 +00:00
Janusz Krzysztofik
0ffc11800c ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1
After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c,
omap_pcm_prepare() unconditionally calls:

        omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
        omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);

Current implementation of those two functions found in
arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at
all, so they both end with BUG() on that machine. That results in
ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta.

The patch corrects the problem by not calling those two functions when run on
OMAP1 class based machines.

Created against linux-2.6.32-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>
2009-10-22 11:47:14 +01:00
Peter Ujfalusi
017deee639 ASoC: tlv320dac33: typo fix in the header
Fix the definition of DAC33_LTM field, the LTM bits in
FIFO_IRQ_MODE_B register are starting at bit 6.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-21 19:08:21 +01:00
Janusz Krzysztofik
02624621a5 ASoC: Amstrad Delta minor cleanups
Hi Mark,

Here is a patch that corrects small omissions I have found in my code.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-21 19:08:21 +01:00
Mark Brown
9927f32771 Merge branch 'for-2.6.32' into for-2.6.33 2009-10-19 16:15:35 +01:00
Barry Song
02a06d3042 ASoC: Fix possible codec_dai->ops NULL pointer problems
Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves
then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai
if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc.
access the ops field in these DAIs, panic will happen.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-19 16:15:03 +01:00
Julia Lawall
4f066173fe ASoC: Move dereference after NULL test
If the NULL test on jack is needed, then the derefernce should be after the
NULL test.

A simplified version of the semantic match that detects this problem is as
follows (http://coccinelle.lip6.fr/):

// <smpl>
@match exists@
expression x, E;
identifier fld;
@@

* x->fld
  ... when != \(x = E\|&x\)
* x == NULL
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-19 16:12:35 +01:00
Manuel Lauss
8d567b6b44 ASoC: au1x: psc-ac97: reorganize timeouts
Codec read/write functions: wait 21us between the pokings of hardware.
Add timeouts to unbounded loops waiting for bits to change.

Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-19 16:12:31 +01:00
Manuel Lauss
e697cd410a ASoC: au1x: psc-ac97: verify correct codec register was read
Verify that the correct register has been received from the codec.

Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-19 16:12:30 +01:00
Peter Ujfalusi
d8707cecdf ASoC: TWL4030: Only update the needed bits in *set_dai_sysclk
Do not rewrite the whole register, but only update the needed
bits in set_dai_sysclk functions.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-19 16:12:17 +01:00
Mark Brown
3da8e6885e Merge branch 'for-2.6.32' into for-2.6.33 2009-10-15 15:02:14 +01:00
Peter Ujfalusi
c8bf93f0fe ASoC: Codec driver for Texas Instruments tlv320dac33 codec
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.

TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.

The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.

Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).

b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-15 15:02:04 +01:00
Igor Grinberg
640fb39e38 ASoC: finally enable support for eXeda and CM-X300
Signed-off-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
CC: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: alsa-devel@alsa-project.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-15 15:01:47 +01:00
Mark Brown
d2058b0cd0 ASoC: Remove snd_soc_suspend_device()
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-15 15:01:43 +01:00
Takashi Iwai
4b7348a159 ALSA: hda - Fix capture source checks for ALC662/663 codecs
The ALC662/663 parser calls wrongly alc880_auto_create_input_ctls()
to check the capture source selections.  This should be alc882, instead.

Reference: Novell bnc#546918
	http://bugzilla.novell.com/show_bug.cgi?id=546918

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-14 18:25:23 +02:00
Takashi Iwai
fb66ebd884 Merge branch 'fix/hda' into for-linus 2009-10-13 16:09:56 +02:00
Takashi Iwai
491dc0437d ALSA: hda - Allow all formats as default for Nvidia HDMI
In the commit f0613d5752
    ALSA: hda - Add full rates/formats support for Nvidia HDMI
the flag LIMITIED_RATE_FMT_SUPPORT was set as default, as I forgot
to clear before commit.

Let's enable all formats/rates as default.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-13 16:07:59 +02:00
Philby John
29a4f2d31c ALSA: aaci: ARM1176 aaci-pl041 AC97 register read timeout
After a reboot on an ARM1176 which amounts to a softreset, it has been
noted that the ALSA driver does not get registered and the probe fails
with the error "aaci-pl041 fpga:04: ac97 read back fail". In the process
of reading from a register the SL1TxBusy bit is set indicating that the
transceiver is busy and remains so until the default timeout occurs.
Set the Power down register 0x26 to an arbitrary value as specified in
the PL041 manual (page: 3-18) so that AACISL1TX/AACISL2TX registers take
their default state.

Signed-off-by: Philby John <pjohn@in.mvista.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-13 15:59:55 +02:00
Takashi Iwai
ccca7cdc1b ALSA: hda - Fix volume-knob setup for Dell laptops with STAC9228
The volume-knob widget needs to be set with 0x7f instead of 0xff
for Dell laptops with STAC9228 codec, too, like the previous commit.

Reference: Novell bnc#545013
	http://bugzilla.novell.com/show_bug.cgi?id=545013

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-13 15:32:21 +02:00
Takashi Iwai
54930531a0 ALSA: hda - Fix mute sound with STAC9227/9228 codecs
On FSC laptops, the sound gets muted gradually when the volume is chnaged.
This is due to the wrong volume-knob widget setup.  The delta bit (bit 7)
shouldn't be set for these devices.

This patch adds a new quirk to set the value 0x7f to the widget 0x24
instead of 0xff.

Reference: Novell bnc#546006
	http://bugzilla.novell.com/show_bug.cgi?id=546006

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-13 15:29:34 +02:00
Ben Dooks
ed9d040d40 ASoC: S3C: Remove <plat/audio.h>
Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h
as it provides nothing to the current kernel and is not in any future plans
for the system.

Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Simtec Linux Team <linux@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-13 13:33:53 +01:00
Eero Nurkkala
8e8b2d676f ASoC: Serialize access to dapm_power_widgets()
Access to damp_power_widgets() is assumed to be single-threaded.
Concurrent accesses to dapm_power_widgets() may result in
unpredictable behavior.

Calls from:
close_delayed_work()
soc_codec_close()
soc_pcm_prepare()
soc_suspend()
soc_resume_deferred()
to snd_soc_dapm_stream_event() do not have the codec->mutex
taken to cover the call to dapm_power_widgets(). Thus, take
the mutex in these paths also to assure single-threaded use
of dapm_power_widgets().

Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-13 13:33:02 +01:00
Takashi Iwai
9c6b8dcefe ALSA: bt87x - Add a whitelist for Pinnacle PCTV (11bd:0012)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-13 09:34:28 +02:00
Takashi Iwai
2d9c648295 ALSA: hda - Fix overflow of spec->init_verbs in patch_realtek.c
ALC861-VD lenovo model causes overflow of spec->init_verbs entries due to
the recent changes.  Simply increase the array size to avoid the overflow.

Reported-by: Luca Tettamanti <kronos.it@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-13 08:06:55 +02:00
Peter Ujfalusi
814b7963e5 ASoC: TPA6130A2: Make tpa6130a2_power as static
The power for the amplifier should be handled internally
by the tpa6130a2 driver.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-12 13:40:54 +01:00
David Henningsson
bd3c200e6d ALSA: ice1724 - Make call to set hw params succeed on ESI Juli@
If two streams are started immediately after one another (such as a
playback and a recording stream), the call to set hw params fails with
EBUSY. This patch makes the call succeed, so playback and recording will
work properly.

Signed-off-by: David Henningsson <launchpad.web@epost.diwic.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-11 18:07:21 +02:00
Robert Hancock
43189a38da ALSA: ice1724: Fix surround on Chaintech AV-710
Fix the num_total_dacs setting for Chaintech AV710. The existing comment
that only PSDOUT0 is connected is correct, but since the card is using
packed AC97 mode to send 6 channels to the codec, num_total_dacs should be
set to 6 and not 2. This allows 6-channel surround to work. Also clarify
a comment regarding the additional WM8728 codec on this card (it's connected
to the SPDIF output and always receives the same data).

Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-10 10:53:16 +02:00
Mark Brown
ebab1b1d07 ASoC: Minor fixups to tpa6130a2 driver
- Staticise ttpa6130a2_client.
- Remove unneeded cast from void.
- Use explict NULL rather than 0.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-09 19:13:47 +01:00
Peter Ujfalusi
493b67efff ASoC: TPA6130A2 amplifier driver
Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.

The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.

The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"

From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':

        {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
        {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},

Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-09 18:50:37 +01:00
Takashi Iwai
f0613d5752 ALSA: hda - Add full rates/formats support for Nvidia HDMI
Allow Nvidia HDMI to support more possible sample rates and formats.
At best, the really supported rates and formats should be determined
together with the negotiation with the HDMI receiver, but it's currently
not implemented yet (Nvidia stuff seems incompatible with HDMI 1.3
standard in this regard).  As a compromise, we enable all bits, assuming
that all recent devices do support such rates/formats.

Tested-by: Alan Alan <alanwww1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-09 17:44:08 +02:00
Nicolas Ferre
69d2c2ae1d ASoC: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ek
The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share
with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file
to extend the machine ID checking.

Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-09 12:41:55 +01:00
Takashi Iwai
378e869fd0 Merge branch 'fix/misc' into for-linus 2009-10-08 13:00:02 +02:00
Takashi Iwai
d2a764dd8e Merge branch 'fix/hda' into for-linus 2009-10-08 12:59:58 +02:00
Robert Hancock
1d4efa6650 ALSA: ice1724: increase SPDIF and independent stereo buffer sizes
Increase the default and maximum PCM buffer prellocation size for ice1724's
SPDIF and independent stereo pair outputs to 256K, which is the hardware's
maximum supported size. This allows a reduction in interrupt rate and
potentially power usage when an application is not latency-critical.

Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-08 11:48:11 +02:00
Krzysztof Helt
8dce39b895 ALSA: opl3: circular locking in the snd_opl3_note_on() and snd_opl3_note_off()
Fix following circular locking in the opl3 driver.

=======================================================
[ INFO: possible circular locking dependency detected ]
2.6.32-rc3 #87
-------------------------------------------------------
swapper/0 is trying to acquire lock:
 (&opl3->voice_lock){..-...}, at: [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]

but task is already holding lock:
 (&opl3->sys_timer_lock){..-...}, at: [<cca75169>] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth]

which lock already depends on the new lock.

the existing dependency chain (in reverse order) is:

-> #1 (&opl3->sys_timer_lock){..-...}:
       [<c02461d5>] validate_chain+0xa25/0x1040
       [<c0246aca>] __lock_acquire+0x2da/0xab0
       [<c024731a>] lock_acquire+0x7a/0xa0
       [<c044c300>] _spin_lock_irqsave+0x40/0x60
       [<cca75046>] snd_opl3_note_on+0x686/0x790 [snd_opl3_synth]
       [<cca68912>] snd_midi_process_event+0x322/0x590 [snd_seq_midi_emul]
       [<cca74245>] snd_opl3_synth_event_input+0x15/0x20 [snd_opl3_synth]
       [<cca4dcc0>] snd_seq_deliver_single_event+0x100/0x200 [snd_seq]
       [<cca4de07>] snd_seq_deliver_event+0x47/0x1f0 [snd_seq]
       [<cca4e50b>] snd_seq_dispatch_event+0x3b/0x140 [snd_seq]
       [<cca5008c>] snd_seq_check_queue+0x10c/0x120 [snd_seq]
       [<cca5037b>] snd_seq_enqueue_event+0x6b/0xe0 [snd_seq]
       [<cca4e0fd>] snd_seq_client_enqueue_event+0xdd/0x100 [snd_seq]
       [<cca4eb7a>] snd_seq_write+0xea/0x190 [snd_seq]
       [<c02827b6>] vfs_write+0x96/0x160
       [<c0282c9d>] sys_write+0x3d/0x70
       [<c0202c45>] syscall_call+0x7/0xb

-> #0 (&opl3->voice_lock){..-...}:
       [<c02467e6>] validate_chain+0x1036/0x1040
       [<c0246aca>] __lock_acquire+0x2da/0xab0
       [<c024731a>] lock_acquire+0x7a/0xa0
       [<c044c300>] _spin_lock_irqsave+0x40/0x60
       [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]
       [<cca751f0>] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth]
       [<c022ac46>] run_timer_softirq+0x166/0x1e0
       [<c02269e8>] __do_softirq+0x78/0x110
       [<c0226ac6>] do_softirq+0x46/0x50
       [<c0226e26>] irq_exit+0x36/0x40
       [<c0204bd2>] do_IRQ+0x42/0xb0
       [<c020328e>] common_interrupt+0x2e/0x40
       [<c021092f>] apm_cpu_idle+0x10f/0x290
       [<c0201b11>] cpu_idle+0x21/0x40
       [<c04443cd>] rest_init+0x4d/0x60
       [<c055c835>] start_kernel+0x235/0x280
       [<c055c066>] i386_start_kernel+0x66/0x70

other info that might help us debug this:

2 locks held by swapper/0:
 #0:  (&opl3->tlist){+.-...}, at: [<c022abd0>] run_timer_softirq+0xf0/0x1e0
 #1:  (&opl3->sys_timer_lock){..-...}, at: [<cca75169>] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth]

stack backtrace:
Pid: 0, comm: swapper Not tainted 2.6.32-rc3 #87
Call Trace:
 [<c0245188>] print_circular_bug+0xc8/0xd0
 [<c02467e6>] validate_chain+0x1036/0x1040
 [<c0247f14>] ? check_usage_forwards+0x54/0xd0
 [<c0246aca>] __lock_acquire+0x2da/0xab0
 [<c024731a>] lock_acquire+0x7a/0xa0
 [<cca748fe>] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]
 [<c044c300>] _spin_lock_irqsave+0x40/0x60
 [<cca748fe>] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]
 [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]
 [<c044c307>] ? _spin_lock_irqsave+0x47/0x60
 [<cca751f0>] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth]
 [<c022ac46>] run_timer_softirq+0x166/0x1e0
 [<c022abd0>] ? run_timer_softirq+0xf0/0x1e0
 [<cca75150>] ? snd_opl3_timer_func+0x0/0xc0 [snd_opl3_synth]
 [<c02269e8>] __do_softirq+0x78/0x110
 [<c044c0fd>] ? _spin_unlock+0x1d/0x20
 [<c025915f>] ? handle_level_irq+0xaf/0xe0
 [<c0226ac6>] do_softirq+0x46/0x50
 [<c0226e26>] irq_exit+0x36/0x40
 [<c0204bd2>] do_IRQ+0x42/0xb0
 [<c024463c>] ? trace_hardirqs_on_caller+0x12c/0x180
 [<c020328e>] common_interrupt+0x2e/0x40
 [<c0208d88>] ? default_idle+0x38/0x50
 [<c021092f>] apm_cpu_idle+0x10f/0x290
 [<c0201b11>] cpu_idle+0x21/0x40
 [<c04443cd>] rest_init+0x4d/0x60
 [<c055c835>] start_kernel+0x235/0x280
 [<c055c210>] ? unknown_bootoption+0x0/0x210
 [<c055c066>] i386_start_kernel+0x66/0x70

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-08 11:48:10 +02:00
Pavel Hofman
2bdf66331c ALSA: ICE1712/24 - Change the Multi Track Peak control (level meters) from MIXER to PCM type
* PLEASE NOTE - this change requires the corresponding update of
  envy24control for ice1712 - kind of an ABI change.
* The "Multi Track Peak" control is read-only level meters indicator.
* The control is VERY confusing to most users since it is currently displayed
  in regular mixers. E.g. alsamixer ignores its read-only status
  and allows changing the levels with keys which makes no sense.

Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-08 11:47:56 +02:00
Mark Brown
b727916a1f Merge branch 'for-2.6.32' into for-2.6.33 2009-10-08 10:45:09 +01:00
Takashi Iwai
defb5ab2e0 ALSA: hda - Fix yet another auto-mic bug in ALC268
Since patch_alc268() doesn't call set_capture_mixer() (due to its h/w
design different from other siblings), it needs to call fixup_automic_adc()
explicitly to set up the auto-mic routing.  Otherwise the indices for
int/ext mics aren't set properly.

Reference: Novell bnc#544899
	http://bugzilla.novell.com/show_bug.cgi?id=544899

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-07 15:12:27 +02:00
Mark Brown
6f775ba015 Merge branch 'upstream/wm8350' into for-2.6.32 2009-10-06 19:29:47 +01:00
Mark Brown
5b7dde3468 ASoC: WM8350 capture PGA mutes are inverted
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2009-10-06 19:27:56 +01:00
Mark Brown
b266002abf ASoC: Remove absent SYNC and TDM DAI format options from i.MX SSI
These should be handled via set_tdm_slot() now and cause build
failures as-is.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-06 19:26:57 +01:00
Mark Brown
907bc6c7fc Merge branch 'for-2.6.32' into for-2.6.33 2009-10-06 16:01:27 +01:00
Mark Brown
d2b247a8be ASoC: Add virtual enumeration support for DAPM muxes
Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-06 15:57:02 +01:00
Clemens Ladisch
2fb930b53f sound: via82xx: move DXS volume controls to PCM interface
The "VIA DXS" controls are actually volume controls that apply to the
four PCM substreams, so we better indicate this connection by moving the
controls to the PCM interface.

Commit b452e08e73 in 2.6.30 broke the
restoring of these volumes by "alsactl restore" that most distributions
use; the renaming in this patch cures that regression by preventing
alsactl from applying the old, wrong volume levels to the new controls.
http://bugzilla.kernel.org/show_bug.cgi?id=14151
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-06 14:58:58 +02:00
Mark Brown
3a65577d21 ASoC: Push DAPM enumeration register change test out
Don't assume that enumerations are backed by registers when updating
mux power.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-06 13:04:41 +01:00
Mark Brown
1642e3d42a ASoC: Simplify code for DAPM widget updates
We don't need to check for an event callback since we also check for
an appropriate event flag when applying mux status changes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-06 13:04:30 +01:00
Takashi Iwai
01d4825df6 ALSA: hda - Don't pick up invalid HP pins in alc_subsystem_id()
alc_subsystem_id() tries to pick up a headphone pin if not configured,
but this caused side-effects as the problem in commit
15870f05e9.

This patch fixes the driver behavior to pick up invalid HP pins; at least,
the pins that are listed as the primary outputs aren't taken any more.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-06 13:21:54 +02:00
Mark Brown
2a0f5cb327 Merge branch 'for-2.6.32' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into for-2.6.32 2009-10-06 12:11:09 +01:00
Takashi Iwai
f8f25ba356 ALSA: hda - Add a workaround for ASUS A7K
ASUS A7K needs additional GPIO1 bit setup; it has to be cleared.
Added a new fixup hook for this laptop so that it works as is.

Refernece: Novell bnc#494309
	http://bugzilla.novell.com/show_bug.cgi?id=494309

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-06 08:31:29 +02:00
Mark Brown
d4a8da910e Merge branch 'for-2.6.32' into for-2.6.33 2009-10-05 10:36:28 +01:00
Takashi Iwai
15870f05e9 ALSA: hda - Fix invalid initializations for ALC861 auto mode
The recent auto-parser doesn't work for machines with a single output
with ALC861, such as Toshiba laptops, because alc_subsystem_id() sets
the hp_pins[0] while it's listed in line_outs[0].
This ends up with the doubled initialization of the same mixer widget,
and it mutes the DAC route because hp_pins has no DAC assigned.

To fix this problem, just check spec->autocfg.hp_outs and speaker_outs
so that they are really detected pins.

Reference: Novell bnc#544161
	http://bugzilla.novell.com/show_bug.cgi?id=544161

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-05 08:29:49 +02:00
Linus Torvalds
f0a221ef47 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of ssh://master.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (21 commits)
  ALSA: usb - Use strlcat() correctly
  ALSA: Fix invalid __exit in sound/mips/*.c
  ALSA: hda - Fix / improve ALC66x parser
  ALSA: ctxfi: Swapped SURROUND-SIDE mute
  sound: Make keywest_driver static
  ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-B1VP
  ALSA: hda - Fix digita/analog mic auto-switching with IDT codecs
  ASoC: fix kconfig order of Blackfin drivers
  ALSA: hda - Added quirk to enable sound on Toshiba NB200
  ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2
  ALSA: Don't assume i2c device probing always succeeds
  ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-T350P
  ALSA: echoaudio - Re-enable the line-out control for the Mia card
  ALSA: hda - Resurrect input-source mixer of ALC268 model=acer
  ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
  ALSA: hda - Add HP Pavilion dv4t-1300 to MSI whitelist
  ALSA: hda - CD-audio sound for hda-intel conexant benq laptop
  ASoC: DaVinci: Correct McASP FIFO initialization
  ASoC: Davinci: Fix race with cpu_dai->dma_data
  ASoC: DaVinci: Fix divide by zero error during 1st execution
  ...
2009-10-03 11:25:30 -07:00
Takashi Iwai
7fa9742bf7 Merge branch 'fix/hda' into for-linus 2009-10-03 18:31:33 +02:00
Takashi Iwai
a1cb9cd697 Merge branch 'fix/asoc' into for-linus 2009-10-03 18:31:22 +02:00
Jonathan Cameron
e655a43544 ASoC: wm8940: Fix check on error code form snd_soc_codec_set_cache_io
Fix for typo in commit 8d50e447d1
ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs

Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-02 16:10:55 +01:00
Takashi Iwai
08d1e63508 ALSA: usb - Use strlcat() correctly
Don't pass the advanced position to strlcat() but just gives the buffer
head position so that the max size limit can be checked correctly.
Introduced a new helper function to standaralize strlcat() calls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-02 14:06:08 +02:00
Peter Ujfalusi
ce3e3737a3 ASoC: Improve the debugfs hierarchy
Change the way the debugfs entries are created:
If the codec->dev is valid, than use:
debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/

if the codec->dev is NULL:
debugfs/asoc/{codec->name}/

as root for the debugfs entries.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-02 11:24:21 +01:00
Peter Ujfalusi
eaeae5d9b7 ASoC: Fix SND_SOC_DAPM_LINE handling
Since the SND_SOC_DAPM_LINE can be input or output, additional check is
needed in order to determine if the widget is connected as input or
output.
When checking for connected outputs, if the widget is line, than check
if the sources list is not empty (line is connected as output)
For input endpoint check, when the widget is line, also check if the
sinks list is not empty (line is connected as input).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-02 11:23:21 +01:00
Takashi Iwai
2f229a31aa ALSA: Fix invalid __exit in sound/mips/*.c
The remove callback has to be marked as __devexit, as the dynamic unbind
is possible.

Reported-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-02 11:06:16 +02:00
Takashi Iwai
7085ec12a6 ALSA: hda - Fix / improve ALC66x parser
The auto-parser for ALC662/663/272 codecs doesn't work properly when
a speaker is connected to mono NID 0x17, and doesn't handle the dynamic
DAC assignment properly.

This patch fixes the issues and also improves the assignment of DACs
so that HP and speakers can have independent volume controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-02 09:03:58 +02:00
Sven Eckelmann
3b04691c2b ALSA: ctxfi: Swapped SURROUND-SIDE mute
On Soundblaster X-FI Titenium with emu20k2 the SIDE and SURROUND mute
functions are swapped.
It was checked with 'speaker-test -c 8 -s 3' and (un)mute surround or
'speaker-test -c 8 -s 7' and (un)mute side. The volume seems not
to be affected and works as expected.

Signed-off-by: Sven Eckelmann <sven.eckelmann@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-02 07:45:55 +02:00
Jean Delvare
a656cbf07f sound: Make keywest_driver static
I can't see any reason for struct i2c_driver keywest_driver to not be
static.

Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-02 07:38:37 +02:00
Daniel T Chen
ebb6f6acbc ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-B1VP
BugLink: https://bugs.launchpad.net/bugs/410933

This Sony VAIO model also needs External Amplifier unmuted for audible
playback, so make sure we set the inv_eapd quirk.

Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-02 07:35:26 +02:00
Takashi Iwai
02d3332285 ALSA: hda - Fix digita/analog mic auto-switching with IDT codecs
When the auto-mic switching between an analog and a digital mic is
needed with IDT codecs, the current driver doesn't reset the connection
of the digital mux.

This patch fixes the behavior by checking both mux connections properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-01 16:38:11 +02:00
Peter Ujfalusi
88439ac793 ASoC: add support for multiple cards/codecs in debugfs
In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:

debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
                                                  /dapm_pop_time
                                                  /dapm/{widgets}

With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-01 12:13:04 +01:00
Mark Brown
17c86a3207 Merge branch 'for-2.6.32' into for-2.6.33 2009-10-01 11:35:11 +01:00
Mark Brown
f36c4045db Merge remote branch 'takashi/topic/asoc' into for-2.6.33 2009-10-01 11:33:37 +01:00
Mark Brown
834eb6c599 Merge remote branch 'takashi/fix/asoc' into for-2.6.32 2009-10-01 11:33:26 +01:00
Barry Song
df1246d84a ASoC: fix kconfig order of Blackfin drivers
Some of the Blackfin options don't directly follow the kconfig options
they depend on, so kconfig is unable to display the proper tree.  So sort
the options such they expand/collapse properly.

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-10-01 11:27:27 +01:00
Manoj Iyer
3db6c037c6 ALSA: hda - Added quirk to enable sound on Toshiba NB200
Patch was tested on Toshiba NB200 and is found to enable sound.

Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-01 10:24:08 +02:00
Takashi Iwai
140318aaa9 ASoC: Fix snd_soc_dai_set_pll() calls in neo1973_*.c
Fix the missing argument of snd_soc_dai_set_pll() in neo1973_*.c,
which was forgotten in the commit 85488037bb.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-01 08:42:27 +02:00
Takashi Iwai
c877c25170 ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2
wm8940 requires I2C.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-01 08:33:47 +02:00
Takashi Iwai
18c4078489 ALSA: Don't assume i2c device probing always succeeds
The client->driver pointer can be NULL when i2c-device probing fails
in i2c_new_device().  This patch adds the NULL checks for client->driver
and return the error instead of blind assumption of driver availability.

Reported-by: Tim Shepard <shep@alum.mit.edu>
Cc: Jean Delvare <khali@linux-fr.org>
Cc: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-01 07:46:33 +02:00
Daniel T Chen
5da5b6f9e9 ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-T350P
BugLink: https://bugs.launchpad.net/bugs/410933

This Sony VAIO model needs External Amplifier unmuted for audible
playback, so make sure we set the inv_eapd quirk.

Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-01 07:43:05 +02:00
Takashi Iwai
bb26276744 ASoC: Fix build errors of wm8711.c with SPI
Fix a couple of typos and a missing header file inclusion to build wm8711.c
properly with CONFIG_SPI_MASTER.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-10-01 07:39:45 +02:00
Mark Brown
aa983d9d63 ASoC: Factor out analogue platform data from WM8993
This is also shared with newer CODECs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-30 15:51:37 +01:00
Mark Brown
4c0bccbe66 Merge branch 'upstream/wm8974' into for-2.6.33 2009-09-30 15:48:38 +01:00
Mark Brown
c36b2fc73a ASoC: Clean up WM8974 PLL configuration
Don't use a static for WM8974 PLL factors - we don't support more than
one device so it won't happen but no sense in leaving the race condition
hanging around.  Also, pre_div is a single bit and it's a bit simpler if
we move the handling of the factor of 4 in the output into the
coefficient setup.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-30 15:45:25 +01:00
Chaithrika U S
4fa9c1a595 ASoC: DaVinci: McASP FIFO related updates
The DMA params for McASP with FIFO has been updated so that it works for
various FIFO levels. A member- 'fifo_level' has been added to the DMA
params data structure. The fifo_level can be adjusted by the tx[rx]_numevt
platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This
implementation has been tested for numevt values 1, 2, 4, 8.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-30 13:43:55 +01:00
Giuliano Pochini
392bf2f1ba ALSA: echoaudio - Re-enable the line-out control for the Mia card
Mia has an undocumented line-out control, and it has to be exposed.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-30 08:26:45 +02:00
Takashi Iwai
432fd13359 ALSA: hda - Resurrect input-source mixer of ALC268 model=acer
In the commit fdbc66266c, I mistakenly
replaced the capture mixer array for ALC268_ACER to nosrc version
although this should be kept to alt_mixer.  Now fixed back.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-30 08:13:44 +02:00
Miguel de Barros
a72cb4bc85 ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
Reference: ALSA bug #0004614
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614

port-A (0x11)      - front hp-out
port-D (0x12)      - rear line out
port-E (0x1c)      - front mic-in
port-F (0x16)      - Internal speakers
digital-mic (0x17) - Internal mic

init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware

Signed-off-by: Miguel de Barros <miguel.de.barros@bluewin.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-29 09:15:05 +02:00
Alexey Dobriyan
f0f37e2f77 const: mark struct vm_struct_operations
* mark struct vm_area_struct::vm_ops as const
* mark vm_ops in AGP code

But leave TTM code alone, something is fishy there with global vm_ops
being used.

Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2009-09-27 11:39:25 -07:00
Graeme Gregory
f34762b647 ASoC: pxa-ssp increase max_channels to 8
When running in TDM mode there can be more than 2 channels used. Datasheet has
figures for upto 8 channels so increase max_channels on all SSP interfaces to
this figure.

Signed-off-by: Graeme Gregory <dp@xora.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-25 10:17:33 -07:00
Russell King
baea7b946f Merge branch 'origin' into for-linus
Conflicts:
	MAINTAINERS
2009-09-24 21:22:33 +01:00
Daniel T Chen
3d80dcaca1 ALSA: hda - Add HP Pavilion dv4t-1300 to MSI whitelist
BugLink: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=547994

Enable MSI by default for this Pavilion model.

Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-24 12:14:37 +02:00
Lukasz Marcinowski
22e141300e ALSA: hda - CD-audio sound for hda-intel conexant benq laptop
After puting a cd-audio inside my laptop there was no sound out here,
so I decided to install alsa-driver with debug level and setup a
model=test, it didn't help, but then I look at source code and added
this few lines, now cd-audio is working both when playback/recording.

[Additional minor fixes of mixer element/item names by tiwai]

Signed-off-by: Lukasz Marcinowski <nowymarluk@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-24 09:49:25 +02:00
Mark Brown
2c9ee33d37 Merge branch 'for-2.6.32' into for-2.6.33 2009-09-23 10:54:06 -07:00
Chaithrika U S
539d3d8cbe ASoC: DaVinci: Correct McASP FIFO initialization
McASP write FIFO registers should be modified for playback and read FIFO
registers for capture. Check the PCM mode before manipulating the
FIFO registers. Currently, irrespective of playback/capture both the
FIFOs are enabled or disbaled. This resulted in errors in audio loopback
mode.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-23 10:37:08 -07:00
Troy Kisky
92e2a6f682 ASoC: Davinci: Fix race with cpu_dai->dma_data
This patch removes references to cpu_dai->dma_data.
It makes struct davinci_pcm_dma_params part of
struct davinci_mcbsp_dev or struct davinci_audio_dev.

It removes the unused name variable from davinci_pcm_dma_params.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-23 10:08:57 -07:00
Troy Kisky
81ac55aa14 ASoC: DaVinci: Fix divide by zero error during 1st execution
When both playback and capture stream were open
davinci_i2s_hw_params was setting parameters for
the wrong stream. The fix for davinci_i2s_hw_params
is sufficient, but it looks like a race still happens
in davici_pcm_open. This patch also makes the race smaller
but the next patch provides a better fix.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-23 10:08:56 -07:00
Linus Torvalds
0c9af28074 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:
  ALSA: lx6464es - remove unused struct member
  ALSA: lx6464es - cleanup of rmh message bus function
  ALSA: pcm - Simplify snd_pcm_drain() implementation
2009-09-23 10:04:14 -07:00
Linus Torvalds
fe61c99a12 Merge branch 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ASoC: wm8753: fix mapping when MONOMIX is set to Stereo
  ASoC: some minor changes for AD1836 and AD1938 codec drivers
  ASoC: DaVinci: Fixes to McASP configuration
  ASoC: Blackfin I2S: fix resuming when device hasn't been used
  ASoC: Blackfin I2S: add lost platform_device parameter to resume function
  ASoC: fix typos in Blackfin headers
  ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long
  ASoC: Blackfin AC97: add a few missing multichannel define handling
2009-09-23 10:02:43 -07:00
Cliff Cai
df0fd5e5e1 ASoC: Blackfin: fix inverted handling of SPORT0 on PORT F/G
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-23 09:10:01 -07:00
Barry Song
766df6d98f ASoC: Blackfin I2S: use dai state rather than local counter
Since the active field of the dai already tells us the stream activity,
the local counter variable is redundant and can be replaced.

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-23 09:08:25 -07:00
Russell King
ae19ffbadc Merge branch 'master' into for-linus 2009-09-22 21:01:40 +01:00
Phil Vandry
877ae70763 ASoC: wm8753: fix mapping when MONOMIX is set to Stereo
When MONOMIX is set to Stereo, Left PGA was not powered on but should be.
Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue.

Signed-off-by: Phil Vandry <vandry@TZoNE.ORG>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-21 16:58:43 -07:00
Barry Song
98235a4bb0 ASoC: some minor changes for AD1836 and AD1938 codec drivers
1. delete redundant assignment to bus field in spi_driver structure
2. fix lost assignment to set_bias_level entry in ad1938 codec dai
3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836"

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-21 16:58:33 -07:00
Russell King
28f9f19db9 Merge branch 'devel' of git://git.kernel.org/pub/scm/linux/kernel/git/ycmiao/pxa-linux-2.6 into devel 2009-09-21 16:02:30 +01:00
Joe Perches
a419aef8b8 trivial: remove unnecessary semicolons
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2009-09-21 15:14:58 +02:00
Robert P. J. Day
786d8ca341 trivial: Remove commented out usage of dead MODULE_PARM() in swarm_cs4297a
Get rid of that commented usage of the now defunct MODULE_PARM macro.

Signed-off-by: Robert P. J. Day <rpjday@crashcourse.ca>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2009-09-21 15:14:54 +02:00
Tim Blechmann
8fdc9e870c ALSA: lx6464es - remove unused struct member
we cannot set the sampling rate of the device, but can only read it
from the board, so we don't need the member for it.

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-21 15:13:57 +02:00
Tim Blechmann
95eff499c9 ALSA: lx6464es - cleanup of rmh message bus function
the rmh bus is not used asynchronously, so it is safe to remove the
specific code pieces.

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-21 15:13:53 +02:00
Takashi Iwai
d3a7dcfeeb ALSA: pcm - Simplify snd_pcm_drain() implementation
Simplify snd_pcm_drain() implementation and avoid unneeded array-
allocation for waitqueues.  Instead, one waitqueue is used for the
first draining stream, and wait until all streams finished.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-21 15:13:09 +02:00
Mark Brown
e0274b0a30 Merge branch 'upstream/wm8711' into for-2.6.33 2009-09-21 04:54:21 -07:00
Mark Brown
d62ab35894 ASoC: Convert soc-cache to use C99 style initialisers for the table
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-21 04:21:47 -07:00
Kay Sievers
e454cea20b Driver-Core: extend devnode callbacks to provide permissions
This allows subsytems to provide devtmpfs with non-default permissions
for the device node. Instead of the default mode of 0600, null, zero,
random, urandom, full, tty, ptmx now have a mode of 0666, which allows
non-privileged processes to access standard device nodes in case no
other userspace process applies the expected permissions.

This also fixes a wrong assignment in pktcdvd and a checkpatch.pl complain.

Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2009-09-19 12:50:38 -07:00
jassi brar
d0f5fa17aa ASoC: Support WM8580 based audio subsystem on SMDK64xx machines
New machine driver for WM8580 I2S i/f on SMDK64XX.
By default SoC-Slave is set and WM8580 is configured to use it's
PLLA to generate clocks from a 12MHz crystal attached to WM8580.

[Added dependency on BROKEN since the IISv4 interface hasn't been merged
yet, fixed the PLL API usage and removed the disabling of the PLL in the
hw_free function since that'll break simultaneous playback and record
 -- broonie.]

Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-19 16:28:54 +01:00
Linus Torvalds
6f128fa344 Merge branch 'davinci-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci
* 'davinci-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci: (62 commits)
  DaVinci: DM646x - platform changes for vpif capture and display drivers
  davinci: DM355 - platform changes for vpfe capture
  davinci: DM644x platform changes for vpfe capture
  davinci: audio: move tlv320aic33 i2c setup into board files
  DaVinci: EDMA: Adding 2 new APIs for allocating/freeing PARAMs
  DaVinci: DM365: Adding entries for DM365 IRQ's
  DaVinci: DM355: Adding PINMUX entries for DM355 Display
  davinci: Handle pinmux conflict between mmc/sd and nor flash
  davinci: Add NOR flash support for da850/omap-l138
  davinci: Add NAND flash support for DA850/OMAP-L138
  davinci: Add MMC/SD support for da850/omap-l138
  davinci: Add platform support for da850/omap-l138 GLCD
  davinci: Macro to convert GPIO signal to GPIO pin number
  davinci: Audio support for DA850/OMAP-L138 EVM
  davinci: Audio support for DA830 EVM
  davinci: Correct the number of GPIO pins for da850/omap-l138
  davinci: Configure MDIO pins for EMAC
  DaVinci: DM365: Add Support for new Revision of silicon
  DaVinci: DM365: Fix Compilation issue due to PINMUX entry
  DaVinci: EDMA: Updating default queue handling
  ...
2009-09-18 09:20:37 -07:00
Mark Brown
9f072b7b22 Merge branch 'for-2.6.32' into for-2.6.33 2009-09-18 15:09:44 +01:00
Jassi
b1cd6b9ec7 ASoC: Return correct codec clock in s3c64xx-i2s
Instead of always returnig pointer to the 'audio-bus' clock,
check which clock is used to generate internal clocks and
then return it's pointer.

Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-18 15:09:37 +01:00
Chaithrika U S
0c31cf3e4a ASoC: DaVinci: Fixes to McASP configuration
McASP register settings are not correct for DSP mode of operation.
There is a channel swap initally. This patch provides fixes to
the register values for proper working.

Tested on DA830/OMAP-L137 EVM, DM6467 EVM.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-18 15:08:31 +01:00
Cliff Cai
ad80efc469 ASoC: Blackfin I2S: fix resuming when device hasn't been used
If the sound system hasn't been utilized yet and we suspend, then we
attempt to save/restore using state that doesn't exist.  So use a global
handle instead to reconfigure properly.

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>
2009-09-18 15:07:19 +01:00
Linus Torvalds
b938fb6f49 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 - Fix MSI GX620 mixer
  ASoC: remove unused #include <linux/version.h>
  ASoC: S3C lrsync function made to work with IRQs disabled.
  ALSA: hda - Fix Dell S14 pin setup
  ALSA: hda - Fix IDT92HD83* codec setup
  ASoC: Fix display of stream name in DAPM debugfs
  ALSA: hda - Add support for HP dv6
  ALSA: hda - Fix HP/line-out initialization with IDT/STAC codecs
  ALSA: hda - Set default GPIO for IDT92HD71bxx
  ALSA: hda - Set default GPIO for STAC/IDT codecs
  ASoC: Clean up error handling in MPC5200 DMA setup
  ALSA: hda - Add missing model=auto entry for ALC269
2009-09-17 13:21:52 -07:00
Takashi Iwai
87bfa1dbfb Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Fix MSI GX620 mixer
  ALSA: hda - Fix Dell S14 pin setup
  ALSA: hda - Fix IDT92HD83* codec setup
  ALSA: hda - Add support for HP dv6
  ALSA: hda - Fix HP/line-out initialization with IDT/STAC codecs
  ALSA: hda - Set default GPIO for IDT92HD71bxx
  ALSA: hda - Set default GPIO for STAC/IDT codecs
  ALSA: hda - Add missing model=auto entry for ALC269
2009-09-17 21:08:56 +02:00
Takashi Iwai
673bca1906 Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: remove unused #include <linux/version.h>
  ASoC: S3C lrsync function made to work with IRQs disabled.
  ASoC: Fix display of stream name in DAPM debugfs
  ASoC: Clean up error handling in MPC5200 DMA setup
2009-09-17 21:08:53 +02:00
Takashi Iwai
b99dba34dc ALSA: hda - Fix MSI GX620 mixer
The headphone and speaker mixer elements aren't properly set for
MSI GX620 with targa-8ch-dig quirk.
Also fixed the speaker volume control for other ALC883-targa quirks,
too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-17 18:23:00 +02:00
Barry Song
fab19bae0c ASoC: Blackfin I2S: add lost platform_device parameter to resume function
Commit dc7d7b830e trimmed the platform_device parameter from all of the
suspend functions, but it also accidentally removed it from the resume
function in the Blackfin I2S driver.  So restore it.

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-17 10:51:35 +01:00
Barry Song
7d156a25bd ASoC: fix typos in Blackfin headers
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-17 10:51:35 +01:00
Mike Frysinger
d75150d7c4 ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-17 10:51:35 +01:00
Cliff Cai
79dfc96876 ASoC: Blackfin AC97: add a few missing multichannel define handling
Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was
merged, but two places were missed (the probe/resume functions).  Restore
handling of this option so it gets initialized properly.

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>
2009-09-17 10:51:34 +01:00
Huang Weiyi
d4e54e871f ASoC: remove unused #include <linux/version.h>
Remove unused #include <linux/version.h>('s) in
  sound/soc/codecs/ad1836.c
  sound/soc/codecs/ad1938.c
  sound/soc/codecs/wm8974.c

Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-16 21:08:54 +01:00
Mark Brown
8bb0148955 ASoC: Add S3C64xx IIS CDCLK source selection
CDCLK can either be an output generated by the CPU, intended for use
as the CODEC master clock, or an input (probably from the CODEC)
providing a master clock for the IIS block.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-16 21:07:50 +01:00
Miguel Aguilar
9b95b16678 ASoC: Davinci: Add audio codec support for DM365 EVM
This patch enables tlv320aic3101 support on DM365 EVM and
it was tested on DM365 EVM rev c.

Note: this patch was created based on temp/asoc branch.

Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-15 19:31:05 +01:00
Barry Song
08db48f1ee ASoC: use set_channel_map api to reorder channels for AD1938 and AD1836
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-15 13:33:59 +01:00
Jassi
fd5ad654e6 ASoC: S3C I2S LRCLK polarity option.
1) Explicitly set LRCLK polarity for I2S Vs LSM/MSB modes.
2) Convert from numerical to bit-field values for BCLK selection.
3) Use proper error checking for return value from clk_get

Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-15 13:33:55 +01:00
Jassi
fa68e0025d ASoC: S3C lrsync function made to work with IRQs disabled.
s3c2412_snd_lrsync() maybe reached with IRQs disabled and if LRCLK
is dead due to improper initialization of CPU or CODEC, the system
gets stuck in the loop because jiffies may never get updated.
Implemented counter based wait mechanism for atleast the same
timeout period.

Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-15 13:26:14 +01:00
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