Commit Graph

10381 Commits

Author SHA1 Message Date
Takashi Iwai a353fbb179 ALSA: hda - Remove a superfluous argument of via_auto_init_output()
"force" argument is always true, so let's strip it off.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21 14:24:25 +02:00
Takashi Iwai 020066d1ec ALSA: hda - Fix indep-HP path (de-)activation for VT1708* codecs
This patch fixes non-working indep-HP control on VT1708* codecs.
The problems are that via_independent_hp_put() wasn't fixed to follow
the recent change of three HP paths, and hp_indep_path didn't contain
the amp nids of mixer elements.

Together with the fixes, a few code clean-ups are done.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21 13:55:10 +02:00
Liam Girdwood 4805608ac1 ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
In preparation for ASoC Dynamic PCM (AKA DSP) support.

Provide convenience methods to retrieve the soc_card or snd_card from a
DAPM context.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-20 21:15:51 +01:00
Sangbeom Kim d7c3e9525a ASoC: SAMSUNG: Add I2S0 internal dma driver
I2S in Exynos4 and S5PC110(S5PV210) has a internal dma.
It can be used low power audio mode and 2nd channel transfer.
This patch can support idma.

Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-20 20:58:12 +01:00
Sangbeom Kim 61100f405d ASoC: SAMSUNG: Modify I2S driver to support idma
Previously, I2S driver only can support system dma.
In this patch, i2s driver can support internal dma too.
IDMA h/w configuration is initialized on idma.c

Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-20 20:58:07 +01:00
Rajashekhara, Sudhakar 82d1d52103 ASoC: davinci: add missing break statement
In davinci_vcif_trigger() function, a break() statement was missing
causing the davinci_vcif_stop() function to be called as a fallback
after calling davinci_vcif_start().

Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-20 20:51:35 +01:00
Rajashekhara, Sudhakar 3012f43eaf ASoC: davinci: fix codec start and stop functions
According to DM365 voice codec data sheet at [1], before starting
recording or playback, ADC/DAC modules should follow a reset and
enable cycle. Writing a 1 to the ADC/DAC bit in the register resets
the module and clearing the bit to 0 will enable the module. But the
driver seems to be doing the reverse of it.

[1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf

Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-20 20:51:23 +01:00
Mark Brown 3198b9eb51 ASoC: Acknowledge WM8962 interrupts before acting on them
This closes the small race between a status being read in response to an
interrupt and clearing the interrupt, meaning that if the status changes
between those periods we might not get a reassertion of the interrupt.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-07-20 20:50:08 +01:00
Wolfram Sang 09bddc8eb2 ASoC: sgtl5000: guide user when regulator support is needed
Print a hint when the user has a setup where CONFIG_REGULATOR is really
needed to make the driver work.

Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-19 16:19:30 +01:00
Wolfram Sang e94a4062c8 ASoC: sgtl5000: refactor registering internal ldo
The code for registering the internal ldo was present twice. Turn it
into a function instead. Also, inform the user if LDO is used now.

Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Tested-by: Dong Aisheng <b29396@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-19 16:19:22 +01:00
Wolfram Sang 1c8371d61e ASoC: core: make comments fit the code
In one comment, cpu_dai was mentioned although codec_dai was used in the
code. Also, fix the name for the card dai list which has no seperation
into card_dai and codec_dai.

Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-19 16:19:15 +01:00
Mark Brown 7be4ba24a3 ASoC: Mark cache as dirty when suspending
Since quite a few drivers are not managing to flag the cache as needing
to be resynced after suspend and it's a reasonable thing to do flag the
cache as needing sync automatically when suspending.

The expectation is that systems will mainly only keep the CODEC powered
when doing audio through the CODEC so we won't actually suspend the
device anyway; drivers which want to can override this behaviour when
they resume.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
2011-07-19 16:16:00 +01:00
Linus Torvalds 524196d2ad Merge branch 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ASoC: Correct WM8994 MICBIAS supply widget hookup
  ASoC: Fix shift in WM8958 accessory detection default implementation
  ASoC: sh: fsi-hdmi: fixup snd_soc_card name
  ASoC: sh: fsi-da7210: fixup snd_soc_card name
  ASoC: sh: fsi-ak4642: fixup snd_soc_card name
2011-07-18 09:05:59 -07:00
Takashi Iwai 3b607e3d3a ALSA: hda - Switch HP DAC dynamically with indep-HP switch for VIA
This patch changes the behavior of independent-HP enum switch.  Now
instead of returning a busy error, the driver switches dynamically the
stream of the HP (and shared) DACs according to the current mode.
The logic is similar like the dual-mic ADC switch, but a bit more
complicated because of the presence of shared DAC.

Together with the change, a mutex is introduced to protect against the
possible races for the indep-HP mode setting.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-18 16:54:40 +02:00
Takashi Iwai 3214b9665c ALSA: hda - Implement dynamic loopback control for VIA codecs
This patch adds the dynamic control of analog-loopback for VIA codecs.

When the loopback is enabled, the inputs from line-ins and mics are
mixed with the front DAC, and sent to the front outputs.  The very same
input is routed to the headhpones and speakers in loopback mode.
However, since the loopback mix can't take other than the front DAC,
there is no longer individual volume controls for headphones and
speakers.  Once when the loopback control is off, these volumes take
effect.

Since the individual volumes are more desired in general use caess, the
loopback mode is set to off as default for now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-18 16:47:33 +02:00
Clemens Ladisch c81c6b356b ALSA: virtuoso: fix silent analog output on Xonar Essence ST Deluxe
Commit dd203fa97b (ALSA: virtuoso: remove non-working controls on
Essence ST Deluxe) made it impossible to adjust the volume after the
driver initialized it to muted.

Ensure that those DACs that can be accessed with I2C are initialized
to the same volume that is the reset default of the DAC without I2C.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.38+ <stable@kernel.org>
2011-07-18 09:39:50 +02:00
Arnd Bergmann bc574e190d Merge branches 'omap/prcm' and 'omap/mfd' of git+ssh://master.kernel.org/pub/scm/linux/kernel/git/arm/linux-arm-soc into next/devel-2 2011-07-17 21:48:22 +02:00
Mark Brown 4400855986 Merge branch 'for-3.0' into for-3.1 2011-07-17 18:25:58 +09:00
Mark Brown b793eb60a0 ASoC: Correct WM8994 MICBIAS supply widget hookup
The WM8994 and WM8958 series of devices have two MICBIAS supplies rather
than one, the current widget actually manages the microphone detection
control register bit (which is managed separately by the relevant API).

Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2
widgets.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-17 18:24:29 +09:00
Mark Brown b0b3e6f861 ASoC: Don't use -1 to boostrap subseq so it can be used by drivers
Makes life a little easier if you want to add subsequences to an existing
driver as you can use -1 to put things at the start of sequences.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-17 18:23:31 +09:00
Mark Brown 7d02173cd1 ASoC: Reduce power consumption for idle DAIs in WM8994
If DAIs are idle but their clocks are in use for some reason (eg, as
SYSCLK or for accessory detect) then set the clock dividers to the maximum
to reduce slightly the power consumption of the unclocked circuits.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-17 17:51:10 +09:00
Mark Brown ca1004bab9 ASoC: Report an error for unknown adav80x formats
Not only fixes error handling but also some uninitialized variable
warnings.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
2011-07-17 13:15:20 +09:00
Mark Brown f0f5039c3d ASoC: Handle failed WM8994 FLL lock waits
Try the completion before we start the FLL so that if an interrupt was
delayed long enough for us to miss it we don't wait for the completion
it signalled.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-17 13:15:00 +09:00
Arnd Bergmann 6277839602 Merge branch 'fixes-for-arnd' of git://git.pengutronix.de/git/imx/linux-2.6 into imx/fixes 2011-07-15 21:56:37 +02:00
Mark Brown 1479c3fb5f ASoC: Handle spurious wm_hubs DC servo done interrupts
Don't assume the first fire indicates that we're done.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-15 23:48:05 +09:00
Dimitris Papastamos 6b3860b0a2 ASoC: WM8983: Initial driver
The WM8983 is a low power, high quality stereo CODEC
designed for portable multimedia applications. Highly flexible
analogue mixing functions enable new application features,
combining hi-fi quality audio with voice communication.

Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-15 23:39:53 +09:00
Mark Brown 47d90a03eb Merge branch 'for-3.0' into for-3.1 2011-07-15 22:43:07 +09:00
Mark Brown b35e160a11 ASoC: Fix shift in WM8958 accessory detection default implementation
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-15 22:41:38 +09:00
Daniel T Chen f21169aa87 ALSA: intel8x0: Apply headphones+mute LED quirk for Dell Inspiron 9300
BugLink: https://bugs.launchpad.net/bugs/774895

The original reporter states that his volume keys do not change the
desired Master and PCM mixer elements together, so apply the hp+mute led
quirk for his PCI SSID.

Reported-by: Jeffrey Finkelstein
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-15 07:54:26 +02:00
Takashi Iwai 00ef9610ac ALSA: hda - Fix krealloc() replacement in hda_codec.c
It was obviously wrong, grr....

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-14 15:58:02 +02:00
Takashi Iwai 7b1655f5f2 ALSA: hda - Re-add need_dac_fix check for multi-io jacks of Realtek codecs
During the rewrite, the check of spec->need_dac_fix and the corresponding
num_dacs change was dropped from the channel-mode control.

This patch re-adds it, and also enables need_dac_fix for ALC880 as default,
as this feature was originally introduced to fix h/w bugs of this chip.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-14 15:33:59 +02:00
Axel Lin 58499906c8 ASoC: wm8900: fix a memory leak if wm8900_set_fll fails
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 20:04:26 +09:00
Mark Brown 3b1af3f8c8 ASoC: Log WM8994 FIFO errors from the interrupt
We should spot them anyway on state changes but logging them gives us
better time information about when the misconfiguration happened.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 17:17:23 +09:00
Giridhar Maruthy b3d7615f2a ASoC: SAMSUNG: 24-bit audio playback on Exynos4210
Using 256fs or 512fs will result in distortion of 24-bit
audio samples. This is because the lrclk generated is not
proper. Using 384 fs generates proper output.

Signed-off-by: Giridhar Maruthy <giridhar.maruthy@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 08:25:42 +09:00
Mark Brown f05bdb8bb6 ASoC: Don't warn on low WM8994/58 AIFnCLKs
We can have valid but very low clocks in accessory detection modes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:30 +09:00
Mark Brown c7ebf932e5 ASoC: Use WM8994 FLL lock interrupt
If we have interrupts then wait for the FLL lock interrupt rather than
using dead reckoning when waiting for the FLL to start.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:22 +09:00
Mark Brown b30ead5f39 ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:14 +09:00
Mark Brown d96ca3cd0b ASoC: Implement DC servo completion IRQ handling for wm_hubs devices
The individual devices should set the flag dcs_done_irq in the hubs
shared data structure to indicate that they will flag the interrupt
by calling wm_hubs_dcs_done().

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:38:04 +09:00
Mark Brown b70a51bab9 ASoC: Use late enable handling for direct voice, speaker and headphone
This ensures appropriate clocking for bypass paths to speaker and
headphone and direct voice paths on affected revisions.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:37:52 +09:00
Johannes Stezenbach 889ebae537 ASoC: STA32x: Preserve reserved register bits
Chip documentation explicitly requires that the reset values
of reserved register bits are left untouched.  It is possible
there are differences between STA326 and STA328 or future
chip revisions in these bits, and clobbering them might
cause malfunction.

Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:24:32 +09:00
Johannes Stezenbach 7968843915 ASoC: STA32x: Add mixer controls for biquad coefficients
The STA32x has a number of preset EQ settings, but also
allows full user control of the biquad filter coeffcients
(when "Automode EQ" is set to "User").
Each biquad has five signed, 24bit, fixed-point coefficients
representing the range -1...1.  The five biquad coefficients
can be uploaded in one atomic operation into on-chip
coefficient RAM.
There are also a few prescale, postscale and mixing
coefficients, in the same numeric format and range
(a negative coefficient inverts phase).

These coefficients are made available as SNDRV_CTL_ELEM_TYPE_BYTES
mixer controls.

Signed-off-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-14 00:24:31 +09:00
Paul Menzel cf01b73e26 ALSA: hda - fix up typos in Kconfig help for default buffer size introduced in acfa634f
This commit is a fix up for commit acfa634f.

	commit acfa634f7e
	Author: Takashi Iwai <tiwai@suse.de>
	Date:   Tue Jul 12 17:27:46 2011 +0200

		  ALSA: hda - Add Kconfig for the default buffer size

Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 20:17:23 +02:00
Guillaume Pellerin 0f5733b0c8 ALSA: usb-audio - Add quirks for M-Audio Fast Track Pro and Quattro
This patch gives M-Audio Fast Track Pro and M-Audio Quattro quirks and
endpoints to boot and setup those devices with special options (digital
inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are
just adapted to match the new global M-Audio parameters.

Special configurations can be then loaded through a modprobe conf file.
For example, to set the 24 bits mode on the Fast Track Pro add
/etc/modprobe.d/fast_track_pro.conf :

    options snd_usb_audio   vid=0x763 pid=0x2012 device_setup=0x08

Here is a list of the possibilities in this example :
http://files.parisson.com/debian/fast-track-pro.conf

Signed-off-by: Guillaume Pellerin <yomguy@parisson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 18:15:45 +02:00
Takashi Iwai acfa634f7e ALSA: hda - Add Kconfig for the default buffer size
Add a Kconfig entry to specify the default buffer size.
Distros using PulseAudio can choose a larger value here.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 17:31:46 +02:00
Takashi Iwai 3101ba035c ALSA: Use krealloc() in possible places
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 08:05:16 +02:00
Takashi Iwai 30b4503378 ALSA: hda - Expose secret DAC-AA connection of some VIA codecs
VT1718S and co have a secret connection from DAC to AA-mix, which
doesn't appear in the connection list obtained from the h/w.
Currently the driver fixes the connection index locally at init, but
now we can expose it statically via snd_hda_override_connections()
so that this conection can be checked better by the parser in future.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 07:45:02 +02:00
Takashi Iwai 9e7717c9eb ALSA: hda - Always read raw connections for proc output
In the codec proc outputs, read the raw connections instead of the
cached connection list, i.e. proc files contain only raw values.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 07:45:01 +02:00
Takashi Iwai b2f934a0df ALSA: hda - Add snd_hda_override_conn_list() helper function
Add a function to add/modify the connection-list cache entry.
It'll be useful to fix a buggy hardware result.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 07:44:46 +02:00
Takashi Iwai 19110595c8 ALSA: hda - Turn on extra EAPDs on Conexant codecs
Some machines seem to use EAPD control of the unused pin for controlling
the overall EAPD.  Since the driver currently doesn't check the EAPD of
unused pins, the EAPD isn't enabled.  For avoiding such a problem, turn
all extra EAPDs on as default.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-11 14:46:44 +02:00
Jiri Kosina b7e9c223be Merge branch 'master' into for-next
Sync with Linus' tree to be able to apply pending patches that
are based on newer code already present upstream.
2011-07-11 14:15:55 +02:00
Takashi Iwai 9499473463 ALSA: hda - Preserve input pin-ctl bits in HP-automute for VIA codec
For smart51 pins, we need to preserve the input pin-control bits at
auto-mute controls instead of overwriting zero or pin-out-only.
Otherwise the VREF won't be set properly when smart51 is disabled
again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-11 11:36:44 +02:00
Takashi Iwai 6e969d9155 ALSA: hda - Set line-out pin-ctls properly when indep-HP mode changes
When Independent-HP mode is changed for VIA, the driver needs to
re-issue the auto-mute check so that the line-out pins are set properly
without influence of HP pin state.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-11 11:28:13 +02:00
Takashi Iwai 21ce0b6527 ALSA: hda - Via Fix speaker-mute checks in VIA driver
When the line-jack is plugged/unplugged, the driver must check also
the headphone jack state in addition to the line-out jack.  Currently
it checks only the line-out state and ignores the headphone.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-11 10:33:47 +02:00
Mark Brown 5b7396709e ASoC: Conditionalize the enable of WM8994 ADC TDM mode
Future devices will not benefit from this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-07-09 23:16:48 +09:00
Mark Brown 3db1bbfd4a Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-3.1 2011-07-09 23:16:12 +09:00
Takashi Iwai 017f2a104c ALSA: hda - Implement 44kHz workaround for IdeadPad as fixup
Instead of checking the model quirk, use a fixup table for workaround
of 44kHz-fixed PCM for Lenovo IdeaPad with ALC269.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-09 14:42:25 +02:00
Mark Brown 3f9c42ed6b Merge branch 'for-3.0' into for-3.1 2011-07-09 19:06:33 +09:00
Kuninori Morimoto 2c7beb9285 ASoC: sh: fsi-hdmi: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.

aplay will be "Segmentation fault" without this patch
special thanks to Takashi.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-09 19:06:16 +09:00
Kuninori Morimoto f15c941331 ASoC: sh: fsi-da7210: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.

aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-09 19:06:05 +09:00
Kuninori Morimoto 505b04e0f8 ASoC: sh: fsi-ak4642: fixup snd_soc_card name
it shouldn't contain space letters and
special letters like parentheses.

aplay will be "Segmentation fault" without this patch.
special thanks to Takashi.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-09 19:05:55 +09:00
Takashi Iwai e8fd86efaa Merge branch 'fix/asoc' into for-linus 2011-07-09 11:56:43 +02:00
Takashi Iwai abaead6ac5 ALSA: hda - Fix a copmile warning
It's harmless but annyoing.
  sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’:
  sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-09 11:55:28 +02:00
Takashi Iwai e320bc42be Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2011-07-09 11:43:04 +02:00
Mark Brown 71ae391d45 Merge branch 'for-3.0' into for-3.1 2011-07-09 18:20:36 +09:00
Takashi Iwai 18361bbe31 Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc 2011-07-09 09:44:09 +02:00
Takashi Iwai 3e6179b844 ALSA: hda - Merge alc*_parse_auto_config() functions in patch_realtek.c
Now all alc*_parse_auto_config() do almost same thing except for the
NID list to ignore and the PINs for SSID-check, we can merge all these
to a single function.  A good amount of code reduction.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 16:55:13 +02:00
Takashi Iwai 8452a982fb ALSA: hda - Merge ALC260 auto-parser code
Finally the last one.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 16:19:48 +02:00
Takashi Iwai 4c11398edc ALSA: hda - Merge ALC269 parser code
One more code reduction.  This codec has less DACs, thus the wiring
to DAC can't be filled uniquely for all output pins, i.e. some outputs
share the same volume control.
Except for that, all seems working fine.

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

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 16:01:47 +02:00
Takashi Iwai 72dcd8e76b ALSA: hda - Merge ALC861 auto-parser code
Merge more auto-parser code in patch_realtek.c, now for ALC861.
The topology of this codec is pretty simple, and can be parsed well
by the current starndard parser.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 15:16:55 +02:00
Takashi Iwai 44c0240052 ALSA: hda - Fix amp-cap checks in patch_realtek.c
query_amp_caps() may return non-zero if the amp cap isn't supported
by the codec.  Thus one needs to check widget-caps first, then check
the corresponding amp-caps.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 15:14:19 +02:00
Takashi Iwai a1f649d547 ALSA: hda - Merge ALC861-VD auto-parse to the standard parser
The existing standard auto-parser can work well with this codec, too.
Let's merge.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 14:39:03 +02:00
Takashi Iwai 268ff6fbe7 ALSA: hda - Fix auto-mic detection in Realtek codec-parser
A regression fix from commit 21268961d3
  ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs

The auto-mic wasn't detected properly when no ADC-switch is needed.

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

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

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

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

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

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

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 10:11:35 +02:00
Pavel Roskin 81b85b6bd9 ALSA: usb-audio: replace "void *" with more specific pointers
Signed-off-by: Pavel Roskin <proski@gnu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 10:10:25 +02:00
Lydia Wang a2a870c827 ALSA: hda - Fix Independent-HP detection on VT2002P/1802/1812 codecs
For VT2002P, VT1802 and VT1812 codecs, to create Independent HP
control.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 08:20:05 +02:00
Lydia Wang 5c9a5615de ALSA: hda - Fix DAC checks for VT2002P/1802/1812 codecs
For VT2002P, VT1802 and VT1812 codecs, there're only two DACs. So smart51
control shouldn't be created.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 08:19:28 +02:00
Lydia Wang d69607b3c3 ALSA: hda - Fix VIA output-path init for VT2002P/1802/1812
For VT2002P, VT1802 and VT1812 codecs, the original activate_output_path()
function can't initialize output and hp path correctly, since mixers connected to
output pin widgets are not considered. So modify the activate_output_path()
function to satisify this kind of codec.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 08:18:29 +02:00
Mark Brown b5d5f59be2 Merge branch 'for-3.0' into for-3.1 2011-07-07 09:54:19 -07:00
Axel Lin e12c28a98f ASoC: pxa2xx-pcm: remove unused variable 'dai'
Remove unused variable 'dai' to eliminate below warning.

  CC      sound/soc/pxa/pxa2xx-pcm.o
sound/soc/pxa/pxa2xx-pcm.c: In function 'pxa2xx_soc_pcm_new':
sound/soc/pxa/pxa2xx-pcm.c:91: warning: unused variable 'dai'

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 09:54:09 -07:00
Kuninori Morimoto bd7fdbcaa2 ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2
mask didn't cover update-data

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-07-07 09:46:06 -07:00
Takashi Iwai 1d045db96a ALSA: hda - Split quirk codes from patch_realtek.c
Put the all static quirk codes out of patch_realtek.c, split into the
file for each codec model.  For controlling the build of quirk codes,
a new Kconfig, CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS is introduced.
By setting this off, all quirk codes won't be built, thus you can save
lots of memory.

The codes in patch_realtek.c are also shuffled and more comments are
given, but the contents aren't changed.  This is just a refactoring.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 18:27:29 +02:00
Takashi Iwai 0e4a73ae58 ALSA: hda - Use common paser for digital I/O for ALC260
Avoid open-codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 18:03:12 +02:00
Takashi Iwai 21268961d3 ALSA: hda - More flexible dynamic-ADC switching for Realtek codecs
This patch changes the auto-parser and the auto-mic handling codes to
allow more flexible dynamic ADC-switching with Realtek codecs.

In the new code, the following strategy is taken:

- When a cap-src can't handle all input-sources, either skip it, or
  switch to the ADC-switching mode.  In ADC-switching mode, like the
  former dual-ADC mode for ALC275, it changes ADC on the fly according
  to the current input source.
- When auto-mic is possible, always assign imux.  If the mic pins are
  set statically via a quirk, rebuild imux according to the pins.
  In the auto-mic mode, the driver always changes the imux (although
  the imux isn't exposed as a mixer element).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 18:02:43 +02:00
Takashi Iwai a926757f04 ALSA: hda - Fix warning with ALC882 digital-out detection
The digital out pin on ALC882 may have multiple connections.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 16:08:13 +02:00
Peter Ujfalusi 21385eeb02 ASoC: twl6040: Add back support for legacy mode
The legacy mode has been accidentaly removed by commit:
ASoC: twl6040: add all ABE DAIs

Add back the twl6040-hifi dai.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:47 +03:00
Peter Ujfalusi ff593ca1a4 ASoC: twl6040: No need to convert the PLL ID
Since the PLL handling has been simplified, and
rebased on 0, there is no longer need for converting
the PLL ID.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:46 +03:00
Peter Ujfalusi 753621c215 ASoC: twl6040: Configure PLL only once
Avoid configuring the PLL several times during audio startup.
We can configure the PLL at prepare time with parameters collected
earlier hw_param, and set_dai_sysclk calls.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:45 +03:00
Peter Ujfalusi f53c346c08 ASoC: twl6040: Simplify sample rate constraint handling
We can manage the sample rate constraints without the need
to maintain a variable and a pointer.
This simplifies the handling of the constraint, and makes it
more robust.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:44 +03:00
Peter Ujfalusi af958c72af ASoC: twl6040: Move PLL selection to codec driver
It is better if the selection between the Low power,
and High performance PLL is handled within the codec
driver, not in machine driver(s) to avoid duplicated
code, and also to have consistent tracking of the selected
PLL, and the resulting differences in supported sample
rates.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:44 +03:00
Peter Ujfalusi 7cca606794 ASoC: twl6040: Use neutral name for power mode text/enum
Change the variable names to be neutral (not refering to HS).
This will ease up the introduction of PLL selection, which
going to use the same enum strings.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:43 +03:00
Peter Ujfalusi 2a433b9daf ASoC: twl6040: Do not use wrapper for irq request
The twl6040_request_irq/free_irq inline functions are going
to be removed, so replace them with direct calls.
The irq number is provided by the core driver via resource.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Reviewed-by: Felipe Balbi <balbi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-07-07 14:23:22 +03:00
Sascha Hauer 6584cb8825 ARM i.MX dma: Fix burstsize settings
dmaengine expects the maxburst parameter in words, not bytes.
The imxdma driver and its users do this wrong. Fix this.

As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work
with imx-dma. This broke the driver with imx-sdma support which
correctly takes the maxburst parameter in words. This patch
puts the sdma based sound back to work.

Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
2011-07-07 09:55:50 +02:00
Takashi Iwai c2d986b0d2 ALSA: hda - Clean-up PCM assignments in patch_realtek.c
Instead of assigning each default hda_pcm_stream pointers, do NULL-checks
and assign default values in alc_build_pcms().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:35:17 +02:00
Takashi Iwai f970de2555 ALSA: hda - Unify alc*_auto_init_input_src() in patch_realtek.c
The only different implmentation was alc880_auto_init_input_src(),
and now it covers this variant, and we can use the single function
for all codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-07 09:35:14 +02:00