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6,219 commits

Author SHA1 Message Date
Kuninori Morimoto
b8e583f601 ASoC: Add FSI-AK4642 sound support for SuperH
This patch is tested by ms7724se

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Based and tested on linux-2.6.31-rc6.

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

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

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

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

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

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

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

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

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

The codec differences before/after patch are:

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

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

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

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

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

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-19 00:22:17 +02:00
Mark Brown
59ae07a580 ASoC: WM8993 digital mixing support
The WM8993 provides digital sidetone paths and also allows each
channel on the audio interface to be routed separtately to the
DACs and ADCs.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Created against linux-2.6.31-rc5.

Tested on Amstrad Delta.

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

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

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

Created against linux-2.6.31-rc5.

Tested on Amstrad Delta.

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

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

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 17:52:59 +01:00