7865 Commits

Author SHA1 Message Date
Arnd Bergmann
992cbf7438 sound/oss-msnd-pinnacle: ioctl needs the inode
This broke in sound/oss: convert to unlocked_ioctl, when I missed one
of the ioctl functions still using the inode pointer.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-14 15:14:02 +02:00
Takashi Iwai
840b64c080 ALSA: hda - Add support of dual-ADCs for Realtek ALC275
Some VAIO models with ALC275 have dual ADCs for both internal and external
mics, and the driver needs to switch one of them appropriately.
This patch adds a basic support for this functionality, dynamic switching
between two ADCs per jack plug state.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-13 22:49:01 +02:00
Manuel Lauss
0c74a939d8 ASoC: au1x: fix section mismatch in psc-i2s.c
Annotate platform probe callback with __devinit instead of plain __init.

Signed-off-by: Manuel Lauss <manuel.lauss@googlemail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:39:14 +01:00
arnaud.patard@rtp-net.org
b424ec9533 ASoC: kirkwood-i2s: Handle mute/unmute playback/record
The controller has mute/unmute capability and some bootloader may mute
them at boot. If it's not handled, all things will seem to be working
but no sound will come out of the speaker/headphone.

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:37:09 +01:00
arnaud.patard@rtp-net.org
dfe4c93627 ASoC: Fix kirkwood i2s mono playback
Kirkwood controller needs to be informed if the audio stream is mono
or not. Failing to do so will result in playing at the wrong speed.

Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:37:09 +01:00
Kuninori Morimoto
ccad7b44cc ASoC: fsi: Fixup for master mode
This patch add hw_params to snd_soc_dai_ops,
because board specific set_rate is needed
when FSI was used as master mode.

This patch remove fsi_clk_ctrl from fsi_dai_startup,
because clock should be disabled before set_rate.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:39 +01:00
Kuninori Morimoto
d78541473d ASoC: fsi: Add pr_err for noticing unsupported access
This patch didn't use dev_err,
because it is difficult to get struct device here.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:38 +01:00
Kuninori Morimoto
73b92c1fc0 ASoC: fsi: Change struct fsi_regs to fsi_core
Many registers which were grouped by category were added in FSI2.
To make easy to switch FSI/FSI2, fsi_core was added instead of fsi_regs.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:37 +01:00
Kuninori Morimoto
a7ffb52bb3 ASoC: fsi: remove noisy CR_FMT macro
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:36 +01:00
Kuninori Morimoto
a09370cb8c ASoC: fsi: remove un-used variable on fsi_dai_startup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:35:35 +01:00
Joe Perches
4726a57b8c ASoC: Remove unnecessary casts of private_data
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:34:06 +01:00
Joe Perches
8ff23610a6 ASoC: Remove unnecessary casts of private_data
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:33:59 +01:00
Mark Brown
4d53952a39 Merge branch 'for-2.6.35' into for-2.6.36 2010-07-13 12:29:10 +01:00
Kuninori Morimoto
637727838a ASoC: fsi: fixup wrong value setting order of TDM
channel size should be set before setting register value

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:26:26 +01:00
Kuninori Morimoto
b427b44cc8 ASoC: fsi: fixup clock inversion operation
Clock inversion should be specified by each flags bit.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-13 12:26:26 +01:00
Peter Ujfalusi
27eeb1feed ASoC: TWL4030: DAC power optimization
Restructure the DAPM connections in order to enable
only the needed DAC (out of four in twl4030 series).
I need to keep the 'AIF Enable' supply connected to
the L2/R2 digital path, since the digital loopback
needs AIF and APLL running.
If no valid route available, than none of the DAC will
be powered, but the AIF and APLL is going to be enabled.
Furthermore, if only one audio path have valid route,
than only the corresponding DAC will be powered.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-13 11:30:12 +01:00
Peter Ujfalusi
8b0d31532e ASoC: TWL4030: Fix for digital loopback gain range
When the gain is configured using dB value it was
not possible to use -24dB since the loopback got
muted instead of -24dB.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsomicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-13 11:30:05 +01:00
Linus Torvalds
7e48c02829 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 - Restore cleared pin controls on resume
2010-07-12 14:44:43 -07:00
Arnd Bergmann
d209974cdc sound/oss: convert to unlocked_ioctl
These are the final conversions for the ioctl file operation so we can remove
it in the next merge window.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-12 22:36:47 +02:00
Uwe Kleine-König
a7ce2e0d04 fix comnment/printk typos concerning "empty"
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2010-07-12 18:03:50 +02:00
Arnd Bergmann
90dc763fef sound: push BKL into open functions
This moves the lock_kernel() call from soundcore_open
to the individual OSS device drivers, where we can deal
with it one driver at a time if needed, or just kill
off the drivers.

All core components in ALSA already provide
adequate locking in their open()-functions
and do not require the big kernel lock, so
there is no need to add the BKL there.

Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-12 17:41:05 +02:00
Clemens Ladisch
32e0191d79 ALSA: HDA: VT1708S: fix Smart5.1 mode
Correctly configure bidirectional pins when resuming; do not power down
widgets when they are needed for Smart5.1 output; and on 3-jack boards,
create the streams and controls needed for six channels.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Viliam Kubis <viliam.kubis@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-12 17:25:45 +02:00
Clemens Ladisch
395c61d196 ALSA: via82xx: allow changing the initial DXS volume
As per-stream volume controls, the DXS controls are not intended to
adjust the overall sound level and so are initialized every time
a stream is opened.  However, there are special situations where one
wants to reduce the overall volume in the digital domain, i.e., before
the AC'97 codec's PCM volume control.  To allow this, add a module
parameter that sets the initial DXS volume.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Soeren D. Schulze <soeren.d.schulze@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-12 17:25:27 +02:00
Clemens Ladisch
d32d552e66 ALSA: usb-audio: silence a superfluous warning
It is not advisable to print a warning when a device does not support
setting the sample rate because this is perfectly valid for devices with
a single rate or where rates are implicitly changed by selecting another
alternate setting.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-09 15:08:12 +02:00
Takashi Iwai
f8fb27bd4a Merge branch 'fix/hda' into topic/hda 2010-07-09 10:09:00 +02:00
Takashi Iwai
afbd9b8448 ALSA: hda - Limit the amp value to write
Check the amp max value at put callbacks and set the upper limit
so that the driver won't write any invalid value over the defined
range.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-09 10:08:57 +02:00
Takashi Iwai
3507e2a8f1 ALSA: hda - Add beep mixer support to Conexant codecs
Added the beep mixer controls to Conexant codecs.
They simply control the digital beep generator widget.

For cx5047, I couldn't find any beep generator, so it's not implemented
there.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-09 10:08:56 +02:00
Takashi Iwai
ac0547dc62 ALSA: hda - Restore cleared pin controls on resume
Many codecs now clear the pin controls at suspend via snd_hda_shutup_pins()
for reducing the click noise at power-off.  But this leaves some pins
uninitialized, and they'll be never recovered after resume.

This patch adds the proper recovery of cleared pin controls on resume.
Also it adds a check of bus->shutdown so that pins won't be cleared at
module unloading.

Reference: Kernel bug 16339
	http://bugzilla.kernel.org/show_bug.cgi?id=16339

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-09 08:42:29 +02:00
Mark Brown
66b47fdb85 ASoC: Implement WM8994 OPCLK support
The WM8994 can output a clock derived from its internal SYSCLK, called
OPCLK.  The rate can be selected as a sysclk, with a division from the
SYSCLK rate specified (multiplied by 10 since a division of 5.5 is
supported) and the clock can be disabled by specifying a divisor of
zero.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-09 08:50:12 +09:00
Mark Brown
e88ff1e6db ASoC: Include WM8994 GPIO and interrupt registers in codec_reg
Very handy for debug.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-09 01:37:06 +09:00
Takashi Iwai
7645054f18 Merge branch 'fix/misc' into for-linus 2010-07-08 16:55:26 +02:00
Takashi Iwai
b492c4e895 Merge branch 'fix/hda' into for-linus 2010-07-08 16:55:02 +02:00
Raffaele Recalcati
d9823ed9fa ASoC: DaVinci: More accurate continuous serial clock for McBSP (I2S)
i2s_accurate_sck switch can be used to have a better approximate
    sampling frequency.
    The clock is an externally visible bit clock and it is named
    i2s continuous serial clock (I2S_SCK).
    The trade off is between more accurate clock (fast clock)
    and less accurate clock (slow clock).
    The waveform will be not symmetric.
    Probably it is possible to get a better algorithm for calculating
    the divider, trying to keep a slower clock as possible.

    This patch has been developed against the
        http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
    git tree and has been tested on bmx board (similar to dm365 evm, but using
    uda1345 as external audio codec).

Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-06 23:54:07 +09:00
Raffaele Recalcati
ec63755337 ASoC: DaVinci: Added selection of clk input pin for McBSP
When McBSP peripheral gets the clock from an external pin,
    there are three possible chooses, MCBSP_CLKX, MCBSP_CLKR
    and MCBSP_CLKS.
    evm-dm365 uses MCBSP_CLKR, instead in bmx board I have a different
    hardware connection and I use MCBSP_CLKS, so I have added
    this possibility.

    This patch has been developed against the:
        http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
    git tree and has been tested on bmx board (similar to dm365 evm)

Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-06 23:54:06 +09:00
Raffaele Recalcati
a4c8ea2dda ASoC: DaVinci: Added two clocking possibilities to McBSP (I2S)
Added two clocking options for dm365 McBSP peripheral when used
    with I2S timings, that are SND_SOC_DAIFMT_CBS_CFS (the cpu generates
    clock and frame sync) and SND_SOC_DAIFMT_CBS_CFM (the cpu gets clock
    from external pin and generates frame sync).
    A slave clock management can be important when the external codec needs
    the system clock and the bit clock synchronized (tested with uda1345).
    This patch has been developed against the:
        http://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci.git
    git tree and has been tested on bmx board (similar to dm365 evm, but using
    uda1345 as external audio codec).

Signed-off-by: Raffaele Recalcati <raffaele.recalcati@bticino.it>
Signed-off-by: Davide Bonfanti <davide.bonfanti@bticino.it>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-06 23:54:06 +09:00
Maurus Cuelenaere
088fbab406 ASoC: Invert speaker enabling behaviour in SmartQ sound driver
The speaker was enabled when the headphone was plugged in, which isn't the
wanted behaviour so correct this.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-06 23:54:06 +09:00
Eliot Blennerhassett
f978d36da4 ALSA: asihpi - Remove unneeded ;
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:20:43 +02:00
Eliot Blennerhassett
36ed8bdd86 ALSA: asihpi - Minor HPI error handling fixes
Handle errors in tuner level caching,
Ccorrect error code for aesebu rx status.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:20:21 +02:00
Eliot Blennerhassett
108ccb3f0f ALSA: asihpi - Change compander API and tidy
Compander API changed to one function per parameter.
Factor out some common code for stereo log value reading.
Make some more entity functions static.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:19:56 +02:00
Eliot Blennerhassett
3843914635 ALSA: asihpi - Add ASI5200 family
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:19:35 +02:00
Eliot Blennerhassett
1dd6aaaafc ALSA: asihpi - Use version string instead of printf formatting
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:19:06 +02:00
Eliot Blennerhassett
168f1b07cc ALSA: asihpi - HPI API updates
Remove some deprecated items.
Change compander api to one function per parameter.
Add a version string define.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-06 08:18:27 +02:00
Mark Brown
db059c0f6e ASoC: Automatically manage ALC coefficients for WM8960
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-06 08:46:10 +09:00
John Kacur
171d9f7d78 soundcore_open: Reduce the area BKL coverage
Most of this function is protected by the sound_loader_lock.
We can push down the BKL to this call out err = file->f_op->open(inode,file);

In order to build the sound core without the BKL, we
will need to push the lock_kernel() call into the ~20
device drivers that register their file operations.

Signed-off-by: John Kacur <jkacur@redhat.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Alan Cox <alan@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-05 18:07:30 +02:00
Takashi Iwai
f189efcd1c ALSA: hda - Enable beep on Realtek codecs with PCI SSID override
When the PCI SSID gives an overriding SKU assno, PC-beep bit isn't
detected (since it's located over 16bit), resulting in no PC beep.
Also, many devices seem ignoring the requirement by Realtek's spec
for SSID numbers, and it also confuses the PC beep detection.

This patch assumes the PC beep is available on every machine with
PCI SSID override.  It's a regression fix from 2.6.34.

Reference: Kernel bug 16251
	http://bugzilla.kernel.org/show_bug.cgi?id=16251

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-07-05 17:28:17 +02:00
Mark Brown
afd6d36a0d ASoC: Automatically manage DAC deemphasis rate for WM8960
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-05 23:41:18 +09:00
Mark Brown
4faaa8d968 ASoC: Remove current WM8960 deemphasis control
It will be replaced with automatic deemphasis rate configuration but since
we have an enumeration table in this driver this is done in a separate
commit to make the renumbering of the enumeration items clear.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-05 23:37:17 +09:00
Mark Brown
9af8381023 ASoC: Fix sorting of Makefile and Kconfig
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-07-05 23:35:29 +09:00
Takashi Iwai
65ee2ba310 Merge branch 'devel' of git://git.alsa-project.org/alsa-kernel into topic/misc 2010-07-05 15:37:27 +02:00
Maurus Cuelenaere
ce93a37028 ASoC: Add SmartQ sound driver
This adds sound support for the SmartQ board.

The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750
driver is used for driving the WM8987, as they are register compatible.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-04 18:04:12 +09:00