Commit Graph

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
Maurus Cuelenaere 0d9c15e45b ASoC: codec: Add WM8987 device id to WM8750 driver
The WM8987 codec is register compatible with the WM8750, so just add it to the
SPI and I²C device table.

Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-04 18:02:07 +09:00
Kuninori Morimoto a300de3cff ASoC: ak4642: Add Digital Playback Volume control
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-07-01 17:08:47 +01:00
Vladimir Zapolskiy 338de9d9da ASoC: uda134x: correct bias level setup for codecs family
For UDA1341 codec power control is managed in STATUS1 register, and
for all other codecs in DATA011 register.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-30 14:46:56 +01:00
Vladimir Zapolskiy ed632ad3b8 ASoC: uda134x: add DATA011 register found in codecs family
In UDA1340, UDA1344 and UDA1345 codecs there is one more functional
register in part of DATA0 tranfser. For UDA1341 this register
coincides with EA register.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-30 14:46:56 +01:00
Mark Brown af51b5c0f0 Merge remote branch 'takashi/topic/asoc' into for-2.6.36 2010-06-30 14:46:53 +01:00
Grant Likely 1636f8ac2b sparc/of: Move of_device fields into struct pdev_archdata
This patch moves SPARC architecture specific data members out of
struct of_device and into the pdev_archdata structure.  The reason
for this change is to unify the struct of_device definition amongst
all the architectures.  It also remvoes the .sysdata, .slot, .portid
and .clock_freq properties because they aren't actually used by
anything.

A subsequent patch will replace struct of_device entirely with struct
platform_device and the of_platform support code will share common
routines with the platform bus (but the bus instances themselves can
remain separate).

This patch also adds 'struct resources *resource' and num_resources
to match the fields defined in struct platform_device.  After this
change, 'struct platform_device' can be used as a drop-in replacement
for 'struct of_platform'.

This change is in preparation for merging the of_platform_bus_type
with the platform_bus_type.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: David S. Miller <davem@davemloft.net>
Cc: Stephen Rothwell <sfr@canb.auug.org.au>
2010-06-28 12:41:33 -07:00
David Dillow 08b4509889 sis7019: increase reset delays
A few boards using this controller are reported to need a little extra
time during their reset cycle.

Reported-by: Michael Goeke <michael.goeke@icachip.de>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-06-28 09:42:22 +02:00
David Dillow 3a3d5fd125 sis7019: fix capture issues with multiple periods per buffer
When using a timing voice to clock out periods during capture, the
driver would slowly loose synchronization and never catch up, eventually
reaching a point where it no longer generated interrupts. To avoid
this situation, the virtual period clocking was changed to shorten the
next timing period when our timing voice falls too far behind the
capture voice. In addition, the first virtual period for the timing
voice was slightly too short, causing the timing voice to initially be
ahead of the capture voice.

While tracking down this problem, I noticed that the expected sample
offset was being incorrectly initialized, causing an overrun to be
incorrectly reported when the timing voice happened to be perfectly
synchronized.

Reported-by: Hans Schou <linux@schou.dk>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-06-28 09:42:18 +02:00
David Dillow 5daeba34d2 ALSA: pcm_lib: avoid timing jitter in snd_pcm_read/write()
When using poll() to wait for the next period -- or avail_min samples --
one gets a consistent delay for each system call that is usually just a
little short of the selected period time. However, When using
snd_pcm_read/write(), one gets a jittery delay that alternates between
less than a millisecond and approximately two period times. This is
caused by snd_pcm_lib_{read,write}1() transferring any available samples
to the user's buffer and adjusting the application pointer prior to
sleeping to the end of the current period. When the next period
interrupt occurs, there is then less than avail_min samples remaining to
be transferred in the period, so we end up sleeping until a second
period occurs.

This is solved by using runtime->twake as the number of samples needed
for a wakeup in addition to selecting the proper wait queue to wake in
snd_pcm_update_state(). This requires twake to be non-zero when used
by snd_pcm_lib_{read,write}1() even if avail_min is zero.

Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-06-28 09:42:09 +02:00
Linus Torvalds 29ccb201a2 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: usb/endpoint, fix dangling pointer use
  ALSA: asihpi - Get rid of incorrect "long" types and casts.
  ASoC: DaVinci: Fix McASP hardware FIFO configuration
  ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5,1 / 5,2
  ALSA: usb-audio: fix UAC2 control value queries
  ALSA: usb-audio: parse UAC2 sample rate ranges correctly
  ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACE
  ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2()
  ALSA: hda - Don't check capture source mixer if no ADC is available
2010-06-27 07:39:57 -07:00
Eric Bénard 9c1be7e8cb ASoC: clean i.MX Kconfig
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-25 12:30:48 +01:00
Vladimir Zapolskiy e4295b40ee ASoC: uda134x: fix bias level setup on initialization
On initialization ADC/DAC are enabled only for UDA1341, that's why
bias_level shall be set to off explicitly, otherwise dapm is
misinformed about bias_level on startup.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-25 12:29:02 +01:00
Vladimir Zapolskiy cc3202f5da ASoC: uda134x: replace a macro with a value in platform struct.
This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.

Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-25 12:29:01 +01:00
Takashi Iwai e827e32efc Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2010-06-24 11:11:41 +02:00
Takashi Iwai b415ec7041 ALSA: usb - Fix compile error with CONFIG_SND_DEBUG_VERBOSE=y
Replaced the forgotten cval->mixer->ctrlif.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-24 08:07:28 +02:00
Takashi Iwai d4a86d8194 ALSA: hda - Add missing ALC680_* definitions
Also update the documentation.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-23 17:52:39 +02:00
Kailang Yang d1eb57f47b ALSA: hda - Support ALC680 codec
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-23 16:25:26 +02:00
Daniel Mack 3d8d4dcfd4 ALSA: usb-audio: simplify control interface access
As the control interface is now carried in struct snd_usb_audio, we can
simplify the API a little and also drop the private ctrlif field from
struct usb_mixer_interface.

Also remove a left-over function prototype in pcm.h.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-23 16:10:23 +02:00
Daniel Mack 157a57b6fa ALSA: usb-audio: move and add some comments
Also add a list of open topics.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-23 16:09:50 +02:00
Daniel Mack 21af7d8c0c ALSA: usb-midi: whitespace fixes
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-23 16:09:38 +02:00
Daniel Mack 69da9bcb98 ALSA: usb-audio: unify UAC macros and struct names
Get rid of the last occurances of _v1 suffixes, and move the version
number right after the "uac" string. Now things are consitent again.

Sorry for the forth and back, but it just looks much nicer this way.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-23 16:09:26 +02:00
Daniel Mack f22aa94908 ALSA: usb-audio: clean up includes in clock.c
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-23 16:09:14 +02:00
Takashi Iwai 1240e6b553 Merge branch 'fix/misc' into topic/misc 2010-06-23 16:07:34 +02:00
Alexey Fisher a5c7d797dc ALSA: usb-audio - Add volume resolution quirk for some Logitech webcams
Some programs like Skype trying to set capture volume automatically.
Normally it will tray, carefully step by step lover or higher, set the volume.
In real word it work not really well, because devises and vendors lie about
real audio settings.
For example most Logitech webcams have 6400 or 3500 steps for capture volume.
They do not tell that actual resolution is 384. So we have only 7 or 18 real
steps. In this patch I set real resolution only for tested devices.

Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-23 16:02:07 +02:00
Jarkko Nikula 8c523115ae ASoC: RX-51: Add basic jack detection
This patch adds GPIO jack detection to Nokia N900/RX-51. At the moment only
SND_JACK_VIDEOOUT type is reported. More types could be reported after
getting more audio features supported and necessary drivers integrated for
implementing automated accessory detection.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-23 11:29:14 +01:00
Jarkko Nikula 4eb5470326 ASoC: RX-51: Add Jack Function kcontrol
Nokia RX-51/N900 has multifunction 4-pole audio-video jack that can be used
as headphone, headset or audio-video connector. This patch implements the
control 'Jack Function' which is used to select the desired function.
At the moment only TV-out without audio is supported.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-23 11:29:08 +01:00
Eric Bénard 3d5a451623 codecs/tlv320aic23: fix bias management for suspend/resume
in tlv320aic23_set_bias_level, for the case SND_SOC_BIAS_ON, the
comment says "vref/mid, osc on, dac unmute" but the code doesn't
clear the corresponding bits, thus when resuming, several bits are
not cleared preventing the codec from working.

in tlv320aic23_suspend, clearing the active register is not needed
as it will be done by tlv320aic23_set_bias_level, when setting
bias to SND_SOC_BIAS_OFF

Signed-off-by: Eric Bénard <eric@eukrea.com>
Cc: broonie@opensource.wolfsonmicro.com
Cc: anuj.aggarwal@ti.com
Cc: lrg@slimlogic.co.uk
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-23 11:28:53 +01:00
Lars-Peter Clausen 5898dd9ebd ASoC: JZ4740: Add qi_lb60 board driver
This patch adds ASoC support for the qi_lb60 board.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-23 00:10:57 +01:00
Lars-Peter Clausen 3b097d64ea ASoC: Add JZ4740 codec driver
This patch adds support for the JZ4740 internal codec.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-23 00:10:45 +01:00
Lars-Peter Clausen 11bd3dd1b7 ASoC: Add JZ4740 ASoC support
This patch adds ASoC support for JZ4740 SoCs I2S module.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-23 00:08:06 +01:00
Luke Yelavich 3bfea98ff7 ALSA: hda - Add Macbook 5,2 quirk
BugLink: https://bugs.launchpad.net/bugs/463178

Set Macbook 5,2 (106b:4a00) hardware to use ALC885_MB5

Cc: <stable@kernel.org>
Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-22 11:13:54 +02:00
Takashi Iwai 2f44f84725 ALSA: hda - Fix uninitialized variable
Fix the following compile warning.  kctl should be NULL-initialized.

  sound/pci/hda/patch_realtek.c: In function ‘alc_build_controls’:
  sound/pci/hda/patch_realtek.c:2550:23: warning: ‘kctl’ may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-22 11:12:32 +02:00
Grazvydas Ignotas 4b94dba029 ASoC: pandora: fix CLKX polarity
After mass production started it was found that several boards exhibit
noise problems during sound playback. After some investigation it was
determined that CLKX polarity is set incorrectly, and even if most boards
can tolerate the wrong setting, there are some that don't.

Fix polarity setup in the board file. As the clock settings for input and
output now match, merge in and out functions and structures to simplify
code.

Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-21 18:16:17 +01:00
Takashi Iwai d69f309f04 Merge branch 'fix/misc' into for-linus 2010-06-21 17:08:41 +02:00
Jiri Slaby 272cbc98cf ALSA: usb/endpoint, fix dangling pointer use
Stanse found that in snd_usb_parse_audio_endpoints, there is a
dangling pointer dereference. When snd_usb_parse_audio_format fails,
fp is freed, and continue invoked. On the next loop, there is
"fp && fp->altsetting == 1 && fp->channels == 1" test, but fp is set
from the last iteration (but is bogus) and thus ilegally dereferenced.

Set fp to NULL before "continue".

Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-21 17:07:58 +02:00
Mark Brown b45416656f ASoC: Fix sorting of DA7210 entries in Kconfig
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-20 14:05:46 +01:00
Takashi Iwai 2ac90e990c Merge branch 'fix/misc' into for-linus 2010-06-20 10:38:19 +02:00
Takashi Iwai b2c420657f Merge branch 'fix/asoc' into for-linus 2010-06-20 10:38:14 +02:00
Stuart Longland 20630c7f59 ASoC: Fix overflow bug in SOC_DOUBLE_R_SX_TLV
When SX_TLV widgets are read, if the gain is set to a value below 0dB,
the mixer control is erroniously read as being at maximum volume.

The value read out of the CODEC register is never sign-extended, and
when the minimum value is subtracted (read; added, since the minimum is
negative) the result is a number greater than the maximum allowed value
for the control, and hence it saturates.

Solution: Mask the result so that it "wraps around", emulating
sign-extension.

Signed-off-by: Stuart Longland <redhatter@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-19 02:33:44 +01:00
Eric Bénard 43793207fd ASoC: eukrea-tlv320: add support for our i.MX25 board
* tdm slot has to be configured to get sound working on i.MX25

Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-18 01:56:45 +01:00
Herton Ronaldo Krzesinski f7154de220 ALSA: hda - add ideapad model for Conexant 5051 codec
Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b,
which isn't muted when headphone is plugged in. This adds additional
support to the extra subwoofer via new ideapad model.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-17 20:37:40 +02:00
Andy Shevchenko c9ff921abe ALSA: alsa: riptide: don't use own hex_to_bin() method
Signed-off-by: Andy Shevchenko <ext-andriy.shevchenko@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-17 09:34:58 +02:00
Eliot Blennerhassett 2a383cb3f1 ALSA: asihpi - Get rid of incorrect "long" types and casts.
These give incorrect results for index wrap on 64 bit.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-17 09:33:59 +02:00
Jiri Kosina f1bbbb6912 Merge branch 'master' into for-next 2010-06-16 18:08:13 +02:00
Uwe Kleine-König 421f91d21a fix typos concerning "initiali[zs]e"
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2010-06-16 18:05:05 +02:00
Peter Huewe 66517915e0 ASoC: Fix I2C dependency for SND_FSI_AK4642 and SND_FSI_DA7210
The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn
enables the compilation of ak4642.c - however this codec uses I2C to
communicate with the HW.
Same applies to DA7210.

Consequently when I2C is not set, the compilation fails [1]

This patch fixes this issues, by adding a depencdency on the related HW-
controller.

Signed-off-by: Peter Huewe <peterhuewe@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-16 16:34:17 +01:00
Mark Brown f1df5aec68 ASoC: Pay attention to write errors in volsw_2r_sx
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-16 12:07:35 +01:00
Grant Likely f487537c2b powerpc/5200: Fix build error in sound code.
Compiling in the MPC5200 sound drivers results in the following build error:

sound/soc/fsl/mpc5200_psc_ac97.o: In function `to_psc_dma_stream':
mpc5200_psc_ac97.c:(.text+0x0): multiple definition of `to_psc_dma_stream'
sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here
sound/soc/fsl/efika-audio-fabric.o: In function `to_psc_dma_stream':
efika-audio-fabric.c:(.text+0x0): multiple definition of `to_psc_dma_stream'
sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here
make[3]: *** [sound/soc/fsl/built-in.o] Error 1
make[2]: *** [sound/soc/fsl] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2

This patch fixes it by declaring the inline function in the header file to
also be a static.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Cc: Jon Smirl <jonsmirl@gmail.com>
Tested-by: John Hilmar Linkhorst <John.Linkhorst@rwth-aachen.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-15 14:47:04 -06:00
Mark Brown e71fa37042 ASoC: Default WM2000 ANC and speaker to enabled
The most useful configuration for the WM2000 is to enable the ANC so turn
that on by default, and since we're not reflecting chip default state also
enable the speaker output by default.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-15 15:14:00 +01:00
Mark Brown 67884e215b Merge branch 'for-2.6.35' into for-2.6.36 2010-06-15 11:55:35 +01:00
Sudhakar Rajashekhara 5b61ea4997 ASoC: DaVinci: Fix McASP hardware FIFO configuration
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral
has FIFO support. This FIFO provides additional data buffering. It
also provides tolerance to variation in host/DMA controller response
times. More details of the FIFO operation can be found at

http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=sprufm1&fileType=pdf

Existing sequence of steps for audio playback/capture are:
a. DMA configuration
b. McASP configuration (configures and enables FIFO)
c. Start DMA
d. Start McASP (enables FIFO)

During McASP configuration, while FIFO was being configured, FIFO
was being enabled in davinci_hw_common_param() function of
sound/soc/davinci/davinci-mcasp.c file. This generated a transmit
DMA event, which gets serviced when DMA is started.

https://patchwork.kernel.org/patch/84611/ patch clears the DMA
events before starting DMA, which is the right thing to do. But
this resulted in a state where DMA was waiting for an event from
McASP (after step c above), but the event which was already there,
has got cleared (because of step b above).

The fix is not to enable the FIFO during McASP configuration as
FIFO was being enabled as part of McASP start.

Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-15 11:53:18 +01:00
Kuninori Morimoto 1a01eff1b2 ASoC: header cleanup for da7210
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>
2010-06-15 11:30:38 +01:00
Kuninori Morimoto 3367e452d9 ASoC: header cleanup for ak4642
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>
2010-06-15 11:30:37 +01:00
Kuninori Morimoto c3be0af3d0 ASoC: header cleanup for FSI-DA7210
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>
2010-06-15 11:30:37 +01:00
Kuninori Morimoto 6c8abb4987 ASoC: header cleanup for FSI-AK4642
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>
2010-06-15 11:30:36 +01:00
Kuninori Morimoto 8600d700c0 ASoC: header cleanup for FSI
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>
2010-06-15 11:30:34 +01:00
Takashi Iwai eb6e70417b Merge branch 'fix/misc' into for-linus 2010-06-15 12:24:05 +02:00
Takashi Iwai 8fda43c1a0 Merge branch 'fix/hda' into for-linus 2010-06-15 12:24:01 +02:00
Alex Murray b8f171e7e7 ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5,1 / 5,2
The line-in input is 0x7 not 0x2 for MacBook (Pro) 5,1 / 5,2 models

Signed-off-by: Alex Murray <murray.alex@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-14 09:12:21 +02:00
Grant Likely 4e8680f56b ASoC: Remove unused header from MPC5200 PSC driver
The header contains an extern that isn't used by anything.  Remove.

Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-12 18:06:14 +01:00
Daniel Mack e8bdb6bbab ALSA: usb-audio: fix UAC2 control value queries
For RANGE requests, we should only query as much bytes as we're in fact
interested in.

For CUR requests, we shouldn't confuse the firmware with an overlong
request but just ask for 2 bytes.

This might need fixing in the future as it's not entirely clear when to
dispatch 1-byte, 2-byte and 4-byte request blocks. For now, we assume
everything is coded in 16bit - this works for all firmware
implementations I've seen.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-11 18:06:35 +02:00
Daniel Mack 67c103664a ALSA: usb-audio: parse UAC2 sample rate ranges correctly
A device may report its supported sample rates in ranges rather than in
discrete triplets. The code used to only parse the MIN field instead of
properly paying attention to the MAX and RES values.

Also, handle RES values of 1 correctly and announce a continous sample
rate range in this case.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-11 18:06:12 +02:00
Daniel Mack 11bcbc443a ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACE
Control messages directed to an interface must have the interface number
set in the lower 8 bits of wIndex. This wasn't done correctly for some
clock and mixer messages.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-11 18:05:38 +02:00
Daniel Mack d07140ba7f ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2()
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-11 18:05:05 +02:00
Takashi Iwai fbe618f216 ALSA: hda - Don't check capture source mixer if no ADC is available
With multiple codec configurations, some codec might have no ADC, thus
it keeps spec->adc_nids = NULL.  This causes an Oops in alc_build_controls().

Reference: kernel bug #16156
	https://bugzilla.kernel.org/show_bug.cgi?id=16156

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-11 11:24:58 +02:00
Linus Torvalds e1f38e2cea 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: sound/spi: patch for the unuseful variable removal
  ALSA: hda - Add SSID table for iMac7,1.
  ALSA: hda - Add SSID table for MacBookAir1,1
  ALSA: hda - Add SSID table for MacBookAir2,1
  ALSA: atmel: set "channel A event" output to debug
2010-06-10 09:34:15 -07:00
Linus Torvalds 7c8d20d40f Merge master.kernel.org:/home/rmk/linux-2.6-arm
* master.kernel.org:/home/rmk/linux-2.6-arm:
  ARM: 6164/1: Add kto and kfrom to input operands list.
  ARM: 6166/1: Proper prefetch abort handling on pre-ARMv6
  ARM: 6165/1: trap overflows on highmem pages from kmap_atomic when debugging
  ARM: 6152/1: ux500 make it possible to disable localtimers
  [ARM] pxa/spitz: Correctly register WM8750
  [ARM] pxa/palmtc: storage class should be before const qualifier
  ARM: 6146/1: sa1111: Prevent deadlock in resume path
  ARM: 6145/1: ux500 MTU clockrate correction
  ARM: 6144/1: TCM memory bug freeing bug
  ARM: VFP: Fix vfp_put_double() for d16-d31
2010-06-10 07:35:41 -07:00
Wan ZongShun 019afb581a ASoC: NUC900: patch for fix build error
This patch is to change 'auido.h' to 'audio.h' for
fixing nuc900 alsa driver build error.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-10 14:40:35 +01:00
Takashi Iwai 2d0a1dbf57 Merge branch 'fix/misc' into for-linus 2010-06-10 11:08:53 +02:00
Ryan Mallon 315f7da631 ASoC: EP93xx: Add Snapper CL15 i2s audio support
Add support for i2s audio on Bluewater Systems Snapper CL15 module

Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-09 11:16:18 +01:00
Wan ZongShun ff8bd64eaf ALSA: sound/spi: patch for the unuseful variable removal
The '*bitclk' of structure 'snd_at73c213' seems no use,
so I make a patch to remove the unnecessary variable.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-08 16:51:27 +02:00
Justin P. Mattock ab669967d0 ALSA: hda - Add SSID table for iMac7,1.
This patch add's the iMac7,1 SSID entry to
patch_realtek.c which adds sound support.
bug entry:
    https://bugs.launchpad.net/mactel-support/+bug/360866

Note:I do not have this machine on hand only
codec#0 file for the machine so please
test if you have the appropriate equipment.

Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-08 16:48:56 +02:00
Justin P. Mattock f53dae28cd ALSA: hda - Add SSID table for MacBookAir1,1
This patch add's the MacBookAir1,1 SSID entry to
patch_realtek.c which adds sound support.
bug entry:
    https://bugs.launchpad.net/mactel-support/+bug/268301

Note:I do not have this machine on hand only
codec#0 file for the machine so please
test if you have the appropriate equipment.

Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-08 16:47:47 +02:00
Justin P. Mattock 6e12970bd4 ALSA: hda - Add SSID table for MacBookAir2,1
This adds the SSID number to snd_pci_quirk for the
MacBookAir2,1 taken from codec#0 at:
    http://launchpadlibrarian.net/49455483/Card0.Codecs.codec.0.txt

keep in mind I do not have one of these machines on hand
so please if you do have this machine please test for me..

Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-08 16:46:15 +02:00
Yegor Yefremov f534116308 ALSA: atmel: set "channel A event" output to debug
Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-08 16:42:02 +02:00
Takashi Iwai 9eb3430268 Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2010-06-07 18:38:56 +02:00
Wan ZongShun 04c09a15f5 ASoC: patch for the useless 'break' removal in kirkwood
This patch to remove the 'break;', when the 'switch' jumps to
the 'default' branch, the 'return -EINVAL' will be return with
a error number, so the 'break;' code never be run, it is unuseful
and should be removed here.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-07 14:27:18 +01:00
Wan ZongShun 911ff689ff ASoC: atmel: trivial code cleanup
Remove break after return, it is not needed.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-07 14:25:45 +01:00
Ryan Mallon db5bf412ba ASoC: ep93xx i2s audio driver
Add ep93xx i2s audio driver

Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-07 14:24:19 +01:00
Peter Ujfalusi 9d7db2b2cb ASoC: tlv320dac33: Add support for changing upper threshold
Upper threshold is used in mode7 of DAC33.
Instead of hard wired UTHR, add control to change the upper threshold
value.
Changing upper threshold is not allowed when the playback is already
running, since wrongly timed change in the UTHR can cause problems
with the codec.
With this control the length of the burst in mode7 can be changed.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-07 10:43:35 +01:00
Linus Torvalds bc23416cd4 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-intel - fix wallclk variable update and condition
  ALSA: asihpi - Fix uninitialized variable
  ALSA: hda: Use LPIB for ASUS M2V
  usb/gadget: Replace the old USB audio FU definitions in f_audio.c
  ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1
  ASoC: Add missing Kconfig entry for Phytec boards
  ALSA: usb-audio: export UAC2 clock selectors as mixer controls
  ALSA: usb-audio: clean up find_audio_control_unit()
  ALSA: usb-audio: add UAC2 sepecific Feature Unit controls
  ALSA: usb-audio: unify constants from specification
  ALSA: usb-audio: parse clock topology of UAC2 devices
  ALSA: usb-audio: fix selector unit string index accessor
  include/linux/usb/audio-v2.h: add more UAC2 details
  ALSA: usb-audio: support partially write-protected UAC2 controls
  ALSA: usb-audio: UAC2: clean up parsing of bmaControls
  ALSA: hda: Use LPIB for another mainboard
  ALSA: hda: Use mb31 quirk for an iMac model
  ALSA: hda: Use LPIB for an ASUS device
2010-06-04 09:48:03 -07:00
Eric Bénard 91157888f2 ASoC: imx: add eukrea-tlv320
Add the necessary files to support the TLV320AIC23B wired in I2S
on our i.MX platforms.

Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-03 19:00:38 +01:00
Eric Bénard 0e79612012 ASoC: imx-ssi.c: add new choices to platform configuration
* introduce 3 new flags to allow a more detailed configuration
of the SSI link :
	IMX_SSI_NET : enable Network Mode
	IMX_SSI_SYN : enable Synchronous Mode
	IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode
* new platform can use these settings without breaking actual
platforms.

Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-03 19:00:16 +01:00
Wan ZongShun 749266cd91 ASoC: s3c: patch for the unnecessary variable 'state' removal
The variable 'state' of structure 's3c_ac97_info' seems no use here,
so this patch is to remove the unnecessary variable.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-03 19:00:16 +01:00
Wan ZongShun b07adffbbc ASoC: atmel: patch for the unnecessary variable removal
The variable 'periods' of structure 'atmel_runtime_data'
seems no use in whole atmel alsa driver,so I make a patch
to remove the unnecessary variable.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-03 19:00:16 +01:00
Wan ZongShun c0da5500e9 ASoC: use resource_size for au1x
Use the resource_size function instead of manually calculating the
resource size.This patch can reduce the chance of introducing off-by-one
errors.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-03 19:00:16 +01:00
Peter Ujfalusi ddc29b0104 ASoC: omap-mcbsp: Place correct constraints for streams
OMAP McBSP FIFO is word structured:
McBSP2 has 1024 + 256 = 1280 word long buffer,
McBSP1,3,4,5 has 128 word long buffer

This means, that the size of the FIFO
depends on the McBSP word size configuration.
For example on McBSP3:
16bit samples: size is 128 * 2 = 256 bytes
32bit samples: size is 128 * 4 = 512 bytes
It is simpler to place constraint for buffer and period based on channels.
McBSP3 as example again (16 or 32 bit samples):
1 channel (mono): size is 128 frames (128 words)
2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
4 channels: size is 128 / 4 = 32 frames (4 * 32 words)

Use the second method to place hw_rule on buffer size, and in threshold
mode to period size.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-03 16:12:40 +01:00
Peter Ujfalusi 3f024039e0 ASoC: omap-mcbsp: Save, and use wlen for threshold configuration
Save the word length configuration of McBSP, and use that information
to calculate, and configure the threshold in McBSP.
Previously the calculation was only correct when the stream had 16bit
audio.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-06-03 16:12:40 +01:00
Wan ZongShun 5ef650ae5c ASoC: s6000: use resource_size for {request/release}_mem_region and ioremap
The size calculation is end - start + 1. But,sometimes, the '1' can
be forgotten carelessly, witch will have potential risk, so use resource_size
for {request/release}_mem_region and ioremap here should be good habit.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Daniel Glöckner <dg@emlix.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-02 14:02:39 +01:00
Takashi Iwai d437680299 Merge branch 'fix/asoc' into for-linus 2010-06-02 14:18:13 +02:00
Takashi Iwai c7a441bba9 Merge branch 'fix/hda' into for-linus 2010-06-02 14:18:06 +02:00
Takashi Iwai e854df613f Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc 2010-06-02 14:17:44 +02:00
Takashi Iwai e4caa8bab3 Merge branch 'master' of git.alsa-project.org:alsa-kernel into fix/hda 2010-06-02 14:15:10 +02:00
Jaroslav Kysela 8fc6d4186e ALSA: hda-intel - fix wallclk variable update and condition
This patch fixes thinko introduced in "last minutes" before commiting
of the last wallclk patch.

It also fixes the condition checking if the first period after last
wallclk update is processed. There is a little rounding error in
period_wallclk.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-06-02 13:34:18 +02:00
Jaroslav Kysela edb39935c8 ALSA: hda-intel - fix wallclk variable update and condition
This patch fixes thinko introduced in "last minutes" before commiting
of the last wallclk patch.

It also fixes the condition checking if the first period after last
wallclk update is processed. There is a little rounding error in
period_wallclk.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-06-02 13:34:01 +02:00
Mark Brown 85252b6ae5 Merge branch 'for-2.6.35' into for-2.6.36 2010-06-02 11:47:24 +01:00
Wan ZongShun 08a0b71757 ASoC: nuc900: patch for modifing the ac97 delays to minimum
This patch is to modify the ac97 delays to minimum, all these 1000 micro
seconds delays seem over spec for the AC97 interface.

I deleted some unnecessary delays here and changed the AC97 cold and warm reset
delays from 1000us to 100us.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-02 11:47:06 +01:00
Wan ZongShun 0dc3b44202 ASoC: nuc900: fix a typo and rename the header file
Fix a '#include "nuc900-audio.h' typo, I think it should be 'audio'.
At the same time, this patch renames the 'nuc900-auido.h' file to
'nuc900-audio.h'.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-02 11:47:06 +01:00
Wan ZongShun 8dfb0c7815 ASoC: nuc900: fix a wait loop bug
The current implement meant ACTL_ACCON was only accessed once when read or write
proceeding, which is not right, if so,we have to wait the 'timeout=0x10000' to end
every times.

We need to polling the bit AC_R_FINISH and AC_W_FINISH of ACTL_ACCON
register to identify whether read or write is finished or not,so I make
the patch to fix the issue.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-02 11:47:06 +01:00
Wan ZongShun 018334c045 ASoC: nuc900: patch for SUBSTREAM_TYPE', 'PCM_TX' and 'PCM_RX' removal
This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition.

There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX,
the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be
judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather
than 'if (PCM_TX == stype)', which makes the codes easy to read.

Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-06-02 11:47:06 +01:00
Takashi Iwai ead54d8784 Merge branch 'fix/hda' into for-linus 2010-06-02 12:09:29 +02:00
Takashi Iwai 21896bc010 ALSA: asihpi - Fix uninitialized variable
Initialize prev_ctl properly before reference:
  sound/pci/asihpi/asihpi.c: In function ‘snd_card_asihpi_mixer_new’:
  sound/pci/asihpi/asihpi.c:2568:30: warning: ‘prev_ctl.dst_node_index’ may be used uninitialized in this function

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-02 12:08:37 +02:00
Benjamin Herrenschmidt c2cdf6aba0 powerpc/macio: Fix probing of macio devices by using the right of match table
Grant patches added an of mach table to struct device_driver. However,
while he changed the macio device code to use that, he left the match
table pointer in struct macio_driver and didn't update drivers to use
the "new" one, thus breaking the probing.

This completes the change by moving all drivers to setup the "new"
one, removing all traces of the old one, and while at it (since it
changes the exact same locations), I also remove two other duplicates
from struct driver which are the name and owner fields.

Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
2010-06-02 17:50:38 +10:00
Daniel T Chen 9f75c1b12c ALSA: hda: Use LPIB for ASUS M2V
BugLink: https://launchpad.net/bugs/587546

Symptom: On the reporter's ASUS M2V, using PulseAudio in Ubuntu 10.04 LTS
results in the PA daemon crashing shortly after attempting playback of an
audio file.

Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, attempt playback of an audio file while PulseAudio is
active.

Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.

Reported-and-Tested-By: D Tangman
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-06-01 07:48:43 +02:00
Sascha Hauer 29512c95b5 ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-31 18:11:38 +01:00
Sascha Hauer fc9cbe3998 ASoC: Add missing Kconfig entry for Phytec boards
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-31 18:05:33 +01:00
Daniel Mack 09414207d4 ALSA: usb-audio: export UAC2 clock selectors as mixer controls
The UAC2 clock selectors are fortunately compatible with UAC1 audio
selector units, so we can simply reuse the same approach to get all the
linked units.

Requests to this control need a different CS value though.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 18:18:04 +02:00
Daniel Mack 67e1daa0bb ALSA: usb-audio: clean up find_audio_control_unit()
Use a struct to parse the audio units, and return usable descriptors
for all types. There's no need to limit the result set, except for some
kind of sanity check.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 18:17:49 +02:00
Daniel Mack 2e0281d15c ALSA: usb-audio: add UAC2 sepecific Feature Unit controls
The bits to enable them are always 0 for UAC1 devices, so no additional
checks are required.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 18:17:38 +02:00
Daniel Mack 65f25da44b ALSA: usb-audio: unify constants from specification
Move more definitions from private enums to appropriate header files.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 18:17:22 +02:00
Daniel Mack 79f920fbff ALSA: usb-audio: parse clock topology of UAC2 devices
Audio devices which comply to the UAC2 standard can export complex clock
topologies in its descriptors and set up links between them.

The entities that are defined are

 - clock sources, which define the end-leafs.
 - clock selectors, which act as switch to select one out of many
   possible clocks sources.
 - clock multipliers, which have an input clock source, and act as clock
   source again. They can be used to derive one clock from another.

All sample rate changes, clock validity queries and the like must go to
clock source elements, while clock selectors and multipliers can be used
as terminal clock source.

The following patch adds a parser for these elements and functions to
iterate over the tree and find the leaf nodes (clock sources).

The samplerate set functions were moved to the new clock.c file.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 18:16:59 +02:00
Daniel Mack a6a3325913 ALSA: usb-audio: support partially write-protected UAC2 controls
So far, UAC2 controls are marked read-only if any of the channels are
marked read-only in the descriptors. Change this behaviour and

 - mark them writeable unless all channels are read-only
 - store the read-only mask in usb_mixer_elem_info and
 - check the mask again in set_cur_mix_value(), and bail out for
   write-protected channels.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 18:15:57 +02:00
Daniel Mack dcbe7bcfa3 ALSA: usb-audio: UAC2: clean up parsing of bmaControls
Introduce two new static inline functions for a more readable parsing
of UAC2 bmaControls.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 18:15:45 +02:00
Takashi Iwai 1fab79b8a1 Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc 2010-05-31 18:13:20 +02:00
Takashi Iwai c876ae3eb2 Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2010-05-31 18:12:41 +02:00
Mark Brown 37a5ddf450 ASoC: Fix S/PDIF build
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-31 13:47:26 +01:00
apatard@mandriva.com 2e8693ee79 ASoC: kirkwood: Add audio support to openrd client platforms
This patch is adding support for openrd client platforms. It's using
the cs42l51 codec and has one mic and one speaker plugs.

Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-31 13:16:37 +01:00
apatard@mandriva.com f9b95980f8 ASoC: kirkwood: Add i2s support
This patch enables support for the i2s controller available on kirkwood
platforms

Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-31 13:16:37 +01:00
apatard@mandriva.com 72ed5a8c9b ASoC: Add driver for cs42l51
This patch is adding a ASoC driver for the cs42l51 from Cirrus Logic.
Master mode and spi mode are not supported.

Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.ul>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-31 12:20:02 +01:00
Seungwhan Youn 3a642915ad ASoC: spdif: Add codec driver to use spdif stand-alone
This patch adds spdif dummy codec driver for using spdif-dit as
a stand-alone. Until this, spdif-dit can be used only with other
codecs like tlv320aci3x in davinci platform.

Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-31 12:08:33 +01:00
Peter Ujfalusi a3a29b55c7 ASoC: TWL4030: Add functionalty to reset the registers
Machine driver can instruct the codec driver to reset the
chip registers to their default values at probe time.

If machine driver does not provide setup data, then the
registers are going to be reseted to their defaults, to
be safe.

If the developer on the platform confirms that the register
reset is not needed, than it can be skipped, saving ~20ms
time in probe.

As safety measure do the register reset at remove time also.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-31 11:08:59 +01:00
Peter Ujfalusi 2046f175bc ASoC: TWL4030: Use BIAS_OFF instead of BIAS_STANDBY, when not in use
Restructure the codec power code in order to be able to hit
off when the codec is not in use.

Since the audio registers are accessible while the codec is powered
down, there is no need for additional safety mechanism.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-31 11:08:58 +01:00
Peter Ujfalusi 3c36cc688e ASoC: TWL4030: Correct the ARXR2_APGA_CTL chip default
It seams at least on twl5031 that the ARXR2_APGA_CTL register
does not have the same default value as it is written in
the TRM.
Since the codec part of the PM chip has not been actually
changed according to TI, assuming, that all version has
the same problem, so writing there the TRM value.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-31 11:08:58 +01:00
Peter Ujfalusi 9fdcc0f72a ASoC: TWL4030: Helper to check chip default registers
Since the twl4030 codec driver supports different version
of the PM chip, a helper function can come handy, which
can check the driver's default versus the chip values.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-31 11:08:58 +01:00
Peter Ujfalusi ee4ccac7ce ASoC: TWL4030: Optimize the power up sequence
Since the reg cache now contains the chip default values
for all registers (REG_OPTION is reset to it's default
within this patch), there is no longer need to rewrite
_all_ registers.
Initialize only few selected registers.

According to the latest information, the offset cancellation
need to be done only once, when the codec is powered on, so
there is no need for the power up wrapper.

Move all chip initialization code under chip_init, and do
it when the codec is initialized.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-31 11:08:58 +01:00
Peter Ujfalusi 979bb1f4b8 ASoC: TWL4030: Make offset cancellation path configurable
Add means for machine drivers to select the path for offset
cancellation.
Reset the reg cache value to the chip reset value at the
same time.

Machine drivers can specify which path need to be used for
offset cancellation via the twl4030_setup.offset_cncl_path.
For paths use the defines from
include/linux/mfd/twl4030-codec.h:
TWL4030_OFFSET_CNCL_SEL_*

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-31 11:08:58 +01:00
Peter Ujfalusi cbd2db128f ASoC: TWL4030: Remove wrapper for power down
There is no need for the power down wrapper.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-31 11:08:58 +01:00
Peter Ujfalusi 33f92ed4b3 ASoC: TWL4030: Revisit codec defaults
Reset most of the codec registers to their chip reset
value.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-05-31 11:08:58 +01:00
Mark Brown e37c83c06c Merge commit 'v2.6.35-rc1' into for-2.6.36 2010-05-31 11:07:15 +01:00
Daniel T Chen b90c076424 ALSA: hda: Use LPIB for another mainboard
BugLink: https://launchpad.net/bugs/580749

Symptom: on the original reporter's VIA VT1708-based board, the
PulseAudio daemon dies shortly after the user attempts to play an audio
file.

Test case: boot from Ubuntu 10.04 LTS live cd; attempt to play an audio
file.

Resolution: add SSID for the original reporter's hardware to the
position_fix quirk table, explicitly specifying the LPIB method.

Reported-and-Tested-By: Harald
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 09:23:25 +02:00
Daniel T Chen 26fd74fc01 ALSA: hda: Use mb31 quirk for an iMac model
BugLink: https://launchpad.net/bugs/542550

Symptom: On the reporter's iMac, in Ubuntu 10.04 LTS neither playback
nor capture appear audible out-of-the-box.

Test case: Boot from an Ubuntu 10.04 LTS live cd or from an installed
configuration and attempt to play or capture audio.

Resolution: Specify the mb31 quirk for this machine in the codec SSID
table.

Reported-and-Tested-By: f3a97
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 09:22:59 +02:00
Daniel T Chen dd37f8e865 ALSA: hda: Use LPIB for an ASUS device
BugLink: https://launchpad.net/bugs/465942

Symptom: On the reporter's ASUS device, using PulseAudio in Ubuntu 10.04
LTS results in the PA daemon crashing shortly after attempting to select
capture or to configure the audio hardware profile.

Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, adjust the HDA device's capture volume with PulseAudio.

Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.

Reported-and-Tested-By: Irihapeti
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 09:22:28 +02:00
Ben Collins 15c0cee6c8 ALSA: pcm: Define G723 3-bit and 5-bit formats
This defines the 24bps and 40bps (8khz sample rate) G.723 codec
formats. They are going to be used once I submit the driver for
an mpeg4/g723 compression card.

I've updated the signed value to -1 as per Takashi's comments
since these are non-linear formats.

Signed-off-by: Ben Collins <bcollins@bluecherry.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-31 09:10:03 +02:00
Marek Vasut d30e5d897c [ARM] pxa/spitz: Correctly register WM8750
This patch registers the WM8750 codec on a proper place on the SPITZ machine
after the WM8750 driver was converted to new API.

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
2010-05-31 12:03:45 +08:00
Linus Torvalds 52b0ace7df 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: (26 commits)
  ALSA: snd-usb-caiaq: Bump version number to 1.3.21
  ALSA: Revert "ALSA: snd-usb-caiaq: Set default input mode of A4DJ"
  ALSA: snd-usb-caiaq: Simplify single case to an 'if'
  ALSA: snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
  ALSA: hda: Use LPIB for a Shuttle device
  ALSA: hda: Add support for another Lenovo ThinkPad Edge in conexant codec
  ALSA: hda: Use LPIB for Sony VPCS11V9E
  ALSA: usb-audio: fix feature unit parser for UAC2
  ALSA: asihpi - Minor code cleanup
  ALSA: asihpi - Add support for new ASI8800 family
  ALSA: asihpi - Fix bug preventing outstream_write preload from happening
  ALSA: asihpi - Fix imbalanced lock path in hw_message
  ALSA: asihpi - Remove support for old ASI8800 family
  ALSA: asihpi - Add hd radio blend functions
  ALSA: asihpi - Remove unused io map functions
  ALSA: usb-audio: add support for UAC2 pitch control
  ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
  ALSA: usb-audio: fix return values
  ALSA: usb-audio: parse more format descriptors with structs
  sound: Add missing spin_unlock
  ...
2010-05-29 15:31:57 -07:00
Takashi Iwai d6695f09ea Merge branch 'fix/hda' into for-linus 2010-05-29 21:50:36 +02:00
Takashi Iwai a98d3984c8 Merge branch 'fix/misc' into for-linus 2010-05-29 21:50:33 +02:00
Takashi Iwai 52593de4c1 Merge branch 'fix/asoc' into for-linus 2010-05-29 21:50:27 +02:00
Mark Hills 55567ab70b ALSA: snd-usb-caiaq: Bump version number to 1.3.21
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-29 21:49:11 +02:00
Mark Hills 649233562c ALSA: Revert "ALSA: snd-usb-caiaq: Set default input mode of A4DJ"
Do not explicity set the default input mode. Use the hardware default
of mode 0 ('Control vinyl'), which is now available.

This reverts commit e3ca4c9.

Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-29 21:49:00 +02:00
Mark Hills 4efd7d8f67 ALSA: snd-usb-caiaq: Simplify single case to an 'if'
After removing code, only one case remains. So use an 'if' instead.

Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-29 21:48:47 +02:00
Mark Hills bd4cbf6c76 ALSA: snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
This feature was undocumented on early A4DJ units. It is indicated
by lighting both the 'line' and 'phono' lamps at the same time.
Newer units document this and the newer Windows drivers enable this
for all units, so restore the functionality.

This patch simplifies the code and changes the mode mapping to match
the A8DJ, favouring simpler code and consistency over keeping the
existing mapping.

Both 'Control vinyl' and 'Phono' input modes enable the hardware
preamp. The difference is the input impedance.

This reverts commit 9a9527e.

Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-29 21:48:24 +02:00
Daniel T Chen 61bb42c37d ALSA: hda: Use LPIB for a Shuttle device
BugLink: https://launchpad.net/bugs/551949

Symptom: On the reporter's Shuttle device, using PulseAudio in Ubuntu
10.04 LTS results in "popping clicking" audio with the PA crashing
shortly thereafter.

Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, adjust the HDA device's volume with PulseAudio.

Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.

Reported-and-Tested-By: Christian Mehlis <mehlis@inf.fu-berlin.de>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-29 21:47:59 +02:00
Andreas Herrmann badf18b5f5 ALSA: hda: Add support for another Lenovo ThinkPad Edge in conexant codec
On a Thinkpad Edge 13 "01972NG" I had the problem that speakers played
sound although headphones were plugged in. Using model=ideapad with
latest alsa-git kernel fixed this. So adding this quirk to use ideapad
for another Thinkpad Edge variant seems sensible.

Cc: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Andreas Herrmann <andreas.herrmann3@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-28 10:03:31 +02:00
Daniel T Chen e96d312776 ALSA: hda: Use LPIB for Sony VPCS11V9E
BugLink: https://launchpad.net/bugs/586347

Symptom: On the Sony VPCS11V9E, using GStreamer-based applications with
PulseAudio in Ubuntu 10.04 LTS results in stuttering audio. It appears
to worsen with increased I/O.

Test case: use Rhythmbox under increased I/O pressure. This symptom is
reproducible in the current daily stable alsa-driver snapshots (at least
up until 21 May 2010; later snapshots fail to build from source due to
missing preprocessor directives when compiled against 2.6.32).

Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.

Reported-and-Tested-By: Lauri Kainulainen <lauri@sokkelo.net>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-28 07:51:17 +02:00
Daniel Mack e8d0fee70b ALSA: usb-audio: fix feature unit parser for UAC2
Fix a small off-by-one bug which causes the feature unit to announce a
wrong number of channels. This leads to illegal requests sent to the
firmware eventually.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-28 07:48:17 +02:00
Jassi Brar ce1f7d3076 ASOC: S5PV210: Enable AC97 support
The S5PV210 and S5PC110 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for S5PC110 and
S5PV210 also.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-27 04:05:49 -04:00
Jassi Brar 3dedece4a5 ASOC: S5PC100: Enable AC97 support
The S5PC100 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for
S5PC100 also.

Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-27 04:05:48 -04:00
Eliot Blennerhassett 3ee317fe9c ALSA: asihpi - Minor code cleanup
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-27 09:55:31 +02:00
Eliot Blennerhassett cadae4289d ALSA: asihpi - Add support for new ASI8800 family
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-05-27 09:55:16 +02:00