Joe Sauer [Thu, 10 Jan 2008 13:34:56 +0000 (14:34 +0100)]
[ALSA] Fix inverted Phone volume WM9712 mixer control
Signed-off-by: Joe Sauer <jsauer@vernier.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Matthew Ranostay [Thu, 10 Jan 2008 12:06:26 +0000 (13:06 +0100)]
[ALSA] hda: STAC9228 VT fixes
Moved 2 systems PCI_QUIRK values to STAC_DELL_BIOS. Also the second
front HP jack is incorrect defined in the BIOS VT's for some laptops,
this patch corrects this.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Jiang zhe [Thu, 10 Jan 2008 12:05:47 +0000 (13:05 +0100)]
[ALSA] hda-codec - Device ID for Macbook sound card
Please refer to the [0003680] on ALSA bugtracking system.
The user found that 'model=mbp3' works and provided the ID.
From: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Marcin Ślusarz [Wed, 9 Jan 2008 16:56:07 +0000 (17:56 +0100)]
[ALSA] rawmidi: let sparse know what is going on _for real_
snd_rawmidi_kernel_read1/write1 weren't annotated but used
copy_to_user/copy_from_user when one of parameters (kernel) was equal to 0
remove it and add properly annotated parameter
Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added a mono output mute mixer for the 92hd71bxx family of codecs, this
also removes the need for the mono out node to explicitly unmuted in the
core init.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Tue, 8 Jan 2008 17:13:27 +0000 (18:13 +0100)]
[ALSA] Remove sound/driver.h
This header file exists only for some hacks to adapt alsa-driver
tree. It's useless for building in the kernel. Let's move a few
lines in it to sound/core.h and remove it.
With this patch, sound/driver.h isn't removed but has just a single
compile warning to include it. This should be really killed in
future.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Tue, 8 Jan 2008 17:09:57 +0000 (18:09 +0100)]
[ALSA] Remove PCM sleep_min and tick
The 'tick' in PCM is set (again) via sw_params. And, nobody uses
this feature at all except for a command line option of aplay.
(This is literally 'nobody', as I checked alsa-lib API calls in all
programs in major distros.)
Above all, if we need finer wake-ups for the position update, it's
basically an issue that the driver should solve, not tuned by each
application.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Tue, 8 Jan 2008 17:05:26 +0000 (18:05 +0100)]
[ALSA] Remove PCM xfer_align sw params
The xfer_align sw_params parameter has never been used in a sane manner,
and no one understands what this does exactly. The current
implementation looks also buggy because it allows write of shorter size
than xfer_align. So, if you do partial writes, the write isn't actually
aligned at all.
Removing this parameter will make some pcm_lib_* code more readable
(and less buggy).
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Tue, 8 Jan 2008 17:00:04 +0000 (18:00 +0100)]
[ALSA] Fix PCM write blocking
The snd_pcm_lib_write1() may block in some weird condition:
- the stream isn't started
- avail_min is big (e.g. period size)
- partial write up to buffer_size - avail_min
The patch fixes this invalid blocking problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Tue, 8 Jan 2008 16:57:26 +0000 (17:57 +0100)]
[ALSA] Remove indirect control access
This patch removes the indirect control access to the control elements.
The indirect access has never been used and is even broken on 32bit
ioctl wrapper. Let's clean it up.
The pointers still remain in snd_ctl_elem_* structs just to make sure
that the struct size won't change. Once after checking the size
consistency, we can get rid of them, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Jonathan Woithe [Tue, 8 Jan 2008 11:33:19 +0000 (12:33 +0100)]
[ALSA] hda-codec - Add test model for ALC268
This implements a test model for the ALC268. It depends on the feature
added by alc260-test-eapd-0.2.diff. This patch also adds a mention of
the ALC260 test model to ALSA-Configuration.txt since this seems to have
been missed.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Jonathan Woithe [Tue, 8 Jan 2008 11:16:54 +0000 (12:16 +0100)]
[ALSA] hda-codec - Add EAPD controls for ALC260 test model
This implements a switch control for the EAPD signal output by the ALC26x
chips. Since some laptops may utilise this to activate useful things it
is handy to have a control for this in the ALC26x test models. The patch
includes the control in the ALC260 test model.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Jaroslav Kysela [Tue, 8 Jan 2008 11:24:01 +0000 (12:24 +0100)]
[ALSA] PCM core - remove SNDRV_PCM_TSTAMP_MMAP condition in snd_pcm_status()
The condition caused that the returned ring buffer position does not match
with timestamp when SNDRV_PCM_TSTAMP_MMAP mode was enabled. Removing
condition makes unified behaviour and interrupt based timestamp can be
accessed via PCM_IOCTL_SYNC_PTR or mmaped status area.
[ALSA] hda: Dynamically create digital gain mixers
Dynamically create digital gain mixers for dmics that have out-amp
support. Also some 92HD73xx's codecs don't have DMIC gains, so this also
prevents creating dead mixers.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Mon, 7 Jan 2008 14:16:37 +0000 (15:16 +0100)]
[ALSA] hda-intel - Support multiple devices
It turned out that there can be multiple HD-audio devices on a single
machine (e.g. on-board audio and HDMI on graphic cards), so we need to
support multiple devices with snd-hda-intel driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Julia Lawall [Mon, 7 Jan 2008 12:33:45 +0000 (13:33 +0100)]
[ALSA] sound: Use time_before, time_before_eq, etc.
The functions time_before, time_before_eq, time_after, and time_after_eq
are more robust for comparing jiffies against other values.
A simplified version of the semantic patch making this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@ change_compare_np @
expression E;
@@
(
- jiffies <= E
+ time_before_eq(jiffies,E)
|
- jiffies >= E
+ time_after_eq(jiffies,E)
|
- jiffies < E
+ time_before(jiffies,E)
|
- jiffies > E
+ time_after(jiffies,E)
)
@ include depends on change_compare_np @
@@
#include <linux/jiffies.h>
@ no_include depends on !include && change_compare_np @
@@
#include <linux/...>
+ #include <linux/jiffies.h>
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Krzysztof Helt [Mon, 7 Jan 2008 11:24:45 +0000 (12:24 +0100)]
[ALSA] es18xx: Enable wavetable input from ESS chips
This patch enables wavetable chips ES689/ES69X connected to
ESS ES18xx chips. The wavetable chip uses FM DAC if the clock signal
from the wavetable is active.
It has no effect if there is no ESS wavetable chip present.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Jerone Young [Mon, 7 Jan 2008 11:22:18 +0000 (12:22 +0100)]
[ALSA] hda-codec - Add IEC958 digital out support for Lenovo Thinkpads T61/X61
This patch adds IEC958 digital out support for the AD1984 sound card.
This card can be found in Lenovo Thinkapds T61/X61. The digital out is
not located on the Thinkpad, but optional docking station (it's coxial
digital out). I've add this support as it is done the exact same way
for the AD1983 & AD1884.
I have tested this patch with my Lenovo Thinkpad T61 hooked up to a
docking station (that has the digital coxial) and then run to my home
theater reciever. Works like a charm :-)
Signed-off-by: Jerone Young <jerone@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Some laptops incorrectly assume the front input jack as a line in
instead of a microphone in. Which in turn disables the voltage
reference, in which non-amplified input is not possible. This patch
enables VREF80 for the input jack.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Clemens Ladisch [Fri, 4 Jan 2008 08:22:20 +0000 (09:22 +0100)]
[ALSA] oxygen: use uintptr_t in pointer casts
When we store the DMA channel number in the substream's private_data
pointer, use uintptr_t as an intermediate step when casting from/to
unsigned int to prevent the compiler from whining when the pointer size
is different.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Clemens Ladisch [Sun, 23 Dec 2007 18:50:57 +0000 (19:50 +0100)]
[ALSA] add CMI8788 driver
Add the snd-oxygen driver for the C-Media CMI8788 (Oxygen) chip, used on
the Asound A-8788, AuzenTech X-Meridian, Bgears b-Enspirer,
Club3D Theatron DTS, HT-Omega Claro, Razer Barracuda AC-1,
Sondigo Inferno, and TempoTec HIFIER sound cards.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Jiang Zhe [Thu, 20 Dec 2007 12:13:13 +0000 (13:13 +0100)]
[ALSA] hda-codec - alc268 input_mux should be a selector instead of mixer
According to the [0003659], the node 0x23,0x24 is a selector.
I checked the alc268 spec on the REALTEK website and it showed that they
were selectors indeed.
However, current code implement the alc268 input_mux in a mixer way.
[ALSA] hda-codec - Fix capture mixers of ALC662 models
The commit that added support for ASUS P701 eeepc also changed the
mixers of other ALC662 models, duplicating entries for the Capture
items, making them to not work anymore. This fixes it by removing
duplicated entries using where possible the common alc662_capture_mixer.
Also alc662_capture_mixer should use alc662* functions and not alc882
(I checked /proc/asound/card0/codec* on an eepc model and it's ok).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to
AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added
AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK,
and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet
rev 1.3.
Signed-off-by: Andrew Paprocki <andrew@ishiboo.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Tue, 18 Dec 2007 17:05:52 +0000 (18:05 +0100)]
[ALSA] hda-codec - Fix invalid access to non-existing dmux on STAC
The digital mux on STAC codecs doesn't always exist although the
driver builds dmux enum mixer elements unconditionally.
Now the driver creates 'digital input source' mixer elements only
when dmux is available.
Also, the patch adds the missing dmux definition for STAC925x.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Timur Tabi [Tue, 18 Dec 2007 14:42:53 +0000 (15:42 +0100)]
[ALSA] cs4270: wrong sample rate when CONFIG_SND_SOC_CS4270_VD33_ERRATA is set
When CONFIG_SND_SOC_CS4270_VD33_ERRATA is set, there was a mismatch between
the mclk_ratios[] and cs4270_mode_ratios[] arrays. The two arrays have been
merged and code has been shuffled. One side effect is that the
cs4270_set_dai_sysclk() and cs4270_set_dai_fmt() functions are available only
if I2C has been enabled.
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
[ALSA] at73c213: replace spinlock in mixer functions with a mutex
This patch fixes the locking bug in the at73c213 SPI sound driver. This bug was
triggered because spinlocks were wrapped around the spi_sync call which might
sleep. The fix was to add a mutex to the sound driver and replace the spinlocks
in the mixer functions with mutex lock/unlock.
Tested on STK1000/STK1002.
Takashi Iwai [Mon, 17 Dec 2007 16:14:18 +0000 (17:14 +0100)]
[ALSA] hda-codec - sort pci quirk list
Sort pci quirk list in the order of PCI SSID.
This makes easier to find out the buggy duplicated entries.
Thanks to Andy Shevchenko for providing the sort script.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
David Dillow [Fri, 14 Dec 2007 13:40:23 +0000 (14:40 +0100)]
[ALSA] sis7019: support the SiS 7019 Audio Accelerator
Basic audio support for the SiS 7019 Audio Accelerator as found in the
SiS 55x SoC. There is currently no synth support at the moment, but
audio playback and capture with two periods per buffer has seen
extensive use. Arbitrary period and buffer sizes (with multiple periods
per buffer) have seen light testing, but are believed to be production
ready.
Signed-off-by: David Dillow <dave@thedillows.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Andrew Morton [Fri, 14 Dec 2007 11:13:12 +0000 (12:13 +0100)]
[ALSA] copy_ctl_value_from_user() warning fix
sound/core/control_compat.c: In function 'copy_ctl_value_from_user':
sound/core/control_compat.c:222: warning: 'count' may be used uninitialized in this function
Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Matthew Ranostay [Thu, 13 Dec 2007 16:47:21 +0000 (17:47 +0100)]
[ALSA] hda: Added STAC92HD73 support
Added support for new STAC92HD73 family of codecs. Additionally added
features for multiple analog loopbacks, and multiple dmux mixers.
Regression testing for the analog loopback changes for STAC9205 and
STAC9274D completed with any issues, as well for the dmux changes.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Andy Shevchenko [Thu, 13 Dec 2007 16:32:26 +0000 (17:32 +0100)]
[ALSA] hda-codec - Initial support of the Mitac 8252D (based on ALC883)
The attached patch adds initial support of the Mitac 8252D
(http://www.mitac-mtc.com.tw/English/products/8252Dspec.htm).
Working:
- Front speakers (volume + mute)
- Center/LFE speakers (volume + mute)
- HP out (with Front Volume)
- HP individual mute switch
- HP Jack sense
- Front Mic and its volume
Not tested:
- external mic and its volume
Not working while now:
- Mic Jack sense
Questionable:
- is Mic have Jack sense?
- one or two Mic volume controls?
- CD/Line-in: presense in the mixer
Signed-off-by: Andy Shevchenko <andy@smile.org.ua> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Jaroslav Kysela [Thu, 13 Dec 2007 09:19:42 +0000 (10:19 +0100)]
[ALSA] Use posix clock monotonic for PCM and timer timestamps
We need an accurate and continuous (monotonic) time sources to do
accurate synchronization among more timing sources. This patch allows
to enable monotonic timestamps for ALSA PCM devices and enables monotonic
timestamps for ALSA timer devices.
Pavel Hofman [Mon, 3 Dec 2007 11:44:28 +0000 (12:44 +0100)]
[ALSA] switching rate in STAC9460 codec of Prodigy192
* support for switching rate in STAC9460 - using set_rate_val of the akm
infrastructure
* listing all STAC9460 registers in proc
* disabling mpu401 device for Prodigy192 - otherwise the currently
flawed mpu401 code hangs kernel when opening the midi device
* removing old unused commented-out code
Signed-off-by: Pavel Hofman <dustin@seznam.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Rene Herman [Fri, 30 Nov 2007 16:59:25 +0000 (17:59 +0100)]
[ALSA] sound/isa: kill pnp_resource_change
This removes the pnp_resource_change use from the ALSA ISAPnP drivers. In
2.4 these were useful in providing an easy path to setting the resources,
but in 2.6 they retain function as a layering violation only.
This makes for a nice cleanup (-550 lines) of ALSA but moreover, ALSA is the
only remaining user of pnp_init_resource_table(), pnp_resource_change() and
pnp_manual_config_dev() (and, in fact, of 'struct pnp_resource_table') in
the tree outide of drivers/pnp itself meaning it makes for more cleanup
potential inside the PnP layer.
Thomas Renninger acked their removal from that side, you did from the ALSA
side (CC list just copied from that thread).
Against current alsa-kernel HG. Many more potential cleanups in there, but
this _only_ removes the pnp_resource_change code. Compile tested against
current alsa-kernel HG and compile- and use-tested against 2.6.23.x (few
offsets). Cc: Thomas Renninger <trenn@suse.de> Signed-off-by: Rene Herman <rene.herman@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
There should be a pci_dev_put when breaking out of a loop that iterates
over calls to pci_get_device and similar functions.
In this case, the return under the initial if needs a pci_dev_put in the
same way that the return under the subsequent for loop has a pci_dev_put.
This was fixed using the following semantic patch.
// <smpl>
@@
type T;
identifier d;
expression e;
@@
T *d;
...
while ((d = \(pci_get_device\|pci_get_device_reverse\|pci_get_subsys\|pci_get_class\)(..., d)) != NULL)
{... when != pci_dev_put(d)
when != e = d
(
return d;
|
+ pci_dev_put(d);
? return ...;
)
...}
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Tue, 27 Nov 2007 14:27:17 +0000 (15:27 +0100)]
[ALSA] ice1712 - Fix word clock status control on Delta 1010LT
The 'Word Clock Status' control on Delta 1010LT checks the CS8427
error register too strictly and almost always returns 1 (unlocked).
It should check only the lock status bit.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Ville Syrjala [Mon, 26 Nov 2007 07:58:24 +0000 (08:58 +0100)]
[ALSA] soc/wm8731: Fix stereo mixer controls
Disable the simultaneous load feature for the line in and headphone
out volume registers. This allows left and right volume levels to
be controlled separately.
Signed-off-by: Ville Syrjala <syrjala@sci.fi> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Heikki Lindholm [Fri, 23 Nov 2007 14:37:48 +0000 (15:37 +0100)]
[ALSA] add number of periods constraint to snd-aoa
The aoa driver is not specifying constraints on number of periods, and, it
seems, it might end with a non-integer number, which it cannot deal with.
Fix by adding a proper constraint.
Signed-off-by: Heikki Lindholm <holindho@cs.helsinki.fi> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Fri, 23 Nov 2007 12:14:23 +0000 (13:14 +0100)]
[ALSA] Fix PCM MMAP time-stamp mode
When MMAP time-stamp mode is given, it's supposed to update the time-stamp
only at period boundary. However, it currently updates at each status call
so this is just useless. The patch fixes this misbehavior.
Also it fixes the wrong check of tstamp_mode (don't use bit-and for enum).
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Daniel Mack [Thu, 22 Nov 2007 10:40:04 +0000 (11:40 +0100)]
[ALSA] caiaq - add control API and more input features
- added support for all input controllers on Native Instrument's 'Kore
controller'.
- added ALSA controls to switch LEDs on 'RigKontrol 2', 'RigKontrol3',
'Audio Kontrol 1' and 'Kore controller'.
- added ALSA controls to switch input mode, software lock and ground
lift features on 'Audio 8 DJ'.
Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Dmitry Torokhov [Wed, 21 Nov 2007 15:45:23 +0000 (16:45 +0100)]
[ALSA] caiaq - misc input handling fixes
- link input device with its parent so that it placed in proper spot
in sysfs hierarchy
- drivers that allow changing their keymaps should use private copy
of the keymap so that one instance of a device does not affect
another instance
- it is preferred for drivers to properly set up input_dev->phys to
help userspace locate devices
- drivers should use usb_to_input_id(), or perform endianess conversion,
themselves, otherwise ID is not correct on big-endian boxes
- whitespace and formatting cleanup Acked-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Dmitry Torokhov <dtor@mail.ru> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kamalesh Babulal [Tue, 20 Nov 2007 14:12:33 +0000 (15:12 +0100)]
[ALSA] powermac - Fix typos
The kernel build fails, with following error
CC sound/ppc/tumbler.o
sound/ppc/tumbler.c: In function ‘snapper_get_capture_source’:
sound/ppc/tumbler.c:812: error: ‘union <anonymous>’ has no member named ‘value’
sound/ppc/tumbler.c: In function ‘snapper_put_capture_source’:
sound/ppc/tumbler.c:824: error: ‘union <anonymous>’ has no member named ‘enuemerated’
make[2]: *** [sound/ppc/tumbler.o] Error 1
make[1]: *** [sound/ppc] Error 2
make: *** [sound] Error 2
projects / linux-2.6-omap-h63xx.git / log