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
Takashi Iwai [Mon, 19 Nov 2007 10:56:26 +0000 (11:56 +0100)]
[ALSA] hda-codec - Revert volume knob controls in STAC codecs
Volume knob controls with STAC codecs seem to cause problems with some
devices. Volumes change very slowly or silent suddenly. It's likely
due to conflict between the software and the hardware volume knob
setup.
Since we'll have a virtual master control in future, it's safer to
remove this control completely right now.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
not sleeping for every codec read/write but doing a short udelay and
a conditional reschedule has cut suspend+resume latency by about 1
second on my T60.
Wolke Liu [Fri, 16 Nov 2007 10:06:30 +0000 (11:06 +0100)]
[ALSA] HDA-Intel - Add support for RV6xx HDMI audio
This patch is to add R6xx HDMI audio support. Meanwhile, the device ID
in the previous patch is changed.
I have checked the patch from Herton Ronaldo Krzesinski, it's right as
our spec said. :)
Signed-off-by: Wolke Liu <Wolke.Liu@amd.com> Signed-off-by: Andrea Zhang <Andrea.Zhang@amd.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Takashi Iwai [Thu, 15 Nov 2007 12:16:02 +0000 (13:16 +0100)]
[ALSA] emu10k1 - Check value ranges in ctl callbacks
Check value ranges in ctl callbacks properly. This fixes the unexpected
crash due to wrong value assignment.
Also, remove invalid comments in the last patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Vladimir Barinov [Wed, 14 Nov 2007 16:07:17 +0000 (17:07 +0100)]
[ALSA] ASoC TLV320AIC3X codec driver
This patch adds ALSA SoC support for TI TLV320AIC3X audio codecs.
The features that are supported:
o Capture/Playback/Bypass.
o 16/20/24/32 bit audio.
o 8k - 96k sample rates.
o codec master only mode
o DAPM.
Signed-off-by: Vladimir Barinov <vbarinov@ru.mvista.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Julia Lawall [Wed, 14 Nov 2007 13:30:43 +0000 (14:30 +0100)]
[ALSA] sound/pci: Drop unnecessary continue
Continue is not needed at the bottom of a loop.
The semantic patch implementing this change is as follows:
@@
@@
for (...;...;...) {
...
if (...) {
...
- continue;
}
}
Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Jiang Zhe [Mon, 12 Nov 2007 11:57:03 +0000 (12:57 +0100)]
[ALSA] hda-codec - Add workaround for multiple HPs
Dell laptops have multiple HP jacks that can be used for multi-channel
outputs. The current auto pincfg handles the speaker as the primary
output and thus cannot handle the multi-channel configuration for such
cases. This patch adds a workaround to fix this issue by swapping the
HP and speaker during multi-channel setup routines.
Stanislav Brabec [Mon, 12 Nov 2007 11:11:10 +0000 (12:11 +0100)]
[ALSA] use convenient treble scale on WM8750
On Zaurus SL-C3200 (terrier/spitz) based on WM8750, treble scale is
inconveniently reverted (increase level = decrease treble), in opposite
to bass scale, which uses convenient scale.
Fix ALSA WM8750 mixer treble to use convenient treble scale (increase =
increase treble level)
From: Stanislav Brabec <utx@penguin.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Nicolas Kaiser [Wed, 7 Nov 2007 17:31:43 +0000 (18:31 +0100)]
[ALSA] sound/pci: remove line duplications in defines
Remove line duplications in defines. Acked-by: Thomas Sailer <sailer@ife.ee.ethz.ch> Signed-off-by: Nicolas Kaiser <nikai@nikai.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
[ALSA] cmipci - allow capture of raw spdif subframes
Enable capturing of raw 32bit IEC958_SUBFRAME.
The 24-bits PCM data can be obtained using iec958 plugin.
Known problem: captured stream may begin with either left or right
subframe. Since the iec958 plugin doesn't decode preamble it may swap
the channels sometime.
Setting the ADC48K44K greatly improves capture quality at 48k sampling rate.
With this bit clear ADC does ZOH interpolation of every 22th sample at 48k.
At frequencies higher than 48k there ADC performs a little better with
ADC48K44K bit set.
At 44.1k ADC performs a little better with this bit clear.
At frequencies below 44.1k there is no difference.
projects / linux-2.6 / log