Initialize the capture source properly for auto model.
It's especially important for cases that only mic is detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the beep volume control to ALC268 codec support code.
Since the codec doesn't return the correct AMP caps, we need to override
the value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a private array for TLV entries of virtual master controls instead
of (supposed) static array. This cleans up the existing codes.
Also, now vmaster assumes the simple dB-range TLV that is the only type
it can handle.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the codes for virtual master controls to sound core part so that
not only hda-intel drivers can use it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of 8 channel sound for codecs that are known to work.
So far, only ALC850 is marked as a 8ch-support codec.
This fix is a modified version of the patch on ALSA BTS#2097 by
Martin Ellis:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2097
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds __devinit to the functions used when probing. Will also reduce
the memory footprint a bit if CONFIG_HOTPLUG is not enabled.
Signed-off-by: Hans-Christian Egtvedt <hcegtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support of new AD codecs: AD1883, AD1884A, AD1984A and AD1984B.
These are almost compatible except for additional digital pins, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds an ASoC driver for the WM9713 AC97 codec.
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC262 must have capsrc_nids defined as well as in ALC882.
Also, add a NULL check in alc882_mux_enum_put to avoid Oops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The last patch for fixing the auto-config pin setting breaks the resume
due to a wrong use of snd_hda_codec_amp_stereo(). The code in the init
hook shouldn't touch the amp cache.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new mixer switch to enable/disable the sharing of the default
PCM stream with analog and SPDIF outputs. When "IEC958 Default PCM"
switch is on, the PCM stream is routed both to analog and SPDIF outputs.
This is the behavior in the earlier version.
Turning this switch off has a merit for some codecs, though. Some codec
chips don't support 24bit formats for SPDIF but only for analog outputs.
In this case, you can use 24bit format by disabling this switch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes some bugs in the auto-configurator of Realtek codecs:
- add missing pin set-up for speaker pins
- fix the speaker auto-mute function not to conflict with the existing
"Speaker" mixer switch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In some cases, the BIOS sets up only the HP pins with different assoc
and sequence numbers, e.g. on FSC Esprimo with ALC262.
This patch adds a fix-up for such a case. When multiple HPs are defined
and no line-outs is found, the configurator tries to re-assign some pins
from HP list to line-out, judging from the sequence number.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implemented the auto-mic jack sensing for Samsung laptops with AD1986A
codec chip (model=laptop-eapd).
The hardware uses pin 0x1d and 0x1f for the internal and external
mics, respectively.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the codes of the capture source selection for Realtek codecs.
Now using common helper functions with the new capsrc_nids field.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reprogram the speaker-pin setting at each HP pin plug to make sure
the spekaer auto-muting on AD1981HD hp model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I've just noticed that there are a handful of duplicate controls in the
ALC268 test model mixer. This patch (against alsa-driver 1.0.16) removes
them.
Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
See ALSA bug#3327 for more details. Experimental.
Also fix support for M-Audio Delta 1010E - subdevice check.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The HD-audio hardware usually supports 64bit address for DMA and other
buffers. The patch enables the feature if supported.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On several laptops that have STAC9228 codecs have unused DACs,
this powers them down to a D3 state.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes the problems with Midi In on Hoontech/STA dsp24 cards, for example with
DSP2000 box, without restricting the box configurations available. Also adds
mpu_401 name strings.
Signed-off-by: Alan Horstmann <gineera@aspect135.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the current scheme, PCM device numbers are assigned incrementally
in the order of codecs. This causes problems when the codec number
is irregular, e.g. codec #0 for HDMI and codec #1 for analog. Then
the HDMI becomes the first PCM, which is picked up as the default
output device. Unfortuantely this doesn't work well with normal
setups.
This patch introduced the fixed device numbers for the PCM types,
namely, analog, SPDIF, HDMI and modem. The PCM devices are assigned
according to the corresponding PCM type. After this patch, HDMI will
be always assigned to PCM #3, SPDIF to PCM #1, and the first analog
to PCM #0, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added more fallbacks to OSS PHONEOUT mixer mapping. This corresponds
to the speaker output in general, so now "Mono" and "Speaker" are
assigned.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code doesn't allow multiple SPDIF devices, and causes
errors when multiple SPDIF devices are found (e.g. SPDIF out and HDMI).
This patch allows multiple SPDIF devices by incrementing the index
automatically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the OQO Model 2 Ultra Mobile PC.
Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently you will see an empty "SoC Audio support for SuperH" menu
when building for other archs (example pxa).
This patch adds "depends on SUPERH" to remove that empty menu.
Signed-off-by: Kristoffer Ericson <kristoffer.ericson@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Voice playback volume is in register bits 0:2, not 4:6.
From: Mike Montour <mail@mmontour.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Werner Almesberger <werner@openmoko.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/viro/vfs-2.6:
[patch 7/7] vfs: mountinfo: show dominating group id
[patch 6/7] vfs: mountinfo: add /proc/<pid>/mountinfo
[patch 5/7] vfs: mountinfo: allow using process root
[patch 4/7] vfs: mountinfo: add mount peer group ID
[patch 3/7] vfs: mountinfo: add mount ID
[patch 2/7] vfs: mountinfo: add seq_file_root()
[patch 1/7] vfs: mountinfo: add dentry_path()
[PATCH] remove unused label in xattr.c (noise from ro-bind)
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/net-2.6:
iwlwifi: Fix built-in compilation of iwlcore
net: Unexport move_addr_to_{kernel,user}
rt2x00: Select LEDS_CLASS.
iwlwifi: Select LEDS_CLASS.
leds: Do not guard NEW_LEDS with HAS_IOMEM
[IPSEC]: Fix catch-22 with algorithm IDs above 31
time: Export set_normalized_timespec.
tcp: Make use of before macro in tcp_input.c
hamradio: Remove unneeded and deprecated cli()/sti() calls in dmascc.c
[NETNS]: Remove empty ->init callback.
[DCCP]: Convert do_gettimeofday() to getnstimeofday().
[NETNS]: Don't initialize err variable twice.
[NETNS]: The ip6_fib_timer can work with garbage on net namespace stop.
[IPV4]: Convert do_gettimeofday() to getnstimeofday().
[IPV4]: Make icmp_sk_init() static.
[IPV6]: Make struct ip6_prohibit_entry_template static.
tcp: Trivial fix to correct function name in a comment in net/ipv4/tcp.c
[NET]: Expose netdevice dev_id through sysfs
skbuff: fix missing kernel-doc notation
[ROSE]: Fix soft lockup wrt. rose_node_list_lock
