Added the support of AD1989A and AD1989B codecs.
These codecs can have multiple SPDIF devices, but currently we handle
only one SPDIF. If any real devices with two SPDIF interfaces (likely
one for SPDIF and one for HDMI), we'll fix this rightly.
Otherwise, these codecs are pretty similar with AD1988.
Jarkko Nikula [Mon, 14 Apr 2008 13:28:19 +0000 (15:28 +0200)]
[ALSA] ASoC: Add support for 19.2 MHz MCLK in TLV320AIC3X
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Mon, 14 Apr 2008 13:27:30 +0000 (15:27 +0200)]
[ALSA] wm9713: Don't control touch screen power on suspend
Leave the power bit for the touch screen alone when suspending the WM9713
so that the touch screen driver can handle it. This allows the touch
screen to be used as a wakeup source when the system is suspended.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds support for Quanta IL1 mini-notebook to alsa, defining a new model
for it. It comes with an ALC267 codec chip. Some notes about this model:
* In headphone automute, I use AC_VERB_SET_PIN_WIDGET_CONTROL instead of common
amp mute, to avoid conflict with mixer switch (mixer and automute use the
same nid).
* The only connected capture sources in the hardware are the internal mic and
external mic jack. So instead of using an input source selector like on other
ALC268 models, the mic automute automatically switch between captures.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since 43cc71eed1250755986da4c0f9898f9a635cb3bf, the platform modalias is
prefixed with "platform:". Add MODULE_ALIAS() to the hotpluggable sound
platform drivers, to re-enable auto loading.
[dbrownell@users.sourceforge.net: more drivers, registration fixes]
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org> Signed-off-by: David Brownell <dbrownell@users.sourceforge.net> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tony Vroon [Mon, 14 Apr 2008 11:31:45 +0000 (13:31 +0200)]
[ALSA] hda - Fujitsu Lifebook PC speaker signal
The legacy PC speaker signal was not routed to outputs. The codec is not
prevented from powering down in this patch, although I suppose one could
argue that perhaps it should be. Let me know if anyone feels strongly one
way or the other.
Signed-off-by: Tony Vroon <tony@linx.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
[ALSA] hda - Should use HDA_OUTPUT instead of HDA_INPUT to mute pin 15 of ALC880
To mute the output of Pin widget 15 in ALC880, we should use the
HDA_OUTPUT. However, current code looks like :
snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits);
It may be a misspelling.
Andrew Morton [Mon, 14 Apr 2008 11:09:33 +0000 (13:09 +0200)]
[ALSA] es1968 - fix coding style in the last patch
WARNING: braces {} are not necessary for single statement blocks
#40: FILE: sound/pci/es1968.c:1831:
+ if (diff > 1) {
+ __maestro_write(chip, IDR0_DATA_PORT, cp1);
+ }
total: 0 errors, 1 warnings, 35 lines checked
./patches/es1968-fix-jitter-on-some-maestro-cards.patch has style problems, please review. If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.
Please run checkpatch prior to sending patches
Cc: Andreas Mueller <andreas@stapelspeicher.org> Tested-by: Rene Herman <rene.herman@keyaccess.nl> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Wed, 9 Apr 2008 07:16:14 +0000 (09:16 +0200)]
[ALSA] virtuoso: correctly switch input jack on Xonar DX
When selecting the capture source on the Xonar DX, the input jack must
be routed to either the line input or the microphone input by setting a
GPIO pin. This requires an additional callback so that the model driver
can hook into the toggling of AC97 switches.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
[ALSA] hda-codec - Support mic automute for Clevo M720R/SR
Add support for mic automute in clevo-m720r ALC883 model, and rename it
to more generic clevo-m720. Also change model entry in ALSA-Configuration.txt
accordingly.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
[ALSA] hda-codec - Use common 3stack-6ch mixer for 3stack-hp model
Forgot one more: 3stack-hp model also have now the same mixer as
3stack-6ch (after DAC assignment fix in ALC883), so use it avoiding
duplicating the same mixer definition.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
[ALSA] hda-codec - Use base ALC883 mixer for 6stack-dell model
After DAC assignment fix in ALC883, alc888_6st_dell_mixer is now the
same as alc883_base_mixer. Avoid duplicated code and use
alc883_base_mixer in 6stack-dell model, removing alc888_6st_dell_mixer
definition.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
[ALSA] hda-codec - Remove now uneeded 6stack-hp model from ALC883
After DAC assignment fix in ALC883, the 6stack-hp model is now the same
as 6stack-dig. So just remove 6stack-hp model and replace its use with
6stack-dig.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Andrew Morton [Thu, 20 Mar 2008 11:07:31 +0000 (12:07 +0100)]
[ALSA] sound/pci/pcxhr/pcxhr.c: fix warnings
sparc64:
sound/pci/pcxhr/pcxhr.c: In function `pcxhr_update_r_buffer':
sound/pci/pcxhr/pcxhr.c:459: warning: cast to pointer from integer of different size
sound/pci/pcxhr/pcxhr.c: In function `pcxhr_trigger_tasklet':
sound/pci/pcxhr/pcxhr.c:628: warning: long int format, different type arg (arg 4)
Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/pcxhr/pcxhr_core.c: In function `pcxhr_set_pipe_state':
sound/pci/pcxhr/pcxhr_core.c:899: warning: long int format, different type arg (arg 4)
suseconds_t is int on sparc64.
Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/pci/aw2/aw2-alsa.c: In function 'snd_aw2_create':
sound/pci/aw2/aw2-alsa.c:282: error: 'DMA_32BIT_MASK' undeclared (first use in this function)
sound/pci/aw2/aw2-alsa.c:282: error: (Each undeclared identifier is reported only once
sound/pci/aw2/aw2-alsa.c:282: error: for each function it appears in.)
Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
When specifying which PCM devices to use, model drivers now use flags
that also specify the routing between PCM devices and DMA channels
instead of just DMA channel bits. This simplifies some code that checks
for these flags.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Wed, 19 Mar 2008 07:17:33 +0000 (08:17 +0100)]
[ALSA] oxygen: add monitor controls
Add controls to enable monitoring of the analog and digital inputs.
To allow monitoring after loading the driver when nothing has been
played back or recorded yet, the I2S input and outputs are initialized
to a valid configuration.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Robert Jarzmik [Tue, 18 Mar 2008 11:08:35 +0000 (12:08 +0100)]
[ALSA] soc - Add missing audio path between Mono Mixer and Mic PGAs
Signed-off-by: Robert Jarzmik <rjarzmik@free.fr> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
[ALSA] hda-codec - Fix DAC assignment order in ALC883
Actually clfe and surround DACs are inverted in alc883_dac_nids array
(see ALC883 datasheet). I discovered this while testing multi-channel
setup (using 3stack-6ch-dig model) on MSI 945GCM5 V2 motherboard that
has an ALC883 codec. Simply Rear Left/Right and Center/LFE were swapped
in 6 channel mode (also in 4 channel mode you didn't get rear left/right
output). Other models also were affected by this bug, as can be seen by
the mixer layouts that "workaround" this (the real bug was not noticed,
and some other models simply played with mixer and initial verbs). Thus
along with fixing the order of dac nids, also change the models that
relied on previous dac ordering properly.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/drivers/pcsp/pcsp.c: In function 'snd_pcsp_create':
sound/drivers/pcsp/pcsp.c:54: error: 'loops_per_jiffy' undeclared (first use in\ this function)
sound/drivers/pcsp/pcsp.c:54: error: (Each undeclared identifier is reported on\ ly once
sound/drivers/pcsp/pcsp.c:54: error: for each function it appears in.)
Takashi Iwai [Fri, 14 Mar 2008 16:17:09 +0000 (17:17 +0100)]
[ALSA] ice1724 - Fix the SPDIF input sample-rate on Juli@
AK4114 on Juli@ has the SPDIF input sample rate detection and
causes errors when an incompatible sample rate is chosen.
The patch adds the open hook to check the current rate and limit
the hw constraints.
Takashi Iwai [Fri, 14 Mar 2008 14:52:20 +0000 (15:52 +0100)]
[ALSA] hda-intel - Fix power-off hang on ASUS P5AD2
The hda-intel driver has a problem at power-off on ASUS P5AD2.
It's caused when the position-buffer is enabled -- most likely a
hardware-specific problem.
This patch adds a quirk to avoid the unnecessary enablement of
position-buffer.
Currently ALC662 doesn't suport amp mute for AmpOut in nids 0x02, 0x03,
0x04 (see block diagram in ALC662 datasheet page 3, does M correspond to
mute?). The result is that currently mute for "Front Playback Switch",
"Surround Playback Switch", "Center Playback Switch" and "LFE Playback
Switch" mixer items doesn't work (tested on Asus P5GC-MX motherboard
with 3stack-6ch model).
The solution I found for this is to mute the proper inputs in 0x0c,
0x0d, 0x0e audio mixers.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 14 Mar 2008 08:18:32 +0000 (09:18 +0100)]
[ALSA] hda-codec - Fix orphan Headphone controls in STAC codecs
Currently, the headphone controls are created as Master wrongly in
some cases, and this prevents the virtual master controls.
The patch fixes the problem by simply using "Headphone" always for
headphone controls.
Matthew Ranostay [Fri, 14 Mar 2008 07:46:51 +0000 (08:46 +0100)]
[ALSA] hda: 92HD73xxx distortion fix
Fixed issue on some laptops that if the Master mixer and DAC mixers are
turned all the way up that will cause distortion. This is fixed by limiting
the max volume with the volume knob nid.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_pcm_lib_preallocate_free_for_all() is called from snd_pcm_free() just
after calling the private_free routine. So there should be no need to call
it in driver's private_free routine.
Atsushi Nemoto [Sat, 8 Mar 2008 10:07:26 +0000 (11:07 +0100)]
[ALSA] at73c213: fix error checking for clk API
The clk_round_rate() and clk_set_rate() will return int, so not store thier
return value to unsigned long variable. This bug hides real error on these
API.
projects / linux-2.6-omap-h63xx.git / log