Rob Sims wrote:
"I can't seem to turn on register 0x17, bit 3 in the sound chip, except
by codec_reg_write; the mixer lacks direct or indirect control. It
seems there are two names for the output of the rec mixer:
Capture ST Mixer
Playback Mixer
Would the following do the trick?"
I confirm that this solves the audio problems I was having.
Signed-off-by: Jonas Bonn <jonas.bonn@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: ASoC: Fix double free and memory leak in many codec drivers
ALSA: CA0106 on MSI K8N Diamond PLUS Motherboard
Many SoC audio codec drivers have improper freeing of memory in error
paths.
* codec is allocated in the platform device probe function, but is not
freed there in case of error. Instead it is freed in the i2c device
probe function's error path. However the success or failure of both
functions is not linked, so this could result in a double free (if
the platform device is successfully probed, the i2c device probing
fails and then the platform driver is unregistered.)
* codec->private_data is allocated in many platform device probe
functions but not freed in their error paths.
This patch hopefully solves all these problems.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch lets the files using linux/version.h match the files that
#include it.
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This patch merges struct snd_soc_codec_dai and struct
snd_soc_cpu_dai into struct snd_soc_dai for the codec drivers.
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Several ASoC codec drivers use custom macros equivalent to the standard
pr_ macros, most of which are not actually used. Replace these custom
macros with the standard ones.
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>
Signed-off-by: Mike Montour <mail@mmontour.net>
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>
The register cache size is used by the codec_reg sysfs file which works in
terms of the register cache access functions rather than in terms of raw
access to the cache so the size specified needs to be in terms of the
number of elements.
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>
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>
Currently the ASoC core configures the bias levels in the system using
a callback on codecs and machines called 'dapm_event', passing it PCI
style power levels as SNDRV_CTL_POWER_ constants. This is more obscure
than it needs to be and has caused confusion to driver authors,
especially given that DAPM is also performing power management.
Address this by renaming the callback function to 'set_bias_level' and
using constants explicitly representing the off, standby, pre-on and on
states which DAPM transitions through.
Also unexport the API for setting bias level: there are currently no
in-tree users of this API other than the core itself and it is likely
that the core would need to be extended to cater for any users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
GTA02 device has a speaker between LOUT2 & ROUT2 and in this mode ROUT2
needs to be inverted. This patch adds a mixer control for this.
Signed-off-by: Graeme Gregory <graeme@openmoko.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.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>
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>
This patch series adds support for the WM8753 codec as found on the
OpenMoko Neo 1973 (other Neo 1973 and Samsung S3C24xx patches to follow
today) as well other new devices.
Features:-
o HiFi and Voice DAI supported (inc runtime switching of DAI mode)
o DAPM
o All mixers
o PLL calculator
o 16,20 and 24bit samples.
o WM8753 I2C ID added to include/linux/i2c-id.h
From: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Harald Welte <laforge@openmoko.org>
Signed-off-by: Graeme Gregory <gg@opensource.wolfsonmicro.com>
Signed-off-by: Seth Forshee <seth.forshee@gmail.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
