Set the invalid dma channel to -1 (and check properly for it) in
pxa2xx_pcm_hw_free(). Was assuming 0 is an invalid channel number but 0
is a valid pxa dma channel num.
Signed-off-by: stephen <stephen.ware@eqware.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the more specific preset for ALC1200 above the general one for
ALC888, so that it will have the chance to get matched and selected.
Reported-by: Thomas Schneider <nailstudio@gmx.net>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This set of patches introduces calls to the following set of functions:
usb_endpoint_dir_in(epd)
usb_endpoint_dir_out(epd)
usb_endpoint_is_bulk_in(epd)
usb_endpoint_is_bulk_out(epd)
usb_endpoint_is_int_in(epd)
usb_endpoint_is_int_out(epd)
usb_endpoint_num(epd)
usb_endpoint_type(epd)
usb_endpoint_xfer_bulk(epd)
usb_endpoint_xfer_control(epd)
usb_endpoint_xfer_int(epd)
usb_endpoint_xfer_isoc(epd)
In some cases, introducing one of these functions is not possible, and it
just replaces an explicit integer value by one of the following constants:
USB_ENDPOINT_XFER_BULK
USB_ENDPOINT_XFER_CONTROL
USB_ENDPOINT_XFER_INT
USB_ENDPOINT_XFER_ISOC
An extract of the semantic patch that makes these changes is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r1@ struct usb_endpoint_descriptor *epd; @@
- ((epd->bmAttributes & \(USB_ENDPOINT_XFERTYPE_MASK\|3\)) ==
- \(USB_ENDPOINT_XFER_CONTROL\|0\))
+ usb_endpoint_xfer_control(epd)
@r5@ struct usb_endpoint_descriptor *epd; @@
- ((epd->bEndpointAddress & \(USB_ENDPOINT_DIR_MASK\|0x80\)) ==
- \(USB_DIR_IN\|0x80\))
+ usb_endpoint_dir_in(epd)
@inc@
@@
#include <linux/usb.h>
@depends on !inc && (r1||r5)@
@@
+ #include <linux/usb.h>
#include <linux/usb/...>
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds DAPM implementaion for the capture path
on twlx030.
TWL has two physical ADC and two digital microphone (stereo) connections.
The CPU interface has four microphone channels.
For simplicity the microphone channel paths are named as:
TX1 (Left/Right) - when using i2s mode, only the TX1 data is valid
TX2 (Left/Right)
Input routing (simplified version):
There is two levels of mux settings for TWL in input path:
Analog input mux:
ADCL <- {Off, Main mic, Headset mic, AUXL, Carkit mic}
ADCR <- {Off, Sub mic, AUXR}
Analog/Digital mux:
TX1 Analog mode:
TX1L <- ADCL
TX1R <- ADCR
TX1 Digital mode:
TX1L <- Digimic0 (Left)
TX1R <- Digimic0 (Right)
TX2 Analog mode:
TX2L <- ADCL
TX2R <- ADCR
TX2 Digital mode:
TX2L <- Digimic1 (Left)
TX2R <- Digimic1 (Right)
The patch provides the following user controls for the capture path:
Mux settings:
"TX1 Capture Route": {Analog, Digimic0}
"TX2 Capture Route": {Analog, Digimic1}
"Analog Left Capture Route": {Off, Main Mic, Headset Mic, AUXL, Carkit Mic}
"Analog Right Capture Route": {Off, Sub Mic, AUXR}
Volume/Gain controls:
"TX1 Digital Capture Volume": Stereo gain control for TX1 path
"TX2 Digital Capture Volume": Stereo gain control for TX2 path
"Analog Capture Volume": Stereo gain control for the analog path only
Important things for the board files:
Microphone bias:
"Mic Bias 1": Bias for Main mic or for digimic0 (analog or digital path)
"Mic Bias 2": Bias for Sub mic or for digimic1 (analog or digital path)
"Headset Mic Bias": Bias for Headset mic
When the routing configured correctly only the needed components will be
powered/enabled.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Modify the enum filter to more generic that it will filter
out the enums with text "Invalid".
The enum filter also required for the capture path.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (583 commits)
V4L/DVB (10130): use USB API functions rather than constants
V4L/DVB (10129): dvb: remove deprecated use of RW_LOCK_UNLOCKED in frontends
V4L/DVB (10128): modify V4L documentation to be a valid XHTML
V4L/DVB (10127): stv06xx: Avoid having y unitialized
V4L/DVB (10125): em28xx: Don't do AC97 vendor detection for i2s audio devices
V4L/DVB (10124): em28xx: expand output formats available
V4L/DVB (10123): em28xx: fix reversed definitions of I2S audio modes
V4L/DVB (10122): em28xx: don't load em28xx-alsa for em2870 based devices
V4L/DVB (10121): em28xx: remove worthless Pinnacle PCTV HD Mini 80e device profile
V4L/DVB (10120): em28xx: remove redundant Pinnacle Dazzle DVC 100 profile
V4L/DVB (10119): em28xx: fix corrupted XCLK value
V4L/DVB (10118): zoran: fix warning for a variable not used
V4L/DVB (10116): af9013: Fix gcc false warnings
V4L/DVB (10111a): usbvideo.h: remove an useless blank line
V4L/DVB (10111): quickcam_messenger.c: fix a warning
V4L/DVB (10110): v4l2-ioctl: Fix warnings when using .unlocked_ioctl = __video_ioctl2
V4L/DVB (10109): anysee: Fix usage of an unitialized function
V4L/DVB (10104): uvcvideo: Add support for video output devices
V4L/DVB (10102): uvcvideo: Ignore interrupt endpoint for built-in iSight webcams.
V4L/DVB (10101): uvcvideo: Fix bulk URB processing when the header is erroneous
...
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (407 commits)
[ARM] pxafb: add support for overlay1 and overlay2 as framebuffer devices
[ARM] pxafb: cleanup of the timing checking code
[ARM] pxafb: cleanup of the color format manipulation code
[ARM] pxafb: add palette format support for LCCR4_PAL_FOR_3
[ARM] pxafb: add support for FBIOPAN_DISPLAY by dma braching
[ARM] pxafb: allow pxafb_set_par() to start from arbitrary yoffset
[ARM] pxafb: allow video memory size to be configurable
[ARM] pxa: add document on the MFP design and how to use it
[ARM] sa1100_wdt: don't assume CLOCK_TICK_RATE to be a constant
[ARM] rtc-sa1100: don't assume CLOCK_TICK_RATE to be a constant
[ARM] pxa/tavorevb: update board support (smartpanel LCD + keypad)
[ARM] pxa: Update eseries defconfig
[ARM] 5352/1: add w90p910-plat config file
[ARM] s3c: S3C options should depend on PLAT_S3C
[ARM] mv78xx0: implement GPIO and GPIO interrupt support
[ARM] Kirkwood: implement GPIO and GPIO interrupt support
[ARM] Orion: share GPIO IRQ handling code
[ARM] Orion: share GPIO handling code
[ARM] s3c: define __io using the typesafe version
[ARM] S3C64XX: Ensure CPU_V6 is selected
...
* 'timers-core-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
hrtimers: fix warning in kernel/hrtimer.c
x86: make sure we really have an hpet mapping before using it
x86: enable HPET on Fujitsu u9200
linux/timex.h: cleanup for userspace
posix-timers: simplify de_thread()->exit_itimers() path
posix-timers: check ->it_signal instead of ->it_pid to validate the timer
posix-timers: use "struct pid*" instead of "struct task_struct*"
nohz: suppress needless timer reprogramming
clocksource, acpi_pm.c: put acpi_pm_read_slow() under CONFIG_PCI
nohz: no softirq pending warnings for offline cpus
hrtimer: removing all ur callback modes, fix
hrtimer: removing all ur callback modes, fix hotplug
hrtimer: removing all ur callback modes
x86: correct link to HPET timer specification
rtc-cmos: export second NVRAM bank
Fixed up conflicts in sound/drivers/pcsp/pcsp.c and sound/core/hrtimer.c
manually.
The card based on stv0299 or stv0288 demodulators.
Signed-off-by: Igor M. Liplianin <liplianin@me.by>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
Thanks to Troy Kisky <troy.kisky@boundarydevices.com> for noticing.
- DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2
- DSP_B has 0-bit data delay which corresponds to submode 1
- Currently driver sets them opposite so swap the submode setting
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- OMAP McBSP DAI driver claims to support DSP_A format which has 1-bit data
delay but configures link for 0-bit data delay which is in fact DSP_B
- Fix this by changing format from DSP_A to DSP_B
- Fix also TLV320AIC23 codec and OSK5912 machine drivers since the same
error is populated also there
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add check to determine if dinput_mux is set by any of patch_stac*() functions,
otherwise a invalid pointer my be referenced causing gibberish to mixer values.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing __devexit annotation to wm8350_codec_remove():
sound/soc/codecs/wm8350.c:1546: warning: 'wm8350_codec_remove' defined but not used
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sense DaVinci does not support true I2S mode and
we don't have to use the hack, use dsp_b mode instead
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the meaning of SND_SOC_DAIFMT_NB_NF to match that
used in the codec.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DaVinci does not support true I2S or right justified
mode so not all I2S codecs will work with it when the codec is
master. Document this limitation.
Add dsp_a, dsp_b mode options
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Minor, just move a block of code to make next patch clearer.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Just at little cleanup of davinci_i2s_set_dai_fmt
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Document the current polarity choices.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add constants with a value of 0 to show more explicitly
what is being requested.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The capture with 44.1kHz on ca0106 seems to cause loud noises on
later playbacks, which doesn't support 44.1kHz. A simple fix is to
disable 44.1kHz, as the "default" PCM with dsnoop is anyway only with
48kHz.
Reference: Novell bnc#447624
https://bugzilla.novell.com/show_bug.cgi?id=447624
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When no jack detection is available, the pins should be always
turned on since it can't be turned on/off dynamically via unsol
events.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There will be a Oops or frequent underrun messages when playing music with
omap soc driver, this is because a data region is incorretly sized, other data
region will be overwriten when writing to this data region.
Signed-off-by: Stanley Miao <stanley.miao@windriver.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added probe_only module option to hd-audio driver.
This option specifies whether the driver creates and initializes the
codec-parser after probing. When this option is set, the driver skips
the codec parsing and initialization but gives you proc and other
accesses. It's useful to see the initial codec state for debugging.
The default of this value is off, so the default behavior is as same
as before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the line_out has only one DAC and it's unique (i.e. not shared
by other outputs), assign a more reasonable and distinct mixer name
such as "Headphone" or "Speaker".
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current auto-configuration code has several problems especially
for the new IDT codecs, e.g. wrong assignment of pins and DACs or
coupled volume for speaker and headphone.
This patch is a fairly large rewrite of the auto-configuration code.
Some remaks
- mic_switch and line_switch contain NIDs instead of bool
- dac_list isn't fixed for IDT 92HD* codecs now, they are all probed
- extra HP and speakers are stored in extra_dacs[].
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit re-enabled hp_nid setup for IDT92HD73*, but
it's unneeded indeed for Dell laptops that have multiple headphones.
Setting the extra hp_nid results in a non-working "Headpohne" mixer
control. Thus hp_nid should be 0 for these dell models.
Also, the automatic addition of hp_nid should check whether it's
a dual-HP model or not. For dual-HPs, the pins are already checked
by the early workaround.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added "IEC958 PCM Stream" controls for the per-stream IEC958 status
bits. Using this instead of "IEC958 Default" is safer since the status
bits will be recovered to the default states after closing the PCM
stream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the call of snd_ctl_add() by replacing with snd_hda_ctl_add()
so that this mixer element can be tracked for re-configuration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The re-initializations of codec amp and verb caches are missing
at reconfig, which may cause Oops occasionally.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the model without the jack-detection for some desktops that
have really no jack-detection. The recent driver caused regressions
regarding the sound output on such machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 07f455f779.
ALSA: hda: removed unneeded hp_nid references
Removed unneeded hp_nid references for 92hd73xx codec family.
This caused the silent output on some Intel desktops due to missing
routing of widget 0x0a and 0x0d.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8350 is an integrated audio and power management subsystem which
provides a single-chip solution for portable audio and multimedia systems.
The integrated audio CODEC provides all the necessary functions for
high-quality stereo recording and playback. Programmable on-chip
amplifiers allow for the direct connection of headphones and microphones
with a minimum of external components. A programmable low-noise bias
voltage is available to feed one or more electret microphones.
Additional audio features include programmable high-pass filter in the
ADC input path.
This driver was originally written by Liam Girdwood with further updates
from me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This should never happen and it's helpful to identify the specific control
that failed when it does happen.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than listing lots of architectures per line in Kconfig and
Makefile, causing merge conflicts all the time, have one per line
in alphabetical order.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace all tasklet_hi_schedule() callers with the normal
tasklet_schedule(). The former often causes troubles with
RT-kernels, and has actually no merit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove codec vendor names from the codec name strings.
The vendor name is already given from the vendor name table, so
displayed doubly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some desktops seems to have no HP/mic jack detection on the front panel,
which results in the silent output in the recent driver, because the
driver mutes the output (to save power) when no plug is detected.
This patch adds a new model that disables the jack-detection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch removes an inconsistency that became apparent when I
documented the fields of snd_ca0106_details. spi_dac is always
used in a 'boolean' sense, so this cleanup should make no difference.
[Actually, there is one place checking explicitly spi_dac == 1, so
this will change the behavior. But, supposing it's rather a typo,
I apply this clean-up patch -- tiwai]
Signed-off-by: Ben Stanley <Ben.Stanley@exemail.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi wrote an email [1] explaining the fields of snd_ca0106_details,
so I captured the information into the ca0106.h header file.
[1] http://article.gmane.org/gmane.linux.alsa.devel/56783/match=takashi+gpio_type
Signed-off-by: Ben Stanley <Ben.Stanley@exemail.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Include sound/core.h in sound_core.c so that sound_class is declared
before it is defined, avoiding it looking like it should be static.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Netwinder was using gpio_xxx names which could clash with the GPIO
layer. Add a 'nw_' prefix to ensure that these remain separate.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
A special start-up sequence is required to reduce the pop-noise of Class D
amplifier when enable hands-free on TWL4030.
Signed-off-by: Stanley.Miao <stanley.miao@windriver.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixed a compile warning below:
sound/isa/sb/sb8.c: In function ‘snd_sb8_probe’:
sound/isa/sb/sb8.c:104: warning: ‘err’ may be used uninitialized in this function
by setting the return value correctly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the compile warning regarding the unused function when built
with CONFIG_PM=n:
sound/pci/hda/hda_intel.c:1905: warning: ‘snd_hda_codecs_inuse’ defined but not used
snd_hda_codecs_inuse() is used only in the resume callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the registration of dais in s3c2443-ac97.c.
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_init':
sound/soc/s3c24xx/s3c2443-ac97.c:401: warning: passing argument 1 of 'snd_soc_register_dai' from incompatible pointer type
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_exit':
sound/soc/s3c24xx/s3c2443-ac97.c:407: warning: passing argument 1 of 'snd_soc_unregister_dai' from incompatible pointer type
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver now registers the codec and DAI when probed as an I2C device.
Also convert the driver to use a single dynamic allocation to simplify
error handling.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Redo the instantiation of the WM8900 to do most of the initialisation
work when the I2C driver probes rather than when the ASoC device is
instantiated, registering the codec with the ASoC core when done.
Also move all dynamic allocations into a single kmalloc() to simplify
error handling and rename the I2C driver to make output more sensible.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The GPIO stuff for OLPC in cs5535audio_olpc.c is implemented only for
Geode-LX, and enabled only when CONFIG_MGEODE_LX=y. Without this
config option, the driver gets build errors.
This patch adds a workaround to make it dependent on CONFIG_MGEODE_LX.
Ideally, the OLPC-GPIO stuff should be implemented in a way
independent from CPU type selection...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add copyright info to _olpc.c
- minor layout fixes
- make Makefile more concise
- silence a warning
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Always turn off mic bias; the MIC LED should never come on when the
driver is first loaded.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This drops the AD1888 V_REFOUT control, and replaces it with a MIC Bias
Enable control. It also moves the MIC bias enabling into a separate
function.
Signed-off-by: Andres Salomon <dilinger@debian.org>
The OLPC has a privacy light hooked up in series with the microphone's
V_Ref bias. We want to activate the bias while we are capturing audio.
Signed-off-by: Chris Ball <cjb@laptop.org>
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Checking the HPF register is irrelevant; HPF is secondary to the AI mode.
Instead, check for Analog Input mode via GPIO.
Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown