Most of this function is protected by the sound_loader_lock.
We can push down the BKL to this call out err = file->f_op->open(inode,file);
In order to build the sound core without the BKL, we
will need to push the lock_kernel() call into the ~20
device drivers that register their file operations.
Signed-off-by: John Kacur <jkacur@redhat.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Alan Cox <alan@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the PCI SSID gives an overriding SKU assno, PC-beep bit isn't
detected (since it's located over 16bit), resulting in no PC beep.
Also, many devices seem ignoring the requirement by Realtek's spec
for SSID numbers, and it also confuses the PC beep detection.
This patch assumes the PC beep is available on every machine with
PCI SSID override. It's a regression fix from 2.6.34.
Reference: Kernel bug 16251
http://bugzilla.kernel.org/show_bug.cgi?id=16251
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It will be replaced with automatic deemphasis rate configuration but since
we have an enumeration table in this driver this is done in a separate
commit to make the renumbering of the enumeration items clear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This adds sound support for the SmartQ board.
The hardware consists of a S3C6410 coupled with a WM8987 over I²S. The WM8750
driver is used for driving the WM8987, as they are register compatible.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8987 codec is register compatible with the WM8750, so just add it to the
SPI and I²C device table.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For UDA1341 codec power control is managed in STATUS1 register, and
for all other codecs in DATA011 register.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In UDA1340, UDA1344 and UDA1345 codecs there is one more functional
register in part of DATA0 tranfser. For UDA1341 this register
coincides with EA register.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch moves SPARC architecture specific data members out of
struct of_device and into the pdev_archdata structure. The reason
for this change is to unify the struct of_device definition amongst
all the architectures. It also remvoes the .sysdata, .slot, .portid
and .clock_freq properties because they aren't actually used by
anything.
A subsequent patch will replace struct of_device entirely with struct
platform_device and the of_platform support code will share common
routines with the platform bus (but the bus instances themselves can
remain separate).
This patch also adds 'struct resources *resource' and num_resources
to match the fields defined in struct platform_device. After this
change, 'struct platform_device' can be used as a drop-in replacement
for 'struct of_platform'.
This change is in preparation for merging the of_platform_bus_type
with the platform_bus_type.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: David S. Miller <davem@davemloft.net>
Cc: Stephen Rothwell <sfr@canb.auug.org.au>
A few boards using this controller are reported to need a little extra
time during their reset cycle.
Reported-by: Michael Goeke <michael.goeke@icachip.de>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When using a timing voice to clock out periods during capture, the
driver would slowly loose synchronization and never catch up, eventually
reaching a point where it no longer generated interrupts. To avoid
this situation, the virtual period clocking was changed to shorten the
next timing period when our timing voice falls too far behind the
capture voice. In addition, the first virtual period for the timing
voice was slightly too short, causing the timing voice to initially be
ahead of the capture voice.
While tracking down this problem, I noticed that the expected sample
offset was being incorrectly initialized, causing an overrun to be
incorrectly reported when the timing voice happened to be perfectly
synchronized.
Reported-by: Hans Schou <linux@schou.dk>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
When using poll() to wait for the next period -- or avail_min samples --
one gets a consistent delay for each system call that is usually just a
little short of the selected period time. However, When using
snd_pcm_read/write(), one gets a jittery delay that alternates between
less than a millisecond and approximately two period times. This is
caused by snd_pcm_lib_{read,write}1() transferring any available samples
to the user's buffer and adjusting the application pointer prior to
sleeping to the end of the current period. When the next period
interrupt occurs, there is then less than avail_min samples remaining to
be transferred in the period, so we end up sleeping until a second
period occurs.
This is solved by using runtime->twake as the number of samples needed
for a wakeup in addition to selecting the proper wait queue to wake in
snd_pcm_update_state(). This requires twake to be non-zero when used
by snd_pcm_lib_{read,write}1() even if avail_min is zero.
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: usb/endpoint, fix dangling pointer use
ALSA: asihpi - Get rid of incorrect "long" types and casts.
ASoC: DaVinci: Fix McASP hardware FIFO configuration
ALSA: hda - Fix line-in for mb5 model MacBook (Pro) 5,1 / 5,2
ALSA: usb-audio: fix UAC2 control value queries
ALSA: usb-audio: parse UAC2 sample rate ranges correctly
ALSA: usb-audio: fix control messages for USB_RECIP_INTERFACE
ALSA: usb-audio: add check for faulty clock in parse_audio_format_rates_v2()
ALSA: hda - Don't check capture source mixer if no ADC is available
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On initialization ADC/DAC are enabled only for UDA1341, that's why
bias_level shall be set to off explicitly, otherwise dapm is
misinformed about bias_level on startup.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This change wipes out a hardcoded macro, which enables codec bias
level control. Now is_powered_on_standby value shall be used instead.
Signed-off-by: Vladimir Zapolskiy <vzapolskiy@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As the control interface is now carried in struct snd_usb_audio, we can
simplify the API a little and also drop the private ctrlif field from
struct usb_mixer_interface.
Also remove a left-over function prototype in pcm.h.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Get rid of the last occurances of _v1 suffixes, and move the version
number right after the "uac" string. Now things are consitent again.
Sorry for the forth and back, but it just looks much nicer this way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some programs like Skype trying to set capture volume automatically.
Normally it will tray, carefully step by step lover or higher, set the volume.
In real word it work not really well, because devises and vendors lie about
real audio settings.
For example most Logitech webcams have 6400 or 3500 steps for capture volume.
They do not tell that actual resolution is 384. So we have only 7 or 18 real
steps. In this patch I set real resolution only for tested devices.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds GPIO jack detection to Nokia N900/RX-51. At the moment only
SND_JACK_VIDEOOUT type is reported. More types could be reported after
getting more audio features supported and necessary drivers integrated for
implementing automated accessory detection.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Nokia RX-51/N900 has multifunction 4-pole audio-video jack that can be used
as headphone, headset or audio-video connector. This patch implements the
control 'Jack Function' which is used to select the desired function.
At the moment only TV-out without audio is supported.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
in tlv320aic23_set_bias_level, for the case SND_SOC_BIAS_ON, the
comment says "vref/mid, osc on, dac unmute" but the code doesn't
clear the corresponding bits, thus when resuming, several bits are
not cleared preventing the codec from working.
in tlv320aic23_suspend, clearing the active register is not needed
as it will be done by tlv320aic23_set_bias_level, when setting
bias to SND_SOC_BIAS_OFF
Signed-off-by: Eric Bénard <eric@eukrea.com>
Cc: broonie@opensource.wolfsonmicro.com
Cc: anuj.aggarwal@ti.com
Cc: lrg@slimlogic.co.uk
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch adds ASoC support for the qi_lb60 board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for the JZ4740 internal codec.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds ASoC support for JZ4740 SoCs I2S module.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://bugs.launchpad.net/bugs/463178
Set Macbook 5,2 (106b:4a00) hardware to use ALC885_MB5
Cc: <stable@kernel.org>
Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following compile warning. kctl should be NULL-initialized.
sound/pci/hda/patch_realtek.c: In function ‘alc_build_controls’:
sound/pci/hda/patch_realtek.c:2550:23: warning: ‘kctl’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After mass production started it was found that several boards exhibit
noise problems during sound playback. After some investigation it was
determined that CLKX polarity is set incorrectly, and even if most boards
can tolerate the wrong setting, there are some that don't.
Fix polarity setup in the board file. As the clock settings for input and
output now match, merge in and out functions and structures to simplify
code.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Stanse found that in snd_usb_parse_audio_endpoints, there is a
dangling pointer dereference. When snd_usb_parse_audio_format fails,
fp is freed, and continue invoked. On the next loop, there is
"fp && fp->altsetting == 1 && fp->channels == 1" test, but fp is set
from the last iteration (but is bogus) and thus ilegally dereferenced.
Set fp to NULL before "continue".
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When SX_TLV widgets are read, if the gain is set to a value below 0dB,
the mixer control is erroniously read as being at maximum volume.
The value read out of the CODEC register is never sign-extended, and
when the minimum value is subtracted (read; added, since the minimum is
negative) the result is a number greater than the maximum allowed value
for the control, and hence it saturates.
Solution: Mask the result so that it "wraps around", emulating
sign-extension.
Signed-off-by: Stuart Longland <redhatter@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* tdm slot has to be configured to get sound working on i.MX25
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b,
which isn't muted when headphone is plugged in. This adds additional
support to the extra subwoofer via new ideapad model.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These give incorrect results for index wrap on 64 bit.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn
enables the compilation of ak4642.c - however this codec uses I2C to
communicate with the HW.
Same applies to DA7210.
Consequently when I2C is not set, the compilation fails [1]
This patch fixes this issues, by adding a depencdency on the related HW-
controller.
Signed-off-by: Peter Huewe <peterhuewe@gmx.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Compiling in the MPC5200 sound drivers results in the following build error:
sound/soc/fsl/mpc5200_psc_ac97.o: In function `to_psc_dma_stream':
mpc5200_psc_ac97.c:(.text+0x0): multiple definition of `to_psc_dma_stream'
sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here
sound/soc/fsl/efika-audio-fabric.o: In function `to_psc_dma_stream':
efika-audio-fabric.c:(.text+0x0): multiple definition of `to_psc_dma_stream'
sound/soc/fsl/mpc5200_dma.o:mpc5200_dma.c:(.text+0x0): first defined here
make[3]: *** [sound/soc/fsl/built-in.o] Error 1
make[2]: *** [sound/soc/fsl] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
This patch fixes it by declaring the inline function in the header file to
also be a static.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Cc: Jon Smirl <jonsmirl@gmail.com>
Tested-by: John Hilmar Linkhorst <John.Linkhorst@rwth-aachen.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The most useful configuration for the WM2000 is to enable the ANC so turn
that on by default, and since we're not reflecting chip default state also
enable the speaker output by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral
has FIFO support. This FIFO provides additional data buffering. It
also provides tolerance to variation in host/DMA controller response
times. More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=sprufm1&fileType=pdf
Existing sequence of steps for audio playback/capture are:
a. DMA configuration
b. McASP configuration (configures and enables FIFO)
c. Start DMA
d. Start McASP (enables FIFO)
During McASP configuration, while FIFO was being configured, FIFO
was being enabled in davinci_hw_common_param() function of
sound/soc/davinci/davinci-mcasp.c file. This generated a transmit
DMA event, which gets serviced when DMA is started.
https://patchwork.kernel.org/patch/84611/ patch clears the DMA
events before starting DMA, which is the right thing to do. But
this resulted in a state where DMA was waiting for an event from
McASP (after step c above), but the event which was already there,
has got cleared (because of step b above).
The fix is not to enable the FIFO during McASP configuration as
FIFO was being enabled as part of McASP start.
Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The line-in input is 0x7 not 0x2 for MacBook (Pro) 5,1 / 5,2 models
Signed-off-by: Alex Murray <murray.alex@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The header contains an extern that isn't used by anything. Remove.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For RANGE requests, we should only query as much bytes as we're in fact
interested in.
For CUR requests, we shouldn't confuse the firmware with an overlong
request but just ask for 2 bytes.
This might need fixing in the future as it's not entirely clear when to
dispatch 1-byte, 2-byte and 4-byte request blocks. For now, we assume
everything is coded in 16bit - this works for all firmware
implementations I've seen.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A device may report its supported sample rates in ranges rather than in
discrete triplets. The code used to only parse the MIN field instead of
properly paying attention to the MAX and RES values.
Also, handle RES values of 1 correctly and announce a continous sample
rate range in this case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Control messages directed to an interface must have the interface number
set in the lower 8 bits of wIndex. This wasn't done correctly for some
clock and mixer messages.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With multiple codec configurations, some codec might have no ADC, thus
it keeps spec->adc_nids = NULL. This causes an Oops in alc_build_controls().
Reference: kernel bug #16156https://bugzilla.kernel.org/show_bug.cgi?id=16156
Cc: <stable@kernel.org>
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: sound/spi: patch for the unuseful variable removal
ALSA: hda - Add SSID table for iMac7,1.
ALSA: hda - Add SSID table for MacBookAir1,1
ALSA: hda - Add SSID table for MacBookAir2,1
ALSA: atmel: set "channel A event" output to debug
* master.kernel.org:/home/rmk/linux-2.6-arm:
ARM: 6164/1: Add kto and kfrom to input operands list.
ARM: 6166/1: Proper prefetch abort handling on pre-ARMv6
ARM: 6165/1: trap overflows on highmem pages from kmap_atomic when debugging
ARM: 6152/1: ux500 make it possible to disable localtimers
[ARM] pxa/spitz: Correctly register WM8750
[ARM] pxa/palmtc: storage class should be before const qualifier
ARM: 6146/1: sa1111: Prevent deadlock in resume path
ARM: 6145/1: ux500 MTU clockrate correction
ARM: 6144/1: TCM memory bug freeing bug
ARM: VFP: Fix vfp_put_double() for d16-d31
This patch is to change 'auido.h' to 'audio.h' for
fixing nuc900 alsa driver build error.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for i2s audio on Bluewater Systems Snapper CL15 module
Signed-off-by: Ryan Mallon <ryan@bluewatersys.com>
Acked-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The '*bitclk' of structure 'snd_at73c213' seems no use,
so I make a patch to remove the unnecessary variable.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add's the iMac7,1 SSID entry to
patch_realtek.c which adds sound support.
bug entry:
https://bugs.launchpad.net/mactel-support/+bug/360866
Note:I do not have this machine on hand only
codec#0 file for the machine so please
test if you have the appropriate equipment.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add's the MacBookAir1,1 SSID entry to
patch_realtek.c which adds sound support.
bug entry:
https://bugs.launchpad.net/mactel-support/+bug/268301
Note:I do not have this machine on hand only
codec#0 file for the machine so please
test if you have the appropriate equipment.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds the SSID number to snd_pci_quirk for the
MacBookAir2,1 taken from codec#0 at:
http://launchpadlibrarian.net/49455483/Card0.Codecs.codec.0.txt
keep in mind I do not have one of these machines on hand
so please if you do have this machine please test for me..
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch to remove the 'break;', when the 'switch' jumps to
the 'default' branch, the 'return -EINVAL' will be return with
a error number, so the 'break;' code never be run, it is unuseful
and should be removed here.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove break after return, it is not needed.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Upper threshold is used in mode7 of DAC33.
Instead of hard wired UTHR, add control to change the upper threshold
value.
Changing upper threshold is not allowed when the playback is already
running, since wrongly timed change in the UTHR can cause problems
with the codec.
With this control the length of the burst in mode7 can be changed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda-intel - fix wallclk variable update and condition
ALSA: asihpi - Fix uninitialized variable
ALSA: hda: Use LPIB for ASUS M2V
usb/gadget: Replace the old USB audio FU definitions in f_audio.c
ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1
ASoC: Add missing Kconfig entry for Phytec boards
ALSA: usb-audio: export UAC2 clock selectors as mixer controls
ALSA: usb-audio: clean up find_audio_control_unit()
ALSA: usb-audio: add UAC2 sepecific Feature Unit controls
ALSA: usb-audio: unify constants from specification
ALSA: usb-audio: parse clock topology of UAC2 devices
ALSA: usb-audio: fix selector unit string index accessor
include/linux/usb/audio-v2.h: add more UAC2 details
ALSA: usb-audio: support partially write-protected UAC2 controls
ALSA: usb-audio: UAC2: clean up parsing of bmaControls
ALSA: hda: Use LPIB for another mainboard
ALSA: hda: Use mb31 quirk for an iMac model
ALSA: hda: Use LPIB for an ASUS device
Add the necessary files to support the TLV320AIC23B wired in I2S
on our i.MX platforms.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* introduce 3 new flags to allow a more detailed configuration
of the SSI link :
IMX_SSI_NET : enable Network Mode
IMX_SSI_SYN : enable Synchronous Mode
IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode
* new platform can use these settings without breaking actual
platforms.
Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The variable 'state' of structure 's3c_ac97_info' seems no use here,
so this patch is to remove the unnecessary variable.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The variable 'periods' of structure 'atmel_runtime_data'
seems no use in whole atmel alsa driver,so I make a patch
to remove the unnecessary variable.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the resource_size function instead of manually calculating the
resource size.This patch can reduce the chance of introducing off-by-one
errors.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Manuel Lauss <manuel.lauss@googlemail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
OMAP McBSP FIFO is word structured:
McBSP2 has 1024 + 256 = 1280 word long buffer,
McBSP1,3,4,5 has 128 word long buffer
This means, that the size of the FIFO
depends on the McBSP word size configuration.
For example on McBSP3:
16bit samples: size is 128 * 2 = 256 bytes
32bit samples: size is 128 * 4 = 512 bytes
It is simpler to place constraint for buffer and period based on channels.
McBSP3 as example again (16 or 32 bit samples):
1 channel (mono): size is 128 frames (128 words)
2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
Use the second method to place hw_rule on buffer size, and in threshold
mode to period size.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Save the word length configuration of McBSP, and use that information
to calculate, and configure the threshold in McBSP.
Previously the calculation was only correct when the stream had 16bit
audio.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The size calculation is end - start + 1. But,sometimes, the '1' can
be forgotten carelessly, witch will have potential risk, so use resource_size
for {request/release}_mem_region and ioremap here should be good habit.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Daniel Glöckner <dg@emlix.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes thinko introduced in "last minutes" before commiting
of the last wallclk patch.
It also fixes the condition checking if the first period after last
wallclk update is processed. There is a little rounding error in
period_wallclk.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch fixes thinko introduced in "last minutes" before commiting
of the last wallclk patch.
It also fixes the condition checking if the first period after last
wallclk update is processed. There is a little rounding error in
period_wallclk.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This patch is to modify the ac97 delays to minimum, all these 1000 micro
seconds delays seem over spec for the AC97 interface.
I deleted some unnecessary delays here and changed the AC97 cold and warm reset
delays from 1000us to 100us.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix a '#include "nuc900-audio.h' typo, I think it should be 'audio'.
At the same time, this patch renames the 'nuc900-auido.h' file to
'nuc900-audio.h'.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current implement meant ACTL_ACCON was only accessed once when read or write
proceeding, which is not right, if so,we have to wait the 'timeout=0x10000' to end
every times.
We need to polling the bit AC_R_FINISH and AC_W_FINISH of ACTL_ACCON
register to identify whether read or write is finished or not,so I make
the patch to fix the issue.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition.
There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX,
the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be
judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather
than 'if (PCM_TX == stype)', which makes the codes easy to read.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Initialize prev_ctl properly before reference:
sound/pci/asihpi/asihpi.c: In function ‘snd_card_asihpi_mixer_new’:
sound/pci/asihpi/asihpi.c:2568:30: warning: ‘prev_ctl.dst_node_index’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Grant patches added an of mach table to struct device_driver. However,
while he changed the macio device code to use that, he left the match
table pointer in struct macio_driver and didn't update drivers to use
the "new" one, thus breaking the probing.
This completes the change by moving all drivers to setup the "new"
one, removing all traces of the old one, and while at it (since it
changes the exact same locations), I also remove two other duplicates
from struct driver which are the name and owner fields.
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
BugLink: https://launchpad.net/bugs/587546
Symptom: On the reporter's ASUS M2V, using PulseAudio in Ubuntu 10.04 LTS
results in the PA daemon crashing shortly after attempting playback of an
audio file.
Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, attempt playback of an audio file while PulseAudio is
active.
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: D Tangman
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAC2 clock selectors are fortunately compatible with UAC1 audio
selector units, so we can simply reuse the same approach to get all the
linked units.
Requests to this control need a different CS value though.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a struct to parse the audio units, and return usable descriptors
for all types. There's no need to limit the result set, except for some
kind of sanity check.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The bits to enable them are always 0 for UAC1 devices, so no additional
checks are required.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move more definitions from private enums to appropriate header files.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audio devices which comply to the UAC2 standard can export complex clock
topologies in its descriptors and set up links between them.
The entities that are defined are
- clock sources, which define the end-leafs.
- clock selectors, which act as switch to select one out of many
possible clocks sources.
- clock multipliers, which have an input clock source, and act as clock
source again. They can be used to derive one clock from another.
All sample rate changes, clock validity queries and the like must go to
clock source elements, while clock selectors and multipliers can be used
as terminal clock source.
The following patch adds a parser for these elements and functions to
iterate over the tree and find the leaf nodes (clock sources).
The samplerate set functions were moved to the new clock.c file.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, UAC2 controls are marked read-only if any of the channels are
marked read-only in the descriptors. Change this behaviour and
- mark them writeable unless all channels are read-only
- store the read-only mask in usb_mixer_elem_info and
- check the mask again in set_cur_mix_value(), and bail out for
write-protected channels.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce two new static inline functions for a more readable parsing
of UAC2 bmaControls.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is adding support for openrd client platforms. It's using
the cs42l51 codec and has one mic and one speaker plugs.
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch enables support for the i2s controller available on kirkwood
platforms
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is adding a ASoC driver for the cs42l51 from Cirrus Logic.
Master mode and spi mode are not supported.
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.ul>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds spdif dummy codec driver for using spdif-dit as
a stand-alone. Until this, spdif-dit can be used only with other
codecs like tlv320aci3x in davinci platform.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Machine driver can instruct the codec driver to reset the
chip registers to their default values at probe time.
If machine driver does not provide setup data, then the
registers are going to be reseted to their defaults, to
be safe.
If the developer on the platform confirms that the register
reset is not needed, than it can be skipped, saving ~20ms
time in probe.
As safety measure do the register reset at remove time also.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Restructure the codec power code in order to be able to hit
off when the codec is not in use.
Since the audio registers are accessible while the codec is powered
down, there is no need for additional safety mechanism.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
It seams at least on twl5031 that the ARXR2_APGA_CTL register
does not have the same default value as it is written in
the TRM.
Since the codec part of the PM chip has not been actually
changed according to TI, assuming, that all version has
the same problem, so writing there the TRM value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the twl4030 codec driver supports different version
of the PM chip, a helper function can come handy, which
can check the driver's default versus the chip values.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the reg cache now contains the chip default values
for all registers (REG_OPTION is reset to it's default
within this patch), there is no longer need to rewrite
_all_ registers.
Initialize only few selected registers.
According to the latest information, the offset cancellation
need to be done only once, when the codec is powered on, so
there is no need for the power up wrapper.
Move all chip initialization code under chip_init, and do
it when the codec is initialized.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add means for machine drivers to select the path for offset
cancellation.
Reset the reg cache value to the chip reset value at the
same time.
Machine drivers can specify which path need to be used for
offset cancellation via the twl4030_setup.offset_cncl_path.
For paths use the defines from
include/linux/mfd/twl4030-codec.h:
TWL4030_OFFSET_CNCL_SEL_*
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
There is no need for the power down wrapper.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Reset most of the codec registers to their chip reset
value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
BugLink: https://launchpad.net/bugs/580749
Symptom: on the original reporter's VIA VT1708-based board, the
PulseAudio daemon dies shortly after the user attempts to play an audio
file.
Test case: boot from Ubuntu 10.04 LTS live cd; attempt to play an audio
file.
Resolution: add SSID for the original reporter's hardware to the
position_fix quirk table, explicitly specifying the LPIB method.
Reported-and-Tested-By: Harald
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/542550
Symptom: On the reporter's iMac, in Ubuntu 10.04 LTS neither playback
nor capture appear audible out-of-the-box.
Test case: Boot from an Ubuntu 10.04 LTS live cd or from an installed
configuration and attempt to play or capture audio.
Resolution: Specify the mb31 quirk for this machine in the codec SSID
table.
Reported-and-Tested-By: f3a97
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/465942
Symptom: On the reporter's ASUS device, using PulseAudio in Ubuntu 10.04
LTS results in the PA daemon crashing shortly after attempting to select
capture or to configure the audio hardware profile.
Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, adjust the HDA device's capture volume with PulseAudio.
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: Irihapeti
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This defines the 24bps and 40bps (8khz sample rate) G.723 codec
formats. They are going to be used once I submit the driver for
an mpeg4/g723 compression card.
I've updated the signed value to -1 as per Takashi's comments
since these are non-linear formats.
Signed-off-by: Ben Collins <bcollins@bluecherry.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch registers the WM8750 codec on a proper place on the SPITZ machine
after the WM8750 driver was converted to new API.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (26 commits)
ALSA: snd-usb-caiaq: Bump version number to 1.3.21
ALSA: Revert "ALSA: snd-usb-caiaq: Set default input mode of A4DJ"
ALSA: snd-usb-caiaq: Simplify single case to an 'if'
ALSA: snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
ALSA: hda: Use LPIB for a Shuttle device
ALSA: hda: Add support for another Lenovo ThinkPad Edge in conexant codec
ALSA: hda: Use LPIB for Sony VPCS11V9E
ALSA: usb-audio: fix feature unit parser for UAC2
ALSA: asihpi - Minor code cleanup
ALSA: asihpi - Add support for new ASI8800 family
ALSA: asihpi - Fix bug preventing outstream_write preload from happening
ALSA: asihpi - Fix imbalanced lock path in hw_message
ALSA: asihpi - Remove support for old ASI8800 family
ALSA: asihpi - Add hd radio blend functions
ALSA: asihpi - Remove unused io map functions
ALSA: usb-audio: add support for UAC2 pitch control
ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
ALSA: usb-audio: fix return values
ALSA: usb-audio: parse more format descriptors with structs
sound: Add missing spin_unlock
...
Do not explicity set the default input mode. Use the hardware default
of mode 0 ('Control vinyl'), which is now available.
This reverts commit e3ca4c9.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After removing code, only one case remains. So use an 'if' instead.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This feature was undocumented on early A4DJ units. It is indicated
by lighting both the 'line' and 'phono' lamps at the same time.
Newer units document this and the newer Windows drivers enable this
for all units, so restore the functionality.
This patch simplifies the code and changes the mode mapping to match
the A8DJ, favouring simpler code and consistency over keeping the
existing mapping.
Both 'Control vinyl' and 'Phono' input modes enable the hardware
preamp. The difference is the input impedance.
This reverts commit 9a9527e.
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/551949
Symptom: On the reporter's Shuttle device, using PulseAudio in Ubuntu
10.04 LTS results in "popping clicking" audio with the PA crashing
shortly thereafter.
Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, adjust the HDA device's volume with PulseAudio.
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: Christian Mehlis <mehlis@inf.fu-berlin.de>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On a Thinkpad Edge 13 "01972NG" I had the problem that speakers played
sound although headphones were plugged in. Using model=ideapad with
latest alsa-git kernel fixed this. So adding this quirk to use ideapad
for another Thinkpad Edge variant seems sensible.
Cc: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Andreas Herrmann <andreas.herrmann3@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/586347
Symptom: On the Sony VPCS11V9E, using GStreamer-based applications with
PulseAudio in Ubuntu 10.04 LTS results in stuttering audio. It appears
to worsen with increased I/O.
Test case: use Rhythmbox under increased I/O pressure. This symptom is
reproducible in the current daily stable alsa-driver snapshots (at least
up until 21 May 2010; later snapshots fail to build from source due to
missing preprocessor directives when compiled against 2.6.32).
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: Lauri Kainulainen <lauri@sokkelo.net>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a small off-by-one bug which causes the feature unit to announce a
wrong number of channels. This leads to illegal requests sent to the
firmware eventually.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The S5PV210 and S5PC110 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for S5PC110 and
S5PV210 also.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S5PC100 has the AC97 controller same as S3C6410.
Simply enable the options to build the drivers for
S5PC100 also.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add hd radio blend functions. HPI version inc to 4.03.25.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This request is again handled differently in comparison to UAC1.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 devices have their information about pitch control stored in a
different field. Parse it, and emulate the bits for a v1 device.
A new struct uac2_iso_endpoint_descriptor is added.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
-1 is not a good return value as it means -EPERM, "not permitted".
Choose -ENOTSUPP instead, which is what the code really wants to tell
its callers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a spin_unlock missing on the error path.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression E1;
@@
* spin_lock(E1,...);
<+... when != E1
if (...) {
... when != E1
* return ...;
}
...+>
* spin_unlock(E1,...);
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These scales should be regular, not linear.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
These should be regular rather than linear scales.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
This updates the i.MX SSI driver to make it compatible with the ASoC tree
following the move of DMA parameters from the DAI to the audio substream
object.
Signed-off-by: Stuart Longland <redhatter@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Add a module option to allow the GPR mixer controls to have the full
resolution of the hardware, i.e., 0...2^31-1 instead of 0...100.
Because of bugs in userspace tools like alsactl and alsamixer, this is
not yet enabled by default.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the cleanup of the hw_ptr update functions in 2.6.33, the calculation
of the delta value was changed to use the modulo operator to protect
against a negative difference due to the pointer wrapping around at the
boundary.
However, the ptr variables are unsigned, so a negative difference would
result in the two complement's value which has no relation to the actual
difference relative to the boundary; the result is typically some value
near LONG_MAX-boundary. Furthermore, even if the modulo operation would
be done with signed types, the result of a negative dividend could be
negative.
The invalid delta value is then caught by the following checks, but this
means that the pointer update is ignored.
To fix this, use a range check as in the other pointer calculations.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check that the interrupt raised for a stream is actually a buffer
completion interrupt before handling it as one. Otherwise, memory
errors or FIFO xruns would be interpreted as a pointer update and could
break the stream timing.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/576160
Symptom: Currently (2.6.32.12) the Dell M1730 uses the 3stack model
quirk. Unfortunately this means that capture is not functional out-
of-the-box despite ensuring that capture settings are unmuted and
raised fully.
Test case: boot from Ubuntu 10.04 LTS live cd; capture does not
work.
Resolution: Correct the model quirk for Dell M1730 to rely on the
BIOS configuration.
This patch also trivially sorts the quirk into the correct section
based on the comments.
Reported-and-Tested-By: <picdragon99@msn.com>
Tested-By: Daren Hayward
Tested-By: Tobias Krais
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
First issue:
With the original patch, I've noticed by unmuting the mic
(and even having it muted), there is a distorted("Noise")
coming from the internal speakers, even when the headphones are plugged in.
What my finding's revealed is:
/* Mic (rear) pin: input vref at 80% */
{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
From the original patch. Looking at codec#0 0x18/0x1a is listed as:
Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0000373c: IN OUT HP Detect
Vref caps: HIZ 50 GRD 80 100
Pin Default 0x90100141: [Fixed] Speaker at Int N/A
Conn = Unknown, Color = Unknown
DefAssociation = 0x4, Sequence = 0x1
Misc = NO_PRESENCE
Pin-ctls: 0x41: OUT VREF_50
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c* 0x0d 0x0e 0x0f 0x26
seems this Node is listed as: [Fixed] Speaker while 0x15
Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out
Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0
Amp-In vals: [0x00 0x00]
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x80 0x80]
Pincap 0x0000373c: IN OUT HP Detect
Vref caps: HIZ 50 GRD 80 100
Pin Default 0x018b3020: [Jack] Line In at Ext Rear
Conn = Comb, Color = Blue
DefAssociation = 0x2, Sequence = 0x0
Pin-ctls: 0x01: VREF_50
Unsolicited: tag=00, enabled=0
Connection: 5
0x0c 0x0d* 0x0e 0x0f 0x26
is [Jack] Line In at Ext Rear.
(looking at the other apple products as examples
I came up with the fix below).
Second issue:
alc885_mbp_4ch_modes
The original patch does a good job with the
HP pin automute function, but from what I noticed is I would have to manually
change the channel form 2 to 4 after plugging the headphones in.
And not to mention having odd moments to where I was jamming out
with the headphones on, then later realized I had sound blasting out
of the speakers as well. My findings revealed that changing
alc885_mbp_4ch_modes to alc885_mba21_ch_modes and setting
- spec->autocfg.speaker_pins[0] = 0x15;
+ spec->autocfg.speaker_pins[0] = 0x18;
gets the automute function when the headphones plugged in working
flawlessly(and the no need to manually change the channel number
afterwards).
Third issue:
alc885_imac91_mixer
There probably doesnt need to be anything changed with this
(esspecially if your one to like lots of sliders),but my findings
revealed that mac osx only has a master on the top right,
another switch on itunes, and then a slider for the mic.
So the changes I did below try and mimic osx as much as possible
(only thing I had an issue with is just having one mute switch
on the master, instead of having two(still investigating)).
fourth issue:
alc882_capture_source
I endeded up creating alc889A_imac91_capture_source()
only because looking at alc882_capture_source I see
that the mic is set to 0x1 while this works, I also noticed
that adding 0x1 and 0x01 and testing that 0x1 somehow
stops working, and 0x01 works(so I figured 0x01 was more
of the alpha of the numbers(still need to figure out
where that valuse is)). In any case the microphone
does work with the original, and with the below patch, but both
still record not as clean(lots of "Noise", which I would like to
look into too).
Note: using alsamixer -Va reveals the capture switches.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/549560
Symptom: on a significant number of hardware, booting from a live cd
results in capture working correctly, but once the distribution is
installed, booting from the install results in capture not working.
Test case: boot from Ubuntu 10.04 LTS live cd; capture works correctly.
Install to HD and reboot; capture does not work. Reproduced with 2.6.32
mainline build (vanilla kernel.org compile)
Resolution: add SSID for Toshiba A100-259 to the position_fix quirk
table, explicitly specifying the LPIB method.
I'll be sending additional patches for these SSIDs as bug reports are
confirmed.
This patch also trivially sorts the quirk table in ascending order by
subsystem vendor.
Reported-and-Tested-by: <davide.molteni@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/583983
Symptom: on a significant number of hardware, booting from a live cd
results in capture working correctly, but once the distribution is
installed, booting from the install results in capture not working.
Test case: boot from Ubuntu 10.04 LTS live cd; capture works correctly.
Install to HD and reboot; capture does not work. Reproduced with 2.6.32
mainline build (vanilla kernel.org compile).
Resolution: add SSID for Acer Aspire 5110 to the position_fix quirk
table, explicitly specifying the LPIB method.
I'll be sending additional patches for these SSIDs as bug reports are
confirmed.
Reported-and-Tested-By: Leo
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the VENDOR/DEVICE ids provided in pci_ids.h instead of creating
local ids of the same values.
Also, fix the following checkpatch.pl warnings:
WARNING: Use #include <linux/io.h> instead of <asm/io.h>
WARNING: unnecessary whitespace before a quoted newline
Signed-off-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* git://git.kernel.org/pub/scm/linux/kernel/git/lethal/sh-2.6: (25 commits)
sh: fix up sh7785lcr_32bit_defconfig.
arch/sh/lib/strlen.S: Checkpatch cleanup
sh: fix up sh7786 dmaengine build.
sh: guard cookie consistency across termination in the DMA driver
sh: prevent the DMA driver from unloading, while in use
sh: fix Oops in the serial SCI driver
sh: allow platforms to specify SD-card supported voltages
mmc: let MFD's provide supported Vdd card voltages to tmio_mmc
sh: disable SD-card write-protection detection on kfr2r09
mfd: pass platform flags down to the tmio_mmc driver
tmio: add a platform flag to disable card write-protection detection
sh: Add SDHI DMA support to migor
sh: Add SDHI DMA support to kfr2r09
sh: Add SDHI DMA support to ms7724se
sh: Add SDHI DMA support to ecovec
mmc: add DMA support to tmio_mmc driver, when used on SuperH
sh: prepare the SDHI MFD driver to pass DMA configuration to tmio_mmc.c
mmc: prepare tmio_mmc for passing of DMA configuration from the MFD cell
sh: add DMA slave definitions to sh7724
sh: add DMA slaves for two SDHI controllers to sh7722
...
Now that DMA slave IDs are only used used in platform specific code and have
become opaque cookies for the rest of the code, we can make the, CPU specific
too.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Merging in current state of Linus' tree to deal with merge conflicts and
build failures in vio.c after merge.
Conflicts:
drivers/i2c/busses/i2c-cpm.c
drivers/i2c/busses/i2c-mpc.c
drivers/net/gianfar.c
Also fixed up one line in arch/powerpc/kernel/vio.c to use the
correct node pointer.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
.name, .match_table and .owner are duplicated in both of_platform_driver
and device_driver. This patch is a removes the extra copies from struct
of_platform_driver and converts all users to the device_driver members.
This patch is a pretty mechanical change. The usage model doesn't change
and if any drivers have been missed, or if anything has been fixed up
incorrectly, then it will fail with a compile time error, and the fixup
will be trivial. This patch looks big and scary because it touches so
many files, but it should be pretty safe.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Sean MacLennan <smaclennan@pikatech.com>
By moving dma_mask into pdev_archdata, and adding archdata to
struct of_device, it makes it possible to substitute of_device
with struct platform_device, which is a stepping stone to
removing the of_platform bus entirely.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
* 'next' of git://git.kernel.org/pub/scm/linux/kernel/git/djbw/async_tx:
DMAENGINE: extend the control command to include an arg
async_tx: trim dma_async_tx_descriptor in 'no channel switch' case
DMAENGINE: DMA40 fix for allocation of logical channel 0
DMAENGINE: DMA40 support paused channel status
dmaengine: mpc512x: Use resource_size
DMA ENGINE: Do not reset 'private' of channel
ioat: Remove duplicated devm_kzalloc() calls for ioatdma_device
ioat3: disable cacheline-unaligned transfers for raid operations
ioat2,3: convert to producer/consumer locking
ioat: convert to circ_buf
DMAENGINE: Support for ST-Ericssons DMA40 block v3
async_tx: use of kzalloc/kfree requires the include of slab.h
dmaengine: provide helper for setting txstate
DMAENGINE: generic channel status v2
DMAENGINE: generic slave control v2
dma: timb-dma: Update comment and fix compiler warning
dma: Add timb-dma
DMAENGINE: COH 901 318 fix bytesleft
DMAENGINE: COH 901 318 rename confusing vars
* 'next' of git://git.kernel.org/pub/scm/linux/kernel/git/benh/powerpc: (92 commits)
powerpc: Remove unused 'protect4gb' boot parameter
powerpc: Build-in e1000e for pseries & ppc64_defconfig
powerpc/pseries: Make request_ras_irqs() available to other pseries code
powerpc/numa: Use ibm,architecture-vec-5 to detect form 1 affinity
powerpc/numa: Set a smaller value for RECLAIM_DISTANCE to enable zone reclaim
powerpc: Use smt_snooze_delay=-1 to always busy loop
powerpc: Remove check of ibm,smt-snooze-delay OF property
powerpc/kdump: Fix race in kdump shutdown
powerpc/kexec: Fix race in kexec shutdown
powerpc/kexec: Speedup kexec hash PTE tear down
powerpc/pseries: Add hcall to read 4 ptes at a time in real mode
powerpc: Use more accurate limit for first segment memory allocations
powerpc/kdump: Use chip->shutdown to disable IRQs
powerpc/kdump: CPUs assume the context of the oopsing CPU
powerpc/crashdump: Do not fail on NULL pointer dereferencing
powerpc/eeh: Fix oops when probing in early boot
powerpc/pci: Check devices status property when scanning OF tree
powerpc/vio: Switch VIO Bus PM to use generic helpers
powerpc: Avoid bad relocations in iSeries code
powerpc: Use common cpu_die (fixes SMP+SUSPEND build)
...
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The decoding/encoding is based on own reverse-engineering. Both control and
data ports are handled. Writing to control port supports SysEx events only,
as this is the only type of messages that MPD16 recognizes.
Signed-off-by: Krzysztof Foltman <wdev@foltman.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 7910b4a1db in 2.6.34 changed the
runtime->boundary calculation to make this value a multiple of both the
buffer_size and the period_size, because the latter is assumed by the
runtime->hw_ptr_interrupt calculation.
However, due to the lack of a ioctl that could read the software
parameters before they are set, the kernel requires that alsa-lib
calculates the boundary value, too. The changed algorithm leads to
a different boundary value used by alsa-lib, which makes, e.g., mplayer
fail to play a 44.1 kHz file because the silence_size parameter is now
invalid; bug report:
<https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5015>.
This patch reverts the change to the boundary calculation, and instead
fixes the hw_ptr_interrupt calculation to be period-aligned regardless
of the boundary value.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When using MCLK is configured for 19.2 Mhz, clock slicer should be
enabled and HPPLL should be bypassed in clock path.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
This is needed before the USB merge.
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
For more clearance what the functions actually do,
usb_buffer_alloc() is renamed to usb_alloc_coherent()
usb_buffer_free() is renamed to usb_free_coherent()
They should only be used in code which really needs DMA coherency.
All call sites have been changed accordingly, except for staging
drivers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Alan Stern <stern@rowland.harvard.edu>
Cc: Pedro Ribeiro <pedrib@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits)
ALSA: hda: Storage class should be before const qualifier
ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies
ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT
ASoC: sdp4430 - add sdp4430 pcm ops to DAI.
ASoC: TWL6040: Enable earphone path in codec
ASoC: SDP4430: Add support for Earphone speaker
ASoC: SDP4430: Add sdp4430 machine driver
ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF
ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function
ALSA: sound/pci/asihpi: Use kzalloc
ALSA: hdmi - dont fail on extra nodes
ALSA: intelhdmi - add id for the CougarPoint chipset
ALSA: intelhdmi - user friendly codec name
ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS
ALSA: asihpi: incorrect range check
ALSA: asihpi: testing the wrong variable
ALSA: es1688: add pedantic range checks
ARM: McBSP: Add support for omap4 in McBSP driver
ARM: McBSP: Fix request for irq in OMAP4
OMAP: McBSP: Add 32-bit mode support
...
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (44 commits)
vlynq: make whole Kconfig-menu dependant on architecture
add descriptive comment for TIF_MEMDIE task flag declaration.
EEPROM: max6875: Header file cleanup
EEPROM: 93cx6: Header file cleanup
EEPROM: Header file cleanup
agp: use NULL instead of 0 when pointer is needed
rtc-v3020: make bitfield unsigned
PCI: make bitfield unsigned
jbd2: use NULL instead of 0 when pointer is needed
cciss: fix shadows sparse warning
doc: inode uses a mutex instead of a semaphore.
uml: i386: Avoid redefinition of NR_syscalls
fix "seperate" typos in comments
cocbalt_lcdfb: correct sections
doc: Change urls for sparse
Powerpc: wii: Fix typo in comment
i2o: cleanup some exit paths
Documentation/: it's -> its where appropriate
UML: Fix compiler warning due to missing task_struct declaration
UML: add kernel.h include to signal.c
...
The C99 specification states in section 6.11.5:
The placement of a storage-class specifier other than at the beginning
of the declaration specifiers in a declaration is an obsolescent
feature.
Signed-off-by: Tobias Klauser <tklauser@distanz.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These pins are for decoupling capacitors for the internal charge pumps
in TPA6130A2 and TPA6140A2 and not for connecting external supply.
Thanks to Eduardo Valentin <eduardo.valentin@nokia.com> for pointing out the
issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen@nokia.com> with
TPA6140A2.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Reviewed-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (224 commits)
ARM: remove 'select GENERIC_TIME'
ARM: 6136/1: ARCH_REQUIRE_GPIOLIB selects GENERIC_GPIO
ARM: 6074/1: oprofile: convert from sysdev to platform device
ARM: 6073/1: oprofile: remove old files and update KConfig
ARM: 6072/1: oprofile: use perf-events framework as backend
ARM: 6071/1: perf-events: allow modules to query the number of hardware counters
ARM: 6070/1: perf-events: add support for xscale PMUs
ARM: 6069/1: perf-events: use numeric ID to identify PMU
ARM: 6064/1: pmu: register IRQs at runtime
ARM: Optionally allow ARMv6 to use 'normal, bufferable' memory for DMA
ARM: 6134/1: Handle instruction cache maintenance fault properly
ARM: nwfpe: allow debugging output to be configured at runtime
ARM: rename mach_cpu_disable() to platform_cpu_disable()
ARM: 6132/1: PL330: Add common core driver
ARM: 6094/1: Extend cache-l2x0 to support the 16-way PL310
ARM: Move memory mapping into mmu.c
ARM: Ensure meminfo is sorted prior to sanity_check_meminfo
ARM: Remove useless linux/bootmem.h includes
ARM: convert /proc/cpu/aligment to seq_file
arm: use asm-generic/scatterlist.h
...
Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise
external widgets doesn't alter the output state.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add support for NUC900 AC97
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add control to enable earphone driver in TWL6040 codec. This driver
is connected to HSDAC Left.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The following structure elements duplicate the information in
'struct device.of_node' and so are being eliminated. This patch
makes all readers of these elements use device.of_node instead.
(struct of_device *)->node
(struct dev_archdata *)->prom_node (sparc)
(struct dev_archdata *)->of_node (powerpc & microblaze)
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
* 'core-locking-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
lockdep: Reduce stack_trace usage
lockdep: No need to disable preemption in debug atomic ops
lockdep: Actually _dec_ in debug_atomic_dec
lockdep: Provide off case for redundant_hardirqs_on increment
lockdep: Simplify debug atomic ops
lockdep: Fix redundant_hardirqs_on incremented with irqs enabled
lockstat: Make lockstat counting per cpu
i8253: Convert i8253_lock to raw_spinlock
This adds an argument to the DMAengine control function, so that
we can later provide control commands that need some external data
passed in through an argument akin to the ioctl() operation
prototype.
[dan.j.williams@intel.com: fix up some missed conversions]
Signed-off-by: Linus Walleij <linus.walleij@stericsson.com>
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
Avoid calling the dac33_hard_power when the codec was
already in BIAS_OFF state.
This could happen in device suspend and module removal
time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Since the cases when the same power state would be set again
handled gracefully, we do not need to use dev_warn.
Signed-off-by: Felipe Balbi <felipe.balbi@nokia.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Use kzalloc rather than the combination of kmalloc and memset.
The semantic patch that makes this change is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x,size,flags;
statement S;
@@
-x = kmalloc(size,flags);
+x = kzalloc(size,flags);
if (x == NULL) S
-memset(x, 0, size);
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The number of HDMI nodes is expected to go up in future.
So don't fail hard on seeing extra converter/pin nodes.
We can still operate safely on the nodes within
MAX_HDMI_CVTS/MAX_HDMI_PINS.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the full chipset codename as codec name.
They are more user friendly than the spec abbrs.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is necessary to support >=3 HDMI playback devices
starting from the CougarPoint codec.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The entity_type_to_size[] array has LAST_ENTITY_TYPE (11) number of elements,
not LAST_ENTITY_ROLE (17). This only affects the debug output.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Smatch complains that if (dev == SNDRV_CARDS) we're one past the end of
the array. That's unlikely to happen in real life, I suppose.
Also smatch complains about "strcpy(card->shortname, pcm->name);"
The "pcm->name" buffer is 80 characters and "card->shortname" is 32
characters. If you follow the call paths it turns out we never actually
use more than 16 characters so it's not a problem. But anyway, let's
make it easy for people auditing this in the future.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is adding a new control which has the following capabilities:
- tlv
- variable data size (for instance, 7 ou 8 bit)
- double mixer
- data range centered around 0
Signed-off-by: Arnaud Patard <apatard@mandriva.com>
Acked-by: Liam Girdwood <lrg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec,
or others.
Signed-off-by: Sergey Lapin <slapin@ossfans.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The capture source control of maya44 was wrongly coded with the bit
shift instead of the bit mask. Also, the slot for line-in was
wrongly assigned (slot 5 instead of 4).
Reported-by: Alex Chernyshoff <alexdsp@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec has support for swapping the left and right
channels in the digimic interface.
New kcontrol to handle this bit.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
MIPS non-coherent archs need the noncached pgprot in mmap of PCM buffers.
But, since the coherency needs to be checked dynamically via
plat_device_is_coherent(), we need an ugly check dependent on MIPS
in ALSA core code.
This should be cleaned up in MIPS arch side (e.g. creating
dma_mmap_coherent()) in near future.
Tested-by: Wu Zhangjin <wuzhangjin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 65c3ac885c in 2.6.33 accidentally
left out the initialization of the AC97 codec FMIC2MIC bit, which broke
recording from the front panel microphone.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that HP dv series have inconsistent the mute-LED GPIO
mapping among various models. dv4/7 seem to use GPIO 0 while dv 5/6
seem to use GPIO 3. The previous commit
26ebe0a289
ALSA: hda - Fix mute-LED GPIO pin for HP dv series
breaks dv5/6.
This patch adds the new quirk model, hp-dv4, to handle HP dv4/7
separately from HP dv5/6.
Tested-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com> (for dv6-1110ax)
Acked-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For both UAC1 and UAC2, interrupt endpoint messages are now parsed with
structs rather that with anonymous buffer array accesses.
For UAC2, only CUR interrupt notifications are supported for now.
snd_usb_mixer_status_complete() was renamed to
snd_usb_mixer_interrupt().
Fixed one indentation flaw on the way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to prevent code ambiguous, add namespace on functions in ssp driver.
Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
PXA_SSP is actually used by drivers like drivers/spi/pxa2xx_spi.c and
sound/soc/pxa/pxa-ssp.c, not by boards. Remove those incorrect 'select'
from Kconfig and make SOC_PXA_SSP to select.
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
