The widget connections of ADC of ALC880 and ALC2260 aren't initialized,
thus it might point to invalid pin. This can be a problem when mode=auto
and there is only one input pin. Then user can't change the connection
at all.
This patch adds the code to initialize the input pin connection of these
codecs.
Reference: Novell bnc#594363
https://bugzilla.novell.com/show_bug.cgi?id=594363
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Register the WM8750 as a SPI or I2C device. This patch mostly shuffles code
around. Hugely inspired by WM8753 which was already converted.
Also, this patch fixes the Jive and Spitz machine.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Corrected HP and mic pins for ALC269vb amic and dmic models.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269vb has an alternative HP pin 0x21 in addition.
Fix the parser to recognize it.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a local mutex instead of BKL. This should suffice since each device
type has also its open_mutex.
Also, a bit of clean-up of the legacy device auto-loading code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using a regular timer results in poll times < 1 jiffie with small
buffers, so we loaded the timer with the actual jiffie value. We can
be more accurate using a hrtimer. Also, we have to call
snd_pcm_period_elapsed after playing period_bytes and not
runtime->period_size (which is in samples and not in bytes).
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When checking if we are DMA capable we have to check for the
IMX_SSI_DMA flag which is already set from platform_data instead
of setting it again when we want to do DMA.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@Slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: mixart: range checking proc file
ALSA: hda - Fix a wrong array range check in patch_realtek.c
ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream
ALSA: hda - Enable amplifiers on Acer Inspire 6530G
ASoC: Only do WM8994 bias off transition from standby
ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices
ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction
ASoC: Support second DC servo readback method for wm_hubs
ASoC: Avoid wraparound in wm_hubs DC servo correction
ALSA: echoaudio - Eliminate use after free
ALSA: i2c: cleanup: change parameter to pointer
ALSA: hda - Add MSI blacklist for Aopen MZ915-M
ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code
ALSA: hda - Update document about MSI and interrupts
ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981
ALSA: hda - Add missing printk argument in previous patch
ASoC: Fix passing platform_data to ac97 bus users and fix a leak
ALSA: hda - Fix ADC/MUX assignment of ALC269 codec
ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo()
ASoC: wm8994: playback => capture
This adds support for the Medion WIM2160 soundcard.
There's no PCI quirk added because it has the same PCI id as the
Medion MD2.
Signed-off-by: Maurus Cuelenaere <mcuelenaere@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Assign DACs properly to each output. Currently, the front output is bound
to HP/speaker outputs blindly, but they should be assigned to individual
DACs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The original code doesn't take into consideration that the value of
MIXART_BA0_SIZE - pos can be less than zero which would lead to a large
unsigned value for "count".
Also I moved the check that read size is a multiple of 4 bytes below
the code that adjusts "count".
Signed-off-by: Dan Carpenter <error27@gmail.com>
Cc: <stable@kernel.org>
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 6a4f2ccb46 introduced a wrong
comparision for the array range check, which effectively skips the whole
initialization of DAC connections. Fixed now.
Reference: bko#15689
https://bugzilla.kernel.org/show_bug.cgi?id=15689
Reported-by: Adrian Ulrich <kernel@blinkenlights.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support interrupt based microphone bias detection. The WM8994 has two
microphone bias supplies, with detection supported on both. Detection
using GPIOs together with the standard GPIO based jack framework is
already supported via the platform data for the WM8994 core driver.
Note that as well as the microphone bias itself the system clock and
whichever AIF clock is supplying the system clock will need to be
enabled for detection to function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
[Note that this is a backported version for 2.6.34.
Upstream commit is fd23b7dee]
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After more tests it appears that EAPD needs to be enabled
on both the 0x14 and 0x15 NIDs to enable the main speaker
and headphone amplifiers. The maximum volume setting is
now equal to what the machine achieves under other operating
systems.
Disabling Front or LFE playback triggers EAPD and disables
the amplifier. As such, these two playback switches have
been removed from the mixer.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Otherwise we may try to power down multiple times when the using
idle bias off and the driver is removed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The DCS_DATAPATH_BUSY bit used to monitor the completion of DC servo
operations has been deprecated and with some more recente revisions
may perform incorrectly, especially when only analogue bypass paths
are in use. Switch to using readback from the DC servo command
register instead, which is supported for all devices. Without this
unacceptably long timeouts may be observed in some circumstances.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
If we need to offset correct the DC servo then don't use runtime
recalibration since that is likely to introduce further offsets
which will be evident on powerdown.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
More recent Wolfson hubs devices add the ability to read back the DC
servo calibration information from the register used to write offsets,
and later still ones remove the old readback registers. Add support
for the new scheme, and use it for WM8994 device revisions that
support it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
If the correction wraps around then a substantial offset would be
introduced.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Reverse headphone detection bit on PowerMac G4 Digital Audio (Tumbler).
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the call to snd_card_free in the error handling code at the end of the
function, as in the other error cases.
A simplified version of the semantic patch that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@@
expression E,E2;
@@
snd_card_free(E)
...
(
E = E2
|
* E
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We actually pass an array of 7 chars not 5.
This silences a smatch warning.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device needs MSI disablement. Added to the quirk list.
Reported-by: Harald Dunkel <harri@afaics.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With recent (2.6.34) chnages in PCM handling, capture stopped working on my
OMAP1510 based Amstrad Delta videophone.
Using 2.6.34-rc2, I was able to correct the problem in 3 different ways:
1. reverting commit 7b3a177b0d,
2. enabling additional jiffies check with
echo 4 >/proc/asound/card0/pcm0c0/xrun_debug
3. applying the patch below.
Since I wasn't able to reproduce the problem on my i686 PC, I guess the
problem is probably machine specific.
The patch reuses the method for software emulation of missing hardware
pointer, already implemented for playback on OMAP1510. It's possible that
event if a hardware pointer is available for capture on this machine, its
behaviour may be not compatible with what upper layer expects.
If you think the problem may be more general and should be solved differently,
on a higher level, I can try to work more on it if you give me a hint.
If the patch gets accepted, I suggest it goes as a fix in the current release
cycle.
Created and tested against linux-2.6.34-rc2.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://launchpad.net/bugs/551606
The OR's hardware distorts at PCM 100% because it does not correspond to
0 dB. Fix this in patch_ad1981() for all models using the Thinkpad
quirk.
Reported-by: Jane Silber
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new helper, snd_hda_codec_update_cache(), for reducing the unneeded
verbs. This function checks the cached value and skips if it's identical
with the given one. Otherwise it works like snd_hda_codec_write_cache().
The alc269 code uses this function as an example.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some HP laptops have a mute LED that is controlled over the unused
MIC2 VREF pin. Implement the LED updater like patch_sigmatel.c for this
model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
percpu.h is included by sched.h and module.h and thus ends up being
included when building most .c files. percpu.h includes slab.h which
in turn includes gfp.h making everything defined by the two files
universally available and complicating inclusion dependencies.
percpu.h -> slab.h dependency is about to be removed. Prepare for
this change by updating users of gfp and slab facilities include those
headers directly instead of assuming availability. As this conversion
needs to touch large number of source files, the following script is
used as the basis of conversion.
http://userweb.kernel.org/~tj/misc/slabh-sweep.py
The script does the followings.
* Scan files for gfp and slab usages and update includes such that
only the necessary includes are there. ie. if only gfp is used,
gfp.h, if slab is used, slab.h.
* When the script inserts a new include, it looks at the include
blocks and try to put the new include such that its order conforms
to its surrounding. It's put in the include block which contains
core kernel includes, in the same order that the rest are ordered -
alphabetical, Christmas tree, rev-Xmas-tree or at the end if there
doesn't seem to be any matching order.
* If the script can't find a place to put a new include (mostly
because the file doesn't have fitting include block), it prints out
an error message indicating which .h file needs to be added to the
file.
The conversion was done in the following steps.
1. The initial automatic conversion of all .c files updated slightly
over 4000 files, deleting around 700 includes and adding ~480 gfp.h
and ~3000 slab.h inclusions. The script emitted errors for ~400
files.
2. Each error was manually checked. Some didn't need the inclusion,
some needed manual addition while adding it to implementation .h or
embedding .c file was more appropriate for others. This step added
inclusions to around 150 files.
3. The script was run again and the output was compared to the edits
from #2 to make sure no file was left behind.
4. Several build tests were done and a couple of problems were fixed.
e.g. lib/decompress_*.c used malloc/free() wrappers around slab
APIs requiring slab.h to be added manually.
5. The script was run on all .h files but without automatically
editing them as sprinkling gfp.h and slab.h inclusions around .h
files could easily lead to inclusion dependency hell. Most gfp.h
inclusion directives were ignored as stuff from gfp.h was usually
wildly available and often used in preprocessor macros. Each
slab.h inclusion directive was examined and added manually as
necessary.
6. percpu.h was updated not to include slab.h.
7. Build test were done on the following configurations and failures
were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my
distributed build env didn't work with gcov compiles) and a few
more options had to be turned off depending on archs to make things
build (like ipr on powerpc/64 which failed due to missing writeq).
* x86 and x86_64 UP and SMP allmodconfig and a custom test config.
* powerpc and powerpc64 SMP allmodconfig
* sparc and sparc64 SMP allmodconfig
* ia64 SMP allmodconfig
* s390 SMP allmodconfig
* alpha SMP allmodconfig
* um on x86_64 SMP allmodconfig
8. percpu.h modifications were reverted so that it could be applied as
a separate patch and serve as bisection point.
Given the fact that I had only a couple of failures from tests on step
6, I'm fairly confident about the coverage of this conversion patch.
If there is a breakage, it's likely to be something in one of the arch
headers which should be easily discoverable easily on most builds of
the specific arch.
Signed-off-by: Tejun Heo <tj@kernel.org>
Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
The way i've factored out the bus probe and removal functions so
that there's no code in the individual I2C and SPI functions means
that the register() and unregister() functions could just be squashed
into the bus_probe() and bus_remove() functions.
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 issue is an attempt to write the pdata without the AC97 device
allocated when using ac97.c - also added a comment in soc-core.c for the
special case for ac97. -- broonie]
Signed-off-by: Graham Gower <graham.gower@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Implicit slab.h inclusion via percpu.h is about to go away. Make sure
gfp.h or slab.h is included as necessary.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implicit slab.h inclusion via percpu.h is about to go away. Make sure
gfp.h or slab.h is included as necessary.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 codec has a few different variants, and each of them may have
different ADC and MUX widgets. For example, one model has ADC 0x08
with MUX 0x23 while others has ADC 0x09 or ADC 0x07 with MUX 022 or
0x24. The difference of ADC appears usually as the capability of
the digital mic pin (0x12), and the current driver sometimes misses
the internal mic pin due to the mismatching ADC.
This patch adds a bit more clever way to find the matching ADC instead
of the static list. Now the driver checks all active input pins and
fills only the ADC/MUX's that contain all of them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The mask and value parameters passed to snd_hda_codec_amp_stereo()
should be 8-bit values for mute and volume. Passing AMP_IN_MUTE() is
wrong, which is found in many places in patch_realtek.c as a left-over
from the conversion to snd_hda_codec_amp_stereo().
Reported-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=575669
The OR states that position_fix=1 is necessary to work around glitching
during volume adjustments using PulseAudio.
Reported-by: Carlos Laviola <claviola@debian.org>
Tested-by: Carlos Laviola <claviola@debian.org>
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/481058
The OR has verified that both 'Headphone Jack Sense' and 'Line Jack Sense'
need to be muted for sound to be audible, so just add the machine's SSID
to the ac97 jack sense blacklist.
Reported-by: Richard Gagne
Tested-by: Richard Gagne
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert the device_terminate_all() operation on the
DMA engine to a generic device_control() operation
which can now optionally support also pausing and
resuming DMA on a certain channel. Implemented for the
COH 901 318 DMAC as an example.
[dan.j.williams@intel.com: update for timberdale]
Signed-off-by: Linus Walleij <linus.walleij@stericsson.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Maciej Sosnowski <maciej.sosnowski@intel.com>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Cc: Pavel Machek <pavel@ucw.cz>
Cc: Li Yang <leoli@freescale.com>
Cc: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Cc: Paul Mundt <lethal@linux-sh.org>
Cc: Ralf Baechle <ralf@linux-mips.org>
Cc: Haavard Skinnemoen <haavard.skinnemoen@atmel.com>
Cc: Magnus Damm <damm@opensource.se>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Joe Perches <joe@perches.com>
Cc: Roland Dreier <rdreier@cisco.com>
Signed-off-by: Dan Williams <dan.j.williams@intel.com>
The commit 4d96eb255c broke the interrupt
time xrun functionality (stream stop etc.) if the CONFIG_SND_PCM_XRUN_DEBUG
is not set. This is because the xrun() is null defined without it.
Fix this by letting the function xrun() to be always defined as it was
before.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ARM-SHMOBILE series have FIFO-buffered serial interface 2 (FSI2)
device which is advanced version of FSI.
This patch add simple support for it.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The probe_only module parameter skips the codec initialization, too.
Remove the model=hwio code and use second bit in probe_only to
skip the HDA codec reset procedure.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
For Lenovo Thinkpad T61/X61, the analog beep input is connected
to node 0x20, index 3. Move the digital beep mute/volume controls
as "Digital Beep" and create analog beep controls for mentioned node.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Using the 'model=hwio' option, the driver bypasses any codec
initialization and the reset procedure for codecs is also
bypassed. This mode is usefull to enable direct access using
hwdep interface (using hdaverb or hda-analyzer tools) and
retain codec setup from BIOS.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
BugLink: https://launchpad.net/bugs/303789
This model needs both 'Headphone Jack Sense' and 'Line Jack Sense'
muted for audible audio, so just add its SSID to the blacklist and
don't enumerate the controls.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add full duplex support on AT91 and AVR.
It was a bug: we needed to check first if there are some chips opened so we
could enable both reception and sending of the data.
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add AC97 support for ATMEL AT91, using the AVR32 code.
While AVR is using a DMA, the AT91 chips are using a Peripheral Data
Controller.
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Skip initialization of connections of DAC widgets that aren't used,
which resulted in invalid verb parameters.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds basic support for M-Audio's Fast Track Ultra series of USB
audio interfaces. It is a refactored version of the patch Clemens
Ladisch posted some time ago. Neither playback nor capturing work
properly at 44100 Hz (don't know why).
The other sampling rates work properly. There's no support for the DSP
mixer, yet.
Signed-off-by: Felix Homann <fexpop@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sparse caught that initialize "playback" two times instead of
initializing "capture".
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Bit operation for fsi_master should be done inside master lock.
But soft-reset/interrupt operation were outside of it.
This patch modify this problem.
It still allow to INT_ST outside-operation on fsi_interrupt,
but it is not problem.
Because this register doesn't need the bit operation.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When the CMI8738 FRAME2 register is read, the chip sometimes (probably
when wrapping around) returns an invalid value that would be outside the
programmed DMA buffer. This leads to an inconsistent PCM pointer that is
likely to result in an underrun.
To work around this, read the register multiple times until we get a
valid value; the error state seems to be very short-lived.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Matija Nalis <mnalis-alsadev@voyager.hr>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper suspend/resume code for Terratec Aureon cards.
Based on ice1724 suspend/resume work of Igor Chernyshev.
Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4944
Tested on linux-2.6.32.9
Signed-off-by: Bernhard Urban <lewurm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current ak4642 was not able to select pll.
This patch add support it.
It still expect PLL base input pin is MCKI.
see Table 5 "setting of PLL Mode" of datasheet
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This also adds the first DAI operation for AIF3 so fill out the ID and
the ops for that too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Now that the EHCI driver copes with small iso packets without blowing
up, take the snd-ua101 driver out of the alpha-test stage.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Don't force enable the microphone bias on WM8903 when doing jack
detection, and don't force enable microphone bias. This allows
platforms to only enable microphone detection when a jack has been
inserted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
If no report is specified then disable detection. Note that we don't
disable the slow clock, though the power consumption from it should
be negligable. That should be reference counted, ideally through DAPM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow machines to control exactly when the bias is turned on and off.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some systems, such as those with mechanical jack detection, may wish
to force enable a pin (typically mic bias) only some of the time.
Support such systems by having disable_pin() also coveer force enabled
pins.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some systems provide both mechanical and electrical detection of jack
status changes. On such systems power savings can be achieved by only
enabling the electrical detection methods when physical insertion has
been detected.
Begin supporting such systems by providing a notifier for jack status
changes which can be used to trigger any reconfiguration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Gain controls on outputs affect the power consumption
when the gain is set to non 0 value.
Outputs with amps have one register to configure the
routing and the gain:
PREDL_CTL (0x25):
bit 0: Voice enable
bit 1: Audio L1 enable
bit 2: Audio L2 enable
bit 3: Audio R2 enable
bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB)
bit 0 - 3: is handled in DAPM domain (DAPM_MIXER)
bit 4 - 5: has simple volume control
If there is no audio activity (BIAS_STANDBY), and
user changes the volume, than the output amplifier will
be enabled.
If the user changes the routing (but the codec remains in
BIAS_STANDBY), than the cached gain value also be written
to the register, which enables the amplifier.
The existing workaround for this is to have virtual
PGAs associated with the outputs, and whit DAPM PMD
the gain on the output will be forced to 0 (off) by
bypassing the regcache.
This failed to disable the amplifiers in several
scenario (as mentioned above).
Also if the codec is in BIAS_ON state, and user modifies
a volume control, which path is actually not enabled, than
that amplifier will be enabled as well, but it will
be not turned off, since there is no DAPM path, which
would make mute it.
To prevent amps being enabled, when they are not
needed, introduce the following workaround:
Track the state of each of this type of output.
In twl4030_write only allow actual write, when the
given output is enabled, otherwise only update
the reg_cache.
The PGA event handlers on power up will write the cached
value to the chip (restoring gain, routing selection).
On power down 0 is written to the register (disabling
the amp, and also just in case clearing the routing).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build errors when CONFIG_PM is not enabled:
sound/usb/card.c:629: error: 'usb_audio_suspend' undeclared here (not in a function)
sound/usb/card.c:630: error: 'usb_audio_resume' undeclared here (not in a function)
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device does not have audio controllers and backlit buttons only.
Input data is handled over a dedicated USB endpoint.
All functions are supported by the driver now.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the device id for Nvidia GT220 cards to the nvhdmi
driver. I have tested it and confirmed it to be working.
Original patch download link:
https://gist.github.com/324070/
Signed-off-by: Derek Kelly <user.vdr@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/420578
The OR has verified that his hardware distorts because of the 0 dB
offset not corresponding to the highest PCM level. Fix this by capping
said PCM level to 0 dB similarly to what we do for CX20549 (Venice).
Reported-by: Mike Pontillo <pontillo@gmail.com>
Tested-by: Mike Pontillo <pontillo@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding this PCI quirk fixes the board config detection.
This also fixes jack sensing by using "hp_detect=1" via properly detected
board config.
Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The constant DMA_ACTIVE is defined with the dma_buffparams structure rather
than with the audio_operations structure. Takashi Iwai suggested that the
dmap_out field of the audio_operations structure should be used instead.
This is not tested.
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If instantiation of a card failed, we still have to remove it from the
card list on unregistration. This fixes an Oops on Migo-R, triggering,
when after a failed firmware load attempt the driver modules are removed
and re-inserted again.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a warning ("pxa_free_dma: trying to free channel 0 which is
already freed") when a device was opened but the hw_params() call
failed.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Note that since all the microphones share a bias there is a single
jack exported for all three, even though there are two physical
connectors plus the soldered down silicon mic. Note also that the SiMic
is always present by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
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 export is not needed since the per-bus code lives in the same
module.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Yi Li <yi.li@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Initial version of TWL6040 codec driver.
The TWL6040 codec uses a proprietary PDM-based digital audio interface.
Audio paths supported are:
- Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right
- Output: Headset Left/Right, Handsfree Left/Right
TWL6040 codec supports power-up/down manual and automatic sequence.
Manual sequence is done through a specific register writes sequence.
Automatic sequence is done when the codec is powered-up through the
external AUDPWRON line. The completion of the sequence is signaled
through the audio interrupt.
TWL6040 codec sysclk can be provided by: low-power or high
performance PLL:
- The low-power PLL takes a low-frequency input at 32,768 Hz and
generates an approximate of 17.64 or 19.2 MHz (for 44.1 KHz and 48 KHz
respectively)
- The high-performance PLL generates an exact 19.2 MHz clock signal
from high-frequency input at 12/19.2/26/38.4 MHz.
Low-power playback mode is a special scenario where only headset path
(headset DAC and driver) is active.
For the particular case of headset path, PLL being used defines the
headset power mode: low-power, high-performance.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're keeping track of the number of times we've iterated but never
actually using this to bail out if the chip looks stuck.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
During validation of the internal clocking setup it has
been found that the following settings were not configured
in an optimal way:
ASRC_CTRL_A: SRCLKDIV was incorrect, instad of divide ratio 3,
ratio of 2 has to be used (as the comment stated)
DAC_CTRL_A: Fs = Fsref is the desired configuration instead of
Fs = Fsref / 1.5
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To make DSP_A mode working correctly the data delay should be
configured to 0. DSP_B mode thus can not be used with DAC33,
so remove it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Basic support for Left Justified coding for OMAP McBSP.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Indentation in initial support for McPDM driver was converted to spaces.
Use tabs to comply with open source coding-style.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added alc_codec_rename() helper for renaming codec->chip_name.
Added Acer-specific codec naming for ALC269/662.
[Clean-up and refactoring by tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added alc_auto_parse_customize_define() to parse the Realtek-specific
attributes from SKU. Also enable beep controls only when the proper
attribute bit is set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new check for MIC. Do the internal DMIC as the Front MIC.
It could solve the default record source index issue.
[Fix the check properly using the bitmask by tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I think this should be automatic pin instead of ping.
(but could be wrong).
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
sound: sequencer: clean up remove bogus check
ALSA: hda: Use LPIB and 6stack-dig for eMachines T5212
ALSA: hda - Disable MSI for Nvidia controller
ALSA: hda - Add PCI quirks for MSI NetOn AP1900 and Wind Top AE2220
ALSA: hda - Fix secondary ADC of ALC260 basic model
ALSA: hda - Add an error message for invalid mapping NID
ALSA: hda - New Intel HDA controller
The SIU ASoC driver must load firmware to program the DSP, therefore it
has to select FW_LOADER in its Kconfig entry.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The registers for AD193X are defined as 0x800-0x810 for spi which uses
16_8 mode, for i2c to support AD1937, we will use 8_8 mode, only the low
byte of 0x800-0x810 is valid. The patch will not destory other codecs,
but make soc cache interface more useful.
Signed-off-by: Barry Song <barry.song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some devices provide support for detection of a small number of
buttons on their jacks. One common implementation provides a single
button, implemented by shorting the microphone to ground and detected
along with microphone presence detection by detecting varying current
draws on the microphone bias signal.
Provide support for up to three buttons via the jack interface. These
default to reporting BTN_n but an API is provided to allow these to
be remapped to other keys by the machine driver where it knows what
the keys are. More keys can be added with ease if required.
This is only intended to support simple accessory button designs. If
the interface is limiting then either creating a child device for the
accessory or accessing the input device in the jack directly is
recommended.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8750 is using some delayed work to manage the ramping of the bias
at startup and resume out of line from the normal flow. This predates
the support within ASoC core for moving the resume out of line from the
main system resume which provides equivalent functionality with better
interaction with applications. Change to doing the ramp in line to make
use of the core functionality.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A code audit reveals that there are currently no users of the widget
controls on PGAs. This is likely to continue to be the case since
while there are useful things that can be done with integrating the
PGA gain and mute controls with the power sequencing userspace
generally wants stereo controls for output stages which this doesn't
map onto well.
In preparation for implementing something more useful strip out the
existing code, leaving the parameters there for use by the new code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8350 provides microphone presence and short circuit detection.
Integrate this with the ASoC jack reporting API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8904 allows microphone detection signals to be brought out as
alternate functions of the GPIO signals which can be detected using
interrupt inputs on the CPU. Allow this to be configured using
platform data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide platform data allowing the configuration of the GPIO pins
on the WM8904 to be selected, allowing alternate functions to be
enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Support use of the WM8903 IRQ for reporting of microphone presence
and short detection.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently used to detect completion of the write sequencer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide support for WM8903 microphone presence and short detection
using the GPIOs to route out a logic signal suitable for handling
using snd_soc_jack_add_gpios() on the processor GPIOs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow users to pass in a default configuration for the GPIOs of
the WM8903 as platform data. This allows configuration of the pin
muxing of the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow pins to be forced on regardless of their power state. This is
intended for use with microphone bias supplies which need to be
enabled in order to support microphone detection - in systems without
appropriate hardware leaving the microphone unbiased when not in use
saves power.
The force done at power check time in order to avoid disrupting other
power detection logic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
A few lines earlier bend is limited to 2399. So semitones is always
less than 24 here.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/538895
The OR has verified that both position_fix=1 and model=6stack-dig are
necessary to have capture function properly. (The existing 3stack-6ch
model quirk seems to be incorrect.)
Reported-by: Reuben Bailey <reuben.e.bailey@gmail.com>
Tested-by: Reuben Bailey <reuben.e.bailey@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Judging from the member of enable_msi white-list, Nvidia controller
seems to cause troubles with MSI enabled, e.g. boot hang up or other
serious issue may come up. It's safer to disable MSI as default for
Nvidia controllers again for now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
This should make the speakers and jack detection work on MSI all-in-one
computers NetOn AP1900 and Wind Top AE2220.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix adc_nids[] for ALC260 basic model to match with num_adc_nids.
Otherwise you get an invalid NID in the secondary "Input Source" mixer
element.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Add an error message to snd_hda_add_nid() for invalid mapping NID to make
easier to hunt the buggy code.
Also added a missing space to the error message in snd_hda_build_controls()
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: hda - Sound MSI fallout on a Asus mobo NVIDIA MCP55
sound: fix opti92x-ad1848 build
ALSA: hda - Fix input source elements of secondary ADCs on Realtek
ALSA: hda - Fix wrong model range check for ALC268
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (56 commits)
doc: fix typo in comment explaining rb_tree usage
Remove fs/ntfs/ChangeLog
doc: fix console doc typo
doc: cpuset: Update the cpuset flag file
Fix of spelling in arch/sparc/kernel/leon_kernel.c no longer needed
Remove drivers/parport/ChangeLog
Remove drivers/char/ChangeLog
doc: typo - Table 1-2 should refer to "status", not "statm"
tree-wide: fix typos "ass?o[sc]iac?te" -> "associate" in comments
No need to patch AMD-provided drivers/gpu/drm/radeon/atombios.h
devres/irq: Fix devm_irq_match comment
Remove reference to kthread_create_on_cpu
tree-wide: Assorted spelling fixes
tree-wide: fix 'lenght' typo in comments and code
drm/kms: fix spelling in error message
doc: capitalization and other minor fixes in pnp doc
devres: typo fix s/dev/devm/
Remove redundant trailing semicolons from macros
fix typo "definetly" -> "definitely" in comment
tree-wide: s/widht/width/g typo in comments
...
Fix trivial conflict in Documentation/laptops/00-INDEX
* 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (370 commits)
ARM: S3C2443: Add set_rate and round_rate calls for armdiv clock
ARM: S3C2443: Remove #if 0 for clk_mpll
ARM: S3C2443: Update notes on MPLLREF clock
ARM: S3C2443: Further clksrc-clk conversions
ARM: S3C2443: Change to using plat-samsung clksrc-clk implementation
USB: Fix s3c-hsotg build following Samsung platform header moves
ARM: S3C64XX: Reintroduce unconditional build of audio device
ARM: 5961/1: ux500: fix CLKRST addresses
ARM: 5977/1: arm: Enable backtrace printing on oops when PC is corrupted
ASoC: Fix S3C64xx IIS driver for Samsung header reorg
ARM: S3C2440: Fix plat-s3c24xx move of s3c2440/s3c2442 support
[ARM] pxa: fix typo in mxm8x10.h
[ARM] pxa/raumfeld: set GPIO drive bits for LED pins
[ARM] pxa/zeus: Add support for mcp2515 CAN bus
[ARM] pxa/zeus: Add support for onboard max6369 watchdog
[ARM] pxa/zeus: Add Eurotech as the manufacturer
[ARM] pxa/zeus: Correct the USB host initialisation flags
[ARM] pxa/zeus: Allow usage of 8250-compatible UART in uncompress
[ARM] pxa: refactor uncompress.h for non-PXA uarts
[ARM] mmp2: fix incorrect calling of chip->mask_ack() for 2nd level cascaded IRQs
...
USB Audio Class v2.0 compliant devices have different descriptors and a
different way of setting/getting min/max/res/cur properties. This patch
adds support for them.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a number of new structs for mixer, selector, feature and
processing units and some static inline helpers to access fields which
have dynamic offsets. Use them in mixer.c to parse the descriptors. This
is necessary for the upcoming audio v2 parsers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For clearer namespace, also rename usbmixer_maps.c -> mixer_maps.c
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move all non-standard mixer controls and vendor-specific extensions to a
separate file. Some structs need to be exported now.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No need for the private enum.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Split the audio.h file in two to clearly denote the differences
between the standards.
- Add many more defines to audio-v2.h. Most of them are not currently
used.
- Replaced a magic value with a proper define
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new delay calback function to report the delay through
ALSA for application caused by the internal FIFO.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Platform data option for the codec to keep the BCLK clock
continuously running in FIFO modes (codec master).
OMAP3 McBSP when in slave mode needs continuous BCLK running
on the serial bus in order to operate correctly.
Since in FIFO mode the DAC33 can also shut down the BCLK clock
and enable it only when it is needed, let the platforms decide
if the CPU side needs the BCLK running or not.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To avoid race condition especially in FIFO modes the
sequence for enabling and disabling the codec need to
be changed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DM365 EVM has two codecs: the Audio Codec (AIC3x) and the Voice Codec,
the idea is to have both enabled in the same kernel simultaneously. However,
the current soc-core doesn't support simultaneous codecs, once that
support will have added, a patch will be posted to enable both codecs in
the DM365 EVM.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the DM365 is the only SoC that includes this Voice Codec.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the support for the interface needed by the DaVinci
Voice Codec CQ93VC.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For ASoC, if either CPU or CODEC driver has set the flag, the MACHINE driver
should be given a chance to figure out if the dai, that set the flag, can
accomodate a rate that it does not explicitly specify but is specified
by the dai at the other end of the link.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver USE PLL for 11025/22050/44100/88200 rate.
To enable switching to bypass mode, PLL is always turned on.
Special thanks to Phil
Signed-off-by: Phil Edworthy <Phil.Edworthy@renesas.com>
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Towards having build for multiple SoCs segregate hw_params callback
for s3c2412 and s3c64xx.
Since, all new SoCs have s3c64xx like register map, we keep that as
default implementation if no SoC specific callback is already defined.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For some CPU-CODEC and source clock combination we might need to set
BCLK to N*Sample_size*LRCLK, where N may be even 3 or 4, not just 2.
We can simply remove the dependency of BCLK on sample size as there
is already a callback(S3C_I2SV2_DIV_BCLK) available to set required BCLK.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Towards generalizing CPU driver interface, do not accept direct field
values for the BCLK and RCLK.
The machine driver should simply request the FS-multiple and not provide
the value to be set in divide field of IISMOD.
[Confirmed by Jassi that no existing machine drivers are affected --
broonie]
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order for the RATE and FMT defines to be reuseable in future by the
i2sv4 driver, move the MACROs out to the header file.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having the multiple definitions of the same clocks,
define them in one common place and refer by SoC specific names.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No point in duplicating this structure layout in each driver.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On TI DM6467 EVM, S/PDIF DIT codec fails to open as it is unable to install
hardware params. This dummy codec has no set_fmt and set_sysclk implementations
and calls from the application to these functions cause errors. This patch adds
a new hardware params callback function for S/PDIF transciever codec.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Shared memory mappings on nommu machines require a get_unmapped_area
file operation that suggests an address for the mapping. The current
implementation returns 0 and thus forces the driver to implement an
mmap handler that fixes up the start and end address of the vma.
This patch returns the address of the dma buffer, so it should work
out of the box for all drivers that use the snd_pcm_runtime->dma_area
pointer.
Addresses for mapping the status and control pages are returned as
well, but to make those work the conditional compilation of
snd_pcm_mmap_{status,control} would need to be revised.
URL: http://thread.gmane.org/gmane.linux.alsa.devel/61230
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
without the following patch audio ssttuutteerrs on
ASUS M2N32-SLI PREMIUM ACPI BIOS Revision 1304
the sound device is:
00:0e.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2)
worked with 2.6.32
Signed-off-by: Ralf Gerbig <rge@quengel.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix 'else' placement in ifdef block so that build succeeds:
sound/isa/opti9xx/opti92x-ad1848.c:221: error: 'else' without a previous 'if'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (26 commits)
ALSA: hdmi - show debug message on changing audio infoframe
ALSA: hdmi - merge common code for intelhdmi and nvhdmi
ALSA: hda - Add ASRock mobo to MSI blacklist
ALSA: hda: uninitialized variable fix
ALSA: hda: Use LPIB for a Biostar Microtech board
ALSA: usb/audio.h: Fix field order
ALSA: fix jazz16 compile (udelay)
ALSA: hda: Use LPIB for Dell Latitude 131L
ALSA: hda - Build hda_eld into snd-hda-codec module
ALSA: hda - Support NVIDIA MCP89 and GT21x hdmi audio
ALSA: hda - Support max codecs to 8 for nvidia hda controller
ALSA: riptide: clean up while loop
ALSA: usbaudio - remove debug "SAMPLE BYTES" printk line
ALSA: timer - pass real event in snd_timer_notify1() to instance callback
ALSA: oxygen: change || to &&
ALSA: opti92x: use PnP data to select Master Control port
ASoC: fix ak4104 register array access
ASoC: soc_pcm_open: Add missing bailout tag
ALSA: usbaudio: Fix wrong bitrate for Creative Creative VF0470 Live Cam
ALSA: ua101: removing debugging code
...
Since alc_auto_create_input_ctls() doesn't set the elements for the
secondary ADCs, "Input Source" elemtns for these also get empty, resulting
in buggy outputs of alsactl like:
control.14 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
iface MIXER
name 'Input Source'
index 1
value 0
}
This patch fixes alc_mux_enum_*() (and others) to fall back to the
first entry if the secondary input mux is empty.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Fix a wrong value passed to snd_hda_check_board_codec_sid_config() as
the upper-limit in parse_alc268(), so that any wrong value can't be
passed.
So far, no bogus value was set in the quirk entries, so this won't give
any behavioral changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create patch_hdmi.c to hold common code from intelhdmi and nvhdmi.
For now the patch_hdmi.c file is simply included by patch_intelhdmi.c
and patch_nvhdmi.c, and does not represent a real codec.
There are no behavior changes to intelhdmi. However nvhdmi made several
changes when copying code out of intelhdmi, which are all reverted in
this patch. Wei Ni confirmed that the reverted code actually works fine.
Tested-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* git://git.kernel.org/pub/scm/linux/kernel/git/lethal/sh-2.6: (26 commits)
sh: Convert sh to use read/update_persistent_clock
sh: Move PMB debugfs entry initialization to later stage
sh: Fix up flush_cache_vmap() on SMP.
sh: fix up MMU reset with variable PMB mapping sizes.
sh: establish PMB mappings for NUMA nodes.
sh: check for existing mappings for bolted PMB entries.
sh: fixed virt/phys mapping helpers for PMB.
sh: make pmb iomapping configurable.
sh: reworked dynamic PMB mapping.
sh: Fix up cpumask_of_pcibus() for the NUMA build.
serial: sh-sci: Tidy up build warnings.
sh: Fix up ctrl_read/write stragglers in migor setup.
serial: sh-sci: Add DMA support.
dmaengine: shdma: extend .device_terminate_all() to record partial transfer
sh: merge sh7722 and sh7724 DMA register definitions
sh: activate runtime PM for dmaengine on sh7722 and sh7724
dmaengine: shdma: add runtime PM support.
dmaengine: shdma: separate DMA headers.
dmaengine: shdma: convert to platform device resources
dmaengine: shdma: fix DMA error handling.
...
The headphone detect and charger are using the IRQ numbers so need
to take account of irq_base with the genirq conversion. I obviously
picked the wrong system for initial testing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
Rename for_each_bit to for_each_set_bit in the kernel source tree. To
permit for_each_clear_bit(), should that ever be added.
The patch includes a macro to map the old for_each_bit() onto the new
for_each_set_bit(). This is a (very) temporary thing to ease the migration.
[akpm@linux-foundation.org: add temporary for_each_bit()]
Suggested-by: Alexey Dobriyan <adobriyan@gmail.com>
Suggested-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Russell King <rmk@arm.linux.org.uk>
Cc: David Woodhouse <dwmw2@infradead.org>
Cc: Artem Bityutskiy <dedekind@infradead.org>
Cc: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Currently during pop/click debug we're inserting a delay both after
every log message we generate and at explicit points in the sequence,
slowing things down even further than they need to be especially when
many writes get coalesced by the sequence generation code.
Remove the per-printk delay and ensure that we have explicit delays
where we say we want them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Commit eaa9b3a748 introduced the following
uninitialized warning:
sound/pci/hda/patch_realtek.c: In function 'set_capture_mixer':
sound/pci/hda/patch_realtek.c:4928: warning: 'pin' is used uninitialized in this function
sound/pci/hda/patch_realtek.c:4918: note: 'pin' was declared here
It appears indeed that 'pin' needs to be initialized to 0.
Signed-off-by: Frederik Deweerdt <frederik.deweerdt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/523953
The OR has verified that position_fix=1 is necessary to work around
errors on his machine.
Reported-by: MMarking
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sample rate setting is done with a 4-byte long class request that
addresses the interface.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the parser to correctly handle v2 descriptors with multiple
format bits set.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for USB audio 2.0 support, change the audioformat
structure so that it uses a bitmask to specify possible formats.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_usb_substream::format field actually contains the index of the
current alternate setting, so rename it to altset_idx to avoid
confusion.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the usb audio driver by factoring out a lot of functions to
separate files. Code for procfs, quirks, urbs, format parsers etc all
got a new home now.
Moved almost all special quirk handling to quirks.c and introduced new
generic functions to handle them, so the exceptions do not pollute the
whole driver.
Renamed usbaudio.c to card.c because this is what it actually does now.
Renamed usbmidi.c to midi.c for namespace clarity.
Removed more things from usbaudio.h.
The non-standard drivers were adopted accordingly.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename snd-usb-lib to snd-usbmidi-lib as MIDI functions are the only
thing it actually contains. Introduce a new header file to only declare
these functions.
Introduced usbmixer.h for all functions exported by usbmixer.c.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As part of the USB audio code cleanup, move the non-standard ua101
driver out of the way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While trying to compile jazz16 isa sound driver on alpha (2.6.33+git), I
found a compile failure in jazz16.c (udelay is unknown). Fix it by
including delay.h.
Signed-foo-by: Meelis Roos <mroos@linux.ee>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reorgs of the Samsung headers have moved the GPIO bank definitions
from plat/ to mach/ - the IIS driver needs to be updated to take care
of this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
BugLink: https://launchpad.net/bugs/530346
The OR has verified that position_fix=1 is necessary to work around
errors on his machine.
Reported-by: Tom Louwrier
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now two modules require hda_eld.o, so we need to put it to the common
place instead of building into two individual modules.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Report the current FIFO depth when delay is queried. The FIFO is only
16 frames deep so the latency will be at most a couple of miliseconds
(and we tend to end up reporting zero most of the time) but it may
help some applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Support nvidia MCP89 and GT21x 8ch hdmi audio.
Add some eld support.
Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support max codecs to 8 for nvidia hda controller.
Change AZX_MAX_CODECS to 8, and add
"#define AZX_DEFAULT_CODECS 4" for default driver.
Set azx_max_codecs to 8 for nvidia controller.
Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If getpaths() returned an odd number this would be a buffer under-run and an
endless loop. It turns out that getpaths() can only return even numbers, but
let's make it easy for people auditing code. With the new code you don't
need to look at getpaths().
This silences a smatch warning.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the original code the condition was always true (hopefully) because
WM8776_HPLVOL is zero.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Master Control port (MC) is available as the last
PnP resource (OPT005). Use this value instead fo guessing.
Also, add some comments to the code.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't touch the variable 'reg' to construct the value for the actual SPI
transport. This variable is again used to access the driver's register
cache, and so random memory is overwritten.
Compute the value in-place instead.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The codec_dai needs to be shutdown should the machine startup fails.
This patch adds another bailout tag for that case and rename the tag
for configuration failures.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8960 headphone outputs can be run in capless mode with OUT3
used to drive a pseudo ground for the headphone drivers. In this
mode the mono mixer is not used, the mixer should be turned on
in concert with the headphone output drivers and the device bias
levels are managed differently.
Also tweak the existing bias management to remove the use of active
discharge while we're at it since that's often audible.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Avoids machine files having to peer into sound/soc which is a bit
rude and icky.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The driver name gets used by dev_() logging so use something a bit
more idiomatic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The delay callback can be used by the core to query the delay
on the dai caused by FIFO or delay in the platform side.
In case if both CPU and CODEC dai has FIFO the delay reported
by each will be added to form the full delay on the chain.
If none of the dai has FIFO, than the delay will be kept as
zero.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Create a soc level wrapper for pcm_pointer callback.
This will facilitate the soc level handling of different
HW buffers in the audio path.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My editor removes the tailing spaces, which causes problems when
changing the soc-core.c
Removing the space.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Initial support for audio using the 1133-EV1 audio and PMIC module for
the i.MX31ADS. Currently only playback is supported, and the FIQ DMA
driver has performance problems at higher sample rates which cause
audible dropouts.
This driver is based heavily on an out of tree one written by Liam
Girdwood.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch works around misbehaviour of Creative Creative VF0470 Live Cam
which reports 16 kHz sample rate for audio capture while actually producing
8 kHz stream.
Signed-off-by: Arseniy Lartsev <arseniy@fizlesh.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some code that is no longer needed now that the relevant parts of
the driver have been tested.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
sound/usb/caiaq/midi.h:6: ERROR: "foo* bar" should be "foo *bar"
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/oss/v_midi.h:5: ERROR: code indent should use tabs where possible
sound/oss/v_midi.h:7: ERROR: trailing whitespace
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In 2.6.33 ACL269 unsol event handler was changed to look up the pre-defined
pins, but the headphone pins aren't defined properly in each quirk.
This patch adds the missing definitions, and fixes the speaker auto-mute
regression on some ASUS (and possibly other) laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
i8253_lock needs to be a real spinlock in preempt-rt, i.e. it can
not be converted to a sleeping lock.
Convert it to raw_spinlock and fix up all users.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Dmitry Torokhov <dmitry.torokhov@gmail.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Cc: Jens Axboe <jens.axboe@oracle.com>
LKML-Reference: <20100217163751.030764372@linutronix.de>
Separate SH DMA headers into ones, commonly used by both drivers, and ones,
specific to each of them. This will make the future development of the
dmaengine driver easier.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Now most (if not all) PXA platforms have been switched to the new MFP
API, it's rather safe to remove these unnecessary pxa_gpio_mode() calls
in pxa2xx-ac97-lib.c now.
Cc: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
This is really pxa27x specific and should be kept in pxa27x.c. With this
newly introduced function, the original set_resetgpio_mode() is deprecated.
Cc: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
MFP registers are saved and restored by the mfp sys_device before all
other platform devices, and it is unnecessary here.
Cc: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.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: (252 commits)
ASoC: Check progress when reporting periods from i.MX FIQ handler
ASoC: Remove a unused variables from i.MX FIQ runtime data
ALSA: hda - Add/fix ALC269 FSC and Quanta models
ALSA: hda - Add ALC670 codec support
OMAP4: PMIC: Add support for twl6030 codec
ALSA: hda - remove unnecessary msleep on power state transitions
usb/gadget/{f_audio,gmidi}.c: follow recent changes in audio.h
ASoC: fsi: Modify over/under run error settlement
ASoC: OMAP4: Add McPDM platform driver
ASoC: OMAP4: Add support for McPDM
ASoC: OMAP: data_type and sync_mode configurable in audio dma
ALSA: hda - Add missing description in HD-Audio-Models.txt
ALSA: add support for Macbook Air 2,1 internal speaker
ALSA: usbaudio: consolidate header files
ALSA: usbmixer: bail out early when parsing audio class v2 descriptors
ALSA: usbaudio: implement basic set of class v2.0 parser
ALSA: usbaudio: introduce new types for audio class v2
ALSA: usbaudio: parse USB descriptors with structs
ALSA: hda - enable snoop for Intel Cougar Point
ALSA: hda - Remove identical definitions for macmini3 model
...
Add support for the Edirol UA-1000 to the UA-101 driver.
Both devices behave the same, so we just have to shuffle around some
interface numbers and name strings.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Machine driver for DB1200 AC97 and I2S audio systems, intended as a proper
reference asoc machine for Alchemy-based systems. AC97/I2S can be selected
at boot time by setting switch S6.7.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Cc: Linux-MIPS <linux-mips@linux-mips.org>
Cc: alsa-devel@alsa-project.org
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
DMA can only be done from physical addresses; move the "virt_to_phys"
source/destination buffer address translation from the dbdma queueing
functions (since the hardware can only DMA to/from physical addresses)
to their respective users.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
Remove dbdma compat macros, move remaining users over to default
queueing functions and -flags.
(Queueing function signature has changed in order to give
a build failure instead of silent functional changes due
to the no longer implicitly specified DDMA_FLAGS_IE flag)
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two
or more dai_links we need to log the number of active users of the dai.
For that, we change semantics of the snd_soc_dai.active flag from indicator
to reference counter.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'next-devicetree' of git://git.secretlab.ca/git/linux-2.6: (41 commits)
of: remove undefined request_OF_resource & release_OF_resource
of/sparc: Remove sparc-local declaration of allnodes and devtree_lock
of: move definition of of_chosen into common code.
of: remove unused extern reference to devtree_lock
of: put default string compare and #a/s-cell values into common header
of/flattree: Don't assume HAVE_LMB
of: protect linux/of.h with CONFIG_OF
proc_devtree: fix THIS_MODULE without module.h
of: Remove old and misplaced function declarations
of/flattree: Make the kernel accept ePAPR style phandle information
of/flattree: endian-convert members of boot_param_header
of: assume big-endian properties, adding conversions where necessary
of: use __be32 for cell value accessors
of/flattree: use OF_ROOT_NODE_{SIZE,ADDR}_CELLS DEFAULT for fdt parsing
of/flattree: use callback to setup initrd from /chosen
proc_devtree: include linux/of.h
of: make set_node_proc_entry private to proc_devtree.c
of: include linux/proc_fs.h
of/flattree: merge early_init_dt_scan_memory() common code
of: add 'of_' prefix to machine_is_compatible()
...
Currently the i.MX FIQ handler is reporting periods as elapsed based
purely on a timer running in the CPU. This means that any clock
mismatch between the CPU and the audio subsystem can result in the
status reported to applications drifting away from the actual status
of the hardware. This is particularly likely at present since the
SSI driver is only capable of operating in slave mode so it's very
likely that the interface will be clocked from a different source.
Instead check the offset reported by the FIQ and only notify when we
have transferred at least one period, re-firing the timer if we didn't
do so. Also factor out the calculation of the timer expiry time for
make it a bit easier to experiment with.
Note that this only improves the situation, problems can still be
triggered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Specify proper quirk models for FSC and Quanta machines with ALC269 codec.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Fixed alc_subsystem_id( ) typo and add new function.
- !(ass & 0x100000)) ==> Delete this check. It is unnecessary check.
- Add porti
- ALC670 support
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will save ~15ms boot time.
The first 10ms sleep was introduced in commit d2595d86e5 for (buggy)
Cxt codecs, so better to limit the sleep to the problem hardware.
For the second 10ms sleep, the HDA spec says:
Power State[1:0]:
00: Node Power state (D0) is fully on.
01: Node Power state (D1) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms, excepting analog pass through circuits (e.g., CD analog
playback) which must remain fully on.
10: Node Power state (D2) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms. For modems, this is the "wake on ring" power state.
11: Node Power state (D3) allows for (does not require) lowest possible power consuming state under software
control. Note that any low power state set by software must retain sufficient operational capability to properly
respond to subsequent software Power State command.
So 10ms is actually the max wait time. It should be safe to
remove/reduce it and rely on the loop of 1ms-sleeps.
CC: Marc Boucher <marc@linuxant.com>
CC: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Zhang Rui <rui.zhang@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add ASoC interface for OMAP McBSP2 and McBSP3 sidetones.
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Tony Lindgren <tony@atomide.com>
In current FSI driver, playback function cares only overrun,
and capture function cares only underrun.
But playback function should had cared about underrun,
and capture function should had cared about overrun too.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McPDM platform driver is configured to use sDMA in order to transfer
to/from memory. Support for interfacing with ABE will be added later.
McPDM dai currently supports up to 4 downlink channels and 2 uplink
channels simultaneously, as well as 88.2 and 96 KHz, and a sample
size of 32 bits.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McPDM is the interface between Phoenix audio codec
and the OMAP4430 processor. It enables data to be transfered
to/from Phoenix at sample rates of 88.4 or 96 KHz.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow client drivers to set the data_type (16, 32) and the
sync_mode (element, packet, etc) of the audio dma transferences.
McBSP dai driver configures it for a data type of 16 bits and
element sync mode.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for Macbook Air 2,1 (late 2008) internal speaker and
headphones. Create a "mba21" model for snd-hda-intel.
Signed-off-by: Reimundo Heluani <rheluani@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the definitions from linux/usb/audio.h all over the ALSA USB audio
driver and add some missing definitions there as well.
Use the endpoint attribute macros from linux/usb/ch9 and remove the own
things from sound/usb/usbaudio.h.
Now things are also nicely prefixed which makes understanding the code
easier.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is just a quick hack that needs to be removed once the new units
defined by the audio class v2.0 standard are supported.
However, it allows using these devices for now, without mixer support.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a number of parsers for audio class v2.0. In particular, the
following internals are different and now handled by the code:
* the number of streaming interfaces is now reported by an interface
association descriptor. The old approach using a proprietary
descriptor is deprecated.
* The number of channels per interface is now stored in the AS_GENERAL
descriptor (used to be part of the FORMAT_TYPE descriptor).
* The list of supported sample rates is no longer stored in a variable
length appendix of the format_type descriptor but is retrieved from
the device using a class specific GET_RANGE command.
* Supported sample formats are now reported as 32bit bitmap rather than
a fixed value. For now, this is worked around by choosing just one of
them.
* A devices needs to have at least one CLOCK_SOURCE descriptor which
denotes a clockID that is needed im the class request command.
* Many descriptors (format_type, ...) have changed their layout. Handle
this by casting the descriptors to the appropriate structs.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some definitions for audio class v2.
Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have
different numerical representations in both standards, so there is need
for a _V1 add-on now. usbmixer.c is changed accordingly.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation of support for v2.0 audio class, use the structs from
linux/usb/audio.h and add some new ones to describe the fields that are
actually parsed by the descriptor decoders.
Also, factor out code from usb_create_streams(). This makes it easier to
adopt the new iteration logic needed for v2.0.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables snoop, eliminating static during playback.
This patch supersedes the previous Cougar Point audio patch.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the card->codec on soc_resume to detect if the soc
device is properly initialized.
If the card->codec is NULL, than do not continue the resume
operation, since the device is not initialized properly.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order for having snd_soc_dais shared among two or more dai_links,
remove the relatively global runtime field from the struct snd_soc_dai
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently only the atmel driver make use of snd_soc_dai.runtime field.
If the dais are to be shared among two or more dai_links, the field
must be got rid of.
So, in atmel driver reach the substream via dai_link->pcm so as to
not depend of snd_soc_dai.runtime field.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Passing pointer to relevant dai_link provides easier reach to the
ASoC tree in suspend/resume of snd_soc_platform. It also provides
direct access to the dai at the other end of the dai_link.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
