The analog mix is disabled now as default (unless "analog_mixer" hint
is given), so it shoudn't appear in the digital input source as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the missing initialization of DMUX connection (to analog input)
for auto-mic mode with STAC/IDT codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We don't need any more static connection to the port F (which is often
used for docking stations) since its connection is done dynamically via
DAC assignment now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
mpu_synth_info[m].name is a char[30], and the minimum length of the data
written by sprintf is 31 bytes including terminating null.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes
including terminating null.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
e->sad[] is declared with size ELD_MAX_SAD=16, but the guard
allows range 0-31.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support the automatic mic-switching with some devices with IDT/STAC
codecs. The condition is that the device has only two inputs, one
for an external mic and one for an internal mic.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since only one event can be associated to a (pin) widget, it's safer
to avoid the multiple mapping. This patch fixes the behavior of the
STAC/IDT codec driver.
Now stac_get_event() doesn't take the type argument but simply returns
the first hit element. Then enable_pin_detect() checks the validity
of the type, and returns non-zero only if a valid entry. The caller
can call stac_issue_unsol_event() after checking the return value.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The I2S DAI driver for blackfin SPORT, but works in TDM mode.
I2S is not a special case of TDM with only left and right two slots for
SPORT interface. I2S coordinates with TDM in SPORT, but not a part of
TDM. TDM require different hardware configuration with I2S, not only
different slot number. One is "Stereo Serial Operation" mode of SPORT,
the other one is "Multichannel Operation" mode. They are incompatible
at the same time.
Hardware and DMA description and data transfer flow are much different
for I2S and TDM. Merging them as a whole will be very ugly and difficult
to maintain.
So we don't define a new DAI type, but give two DAI instances for standard
I2S and TDM, both in I2S-family DAI type. The TDM instance still uses the
I2S-family DAI type.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch fixes some checkpatch identified issues and adds a comment about
line discipline interaction to my driver code, as requested by Mark on my
inital submission (thank you Mark for applying my imperfect patch anyway).
It also fixes MODULE_ALIAS mismatch as used in my machine driver.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The analog mixer unit on IDT 92HD71Bxx codecs is almost useless
since we use only the direct connections from DAC to pin.
Remove the controls to avoid unneeded confusion as default now.
This can be still back via "analog_mixer = 1" hint.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of static snd_kcontrol_new arrays, create "Capture Volume"
and "Capture Switch" controls dynamically based on the mixer attr
values (made via HDA_COMPOSE_AMP_VAL()).
This reduces the code size and gives more flexibility to change
the number of controls later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current driver creates always the digital input source mixer
elements for IDT 92HD71x codecs no matter whether digital mics are
present. This patch adds the proper check to avoid the creation of
these controls if unnecessary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The sentense "Unknown model for xxx, ..." makes people too nervous
and drives them to a direction to a wrong "fix" by giving any
mismatching model option.
Let's rephrase the messages to be more nice and easy (at least that
won't make people suspect conspiracies).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Volume-knob widgets may have connections even if they have no CONN_LIST
cap bit. Allow the query exceptionally in snd_hda_get_connections().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* fix/hda:
ALSA: hda - Fix mute control with some ALC262 models
ALSA: hda - Restore GPIO1 properly at resume with AD1984A
ALSA: hda - Use snprintf() to be safer
The master mute switch is wrongly implemented as checking the pointer
instead of its value, thus it can be never muted. This patch fixes
the issue.
Reference: Novell bnc#404873
https://bugzilla.novell.com/show_bug.cgi?id=404873
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
This patch removes the old method of jack detection from palm27x-asoc
driver and adds jack detection api. It also removes some other (now)
useless stuff from the driver and corrects pin configuration for the
codec.
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch adds a few small enhancements to the ASoC jack handling, as
suggested by Mark in his comments to my Amstrad Delta driver, and a few fixes
for related bugs found while learning Mark's code and testing results.
Enhancements:
1. Update status of an ASoC jack while associating it with new gpios.
2. Really update DAPM pins while associating them with an ASoC jack.
3. Export ASoC jack gpios over gpiolib sysfs for diagnostic purposes.
Fixes:
1. Apply mask on jack status report before using it, just for case.
2. While updating jack associated DAPM pins, use full resulting jack status,
not the status report passed as an argument.
Created and tested on linux-2.6.31-rc3
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for Native Instrument's freshly announced Audio2DJ
sound device hardware. Version number bumped to 1.3.19.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix 79452f0a28 introduced another
bug due to the missing offset for the overlapped hwptr.
When the hwptr goes back to zero, the delta value has to be corrected
with the buffer size. Otherwise this causes looping sounds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch allows passing platform_data to devices attached to AC97 bus
(like touchscreens, battery measurement chips ...).
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MAX9877 needs an address of start register when we write values to
registers through i2c_master_send(), but the code for this was missed in
max9877_write_regs().
If the value of control is 0 in the max9877_set_out_mode(), the value is
not increased to 1, but actually the value to write to the register
should be 1.
And the register bits for out_mode and osc_mode should be cleared before
writing.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds support for Conexant CX20442-11 voice modem codec, suitable
for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Related
sound card driver will follow.
This codec is an optional part of the Conexant SmartV three chip modem design.
As such, documentation for its proprietary digital audio interface is not
available. However, on Amstrad Delta board, thanks to Mark Underwood who
created an initial, omap-alsa based sound driver a few years ago[1], the codec
has been discovered to be accessible not only from the modem side, but also
over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any sound
card that can access the codec DAI directly. The DAI configuration parameters
(sample rate and format, number of channels) has been selected out empirically
for best user experience.
The codec analogue interface consists of two pairs of analogue I/O pins:
speakerphone interface or telephone handset/headset interface. Furthermore, it
seams to provide two operation modes for speakerphone I/O: standard and
advanced, with automatic gain control and echo cancelation. Even if the codec
control interface is unknown and not available, all those interfaces and modes
can be selected over the modem chip using V.253 commands. The driver is able
to issue necessary commands over a suitable hw_write function if provided by a
sound card driver. Otherwise, the codec can be controlled over the modem from
userspace while inactive.
Even if nothig is known about the codec internal power management
capabilities, DAPM widgets has been used to model the codec audio map.
Automatically performed powering up/down of those virtual widgets results in
corresponding V.253 commands being issued.
Some driver features/oddities may be board specific, but I have no way to
verify that with any board other than Amstrad Delta.
[1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.html
Created and tested against linux-2.6.31-rc3.
Applies and works with linux-omap-2.6 commit
7c5cb7862d32cb344be7831d466535d5255e35ac as well.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Here are the new sound enabling patches for IbexPeak.
Summary of tested features:
- playback
- Front Headphone: OK
- 8 channel audio: Front/Rear/CLFE/Side all OK
- recording
- Front Mic/Rear Mic: both OK
(front/rear/line mics are selectable in the "Input source" alsamixer control)
- Line In: not working
(in 6ch mode, its amp/mute, direction and route all looks fine,
so I'm a little puzzled)
(hopefully no one will care this feature)
- digital SPDIF input/output: not tested (no equipment)
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the logging functionality to xrun_debug to record the hwptr
updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt(),
corresponding to 16 and 8, respectively.
For example,
# echo 9 > /proc/asound/card0/pcm0p/xrun_debug
will record the position and other parameters at each period interrupt
together with the normal XRUN debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PLL was not being enabled when it was not bypassed. This patch
enables the PLL when it is used. Additionally, it disables the PLL
when it is bypassed.
Without this patch, the audio on TI DM646x EVM and DM355 EVM
does not work properly. The bit clocks and the frame sync signals
from the codec are not correct and hence the playback/record are faster
than usual for most sample rates. The reason for this was that the PLL
was not enabled when it was not bypassed.
Tested on DM6467 EVM, playback tested on DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit 099db17e66 introduced a
regression at suspend/resume where the GPIO1 bit isn't properly
restored, thus the speaker output gets muted initially after resume.
The fix is simple, use the cached write for storing GPIO data.
Reference: Novell bnc#522764
https://bugzilla.novell.com/show_bug.cgi?id=522764
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a few uninitialized error checks that were introduced recently
mistakenlly during the clean-up:
sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’:
sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’:
sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’:
sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a check to snd_hda_get_connections() routine for
presence of AC_WCAP_CONN_LIST. Also, make sure that negative error
codes from noted route are handled on all places as errors.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- E3500 report cval->max more than it actually can handel, so if you
set 95% capture level it will be silently muted.
- Betwen cval->min and cval-max(real) is 2940 control units,
but real are only 7 with cval->res = 384.
- Alsa can't handel less than 10 controls, so make it more
and set cval->res = 192.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Previous patch used widget type, but the presence flag of the connection
list is in the widget capabilities.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reading node connections for an unknown widget can confuse HDA codec bus.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the logic of ALC861 auto-mode parser for the outputs.
Instead of assuming the fixed DAC list, parse the conection and assign
the DAC dynamically.
Also, unmute the unused output connections to avoid noises on inputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VMware tends to report PCM positions and period updates at utterly
wrong timing. This screws up the recent PCM core code that tries
to correct the position based on the irq timing.
Now, when a backward irq position is detected, skip the update
instead of rebasing. (This is almost the old behavior before
2.6.30.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add some tricks to reduce the click noise at powering down to D3
in the power saving mode on STAC/IDT codecs.
The key seems to be to reset PINs before the power-down, and some
delay before entering D3. The needed delay is significantly long,
but I don't know why.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The callback function to control register was used by whole controls in
MAX9877 driver, but this causes using many if statement for double
register control or invert.
So, the callback function for double register control is separate
differently, and the code for invert is added in the callback function.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This corrects a bug with ADC Inversion Switch in wm8974 codec.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reset was failing with the original udelay(50) between the code in
psc_ac97_cold_reset() and the call to psc_ac97_warm_reset(). Through testing
it was found that a delay of 1ms was necessary for the cold_reset code to
consistently complete successfully.
Signed-off-by: John Bonesio <bones@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* fix/misc:
ALSA: ca0106 - Fix the max capture buffer size
ALSA: OSS sequencer should be initialized after snd_seq_system_client_init
ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
* fix/hda:
ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs
ALSA: hda - Add quirk for Gateway T6834c laptop
ALSA: hda_codec: Check for invalid zero connections
On Soundblaster X-FI Titanium with emu20k2 the SIDE and SURROUND
channels were swapped and wrong.
I double checked it with connector colors and creative soundblaster
windows drivers.
So I swapped them to the true order.
Now "speaker-test -c6" and "speaker-test -c8" are working fine.
Signed-off-by: Frank Roth <frashman@freenet.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The capture buffer size with 64kB seems broken with CA0106.
At least, either the update timing or the DMA position is wrong,
and this screws up pulseaudio badly.
This patch restricts the max buffer size less than that to make life
a bit easier.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
The recent rewrite of the codec parser for STAC9872 caused a regression
for some Sony VAIO models that don't give proper pin default configs
by BIOS. Even using model=vaio doesn't work because the pin definitions
are set after the pin overrides.
This patch fixes the pin definitions in patch_stac9872() to be put
in the right place before the pin overrides. Also the patch adds the
new quirk entry for VAIO F/S to have the correct pin default configs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Added the native timer support for emu20k2, which gives much more
accurate update timing than the system timer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Gateway T6834c laptops need EAPD always on while the default behavior
for the STAC9205 reference board is to turn it off upon every HP plug.
By using the special "eapd" model, which is first introduced for Gateway
T1616 laptops for this same reason, this peculiarity can be properly
handled.
Signed-off-by: Hao Song <baritono.tux@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If spin_lock_irqsave is called twice in a row with the same second
argument, the interrupt state at the point of the second call overwrites
the value saved by the first call. Indeed, the second call does not need
to save the interrupt state, so it is changed to a simple spin_lock.
The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@@
expression lock1,lock2;
expression flags;
@@
*spin_lock_irqsave(lock1,flags)
... when != flags
*spin_lock_irqsave(lock2,flags)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To prevent "Too many connections" message and the error path for some HDMI
codecs (which makes onboard audio unusable), check for invalid zero
connections for CONNECT_LIST verb.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
GCC 4.4.0 doesn't appear to be able to spot that we don't apply any FLL
configuration if the output frequency is zero.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Parse the mono output pin 0x16 correctly even as the primary output
- Create "Speaker" volume control if the primary output is a speaker
- Fix the wrong direction of (optional) "Mono" switch
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The newly added sanity-check for a codec verb can be better written
with logical ORs. Also, the parameter can be more than 8bit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6
line, controlled by register INTBR_PMBR1. Machine driver takes care
of enabling gpio line through i2c and codec driver manipulates the
line during headset ramp up/down sequence.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A recent bug involves passing auto detected >0x7f NID to codec command,
creating an invalid codec addr field, and finally lead to cmd timeout
and fall back into single command mode. Jaroslav fixed that bug in
alc880_parse_auto_config().
It would be safer to further check the bounds of all cmd fields.
Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for new AMD HD audio devices. Use generic driver to detect HD audio
devices with Vendor ID AMD.
Signed-off-by: Andiry Xu <andiry.xu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds support for the Conexant CX20582 codec, based on code from
http://www.linuxant.com/alsa-driver/alsa-driver-linuxant-1.0.19ppch12-1.noarch.rpm.zip
This is the codec to be shipped in the OLPC XO-1.5, so this patch also
includes an XO-specific profile. Resultant configuration:
http://dev.laptop.org/~dsd/20090713/codec0.txthttp://dev.laptop.org/~dsd/20090713/codec0.svg
As the Linuxant code is structured differently to the other codecs,
I was unable to cleanly reimplement everything in the generic and Dell
profiles as more info is needed (e.g. codec graphs). I simplified those
profiles so that hopefully it will not break anyone's audio. If it does,
it may be worth returning -ENODEV from patch_cx5066 on non-OLPC systems,
and then fixing snd_hda_codec_configure() to fall back on the generic
parser, at least until support for other systems is figured out.
Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
clock name strings are no longer passed on platform_data. Instead,
we rely entirely on struct device and clkdev to find the right clock.
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The MAX9877 combines a high-efficiency Class D audio power amplifier
with a stereo Class AB capacitor-less DirectDrive headphone amplifier.
The max9877_add_controls() is called to register the MAX9877 specific
controls on machine specific init() of the machine driver.
The datasheet for the MAX9877 can find at the following url:
http://datasheets.maxim-ic.com/en/ds/MAX9877.pdf
[Slight edit to sort the ALL_CODECS entries -- broonie.]
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We need to check returning error for pci_register_driver(&joystick_driver)
On failure, we should unregister formerly registered audio drivers
This also fixed the compiler warning :
CC [M] sound/pci/riptide/riptide.o
sound/pci/riptide/riptide.c: In function ‘alsa_card_riptide_init’:
sound/pci/riptide/riptide.c:2200: warning: ignoring return value of ‘__pci_register_driver’, declared with attribute warn_unused_result
Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the flexibility of the WM9081 FLL this should never happen
in a real system.
Reported-by: Jaswinder Singh Rajput <jaswinder@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- fix my previous codec activity breakage (_non-warned_ variable assignment
issue)
- convert suspend/resume to 32bit I/O access (I/O is painful; to improve
suspend/resume performance)
- change DEBUG_PLAY_REC to DEBUG_CODEC for consistency
- printk cleanup
- some logging improvements
- minor cleanup/improvements
The variable assignment issue above was a conditional assignment to the
call_function variable (this ended with the non-preinitialized variable
not getting assigned in some cases, thus a dangling stack value, yet gcc 4.3.3
unbelievably did _NOT_ warn about it in this case!!),
needed to change this into _always_ assigning the check result.
Practical result of this bug was that when shutting down
_either_ playback or capture, _both_ streams dropped dead :P
Tested, working (plus resume) and checkpatch.pl:ed on 2.6.30-rc5,
applies cleanly to 2.6.30 proper with my previous (committed)
patches applied.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sending an Active Sensing message when closing a port can interfere with
the following data if the port is reopened and a note-on is sent before
the device's timeout has elapsed. Therefore, it is better to disable
this setting by default.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of using magic numbers for the controlles sent when resetting
a port, use the symbols from asoundef.h.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sending a MIDI reset message when closing a port is wrong because we
only want to shut the device up, not to reset all settings.
Furthermore, many devices ignore this message.
Fortunately, the RawMIDI layer already shuts the device up, so we can
ignore this matter here.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When draining, instead of waiting for fifty milliseconds, just wait for
the currently active URBs to complete. This cuts the usual waiting time
down to one USB frame, or zero in the common case when there is no URB.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some newer USB MIDI interfaces use rather small packet sizes, so to get
enough bandwidth, we have to be able to send multiple packets in one USB
frame, so we have to use multiple URBs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some newer USB MIDI interfaces use rather small packet sizes, so to get
enough bandwidth, we have to be able to receive multiple packets in one
USB frame, so we have to use multiple URBs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turns out that the main cause of output buffer overruns is not slow
drivers but applications that generate too many messages. Therefore, it
makes more sense to make that error message always visible, and to
rate-limit it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Asus Xonar U1. This device is mostly class compliant, but
the digital output requires a vendor-specific request.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Blue Microphones USB devices have an alternate setting that sends two
channels of data to the computer. Unfortunately, the descriptors of
that altsetting have a wrong channel setting, which means that any
recorded data from such a device has twice the sample rate from what
would be expected.
This patch adds a workaround to ignore that altsetting. Since these
devices have only one actual channel, no data is lost.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We need to use the best value we picked, not the last value we
looked at.
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to manually start playback/capture ourselves as the PCM
driver will handle things for us.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no need to manually start playback/capture ourselves as the PCM
driver will handle things for us.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without MODULE_LICENCE("GPL"), when built as a module it will fail
to load because it uses other GPL symbols from kernel.
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In SOC DAPM layer of SOUND subsystem, when add signal route (in the
function snd_soc_dapm_add_route() ), the original code has wrong logic
when dapm layer check each widget whether an external one.
Signed-off-by: Rongrong Cao <rrcao@ambarella.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the last in-kernel direct usage of driver_data, replace it with
the proper dev_get/set_drvdata() calls.
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Jaroslav Kysela <perex@perex.cz>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
The codec read errors in snd_hda_get_connections() are ignored so far,
and it causes a problem like the bug in the commit
9d30937acc
ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checking
Better to check errors in the function and returns a proper error code
rather than passing bogus NID values.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some IbexPeak systems with ALC889A errors like "azx_get_response
timeout, switching to polling mode: last cmd=0xaf9f000b" are produced,
because non-existent codec #10 is wrongly accessed.
The problem is that snd_hda_get_connections() returns out-of-range result
for NID 0x1c (something like 0xf8f9 or 0xffff).
This patch adds a check to alc880_parse_auto_config() to avoid using
of this out-of-range NIDs. A better fix maybe to improve
snd_hda_get_connections() routine to check for valid NID ranges if
NIDs are expected as result.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the merge error at the commit 305355aad8,
an addition of the missing alc880_gpio3_init_verbs to ALC882_TARGA model.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* fix/asoc:
ASoC: Fix wm8753 register cache size and initialization
ASoC: add locking to mpc5200-psc-ac97 driver
ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared
ASoC: Fix register cache initialisation for WM8753
Simplify ALC882_TARGA and return gpio3 to ALC883_TARGA_DIG and
ALC883_TARGA_2ch_DIG, which I accidentally removed in commit id
64a8be7435
Signed-off-by: David Heidelberger <d.okias@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8731 driver has been updated to allow registration via normal
device model methods rather than from within the ASoC driver probe
so update the AT91SAM9G20-EK to make use of this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Andrew Victor <linux@maxim.org.za>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Check for rtd->params->drcmr != NULL before accessing it.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the beep tone calculation for IDT/STAC codecs, lower numbers correspond
to higher frequencies and vice versa. The current code has this backwards,
resulting in beep frequencies which are way too high (and sound bad on
tinny laptop speakers, resulting in complaints).
[Also added hz <= 0 check by tiwai]
Signed-off-by: Paul Vojta <vojta@math.berkeley.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide an interval after the end of DAPM sequencing so that we
can distinguish between a pop in the final step of the sequence
and a pop generated from some other source outside DAPM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The system clock is currently fixed by the driver and this avoids
the need for us to handle errors with enabling and disabling MCLK
(which was incorrect previously so this fixes bugs in error
handling).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 5fd29d6ccb ("printk: clean up
handling of log-levels and newlines") changed printk semantics. printk
lines with multiple KERN_<level> prefixes are no longer emitted as
before the patch.
<level> is now included in the output on each additional use.
Remove all uses of multiple KERN_<level>s in formats.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
As shipped the board does not have inputs but it is relatively
straightforward to modify the board to hook them up so support
is provided in the driver. When these modifications have not
been made enabling the microphone stage can cause problems.
Add an ifdef to disable this by default. Don't put it into
Kconfig since users will have to get their soldering irons
out to change things.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This simplifies the driver by removing the need to manually
configure dividers within the CPU and improve audio performance
by ensuring that the optimal phase relationships between the
clocks in the system are maintained.
Note that currently this means that for playback to work the
Output Mixer HiFi switch must be enabled since otherwise CODEC
will not generate the DAC clock.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While the hardware is capable of some limited asynmmetric modes the
driver does not currently support those modes so tell applications
that only symmetric rates are available.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 is a highly integrated ultra-low power hi-fi CODEC designed
for portable devices such as multimedia phones.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the dma link with correct data as soon as
the master channel has copied it. Otherwise, the
1st period will play twice.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a regression, introduced in aa202455ee
(in alsa-kernel) which I noticed when trying to use the headphone socket on
my EeeCPC 901: the output was *very* quiet, practically silent.
This patch corrects the control types to that which was obviously intended in
the referenced commit.
Signed-off-by: Darren Salt <linux@youmustbejoking.demon.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using SG-buffers with dma_alloc_coherent() is often very inefficient
on non-coherent architectures because a tracking record could be
allocated in addition for each dma_alloc_coherent() call.
Instead, simply disable SG-buffers but just allocate normal continuous
buffers on non-supported (currently all but x86) architectures.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HDA driver disabled HD audio 64bit address support for all AMD
SB600/SB700/SB800 platforms with commit
09240cf429 due to one SB600 issue
reported by community, but we do not see the similar issue on
SB700/SB800 platforms.
This patch is to refine the workaround for SB600 only.
Signed-off-by: Andiry Xu <andiry.xu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the widget type and don't take invalid widgets while parsing
the capture source in patch_via.c.
Also, fixed some compile warnings introduced in the previous commit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding MPU-401 support to cmi8330 driver could cause a regression (non-working
sound) on a system where there is no free IRQ for the MPU-401 device (which
is not very uncommon as this card requires two separate IRQs plus a third one
for MPU-401).
When MPU-401 PnP configuration fails (mostly because of unavailable IRQ), just
ignore MPU-401 and continue without it.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fixed widget NIDs in patch_via.c seem wrong for some codecs,
and it resulted in the invalid capture source selection.
This patch adds the code to parse the topology instead of using
fixed numbers in order to get the right MUX widget id corresponding
to the ADCs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the output pin is used and EAPD capability is present, turn on
the EAPD bit. This fixes the silent output problem on ASUS laptops
with VT1708S codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My CMI8329 had OPL3 port specified in SB16 resources. But now I found out that
it was my modification of the card's PnP EEPROM a couple of years ago (can be
done using C9SETROM.EXE utility). I did it because the OPL3 port was
completely missing from PnP data. It seems to be hardwired to 0x388 on
CMI8329.
Find OPL3 port automatically by searching in WSS and SB16 resources. If not
found, assume that it's hardwired to 0x388.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The parser wasn't called in the proper order.
Split now the parser to be called in patch_cirrus(), and the rest
are just for building PCMs and controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- fully separate codec I/O port handling, enabling the use of a single
function each for all codecs (playback, capture, I2S out)
- add a new separate pcm for I2S out port (UNTESTED, no I2S DAC
available yet)
- switch gameport to low frequency while idle, to try to reduce noise/power
- improve snd_azf3328_codec_setdmaa() calculation
- minor variable type cleanup (u16, bool etc.)
- add some doc updates (help those lost Windows users, debug help, ...)
Note that due to the large cleanup aspect of the codec I/O change,
I was able to fit everything including all improvements into the
same binary size!! (a measly 10 bytes more or so)
This should now be the almost last patch to this driver
(minus some possible kernel clocksource patch and x86_64 fixes or so).
I just felt like taking a break from the usual stuff and wanted to
get this driver's structure finished, and it's rather clean now...
Tested, working and checkpatch.pl:ed on 2.6.30-rc5,
applies cleanly to 2.6.30 proper.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This driver is about as far from being experimental as it can ever get
for an undocumented card, thus create this patch (interestingly it was the only
EXPERIMENTAL remaining in the entire Kconfig file).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Note the slightly tricky cache usage in the volume update function due
to the requirement for a separate write for the VU bit.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The TLV320AIC3x driver is currently the only user of the CODEC hw_read
operation and is jumping through some hoops in order to do so. In order
to support future refactoring to make the hw_read operation more usable
unwrap the usage in this driver to avoid its use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Nothing uses it and the existing hw_read operation needs to be
refectored so it's easier to remove it rather than work with it.
Support can be re-added if the code requires volatile registers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A lot of CODECs share the same register data formats and therefore
replicate the code to manage access to and caching of the register
map. In order to reduce code duplication centralised versions of
this code will be introduced with drivers able to configure the use
of the common code by calling the new snd_soc_codec_set_cache_io()
API call during startup.
As an initial user the 7 bit address/9 bit data format used by many
Wolfson devices is supported for write only CODECs and the drivers
with straightforward register cache implementations are converted to
use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a volatile_register() operation to the CODEC structure providing a
standard operation to query if a register is volatile. This will be used
to factor out the register cache I/O operations for the CODECs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the codec is master then prepare should call
mcbsp_start, not trigger.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Code previously just "ors" in this field without clearing
first. Fix, by never reading this register.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only start sample generator if needed, and more
cleanup on davinci_mcbsp_start.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move variable declaration closer to use.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add davinci_mcbsp_dev as argument to davinci_mcbsp_start
and davinci_mcbsp_stop.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add toggle_clock function to complete i2s reset earlier.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No functional changes. Rename variable w to something
more meaningful. Remove code obfuscating macro MOD_REG_BIT.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add basic support for CMI8329 cards. Makes PCM and OPL3 work.
Does not break CMI8330 (tested).
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* fix/hda:
ALSA: hda - Add sanity check in PCM open callback
ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callback
ALSA: hda - Avoid invalid formats and rates with shared SPDIF
ALSA: hda - Improve ASUS eeePC 1000 mixer
ALSA: hda - Add GPIO1 control at muting with HP laptops
When FLOAT PCM format is available but together with other linear
PCM formats, don't override maxbps value. For FLOAT format, it's always
32, thus it can be better checked in snd_hda_calc_stream_format().
Otherwise the maxbps 32 might be used wrongly even if the linear PCM
doesn't support it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add some sanity checks of struct snd_pcm_hardware fields in the PCM
open callback of hda driver. This makes a bit easier to debug any PCM
setup errors in the codec side.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM rates bit field may have been changed by the codec open callback.
In that case, we need to reset rate_min and rate_max. So, simply call
snd_pcm_lib_hw_rates() again after the codec open callback.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check whether formats and rates don't result in zero due to the
restriction of SPDIF sharing. If any of them can be zero, disable
the SPDIF sharing mode instead. Otherwise it will lead to a PCM
configuration error.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Register cache space was not being allocated for the final register,
causing bugs when it was used. Allocate space for it.
Also ensure that the final register is displayed in sysfs.
[Commit message rewritten to document actual issue. -- broonie]
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The mixer elements created for ASUS eeePC 1000 with ALC269 aren't
standard but strange words like "LineOut". Rename the element names
to follow the standard one like "Headphone" and "Speaker".
Also, split the volumes to each so that the virtual master can control
them.
The alc269_fujitsu_mixer is removed because it's now identical with
the new eeepc mixer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP laptops with AD1984A codecs (at least mobile models) need to set
GPIO1 appropriately to indicate the mute state. The BIOS checks this
bit to judge whether the mute on or off is sent via F8 key.
Without changing this bit, the BIOS can be confused and may toggle
the mute wrongly.
Reference: Novell bnc#515266
https://bugzilla.novell.com/show_bug.cgi?id=515266
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add "set_tristate" callbacks for HiFi and Voice DAIs.
Machine drivers can enable and disable tristate for each
DAI with "snd_soc_dai_set_tristate" function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for EXTMUTE in Zoom2 machine driver. This is necessary
to further reduce pop noise problem. Signal EXTMUTE is connected to
signal GPIO 153 in Zoom2 board.
In addition, change ramp delay value to 3 (218/161/109 ms). With
previous ramp delay value, pop noise was louder. With a longer value
the beep tone can be observed.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AC97 bus register read/write hooks need to provide locking, but the
mpc5200-psc-ac97 driver does not. This patch adds a mutex around
the register access routines.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When doing register reads, it is possible for there to be a stale
data ready bit set which will cause subsequent reads to return
prematurely with incorrect data. This patch fixes the issues by
ensuring stale data is cleared before starting another transaction.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The wrong register cache variable was being used to provide the size for
the memcpy(), resulting in a copy of only a void * of data.
Reported-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Instead of expanding alc882_init_verbs to two elements via a macro,
manually expand to each entry. This makes clear that some have already
the full slot for init_verbs array (currently 5).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After merting patch_alc882() and patch_alc883(), the initialization of
mixer amp 0x0b was missing in alc882_base_init_verbs[].
This is usually no critical problem, but it can disable the power-saving
as the default state, so better to put to mute these channels.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to TRM, an external FET controlled by a 1.8V output signal
can be used to reduce the pop-noise heard when the audio amplifier is
switched on. It is suggested that GPIO6 of TWL4030 be used, but any
other gpio can be used instead. This is indicated in machine driver
with the following twl4030_setup_data members:
-hs_extmute. Set to 1 if board has support for EXTMUTE.
-set_hs_extmute. Set to a callback funcion to control an external gpio
line. Set to NULL if MUTE[GPIO6] pin is used.
Codec driver takes care of enabling and disabling this output during
the headset pop attenuation sequence.
Also add a delay to let VMID settle in ramp up sequence.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The sound device instance needs to be a child of the USB interface, not
the USB device. Newer udev versions pay attention to that.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Lennart Poettering <lennart@poettering.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When resuming, we better take the DACs out of the reset state before
trying to use them.
Reference: kernel bug #13599http://bugzilla.kernel.org/show_bug.cgi?id=13599
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8523 is a high performance stereo DAC with integral charge
pump providing 2Vrms line driver outputs using a single 3.3V power
supply rail.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The clock API can't cope with unbalanced enables and disables and
we only enable in hw_params() but try to disable in shutdown.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
So far, the FLOAT PCM format is used only exclusivley set. But
this can be a combination with other formats.
This patch changes the parser to allow the FLOAT format in addition
to other PCM formats.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device is a mono device but it can read two channel data and
many I2S controllers only understand 2 channels.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
alc882_auto_init_analog_input() sets the input pins to VREF-80 regardless
of the input pin types although it shouldn't be for line-in pins.
This patch fixes the behavior to follow other codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge patch_alc882() and patch_alc883() to the former one since both
codecs have fairly similar connections but just a slight difference.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch tries to work around the problem of broken OMAP1510 PCM playback
pointer calculation by replacing DMA function call that incorrectly tries to
read the value form DMA hardware with a value computed locally from an
already maintained variable omap_runtime_data.period_index.
Tested on OMAP5910 based Amstrad Delta (E3) using work in progress ASoC
driver.
Based on linux-2.6-asoc.git v2.6.31-rc1.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The needed spin_event_timeout() macro is now merged in from the
powerpc tree, so these drivers are no longer broken. This reverts
commit 0c0e09e21a (ASoC: Mark MPC5200
AC97 as BROKEN until PowerPC merge issues are resolved)
Tested against 2.6.31-rc1.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA SoC drivers should be specify SND_SOC_AC97_BUS instead, not AC97_BUS.
Without SND_SOC_AC97_BUS defined, an AC97 device will not get correctly
registered on the AC97 bus, which prevents thinks like the WM9712
touchscreen driver from getting probed.
Tested against 2.6.31-rc1.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Jon Smirl <jonsmirl@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
mpu401_chk_version is called with a spin lock already held. Don't take it
again.
Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During the changes to clean up / fix the realtek codec initialization
routines in commit 4a79ba34ca,
I forgot to add the check for ALC268 and ALC269.
This resulted in the missing EAPD and COEF setup for these codecs.
This patch adds the missing checks for these codecs.
Reference: bko#13633
http://bugzilla.kernel.org/show_bug.cgi?id=13633
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the hint "beep" in snd_hda_attach_beep_device() to avoid the beep
device creation if user doesn't want.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ALSA 1.0.20, the comments were changed to say CMI8330 instead of AD1848.
The CMI8330 chip includes two codecs - AD1848 and SB16, so the comments were
correct and are misleading now. Revert them back.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Line In connector is set up as PIN_IN by default, using
VREF_HIZ. It is connected to both ADCs, so add it to both
input selectors.
Also add the ability to use the input mix (on a SoundBlaster
one would call this "What You Hear").
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls,
removing word "Switch" from HandsfreeL/HandsfreeR widget name
for avoiding to duplicate it.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use kasprintf to allocate temporary devname string instead of a
fixed size string.
This fixes "FIXME" introduced on removal of BUS_ID_SIZE.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The microphone input and its volume register have only one channel, so
we have to make the corresponding mixer control a mono control.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
as long as the io channel number is not set by the driver, the card
is not visible from the ethersound network
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I've built a small HTPC and had to add suspend/resume support in ice1724
driver. There seem to be 3 existing bugs related to that:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4413https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3748https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2314
Due to hardware (un)availability, I only enabled the fix for Audiotrak
Prodigy HD2 card, which is installed in my HTPC. However, most of my code
should be reusable in the future on other ice1724-based cards as well (as
long as people add card-specific peices of code). The fix is currently based
on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel).
Signed-off-by: Igor Chernyshev <igor.ch75+alsa at gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here's a patch on top of the others to use CREATIVE and ECTIVA
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following bugs of acer-aspire-6530g model with ALC888:
- HP jack to mute all speaker outputs including LFE
- Make digital built-in mic working
Signed-off-by: Emilio López <buhitoescolar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Realtek codecs require the pin-sense trigger call before actually
reading the pin-sense. Without this, the pin-detection might not be
done accurately.
This patch adds the pin-capability check and issues the trigger call
if required.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Samsung P50 requires the HP auto-muting unlike other Samsung models.
Added a new model=samsung-p50 to support this.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new flag to ad_spec struct so that the same hack can be used for
any other models (if any). This also allows other models to reuse the
auto-mute functions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split mixer element arrays of AD1986A models to several pieces so that
each model can share the same mixer arrays.
This removes lots of duplicated data.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the patch module option to apply a "patch" as a firmware to
modify pin configurations or give additional hints to the driver
before actually initializing and configuring the codec.
This can be used as a workaround when the BIOS doesn't give sufficient
information or give wrong information that doesn't match with the real
hardware setup, until it's fixed statically in the driver via a quirk.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec setup call via snd_hda_codec_configure() isn't necessarily
called in snd_hda_codec_new(). For the later added feature, it's better
to change the code flow like:
- create all codec instances
- configure each codec
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Ensure that the audio subsystem is powered down cleanly when the system
shuts down by providing a shutdown operation. This ensures that all the
components have been returned to an off state cleanly which should avoid
audio issues from partially charged capacitors or noise on digital inputs
if the system is restarted quickly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Ben Dooks <ben-linux@fluff.org>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Make jack-plug notification selectable
ALSA: ctxfi - Add PM support
sound: seq_midi_event: fix decoding of (N)RPN events
ALSA: hda - Add digital-mic support to ALC262 auto model
ALSA: hda - Fix check of input source type for realtek codecs
ALSA: hda - Add quirk for Sony VAIO Z21MN
ALSA: hda - Get back Input Source for ALC262 toshiba-s06 model
ALSA: hda - Fix unsigned comparison in patch_sigmatel.c
ALSA: via82xx: add option to disable 500ms delay in snd_via82xx_codec_wait
sound: fix check for return value in snd_pcm_hw_refine
ALSA: ctxfi - Allow unknown PCI SSIDs
ASoC: Blackfin: update the bf5xx_i2s_resume parameters
ASoC: Blackfin: keep better track of SPORT configuration state
Event for voice sidetone was being interpreted as an
analog HiFi bypass event because VSTPGA register offset
is less than ARXR2_APGA_CTL offset. Reordering the
register checks allows to handle voice digital bypass
event properly.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AVDAC clk priority allows to determine the path ADC must
be connected when the codec is in option2 and both HiFi
and Voice paths are enabled.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (49 commits)
[ARM] idle: clean up pm_idle calling, obey hlt_counter
[ARM] S3C: Fix gpio-config off-by-one bug
[ARM] S3C64XX: add to_irq() support for EINT() GPIO
[ARM] S3C64XX: clock.c: fix typo in usb-host clock ctrlbit
[ARM] S3C64XX: fix HCLK gate defines
[ARM] Update mach-types
[ARM] wire up rt_tgsigqueueinfo and perf_counter_open
OMAP2 clock/powerdomain: off by 1 error in loop timeout comparisons
OMAP3 SDRC: set FIXEDDELAY when disabling SDRC DLL
OMAP3: Add support for DPLL3 divisor values higher than 2
OMAP3 SRAM: convert SRAM code to use macros rather than magic numbers
OMAP3 SRAM: add more comments on the SRAM code
OMAP3 clock/SDRC: program SDRC_MR register during SDRC clock change
OMAP3 clock: add a short delay when lowering CORE clk rate
OMAP3 clock: initialize SDRC timings at kernel start
OMAP3 clock: remove wait for DPLL3 M2 clock to stabilize
[ARM] Add old Feroceon support to compressed/head.S
[ARM] 5559/1: Limit the stack unwinding caused by a kthread exit
[ARM] 5558/1: Add extra checks to ARM unwinder to avoid tracing corrupt stacks
[ARM] 5557/1: Discard some ARM.ex*.*exit.text sections when !HOTPLUG or !HOTPLUG_CPU
...
* topic/hda:
ALSA: hda - Make jack-plug notification selectable
ALSA: hda - Add digital-mic support to ALC262 auto model
ALSA: hda - Fix check of input source type for realtek codecs
ALSA: hda - Add quirk for Sony VAIO Z21MN
ALSA: hda - Get back Input Source for ALC262 toshiba-s06 model
ALSA: hda - Fix unsigned comparison in patch_sigmatel.c
Make the jack-plug notification via input layer selectable via Kconfig.
This is often unnecessary, and the similr function will be provided
using the ALSA control API in near future anyway.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the suspend/resume support to ctxfi driver.
The team tested on the following seems ok:
AMD Athlon 64 3500+ / ASUS A8N-E / 512MB DDR ATI / Radeon X1300
20k1 & 20k2 cards
Signed-off-by: Wai Yew CHAY <wychay@ctl.creative.com>
Singed-off-by: Ryan RICHARDS <ryan_richards@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Voice interface of twl4030 codec supports: CBM_CFM and
CBS_CFS. It doesn't support CBS_CFM.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-By: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When decoding (N)RPN sequencer events into raw MIDI commands, the
extra_decode_xrpn() function had accidentally swapped the MSB and LSB
controller values of both the parameter number and the data value.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the digital-mic support with ALC262 auto model.
The new ALC262 models have the digital mic at NID 0x12. This widget
isn't checked in the current alc262_auto_create_analog_input_ctls()
since it's under 0x18. So, just reuse the routine for alc269 to fix
the behavior.
But, it doesn't suffice: the digital mic is supported only with the
ADC0, we have to exclude other ADCs when d-mic is detected.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the check of the input-source type by checking the widget type of
each capture-source item. Since some codecs can have both the mixer
and selector types depending on the ADC, alc_mux_enum_put() needs to
check each widget.
With this change, spec->capture_style gets unneeded, so it's removed,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit f9e336f65b
ALSA: hda - Unify capture mixer creation in realtek codes
removed the "Input Source" mixer element creation for toshiba-s06 model
because it contains a digital-mic input.
This patch take the control back.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Fix the comparison of unsigned int that causes a compile warning below
by changing to the right signed type:
patch_sigmatel.c: In function ‘stac92xx_vref_set’:
patch_sigmatel.c:658: warning: comparison of unsigned expression < 0 is always false
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There's a large 500ms delay in snd_via82xx_codec_wait() that, at least
on my hardware, appears to be unnecessary. The rest of the init of
the card works without logging any warnings or errors and both audio
and mixer settings work.
This adds an "nodelay" parameter to disable this (undocumented in the
code) large delay improving bootup time by 489-500ms.
[ 1.034217] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 505757 usecs
vs.
[ 0.533136] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 15915 usecs
Signed-off-by: Simon Arlott <simon@fire.lp0.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'params' is a pointer and looking at the code this probably should be a check
for ioctl return value.
Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow unknown PCI SSIDs for emu20k1 and emu20k2 as "unknown" model.
Also, add a black-list check in case any device has to be listed
as "unsupported". It has a negative value in the pci quirk entry.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The selected 4930G model seemed to keep the subwoofer 'tuba'
function from operating correctly. Removing the existing PCI
ID match made this work again, but it was mapped to 'Side'
instead of to LFE as one would expect.
This attempts to enable all functionality and keep the amount
of available mixer sliders low. Any slider that had no audible
effect on the output audio has been removed, and as such EAPD
is not currently enabled.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These drivers aren't BF52x specific, so don't use bf52x in the names.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Latest ASoC only passes snd_soc_dai to the resume function.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Do not let the SPORT be reconfigured until there are no more active
streams. Then we can let the system reprogram the SPORT state.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No need to define second copies of mode and format, they're declared
with exactly the same type at the head of the function and there's no
conflict in their use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're about to make DMA_nnBIT_MASK() emit `deprecated' warnings. Convert the
remaining stragglers which are visible to the x86_64 build.
Cc: FUJITA Tomonori <fujita.tomonori@lab.ntt.co.jp>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Eric Moore <Eric.Moore@lsil.com>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Alexander Duyck <alexander.h.duyck@intel.com>
Cc: Yi Zou <yi.zou@intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Add basic support for integration with the regulator API to WM8580.
Since the core cannot yet disable biases when the CODEC is idle we
simply request and enable the regulators for the entire time the
driver is active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Seems that nobody recently tried the input on the very first supported
sound card model, RK2. This patch fixes the byte offset to make it
running again.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the use of BUS_ID_SIZE from txx9aclc.c, as BUS_ID_SIZE will be
removed soon later.
Also, use snprintf() instead of sprintf() as a safer operation.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some USB devices give bogus dB information and it screws up PA.
It's better to detect a broken value and correct it in the driver
before exposing the value to the outside.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new types for TLV dB scale specified with min/max values instead
of min/step since the resolution can't match always with the one
a device provides. For example, usb audio devices give 1/256 dB
resolution while ALSA TLV is based on 1/100 dB resolution.
The new min/max types have less problems because the possible
rounding error happens only at min/max.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct some cut+paste typos from 'tagra' to 'targa'.
Signed-off-by: Sasha Alexandr <brina_keith@ns.sympatico.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* akpm: (182 commits)
fbdev: bf54x-lq043fb: use kzalloc over kmalloc/memset
fbdev: *bfin*: fix __dev{init,exit} markings
fbdev: *bfin*: drop unnecessary calls to memset
fbdev: bfin-t350mcqb-fb: drop unused local variables
fbdev: blackfin has __raw I/O accessors, so use them in fb.h
fbdev: s1d13xxxfb: add accelerated bitblt functions
tcx: use standard fields for framebuffer physical address and length
fbdev: add support for handoff from firmware to hw framebuffers
intelfb: fix a bug when changing video timing
fbdev: use framebuffer_release() for freeing fb_info structures
radeon: P2G2CLK_ALWAYS_ONb tested twice, should 2nd be P2G2CLK_DAC_ALWAYS_ONb?
s3c-fb: CPUFREQ frequency scaling support
s3c-fb: fix resource releasing on error during probing
carminefb: fix possible access beyond end of carmine_modedb[]
acornfb: remove fb_mmap function
mb862xxfb: use CONFIG_OF instead of CONFIG_PPC_OF
mb862xxfb: restrict compliation of platform driver to PPC
Samsung SoC Framebuffer driver: add Alpha Channel support
atmel-lcdc: fix pixclock upper bound detection
offb: use framebuffer_alloc() to allocate fb_info struct
...
Manually fix up conflicts due to kmemcheck in mm/slab.c
PIT_TICK_RATE is currently defined in four architectures, but in three
different places. While linux/timex.h is not the perfect place for it, it
is still a reasonable replacement for those drivers that traditionally use
asm/timex.h to get CLOCK_TICK_RATE and expect it to be the PIT frequency.
Note that for Alpha, the actual value changed from 1193182UL to 1193180UL.
This is unlikely to make a difference, and probably can only improve
accuracy. There was a discussion on the correct value of CLOCK_TICK_RATE
a few years ago, after which every existing instance was getting changed
to 1193182. According to the specification, it should be
1193181.818181...
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Cc: Richard Henderson <rth@twiddle.net>
Cc: Ivan Kokshaysky <ink@jurassic.park.msu.ru>
Cc: Ralf Baechle <ralf@linux-mips.org>
Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org>
Cc: Ingo Molnar <mingo@elte.hu>
Cc: Thomas Gleixner <tglx@linutronix.de>
Cc: "H. Peter Anvin" <hpa@zytor.com>
Cc: Len Brown <lenb@kernel.org>
Cc: john stultz <johnstul@us.ibm.com>
Cc: Dmitry Torokhov <dtor@mail.ru>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
* git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/driver-core-2.6: (64 commits)
debugfs: use specified mode to possibly mark files read/write only
debugfs: Fix terminology inconsistency of dir name to mount debugfs filesystem.
xen: remove driver_data direct access of struct device from more drivers
usb: gadget: at91_udc: remove driver_data direct access of struct device
uml: remove driver_data direct access of struct device
block/ps3: remove driver_data direct access of struct device
s390: remove driver_data direct access of struct device
parport: remove driver_data direct access of struct device
parisc: remove driver_data direct access of struct device
of_serial: remove driver_data direct access of struct device
mips: remove driver_data direct access of struct device
ipmi: remove driver_data direct access of struct device
infiniband: ehca: remove driver_data direct access of struct device
ibmvscsi: gadget: at91_udc: remove driver_data direct access of struct device
hvcs: remove driver_data direct access of struct device
xen block: remove driver_data direct access of struct device
thermal: remove driver_data direct access of struct device
scsi: remove driver_data direct access of struct device
pcmcia: remove driver_data direct access of struct device
PCIE: remove driver_data direct access of struct device
...
Manually fix up trivial conflicts due to different direct driver_data
direct access fixups in drivers/block/{ps3disk.c,ps3vram.c}
The wm8731 driver has been converted to register using the standard I2C
device registration mechanism so we can now do the registration in the
arch/arm code as normal.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
This adds support to the sound core to report the proper device name to
userspace for their devices.
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Jan Blunck <jblunck@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
This patch mostly follows commit 5998102b90
"ASoC: Refactor WM8731 device registration" to make UDA1380 use standard
device instantiation. Similarly, the I2C device registration temporarily
moves into the magician machine driver before it will find its final
resting place in the board file.
At the same time, platform specific configuration is moved to platform data
and common power/reset GPIO handling moves into the codec driver.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM x-fi native update routine can cause deadlocks when the
trigger(START) is called while the stream is running.
This patch fixes the deadlock by just postponing the pcm period update
to the next possible wake-up. Also it adds the flip of ti->running
flag (just to be sure as now).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM pointer callback sometimes returns invalid positions and this
screws up the hw_ptr updater in PCM core. Especially since now the
jiffies check is optional with xrun_debug, the invalid position is
handled as is, and causes serious sound skips, etc.
This patch simplifies the position-fix strategy in intel8x0 to be more
robust:
- just falls back to the last position if bogus position is detected
- another sanity check for the backward move of the position due to
a race of register update and the base-index update
This patch is applicable also for 2.6.30.
Tested-by: David Miller <davem@davemloft.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch changes yet again the ID for the TTA cards, resulting in a
more reasonable name:
1 [Aureon51MkII ]: USB-Audio - Aureon5.1MkII
TerraTec Aureon5.1MkII at usb-0000:00:03.0-2, full speed
Signed-off-by: Andrea Borgia <andrea@borgia.bo.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (417 commits)
MAINTAINERS: EB110ATX is not ebsa110
MAINTAINERS: update Eric Miao's email address and status
fb: add support of LCD display controller on pxa168/910 (base layer)
[ARM] 5552/1: ep93xx get_uart_rate(): use EP93XX_SYSCON_PWRCNT and EP93XX_SYSCON_PWRCN
[ARM] pxa/sharpsl_pm: zaurus needs generic pxa suspend/resume routines
[ARM] 5544/1: Trust PrimeCell resource sizes
[ARM] pxa/sharpsl_pm: cleanup of gpio-related code.
[ARM] pxa/sharpsl_pm: drop set_irq_type calls
[ARM] pxa/sharpsl_pm: merge pxa-specific code into generic one
[ARM] pxa/sharpsl_pm: merge the two sharpsl_pm.c since it's now pxa specific
[ARM] sa1100: remove unused collie_pm.c
[ARM] pxa: fix the conflicting non-static declarations of global_gpios[]
[ARM] 5550/1: Add default configure file for w90p910 platform
[ARM] 5549/1: Add clock api for w90p910 platform.
[ARM] 5548/1: Add gpio api for w90p910 platform
[ARM] 5551/1: Add multi-function pin api for w90p910 platform.
[ARM] Make ARM_VIC_NR depend on ARM_VIC
[ARM] 5546/1: ARM PL022 SSP/SPI driver v3
ARM: OMAP4: SMP: Update defconfig for OMAP4430
ARM: OMAP4: SMP: Enable SMP support for OMAP4430
...
Follow-up fix needed since "ASoC: pxa-ssp.c fix clock/frame invert".
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With the recent changes to the DAPM power checks it has become important
to explicitly instantiate all widgets but some drivers were forgetting
to do that. Since everything needs to do it add a call to instantiate
them immediately before the card registration - it does no harm when it
is called repeatedly and saves work in drivers.
Tested-by: pHilipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that ASoC subdevices can be regular devices they can have normal
suspend and resume calls from their buses. However, suspending them
individually is not desirable since this can lead to problems such as
pops and clicks from devices being suspended with their signals being
amplified or clocks being stopped suddenly.
This will be resolved by having the normal device model suspend and
resume calls call into ASoC which will suspend the entire card while any
of its components are suspended. At present this is not yet implemented
but in order to aid the transition of drivers to the standard device
model this patch adds API calls for the notifications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For best performance the DAC sloping stopband filter should be enabled
below 24kHz and not enabled above that so remove the user visible
control for this and do it autonomously in the driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For best performance the DAC sloping stopband filter should be
enabled below 24kHz and not enabled above that so remove the
user visible control for this and do it autonomously in the
driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For best performance the DAC sloping stopband filter should be
enabled below 24kHz and not enabled above that so remove the
user visible control for this and do it autonomously in the
driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Blackfin submission was done as a patch against a different tree
and contained a duplicate hunk which will cause us to loose track of the
substream pointers when shutting down. Remove one of the duplicated
hunks.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The spinlock in atc can cause a sleep in lock:
Kernel failure message 1:
BUG: sleeping function called from invalid context at mm/slub.c:1599
in_atomic(): 0, irqs_disabled(): 1, pid: 2537, name: gstreamer-prope
Pid: 2537, comm: gstreamer-prope Tainted: P
2.6.29.4-167.fc11.x86_64 #1
Call Trace:
[<ffffffff8103ff0f>] __might_sleep+0x10b/0x110
[<ffffffff810cd734>] __kmalloc+0x73/0x130
[<ffffffffa0b4b142>] ? daio_rsc_init+0xaa/0x125 [snd_ctxfi]
[<ffffffffa0b4b212>] dao_rsc_init+0x55/0x1c0 [snd_ctxfi]
[<ffffffffa0b4b3d2>] dao_rsc_reinit+0x55/0x5d [snd_ctxfi]
[<ffffffff813abd6c>] ? _spin_lock_irqsave+0x32/0x3b
[<ffffffffa0b454fe>] atc_spdif_out_passthru+0x92/0x136 [snd_ctxfi]
...
Since the lock path is no critical path, it can be gracefully
replaced with a mutex.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for Zoom2 board. Zoom2 machine driver
connects both codec DAIs (audio and voice) to omap3
McBSP ports in the following way:
HiFi <-> McBSP2
Voice <-> McBSP3
The zoom2 driver has the following DAPM widgets:
* Ext Mic: MAINMIC, SUBMIC (with bias)
* Ext Spk: HFL, HFR
* Headset Stereophone: HSOL, HSOR
* Headset Mic: HSMIC (with bias)
* Aux In: AUXL, AUXR
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
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: (290 commits)
ALSA: pcm - Update document about xrun_debug proc file
ALSA: lx6464es - support standard alsa module parameters
ALSA: snd_usb_caiaq: set mixername
ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205)
ALSA: use card device as parent for jack input-devices
ALSA: sound/ps3: Correct existing and add missing annotations
ALSA: sound/ps3: Restructure driver source
ALSA: sound/ps3: Fix checkpatch issues
ASoC: Fix lm4857 control
ALSA: ctxfi - Clear PCM resources at hw_params and hw_free
ALSA: ctxfi - Check the presence of SRC instance in PCM pointer callbacks
ALSA: ctxfi - Add missing start check in atc_pcm_playback_start()
ALSA: ctxfi - Add use_system_timer module option
ALSA: usb - Add boot quirk for C-Media 6206 USB Audio
ALSA: ctxfi - Fix wrong model id for UAA
ALSA: ctxfi - Clean up probe routines
ALSA: hda - Fix the previous tagra-8ch patch
ALSA: hda - Add 7.1 support for MSI GX620
ALSA: pcm - A helper function to compose PCM stream name for debug prints
ALSA: emu10k1 - Fix minimum periods for efx playback
...
These are not supported since performance can not be guaranteed
when they are in use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
The WM8961 is a low power, high quality stereo CODEC designed for
portable digital applications with headphone and stereo class D speaker
drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a trivial patch that removes an unnecessary void pointer cast.
Signed-off-by: Chris Sanford <crsanford@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Just for the sake of readability, removing extra space
Signed-off-by: Viral Mehta <viral.mehta@einfochips.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
* We don't need to write the registers twice, remove the first write.
* DAIFMT_INV switch is duplicated inside DAIFMT_FORMAT switch, move it
out.
(This patch is for Mark's for-2.6.32 branch, I have not checked if the
code is duplicated on current 2.6.30)
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
I found the PrimeCell/AMBA Bus drivers distrusting the resource
passed in as part of the struct amba_device abstraction. This
patch removes all hard coded resource sizes found in the PrimeCell
drivers and move the responsibility of this definition back to
the platform/board device definition, which already exist and
appear to be correct for all in-tree users of these drivers.
We do this using the resource_size() inline function which was
also replicated in the only driver using the resource size, so
that has been changed too. The KMI_SIZE was left in kmi.h in case
someone likes it. Test-compiled against Versatile and Integrator
defconfigs, seems to work but I don't posess these boards and
cannot test them.
Signed-off-by: Linus Walleij <linus.walleij@stericsson.com>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
trivial patch to support the alsa module parameters `index', `id'
and `enable'
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alsamixer and friends want the mixername to be set. Even though the
driver does not exports a real mixer device, export the name doesn't
harm.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This moves the jack devices from the PCI device into the ALSA card device, which
makes it easier for userspace to find all devices belonging to a specific card
while granting access to logged-in users.
Jack input devices from sound cards can now simply be matched with udev by doing:
SUBSYSTEM="input", SUBSYSTEMS="sound", ...
ls -l /sys/devices/pci0000:00/0000:00:1b.0/sound/card0
controlC0
device -> ../../../0000:00:1b.0
id
input10
input11
input8
input9
number
pcmC0D0c
pcmC0D0p
pcmC0D1p
power
subsystem -> ../../../../../class/sound
uevent
Cc: Lennart Poettering <lennart@0pointer.de>
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sort includes, and reorder code so we can kill the forward declarations
No functional changes
Signed-off-by: Geert Uytterhoeven <Geert.Uytterhoeven@sonycom.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 4eaa9819dc changed semantics of
private_value member of kcontrol. This resulted in inability to control
amplifier and subsequently in very low output volume.
Tested-by: Johannes Schauer <josch@pyneo.org>
Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* topic/pcm-jiffies-check:
ALSA: pcm - A helper function to compose PCM stream name for debug prints
ALSA: pcm - Fix update of runtime->hw_ptr_interrupt
ALSA: pcm - Fix a typo in hw_ptr update check
ALSA: PCM midlevel: lower jiffies check margin using runtime->delay value
ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed
ALSA: PCM midlevel: introduce mask for xrun_debug() macro
ALSA: PCM midlevel: improve fifo_size handling
* topic/oxygen:
sound: virtuoso: add Xonar Essence ST support
sound: virtuoso: enable HDAV S/PDIF input
sound: virtuoso: add another DX PCI ID
sound: oxygen: reset DMA when stream is closed
* topic/misc:
ALSA: sgio2audio.c: clean up checking
ALSA: burgundy: timeout message is off by one.
ALSA: bt87x - Add a quirk entry for Askey Computer Corp. MagicTView'99
ALSA: parisc/harmony: fix printk format warning
ALSA: keywest: Get rid of useless i2c_device_name() macro
* topic/maya44:
ALSA: ice1724 - Add ESI Maya44 support
ALSA: ice1724 - Allow spec driver to create own routing controls
ALSA: ice1724 - Add PCI postint to reset sequence
ALSA: ice1724 - Clean up definitions of DMA records
ALSA: ice1724 - Check error in set_rate function
* topic/core-id-check:
ALSA: Core - clean up snd_card_set_id* calls and remove possible id collision
ALSA: Fix double locking of card list in snd_card_register()
* topic/asoc: (135 commits)
ASoC: Apostrophe patrol
ASoC: codec tlv320aic23 fix bogus divide by 0 message
ASoC: fix NULL pointer dereference in soc_suspend()
ASoC: Fix build error in twl4030.c
ASoC: SSM2602: assign last substream to the master when shutting down
ASoC: Blackfin: document how anomaly 05000250 is handled
ASoC: Blackfin: set the transfer size according the ac97_frame size
ASoC: SSM2602: remove unsupported sample rates
ASoC: TWL4030: Check the interface format for 4 channel mode
ASoC: TWL4030: Use reg_cache in twl4030_init_chip
ASoC: Initialise dev for the dummy S/PDIF DAI
ASoC: Add dummy S/PDIF codec support
ASoC: correct print specifiers for unsigneds
ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
ASoC: Switch FSL SSI DAI over to symmetric_rates
ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
ASoC: Fabric bindings for STAC9766 on the Efika
ASoC: Support for AC97 on Phytec pmc030 base board.
ASoC: AC97 driver for mpc5200
ASoC: Main rewite of the mpc5200 audio DMA code
...
Currently the PCM resources are allocated only once and ever in prepare
callback, assuming that the PCM parameters are never changed. But it's
not true.
This patch adds the call of atc->pcm_release_resources() at hw_params
and hw_free callbacks to assure that the PCM setup is done correctly
for each h/w parameter changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SRC instances may not exist when PCM pointer callback is called at
the state before initialization is finished. Add the NULL check just
to be sure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added use_system_timer module option to force to use the system timer
instead of emu20k1 timer irq for debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added boot quirk for C-Media CM6206 device in snd_usb_audio_probe.
The function snd_usb_cm6206_boot_quirk sets up six internal 16-bit
registers in order to initialize the device. Values for the registers
came from sniffing USB traffic under Windows since only four of the six
are documented in the datasheet for CM106 and some reserved bits were
also being set.
[Minor coding-style fixes by tiwai]
Signed-off-by: Dan Allongo <gongo2k1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CTUAA should be checked instead of CTHENDRIX. The latter is for 20k2 chip.
Also, fixed the detection of UAA/HENDRIX models by fixing the mask bits.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up probe routines and model detection routines so that the driver
won't call and check the PCI subsystem id at each time.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added 7.1 support for MSI GX620 and jack quirk.
Reference: kernel bug#13430
http://bugzilla.kernel.org/show_bug.cgi?id=13430
Signed-off-by: David Heidelberger <d.okias@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Assign proper errors when platform resource claims fail.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
EFX playback stream should have periods_min = 2 to avoid the buffer
position overflow (due to restrictions of the pcm-indirect helper).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Clean up Hungarian coding style
- Don't use static variables for I2C information; this unables to use
multiple instances. Now they are stored in struct hw20k2 fields.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
They are now only accessed within dapm_power_widgets() so can be local
to that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Handle gain ramping for PGAs so we can coalesce their power updates too.
This is not ideal since we can't cope properly with gain ramping for
stereo paths but that was the case without coalescing and gain ramping
is relatively infrequently used so the effects are limited.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The more flexible value muxes and named mixers don't need to be sorted
differently from a power management point of view, they are different
only in terms of the control interface and not in terms of seqencing
behaviour.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reduce the number of register writes we need to set the power state for
a CODEC by coalescing updates to widgets with the same sequence order and
same register into a single write.
This can be a noticable performance improvement with slow or heavily
contended control buses, such as I2C controllers with a low clock
frequency, and is particularly noticable when resuming. It can also
reduce the noticability of and pops and clicks by ensuring that left
and right channels are powered simultaneously if they are in the same
register.
Currently widgets that have events are not coalesced, including PGAs
which may use the volume ramping control.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Optimize the timer update routine to look up wall clock once instead of
checking the position of each stream at each timer update.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the remaining unsigned shorts with unsigned ints.
Tested with pcap2 codec (25 bits registers).
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch does the following:
(1) Add support for the DM646x machine
(2) Modifications required to introduce the platform driver model to get
platform data for all the machines including dm355 and dm644x.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adds driver support for the two instances of McASP on TI's DM646x.
The multichannel audio serial port (McASP) functions as a general-purpose audio
serial port optimized for the needs of multichannel audio application.
(http://www.ti.com/litv/pdf/spruer1b).
There are two instances of McASP on DM646x. The McASP0 module includes up to 4
serializers that can be individually enabled to either transmit or receive
in different modes. The McASP1 module is limited with only 1 pinned-out
serializer that can be enabled to only transmit in DIT mode (neither receiving
in any mode nor transmitting in either Burst or TDM mode is supported).
McASP0 consists of transmit and receive sections that may operate
synchronized, or completely independently with separate master clocks, bit
clocks, and frame syncs, and using different transmit modes with different
bit-stream formats.
Signed-off-by: Steve Chen <schen@mvista.com>
Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com>
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce the platform driver model to get platform data for dm355 and dm644x.
Register platform driver and acquire the resources in the probe function Since
the platform specific code had been moved from machine driver to dm<soc>.c
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Lump the list walk into a single function, and pull in the power
application too so we can do some further refactoring. Pure code
motion.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the past the DAPM power sequencing was done by iterating over the list
of widgets once for each widget type and powering widgets of that type.
Instead of doing that do the sorting at the time we insert the widgets
into the lists of widgets to apply power changes to. This reduces the
amount of computation required for seqencing still further, though the
costs are generally dwarfed by the costs of the register writes
implementing them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit 13f040f9e5 made another
regression, the missing update of runtime->hw_ptr_interrupt.
Since this field is only checked in snd_pcmupdate__hw_ptr_interrupt(),
not in snd_pcm_update_hw_ptr(), it must be updated before the hw_ptr
change check.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
vfree() does it's own 'NULL' check,so no need for check before
calling it.
Signed-off-by: Figo.zhang <figo1802@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a typo in the commit 13f040f9e5
ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed
which causes obvious problems with PA.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some code analyzer software mistakenly gives
divide by 0 error messages for these lines.
This patch will end its confusion.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Use static tables instead of assigining each funciton pointer
- Add __devinit* to appropriate places; pcm, mixer and timer cannot be
marked because they are kept in the function table that lives long
- Move create_alsa_devs function out of struct ct_atc to mark it
__devinit
Signed-off-by: Takashi Iwai <tiwai@suse.de>
emu20k1 has a native timer interrupt based on the audio clock, which
is more accurate than the system timer (from the synchronization POV).
This patch adds the code to handle this with multiple streams.
The system timer is still used on emu20k2, and can be used also for
emu20k1 easily by changing USE_SYSTEM_TIMER to 1 in cttimer.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
with BIOS probing only we offer a non functional headphone swith and
volume slider.
Signed-off-by: Guido Günther <agx@sigxcpu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The UAA-mode check in hwct20k1.c is implemented with the endian-dependent
codes. Fix to be more portable (and readable).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case the initalization of an soc_device failed, there is no codec
associated with it. soc_suspend() will still dereference the pointer
and cause an Ooops when entering the sleep mode.
This happens on our board with a multi-target kernel image when booted
on a machine without audio circuits.
This patch makes the code bail out very early in this special case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix this build error when CONFIG_PM is not set:
ound/pci/hda/hda_intel.c: In function 'azx_bus_reset':
sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all'
sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend'
sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume'
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
3 ISA sound drivers lack their __devexit_p() markers, which would
cause build failures when the kernel is built without hotplug support.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Kyle McMartin <kyle@mcmartin.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the (likely cut-n-paste) error by commit
16a30fbb0d, which causes the error below:
sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache':
sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move locking outside snd_card_set_id_internal() function and rename it
to snd_card_set_id_no_lock() for better function description.
User defined id is just copied to card structure at allocation time.
The real unique id procedure is called in snd_card_register() to
ensure real atomicity.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enable all three capture channels, including the missing nid 7 which is
the only one capable of capturing DMIC input
Enable Headphone amp for the HP jack. This causes a volume boost for
headphones, but does not cause any noticeable effect for light loads
like other amps, so there is no need to make it configurable.
Add Input Mix capture mux setting to capture the output of the playback
input mux (that is, what goes out the speakers except for PCM)
Hack another coef register because the stereo DMIC for some reason
produces a nonstandard sum/difference signal. I found a bit to make it
just use the sum signal for both channels, which makes it behave like a
standard mono microphone. The stereo is useless anyway (they're 1cm apart).
Tested working: Three capture channels, mic in, line in, DMIC.
Tested not working: CD. Not sure why, might be unconnected in the actual
hardware or a CD drive issue.
Also looked at SPDIF. It appears to work (emitter lights up inside the
HP out jack) but I lack a proper miniTOSLINK cable to test it.
Signed-off-by: Hector Martin <hector@marcansoft.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The introduction of snd_card_set_id() added a lock on the card list
to the old choose_default_id() function when using it to implement
the new API call. This lock is needed to allow us to walk the list
and check to see if our new name is a duplicate. Unfortunately this
causes a lockup when called from snd_card_register() (in cases
where no ID is supplied for the card) since the card list is already
locked there.
Fix this fairly hideously by factoring out the implementation and
using a flag to indicate if the lock should be held. A better fix
would probably be to refactor snd_card_register() to move the
_set_id() outside the locking region but I can't immediately see
anything I can convince myself is safe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes crash when shutting down.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Sonic Zhang <sonic.zhang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
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>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
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>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
[I am not sure if this is the correct approach as I don't know if any of
this actual hardware or drivers are really hot pluggable.]
Gets rid of these build warnings:
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x5c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_new()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_new().
If .snd_pmac_new is only used by .snd_pmac_probe then
annotate .snd_pmac_new with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x10c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_burgundy_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_burgundy_init().
If .snd_pmac_burgundy_init is only used by .snd_pmac_probe then
annotate .snd_pmac_burgundy_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x164): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_daca_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_daca_init().
If .snd_pmac_daca_init is only used by .snd_pmac_probe then
annotate .snd_pmac_daca_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1dc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_tumbler_init().
If .snd_pmac_tumbler_init is only used by .snd_pmac_probe then
annotate .snd_pmac_tumbler_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1ec): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_post_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_tumbler_post_init().
If .snd_pmac_tumbler_post_init is only used by .snd_pmac_probe then
annotate .snd_pmac_tumbler_post_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x28c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_awacs_init()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_awacs_init().
If .snd_pmac_awacs_init is only used by .snd_pmac_probe then
annotate .snd_pmac_awacs_init with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2bc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_pcm_new()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_pcm_new().
If .snd_pmac_pcm_new is only used by .snd_pmac_probe then
annotate .snd_pmac_pcm_new with a matching annotation.
WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2f8): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_attach_beep()
The function __devinit .snd_pmac_probe() references
a function __init .snd_pmac_attach_beep().
If .snd_pmac_attach_beep is only used by .snd_pmac_probe then
annotate .snd_pmac_attach_beep with a matching annotation.
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cut'n'paste mistake, whose likely result was nothing at all.
Correct version is "USB_DEVICE", not "USB_DEVICE_VENDOR_SPEC".
Signed-off-by: Andrea Borgia <andrea@borgia.bo.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the limitation of PAGE_SIZE to be 4k by defining the own
page size and macros for 4k. 8kb page size could be natively supported,
but it's disabled right now for simplicity.
Also, clean up using upper_32_bits() macro.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device seems supporting only U8, S16, S24_3LE, S32. Other linear
formats result in bad outputs.
Also, added the support for 32bit float format, which wasn't listed
in the original code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PCM names for surround streams should be also fixed as well as the mixer
element names. Also, a bit clean up for PCM name setup.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lars-Peter Clausen
