The commit 4d96eb255c broke the interrupt
time xrun functionality (stream stop etc.) if the CONFIG_SND_PCM_XRUN_DEBUG
is not set. This is because the xrun() is null defined without it.
Fix this by letting the function xrun() to be always defined as it was
before.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ARM-SHMOBILE series have FIFO-buffered serial interface 2 (FSI2)
device which is advanced version of FSI.
This patch add simple support for it.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The probe_only module parameter skips the codec initialization, too.
Remove the model=hwio code and use second bit in probe_only to
skip the HDA codec reset procedure.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
For Lenovo Thinkpad T61/X61, the analog beep input is connected
to node 0x20, index 3. Move the digital beep mute/volume controls
as "Digital Beep" and create analog beep controls for mentioned node.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Using the 'model=hwio' option, the driver bypasses any codec
initialization and the reset procedure for codecs is also
bypassed. This mode is usefull to enable direct access using
hwdep interface (using hdaverb or hda-analyzer tools) and
retain codec setup from BIOS.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
BugLink: https://launchpad.net/bugs/303789
This model needs both 'Headphone Jack Sense' and 'Line Jack Sense'
muted for audible audio, so just add its SSID to the blacklist and
don't enumerate the controls.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add full duplex support on AT91 and AVR.
It was a bug: we needed to check first if there are some chips opened so we
could enable both reception and sending of the data.
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add AC97 support for ATMEL AT91, using the AVR32 code.
While AVR is using a DMA, the AT91 chips are using a Peripheral Data
Controller.
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Skip initialization of connections of DAC widgets that aren't used,
which resulted in invalid verb parameters.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds basic support for M-Audio's Fast Track Ultra series of USB
audio interfaces. It is a refactored version of the patch Clemens
Ladisch posted some time ago. Neither playback nor capturing work
properly at 44100 Hz (don't know why).
The other sampling rates work properly. There's no support for the DSP
mixer, yet.
Signed-off-by: Felix Homann <fexpop@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sparse caught that initialize "playback" two times instead of
initializing "capture".
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Bit operation for fsi_master should be done inside master lock.
But soft-reset/interrupt operation were outside of it.
This patch modify this problem.
It still allow to INT_ST outside-operation on fsi_interrupt,
but it is not problem.
Because this register doesn't need the bit operation.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When the CMI8738 FRAME2 register is read, the chip sometimes (probably
when wrapping around) returns an invalid value that would be outside the
programmed DMA buffer. This leads to an inconsistent PCM pointer that is
likely to result in an underrun.
To work around this, read the register multiple times until we get a
valid value; the error state seems to be very short-lived.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Matija Nalis <mnalis-alsadev@voyager.hr>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add proper suspend/resume code for Terratec Aureon cards.
Based on ice1724 suspend/resume work of Igor Chernyshev.
Fixes bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4944
Tested on linux-2.6.32.9
Signed-off-by: Bernhard Urban <lewurm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current ak4642 was not able to select pll.
This patch add support it.
It still expect PLL base input pin is MCKI.
see Table 5 "setting of PLL Mode" of datasheet
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This also adds the first DAI operation for AIF3 so fill out the ID and
the ops for that too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Now that the EHCI driver copes with small iso packets without blowing
up, take the snd-ua101 driver out of the alpha-test stage.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Don't force enable the microphone bias on WM8903 when doing jack
detection, and don't force enable microphone bias. This allows
platforms to only enable microphone detection when a jack has been
inserted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
If no report is specified then disable detection. Note that we don't
disable the slow clock, though the power consumption from it should
be negligable. That should be reference counted, ideally through DAPM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow machines to control exactly when the bias is turned on and off.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some systems, such as those with mechanical jack detection, may wish
to force enable a pin (typically mic bias) only some of the time.
Support such systems by having disable_pin() also coveer force enabled
pins.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Some systems provide both mechanical and electrical detection of jack
status changes. On such systems power savings can be achieved by only
enabling the electrical detection methods when physical insertion has
been detected.
Begin supporting such systems by providing a notifier for jack status
changes which can be used to trigger any reconfiguration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Gain controls on outputs affect the power consumption
when the gain is set to non 0 value.
Outputs with amps have one register to configure the
routing and the gain:
PREDL_CTL (0x25):
bit 0: Voice enable
bit 1: Audio L1 enable
bit 2: Audio L2 enable
bit 3: Audio R2 enable
bit 4-5: Gain (0x0 - power down, 0x1 - 6dB, 0x2 - 0dB, 0x3 - -6dB)
bit 0 - 3: is handled in DAPM domain (DAPM_MIXER)
bit 4 - 5: has simple volume control
If there is no audio activity (BIAS_STANDBY), and
user changes the volume, than the output amplifier will
be enabled.
If the user changes the routing (but the codec remains in
BIAS_STANDBY), than the cached gain value also be written
to the register, which enables the amplifier.
The existing workaround for this is to have virtual
PGAs associated with the outputs, and whit DAPM PMD
the gain on the output will be forced to 0 (off) by
bypassing the regcache.
This failed to disable the amplifiers in several
scenario (as mentioned above).
Also if the codec is in BIAS_ON state, and user modifies
a volume control, which path is actually not enabled, than
that amplifier will be enabled as well, but it will
be not turned off, since there is no DAPM path, which
would make mute it.
To prevent amps being enabled, when they are not
needed, introduce the following workaround:
Track the state of each of this type of output.
In twl4030_write only allow actual write, when the
given output is enabled, otherwise only update
the reg_cache.
The PGA event handlers on power up will write the cached
value to the chip (restoring gain, routing selection).
On power down 0 is written to the register (disabling
the amp, and also just in case clearing the routing).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build errors when CONFIG_PM is not enabled:
sound/usb/card.c:629: error: 'usb_audio_suspend' undeclared here (not in a function)
sound/usb/card.c:630: error: 'usb_audio_resume' undeclared here (not in a function)
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This device does not have audio controllers and backlit buttons only.
Input data is handled over a dedicated USB endpoint.
All functions are supported by the driver now.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the device id for Nvidia GT220 cards to the nvhdmi
driver. I have tested it and confirmed it to be working.
Original patch download link:
https://gist.github.com/324070/
Signed-off-by: Derek Kelly <user.vdr@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/420578
The OR has verified that his hardware distorts because of the 0 dB
offset not corresponding to the highest PCM level. Fix this by capping
said PCM level to 0 dB similarly to what we do for CX20549 (Venice).
Reported-by: Mike Pontillo <pontillo@gmail.com>
Tested-by: Mike Pontillo <pontillo@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding this PCI quirk fixes the board config detection.
This also fixes jack sensing by using "hp_detect=1" via properly detected
board config.
Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The constant DMA_ACTIVE is defined with the dma_buffparams structure rather
than with the audio_operations structure. Takashi Iwai suggested that the
dmap_out field of the audio_operations structure should be used instead.
This is not tested.
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If instantiation of a card failed, we still have to remove it from the
card list on unregistration. This fixes an Oops on Migo-R, triggering,
when after a failed firmware load attempt the driver modules are removed
and re-inserted again.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a memory corruption when ASoC devices are used in
full-duplex mode. Specifically for pxa-ssp code, where this pointer
is dynamically allocated for each direction and destroyed upon each
stream start.
All other platforms are fixed blindly, I couldn't even compile-test
them. Sorry for any breakage I may have caused.
Reported-by: Sven Neumann <s.neumann@raumfeld.com>
Reported-by: Michael Hirsch <m.hirsch@raumfeld.com>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a warning ("pxa_free_dma: trying to free channel 0 which is
already freed") when a device was opened but the hw_params() call
failed.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Note that since all the microphones share a bias there is a single
jack exported for all three, even though there are two physical
connectors plus the soldered down silicon mic. Note also that the SiMic
is always present by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The export is not needed since the per-bus code lives in the same
module.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Yi Li <yi.li@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Initial version of TWL6040 codec driver.
The TWL6040 codec uses a proprietary PDM-based digital audio interface.
Audio paths supported are:
- Input: Main Mic, Sub Mic, Headset Mic, Auxiliary-FM Left/Right
- Output: Headset Left/Right, Handsfree Left/Right
TWL6040 codec supports power-up/down manual and automatic sequence.
Manual sequence is done through a specific register writes sequence.
Automatic sequence is done when the codec is powered-up through the
external AUDPWRON line. The completion of the sequence is signaled
through the audio interrupt.
TWL6040 codec sysclk can be provided by: low-power or high
performance PLL:
- The low-power PLL takes a low-frequency input at 32,768 Hz and
generates an approximate of 17.64 or 19.2 MHz (for 44.1 KHz and 48 KHz
respectively)
- The high-performance PLL generates an exact 19.2 MHz clock signal
from high-frequency input at 12/19.2/26/38.4 MHz.
Low-power playback mode is a special scenario where only headset path
(headset DAC and driver) is active.
For the particular case of headset path, PLL being used defines the
headset power mode: low-power, high-performance.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We're keeping track of the number of times we've iterated but never
actually using this to bail out if the chip looks stuck.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
During validation of the internal clocking setup it has
been found that the following settings were not configured
in an optimal way:
ASRC_CTRL_A: SRCLKDIV was incorrect, instad of divide ratio 3,
ratio of 2 has to be used (as the comment stated)
DAC_CTRL_A: Fs = Fsref is the desired configuration instead of
Fs = Fsref / 1.5
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To make DSP_A mode working correctly the data delay should be
configured to 0. DSP_B mode thus can not be used with DAC33,
so remove it.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Basic support for Left Justified coding for OMAP McBSP.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Indentation in initial support for McPDM driver was converted to spaces.
Use tabs to comply with open source coding-style.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added alc_codec_rename() helper for renaming codec->chip_name.
Added Acer-specific codec naming for ALC269/662.
[Clean-up and refactoring by tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added alc_auto_parse_customize_define() to parse the Realtek-specific
attributes from SKU. Also enable beep controls only when the proper
attribute bit is set.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add new check for MIC. Do the internal DMIC as the Front MIC.
It could solve the default record source index issue.
[Fix the check properly using the bitmask by tiwai]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I think this should be automatic pin instead of ping.
(but could be wrong).
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
sound: sequencer: clean up remove bogus check
ALSA: hda: Use LPIB and 6stack-dig for eMachines T5212
ALSA: hda - Disable MSI for Nvidia controller
ALSA: hda - Add PCI quirks for MSI NetOn AP1900 and Wind Top AE2220
ALSA: hda - Fix secondary ADC of ALC260 basic model
ALSA: hda - Add an error message for invalid mapping NID
ALSA: hda - New Intel HDA controller
The SIU ASoC driver must load firmware to program the DSP, therefore it
has to select FW_LOADER in its Kconfig entry.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The registers for AD193X are defined as 0x800-0x810 for spi which uses
16_8 mode, for i2c to support AD1937, we will use 8_8 mode, only the low
byte of 0x800-0x810 is valid. The patch will not destory other codecs,
but make soc cache interface more useful.
Signed-off-by: Barry Song <barry.song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some devices provide support for detection of a small number of
buttons on their jacks. One common implementation provides a single
button, implemented by shorting the microphone to ground and detected
along with microphone presence detection by detecting varying current
draws on the microphone bias signal.
Provide support for up to three buttons via the jack interface. These
default to reporting BTN_n but an API is provided to allow these to
be remapped to other keys by the machine driver where it knows what
the keys are. More keys can be added with ease if required.
This is only intended to support simple accessory button designs. If
the interface is limiting then either creating a child device for the
accessory or accessing the input device in the jack directly is
recommended.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8750 is using some delayed work to manage the ramping of the bias
at startup and resume out of line from the normal flow. This predates
the support within ASoC core for moving the resume out of line from the
main system resume which provides equivalent functionality with better
interaction with applications. Change to doing the ramp in line to make
use of the core functionality.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A code audit reveals that there are currently no users of the widget
controls on PGAs. This is likely to continue to be the case since
while there are useful things that can be done with integrating the
PGA gain and mute controls with the power sequencing userspace
generally wants stereo controls for output stages which this doesn't
map onto well.
In preparation for implementing something more useful strip out the
existing code, leaving the parameters there for use by the new code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8350 provides microphone presence and short circuit detection.
Integrate this with the ASoC jack reporting API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM8904 allows microphone detection signals to be brought out as
alternate functions of the GPIO signals which can be detected using
interrupt inputs on the CPU. Allow this to be configured using
platform data.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide platform data allowing the configuration of the GPIO pins
on the WM8904 to be selected, allowing alternate functions to be
enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Support use of the WM8903 IRQ for reporting of microphone presence
and short detection.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently used to detect completion of the write sequencer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Provide support for WM8903 microphone presence and short detection
using the GPIOs to route out a logic signal suitable for handling
using snd_soc_jack_add_gpios() on the processor GPIOs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow users to pass in a default configuration for the GPIOs of
the WM8903 as platform data. This allows configuration of the pin
muxing of the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Allow pins to be forced on regardless of their power state. This is
intended for use with microphone bias supplies which need to be
enabled in order to support microphone detection - in systems without
appropriate hardware leaving the microphone unbiased when not in use
saves power.
The force done at power check time in order to avoid disrupting other
power detection logic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
A few lines earlier bend is limited to 2399. So semitones is always
less than 24 here.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/538895
The OR has verified that both position_fix=1 and model=6stack-dig are
necessary to have capture function properly. (The existing 3stack-6ch
model quirk seems to be incorrect.)
Reported-by: Reuben Bailey <reuben.e.bailey@gmail.com>
Tested-by: Reuben Bailey <reuben.e.bailey@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Judging from the member of enable_msi white-list, Nvidia controller
seems to cause troubles with MSI enabled, e.g. boot hang up or other
serious issue may come up. It's safer to disable MSI as default for
Nvidia controllers again for now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
This should make the speakers and jack detection work on MSI all-in-one
computers NetOn AP1900 and Wind Top AE2220.
Signed-off-by: Anisse Astier <anisse@astier.eu>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix adc_nids[] for ALC260 basic model to match with num_adc_nids.
Otherwise you get an invalid NID in the secondary "Input Source" mixer
element.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Add an error message to snd_hda_add_nid() for invalid mapping NID to make
easier to hunt the buggy code.
Also added a missing space to the error message in snd_hda_build_controls()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Sound MSI fallout on a Asus mobo NVIDIA MCP55
sound: fix opti92x-ad1848 build
ALSA: hda - Fix input source elements of secondary ADCs on Realtek
ALSA: hda - Fix wrong model range check for ALC268
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (56 commits)
doc: fix typo in comment explaining rb_tree usage
Remove fs/ntfs/ChangeLog
doc: fix console doc typo
doc: cpuset: Update the cpuset flag file
Fix of spelling in arch/sparc/kernel/leon_kernel.c no longer needed
Remove drivers/parport/ChangeLog
Remove drivers/char/ChangeLog
doc: typo - Table 1-2 should refer to "status", not "statm"
tree-wide: fix typos "ass?o[sc]iac?te" -> "associate" in comments
No need to patch AMD-provided drivers/gpu/drm/radeon/atombios.h
devres/irq: Fix devm_irq_match comment
Remove reference to kthread_create_on_cpu
tree-wide: Assorted spelling fixes
tree-wide: fix 'lenght' typo in comments and code
drm/kms: fix spelling in error message
doc: capitalization and other minor fixes in pnp doc
devres: typo fix s/dev/devm/
Remove redundant trailing semicolons from macros
fix typo "definetly" -> "definitely" in comment
tree-wide: s/widht/width/g typo in comments
...
Fix trivial conflict in Documentation/laptops/00-INDEX
* 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (370 commits)
ARM: S3C2443: Add set_rate and round_rate calls for armdiv clock
ARM: S3C2443: Remove #if 0 for clk_mpll
ARM: S3C2443: Update notes on MPLLREF clock
ARM: S3C2443: Further clksrc-clk conversions
ARM: S3C2443: Change to using plat-samsung clksrc-clk implementation
USB: Fix s3c-hsotg build following Samsung platform header moves
ARM: S3C64XX: Reintroduce unconditional build of audio device
ARM: 5961/1: ux500: fix CLKRST addresses
ARM: 5977/1: arm: Enable backtrace printing on oops when PC is corrupted
ASoC: Fix S3C64xx IIS driver for Samsung header reorg
ARM: S3C2440: Fix plat-s3c24xx move of s3c2440/s3c2442 support
[ARM] pxa: fix typo in mxm8x10.h
[ARM] pxa/raumfeld: set GPIO drive bits for LED pins
[ARM] pxa/zeus: Add support for mcp2515 CAN bus
[ARM] pxa/zeus: Add support for onboard max6369 watchdog
[ARM] pxa/zeus: Add Eurotech as the manufacturer
[ARM] pxa/zeus: Correct the USB host initialisation flags
[ARM] pxa/zeus: Allow usage of 8250-compatible UART in uncompress
[ARM] pxa: refactor uncompress.h for non-PXA uarts
[ARM] mmp2: fix incorrect calling of chip->mask_ack() for 2nd level cascaded IRQs
...
USB Audio Class v2.0 compliant devices have different descriptors and a
different way of setting/getting min/max/res/cur properties. This patch
adds support for them.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a number of new structs for mixer, selector, feature and
processing units and some static inline helpers to access fields which
have dynamic offsets. Use them in mixer.c to parse the descriptors. This
is necessary for the upcoming audio v2 parsers.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For clearer namespace, also rename usbmixer_maps.c -> mixer_maps.c
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move all non-standard mixer controls and vendor-specific extensions to a
separate file. Some structs need to be exported now.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No need for the private enum.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Split the audio.h file in two to clearly denote the differences
between the standards.
- Add many more defines to audio-v2.h. Most of them are not currently
used.
- Replaced a magic value with a proper define
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the new delay calback function to report the delay through
ALSA for application caused by the internal FIFO.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Platform data option for the codec to keep the BCLK clock
continuously running in FIFO modes (codec master).
OMAP3 McBSP when in slave mode needs continuous BCLK running
on the serial bus in order to operate correctly.
Since in FIFO mode the DAC33 can also shut down the BCLK clock
and enable it only when it is needed, let the platforms decide
if the CPU side needs the BCLK running or not.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
To avoid race condition especially in FIFO modes the
sequence for enabling and disabling the codec need to
be changed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DM365 EVM has two codecs: the Audio Codec (AIC3x) and the Voice Codec,
the idea is to have both enabled in the same kernel simultaneously. However,
the current soc-core doesn't support simultaneous codecs, once that
support will have added, a patch will be posted to enable both codecs in
the DM365 EVM.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the DM365 is the only SoC that includes this Voice Codec.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the support for the interface needed by the DaVinci
Voice Codec CQ93VC.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For ASoC, if either CPU or CODEC driver has set the flag, the MACHINE driver
should be given a chance to figure out if the dai, that set the flag, can
accomodate a rate that it does not explicitly specify but is specified
by the dai at the other end of the link.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver USE PLL for 11025/22050/44100/88200 rate.
To enable switching to bypass mode, PLL is always turned on.
Special thanks to Phil
Signed-off-by: Phil Edworthy <Phil.Edworthy@renesas.com>
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Towards having build for multiple SoCs segregate hw_params callback
for s3c2412 and s3c64xx.
Since, all new SoCs have s3c64xx like register map, we keep that as
default implementation if no SoC specific callback is already defined.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For some CPU-CODEC and source clock combination we might need to set
BCLK to N*Sample_size*LRCLK, where N may be even 3 or 4, not just 2.
We can simply remove the dependency of BCLK on sample size as there
is already a callback(S3C_I2SV2_DIV_BCLK) available to set required BCLK.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Towards generalizing CPU driver interface, do not accept direct field
values for the BCLK and RCLK.
The machine driver should simply request the FS-multiple and not provide
the value to be set in divide field of IISMOD.
[Confirmed by Jassi that no existing machine drivers are affected --
broonie]
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order for the RATE and FMT defines to be reuseable in future by the
i2sv4 driver, move the MACROs out to the header file.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rather than having the multiple definitions of the same clocks,
define them in one common place and refer by SoC specific names.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No point in duplicating this structure layout in each driver.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On TI DM6467 EVM, S/PDIF DIT codec fails to open as it is unable to install
hardware params. This dummy codec has no set_fmt and set_sysclk implementations
and calls from the application to these functions cause errors. This patch adds
a new hardware params callback function for S/PDIF transciever codec.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Shared memory mappings on nommu machines require a get_unmapped_area
file operation that suggests an address for the mapping. The current
implementation returns 0 and thus forces the driver to implement an
mmap handler that fixes up the start and end address of the vma.
This patch returns the address of the dma buffer, so it should work
out of the box for all drivers that use the snd_pcm_runtime->dma_area
pointer.
Addresses for mapping the status and control pages are returned as
well, but to make those work the conditional compilation of
snd_pcm_mmap_{status,control} would need to be revised.
URL: http://thread.gmane.org/gmane.linux.alsa.devel/61230
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
without the following patch audio ssttuutteerrs on
ASUS M2N32-SLI PREMIUM ACPI BIOS Revision 1304
the sound device is:
00:0e.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2)
worked with 2.6.32
Signed-off-by: Ralf Gerbig <rge@quengel.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix 'else' placement in ifdef block so that build succeeds:
sound/isa/opti9xx/opti92x-ad1848.c:221: error: 'else' without a previous 'if'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (26 commits)
ALSA: hdmi - show debug message on changing audio infoframe
ALSA: hdmi - merge common code for intelhdmi and nvhdmi
ALSA: hda - Add ASRock mobo to MSI blacklist
ALSA: hda: uninitialized variable fix
ALSA: hda: Use LPIB for a Biostar Microtech board
ALSA: usb/audio.h: Fix field order
ALSA: fix jazz16 compile (udelay)
ALSA: hda: Use LPIB for Dell Latitude 131L
ALSA: hda - Build hda_eld into snd-hda-codec module
ALSA: hda - Support NVIDIA MCP89 and GT21x hdmi audio
ALSA: hda - Support max codecs to 8 for nvidia hda controller
ALSA: riptide: clean up while loop
ALSA: usbaudio - remove debug "SAMPLE BYTES" printk line
ALSA: timer - pass real event in snd_timer_notify1() to instance callback
ALSA: oxygen: change || to &&
ALSA: opti92x: use PnP data to select Master Control port
ASoC: fix ak4104 register array access
ASoC: soc_pcm_open: Add missing bailout tag
ALSA: usbaudio: Fix wrong bitrate for Creative Creative VF0470 Live Cam
ALSA: ua101: removing debugging code
...
Since alc_auto_create_input_ctls() doesn't set the elements for the
secondary ADCs, "Input Source" elemtns for these also get empty, resulting
in buggy outputs of alsactl like:
control.14 {
comment.access 'read write'
comment.type ENUMERATED
comment.count 1
iface MIXER
name 'Input Source'
index 1
value 0
}
This patch fixes alc_mux_enum_*() (and others) to fall back to the
first entry if the secondary input mux is empty.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
Fix a wrong value passed to snd_hda_check_board_codec_sid_config() as
the upper-limit in parse_alc268(), so that any wrong value can't be
passed.
So far, no bogus value was set in the quirk entries, so this won't give
any behavioral changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create patch_hdmi.c to hold common code from intelhdmi and nvhdmi.
For now the patch_hdmi.c file is simply included by patch_intelhdmi.c
and patch_nvhdmi.c, and does not represent a real codec.
There are no behavior changes to intelhdmi. However nvhdmi made several
changes when copying code out of intelhdmi, which are all reverted in
this patch. Wei Ni confirmed that the reverted code actually works fine.
Tested-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* git://git.kernel.org/pub/scm/linux/kernel/git/lethal/sh-2.6: (26 commits)
sh: Convert sh to use read/update_persistent_clock
sh: Move PMB debugfs entry initialization to later stage
sh: Fix up flush_cache_vmap() on SMP.
sh: fix up MMU reset with variable PMB mapping sizes.
sh: establish PMB mappings for NUMA nodes.
sh: check for existing mappings for bolted PMB entries.
sh: fixed virt/phys mapping helpers for PMB.
sh: make pmb iomapping configurable.
sh: reworked dynamic PMB mapping.
sh: Fix up cpumask_of_pcibus() for the NUMA build.
serial: sh-sci: Tidy up build warnings.
sh: Fix up ctrl_read/write stragglers in migor setup.
serial: sh-sci: Add DMA support.
dmaengine: shdma: extend .device_terminate_all() to record partial transfer
sh: merge sh7722 and sh7724 DMA register definitions
sh: activate runtime PM for dmaengine on sh7722 and sh7724
dmaengine: shdma: add runtime PM support.
dmaengine: shdma: separate DMA headers.
dmaengine: shdma: convert to platform device resources
dmaengine: shdma: fix DMA error handling.
...
The headphone detect and charger are using the IRQ numbers so need
to take account of irq_base with the genirq conversion. I obviously
picked the wrong system for initial testing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
Rename for_each_bit to for_each_set_bit in the kernel source tree. To
permit for_each_clear_bit(), should that ever be added.
The patch includes a macro to map the old for_each_bit() onto the new
for_each_set_bit(). This is a (very) temporary thing to ease the migration.
[akpm@linux-foundation.org: add temporary for_each_bit()]
Suggested-by: Alexey Dobriyan <adobriyan@gmail.com>
Suggested-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: Russell King <rmk@arm.linux.org.uk>
Cc: David Woodhouse <dwmw2@infradead.org>
Cc: Artem Bityutskiy <dedekind@infradead.org>
Cc: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Currently during pop/click debug we're inserting a delay both after
every log message we generate and at explicit points in the sequence,
slowing things down even further than they need to be especially when
many writes get coalesced by the sequence generation code.
Remove the per-printk delay and ensure that we have explicit delays
where we say we want them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Commit eaa9b3a748 introduced the following
uninitialized warning:
sound/pci/hda/patch_realtek.c: In function 'set_capture_mixer':
sound/pci/hda/patch_realtek.c:4928: warning: 'pin' is used uninitialized in this function
sound/pci/hda/patch_realtek.c:4918: note: 'pin' was declared here
It appears indeed that 'pin' needs to be initialized to 0.
Signed-off-by: Frederik Deweerdt <frederik.deweerdt@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://launchpad.net/bugs/523953
The OR has verified that position_fix=1 is necessary to work around
errors on his machine.
Reported-by: MMarking
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sample rate setting is done with a 4-byte long class request that
addresses the interface.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the parser to correctly handle v2 descriptors with multiple
format bits set.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation for USB audio 2.0 support, change the audioformat
structure so that it uses a bitmask to specify possible formats.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_usb_substream::format field actually contains the index of the
current alternate setting, so rename it to altset_idx to avoid
confusion.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up the usb audio driver by factoring out a lot of functions to
separate files. Code for procfs, quirks, urbs, format parsers etc all
got a new home now.
Moved almost all special quirk handling to quirks.c and introduced new
generic functions to handle them, so the exceptions do not pollute the
whole driver.
Renamed usbaudio.c to card.c because this is what it actually does now.
Renamed usbmidi.c to midi.c for namespace clarity.
Removed more things from usbaudio.h.
The non-standard drivers were adopted accordingly.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename snd-usb-lib to snd-usbmidi-lib as MIDI functions are the only
thing it actually contains. Introduce a new header file to only declare
these functions.
Introduced usbmixer.h for all functions exported by usbmixer.c.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As part of the USB audio code cleanup, move the non-standard ua101
driver out of the way.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While trying to compile jazz16 isa sound driver on alpha (2.6.33+git), I
found a compile failure in jazz16.c (udelay is unknown). Fix it by
including delay.h.
Signed-foo-by: Meelis Roos <mroos@linux.ee>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The reorgs of the Samsung headers have moved the GPIO bank definitions
from plat/ to mach/ - the IIS driver needs to be updated to take care
of this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
BugLink: https://launchpad.net/bugs/530346
The OR has verified that position_fix=1 is necessary to work around
errors on his machine.
Reported-by: Tom Louwrier
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now two modules require hda_eld.o, so we need to put it to the common
place instead of building into two individual modules.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Report the current FIFO depth when delay is queried. The FIFO is only
16 frames deep so the latency will be at most a couple of miliseconds
(and we tend to end up reporting zero most of the time) but it may
help some applications.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Support nvidia MCP89 and GT21x 8ch hdmi audio.
Add some eld support.
Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support max codecs to 8 for nvidia hda controller.
Change AZX_MAX_CODECS to 8, and add
"#define AZX_DEFAULT_CODECS 4" for default driver.
Set azx_max_codecs to 8 for nvidia controller.
Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If getpaths() returned an odd number this would be a buffer under-run and an
endless loop. It turns out that getpaths() can only return even numbers, but
let's make it easy for people auditing code. With the new code you don't
need to look at getpaths().
This silences a smatch warning.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the original code the condition was always true (hopefully) because
WM8776_HPLVOL is zero.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Master Control port (MC) is available as the last
PnP resource (OPT005). Use this value instead fo guessing.
Also, add some comments to the code.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't touch the variable 'reg' to construct the value for the actual SPI
transport. This variable is again used to access the driver's register
cache, and so random memory is overwritten.
Compute the value in-place instead.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The codec_dai needs to be shutdown should the machine startup fails.
This patch adds another bailout tag for that case and rename the tag
for configuration failures.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8960 headphone outputs can be run in capless mode with OUT3
used to drive a pseudo ground for the headphone drivers. In this
mode the mono mixer is not used, the mixer should be turned on
in concert with the headphone output drivers and the device bias
levels are managed differently.
Also tweak the existing bias management to remove the use of active
discharge while we're at it since that's often audible.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Avoids machine files having to peer into sound/soc which is a bit
rude and icky.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The driver name gets used by dev_() logging so use something a bit
more idiomatic.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The delay callback can be used by the core to query the delay
on the dai caused by FIFO or delay in the platform side.
In case if both CPU and CODEC dai has FIFO the delay reported
by each will be added to form the full delay on the chain.
If none of the dai has FIFO, than the delay will be kept as
zero.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Create a soc level wrapper for pcm_pointer callback.
This will facilitate the soc level handling of different
HW buffers in the audio path.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
My editor removes the tailing spaces, which causes problems when
changing the soc-core.c
Removing the space.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Initial support for audio using the 1133-EV1 audio and PMIC module for
the i.MX31ADS. Currently only playback is supported, and the FIQ DMA
driver has performance problems at higher sample rates which cause
audible dropouts.
This driver is based heavily on an out of tree one written by Liam
Girdwood.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch works around misbehaviour of Creative Creative VF0470 Live Cam
which reports 16 kHz sample rate for audio capture while actually producing
8 kHz stream.
Signed-off-by: Arseniy Lartsev <arseniy@fizlesh.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some code that is no longer needed now that the relevant parts of
the driver have been tested.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
sound/usb/caiaq/midi.h:6: ERROR: "foo* bar" should be "foo *bar"
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/oss/v_midi.h:5: ERROR: code indent should use tabs where possible
sound/oss/v_midi.h:7: ERROR: trailing whitespace
Signed-off-by: Andrea Gelmini <andrea.gelmini@gelma.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In 2.6.33 ACL269 unsol event handler was changed to look up the pre-defined
pins, but the headphone pins aren't defined properly in each quirk.
This patch adds the missing definitions, and fixes the speaker auto-mute
regression on some ASUS (and possibly other) laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
i8253_lock needs to be a real spinlock in preempt-rt, i.e. it can
not be converted to a sleeping lock.
Convert it to raw_spinlock and fix up all users.
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Ralf Baechle <ralf@linux-mips.org>
Acked-by: Dmitry Torokhov <dmitry.torokhov@gmail.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Cc: Jens Axboe <jens.axboe@oracle.com>
LKML-Reference: <20100217163751.030764372@linutronix.de>
Separate SH DMA headers into ones, commonly used by both drivers, and ones,
specific to each of them. This will make the future development of the
dmaengine driver easier.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Paul Mundt <lethal@linux-sh.org>
Now most (if not all) PXA platforms have been switched to the new MFP
API, it's rather safe to remove these unnecessary pxa_gpio_mode() calls
in pxa2xx-ac97-lib.c now.
Cc: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
This is really pxa27x specific and should be kept in pxa27x.c. With this
newly introduced function, the original set_resetgpio_mode() is deprecated.
Cc: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
MFP registers are saved and restored by the mfp sys_device before all
other platform devices, and it is unnecessary here.
Cc: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (252 commits)
ASoC: Check progress when reporting periods from i.MX FIQ handler
ASoC: Remove a unused variables from i.MX FIQ runtime data
ALSA: hda - Add/fix ALC269 FSC and Quanta models
ALSA: hda - Add ALC670 codec support
OMAP4: PMIC: Add support for twl6030 codec
ALSA: hda - remove unnecessary msleep on power state transitions
usb/gadget/{f_audio,gmidi}.c: follow recent changes in audio.h
ASoC: fsi: Modify over/under run error settlement
ASoC: OMAP4: Add McPDM platform driver
ASoC: OMAP4: Add support for McPDM
ASoC: OMAP: data_type and sync_mode configurable in audio dma
ALSA: hda - Add missing description in HD-Audio-Models.txt
ALSA: add support for Macbook Air 2,1 internal speaker
ALSA: usbaudio: consolidate header files
ALSA: usbmixer: bail out early when parsing audio class v2 descriptors
ALSA: usbaudio: implement basic set of class v2.0 parser
ALSA: usbaudio: introduce new types for audio class v2
ALSA: usbaudio: parse USB descriptors with structs
ALSA: hda - enable snoop for Intel Cougar Point
ALSA: hda - Remove identical definitions for macmini3 model
...
Add support for the Edirol UA-1000 to the UA-101 driver.
Both devices behave the same, so we just have to shuffle around some
interface numbers and name strings.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Machine driver for DB1200 AC97 and I2S audio systems, intended as a proper
reference asoc machine for Alchemy-based systems. AC97/I2S can be selected
at boot time by setting switch S6.7.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Cc: Linux-MIPS <linux-mips@linux-mips.org>
Cc: alsa-devel@alsa-project.org
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
DMA can only be done from physical addresses; move the "virt_to_phys"
source/destination buffer address translation from the dbdma queueing
functions (since the hardware can only DMA to/from physical addresses)
to their respective users.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
Remove dbdma compat macros, move remaining users over to default
queueing functions and -flags.
(Queueing function signature has changed in order to give
a build failure instead of silent functional changes due
to the no longer implicitly specified DDMA_FLAGS_IE flag)
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
If we are to have a snd_soc_dai i.e, cpu_dai and codec_dai, shared among two
or more dai_links we need to log the number of active users of the dai.
For that, we change semantics of the snd_soc_dai.active flag from indicator
to reference counter.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'next-devicetree' of git://git.secretlab.ca/git/linux-2.6: (41 commits)
of: remove undefined request_OF_resource & release_OF_resource
of/sparc: Remove sparc-local declaration of allnodes and devtree_lock
of: move definition of of_chosen into common code.
of: remove unused extern reference to devtree_lock
of: put default string compare and #a/s-cell values into common header
of/flattree: Don't assume HAVE_LMB
of: protect linux/of.h with CONFIG_OF
proc_devtree: fix THIS_MODULE without module.h
of: Remove old and misplaced function declarations
of/flattree: Make the kernel accept ePAPR style phandle information
of/flattree: endian-convert members of boot_param_header
of: assume big-endian properties, adding conversions where necessary
of: use __be32 for cell value accessors
of/flattree: use OF_ROOT_NODE_{SIZE,ADDR}_CELLS DEFAULT for fdt parsing
of/flattree: use callback to setup initrd from /chosen
proc_devtree: include linux/of.h
of: make set_node_proc_entry private to proc_devtree.c
of: include linux/proc_fs.h
of/flattree: merge early_init_dt_scan_memory() common code
of: add 'of_' prefix to machine_is_compatible()
...
Currently the i.MX FIQ handler is reporting periods as elapsed based
purely on a timer running in the CPU. This means that any clock
mismatch between the CPU and the audio subsystem can result in the
status reported to applications drifting away from the actual status
of the hardware. This is particularly likely at present since the
SSI driver is only capable of operating in slave mode so it's very
likely that the interface will be clocked from a different source.
Instead check the offset reported by the FIQ and only notify when we
have transferred at least one period, re-firing the timer if we didn't
do so. Also factor out the calculation of the timer expiry time for
make it a bit easier to experiment with.
Note that this only improves the situation, problems can still be
triggered.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Specify proper quirk models for FSC and Quanta machines with ALC269 codec.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Fixed alc_subsystem_id( ) typo and add new function.
- !(ass & 0x100000)) ==> Delete this check. It is unnecessary check.
- Add porti
- ALC670 support
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will save ~15ms boot time.
The first 10ms sleep was introduced in commit d2595d86e5 for (buggy)
Cxt codecs, so better to limit the sleep to the problem hardware.
For the second 10ms sleep, the HDA spec says:
Power State[1:0]:
00: Node Power state (D0) is fully on.
01: Node Power state (D1) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms, excepting analog pass through circuits (e.g., CD analog
playback) which must remain fully on.
10: Node Power state (D2) allows for (does not require) the lowest possible power consuming state from which it
can return to the "fully on" state (D0) within 10 ms. For modems, this is the "wake on ring" power state.
11: Node Power state (D3) allows for (does not require) lowest possible power consuming state under software
control. Note that any low power state set by software must retain sufficient operational capability to properly
respond to subsequent software Power State command.
So 10ms is actually the max wait time. It should be safe to
remove/reduce it and rely on the loop of 1ms-sleeps.
CC: Marc Boucher <marc@linuxant.com>
CC: Arjan van de Ven <arjan@linux.intel.com>
Signed-off-by: Zhang Rui <rui.zhang@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add ASoC interface for OMAP McBSP2 and McBSP3 sidetones.
Signed-off-by: Ilkka Koskinen <ilkka.koskinen@nokia.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Tony Lindgren <tony@atomide.com>
In current FSI driver, playback function cares only overrun,
and capture function cares only underrun.
But playback function should had cared about underrun,
and capture function should had cared about overrun too.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McPDM platform driver is configured to use sDMA in order to transfer
to/from memory. Support for interfacing with ABE will be added later.
McPDM dai currently supports up to 4 downlink channels and 2 uplink
channels simultaneously, as well as 88.2 and 96 KHz, and a sample
size of 32 bits.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McPDM is the interface between Phoenix audio codec
and the OMAP4430 processor. It enables data to be transfered
to/from Phoenix at sample rates of 88.4 or 96 KHz.
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya <x0080101@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow client drivers to set the data_type (16, 32) and the
sync_mode (element, packet, etc) of the audio dma transferences.
McBSP dai driver configures it for a data type of 16 bits and
element sync mode.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for Macbook Air 2,1 (late 2008) internal speaker and
headphones. Create a "mba21" model for snd-hda-intel.
Signed-off-by: Reimundo Heluani <rheluani@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the definitions from linux/usb/audio.h all over the ALSA USB audio
driver and add some missing definitions there as well.
Use the endpoint attribute macros from linux/usb/ch9 and remove the own
things from sound/usb/usbaudio.h.
Now things are also nicely prefixed which makes understanding the code
easier.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is just a quick hack that needs to be removed once the new units
defined by the audio class v2.0 standard are supported.
However, it allows using these devices for now, without mixer support.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a number of parsers for audio class v2.0. In particular, the
following internals are different and now handled by the code:
* the number of streaming interfaces is now reported by an interface
association descriptor. The old approach using a proprietary
descriptor is deprecated.
* The number of channels per interface is now stored in the AS_GENERAL
descriptor (used to be part of the FORMAT_TYPE descriptor).
* The list of supported sample rates is no longer stored in a variable
length appendix of the format_type descriptor but is retrieved from
the device using a class specific GET_RANGE command.
* Supported sample formats are now reported as 32bit bitmap rather than
a fixed value. For now, this is worked around by choosing just one of
them.
* A devices needs to have at least one CLOCK_SOURCE descriptor which
denotes a clockID that is needed im the class request command.
* Many descriptors (format_type, ...) have changed their layout. Handle
this by casting the descriptors to the appropriate structs.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds some definitions for audio class v2.
Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have
different numerical representations in both standards, so there is need
for a _V1 add-on now. usbmixer.c is changed accordingly.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In preparation of support for v2.0 audio class, use the structs from
linux/usb/audio.h and add some new ones to describe the fields that are
actually parsed by the descriptor decoders.
Also, factor out code from usb_create_streams(). This makes it easier to
adopt the new iteration logic needed for v2.0.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables snoop, eliminating static during playback.
This patch supersedes the previous Cougar Point audio patch.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the card->codec on soc_resume to detect if the soc
device is properly initialized.
If the card->codec is NULL, than do not continue the resume
operation, since the device is not initialized properly.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In order for having snd_soc_dais shared among two or more dai_links,
remove the relatively global runtime field from the struct snd_soc_dai
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently only the atmel driver make use of snd_soc_dai.runtime field.
If the dais are to be shared among two or more dai_links, the field
must be got rid of.
So, in atmel driver reach the substream via dai_link->pcm so as to
not depend of snd_soc_dai.runtime field.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Passing pointer to relevant dai_link provides easier reach to the
ASoC tree in suspend/resume of snd_soc_platform. It also provides
direct access to the dai at the other end of the dai_link.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a headphones-only quirk for the Fujitsu Siemens D1289.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Marc Haber <mh+alsa201002@zugschlus.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
With PulseAudio and an application accessing an input device like `gnome-volume-manager` both have high CPU load as reported in [1].
Loading `snd-hda-intel` with `position_fix=1` fixes this issue. Therefore add a quirk for ASUS M2V-MX SE.
The only downside is, when now exiting for example MPlayer when it is playing an audio file a high pitched sound is outputted by the speaker.
$ lspci -vvnn | grep -A10 Audio
20:01.0 Audio device [0403]: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) [1106:3288] (rev 10)
Subsystem: ASUSTeK Computer Inc. Device [1043:8290]
Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx-
Latency: 0, Cache Line Size: 64 bytes
Interrupt: pin A routed to IRQ 17
Region 0: Memory at fbffc000 (64-bit, non-prefetchable) [size=16K]
Capabilities: <access denied>
Kernel driver in use: HDA Intel
[1] http://sourceforge.net/mailarchive/forum.php?thread_name=1265550675.4642.24.camel%40mattotaupa&forum_name=alsa-user
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the standard unsol_event callback with each setup callback for
IntelMac models with Realtek ALC885 codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.edge.launchpad.net/ubuntu/+source/linux/+bug/343989
Add a model quirk for the NVIDIA based Macmini hardware, aka Macmini 3,1. The
pinout is almost identical to the mb5 quirk, except for no microphone and
the line-in mixer controls being on a different index. Everything works in
2ch mode, but as I am not sure what needs to be changed for 6ch mode, or
whether the Mac Mini's chip supports 6ch mode, I have simply duplicated
the code from the mb5 quirk for the mac mini chmode management. The new
model parameter for this quirk is "macmini3".
Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/524948
The OR has verified that the existing model=laptop-eapd quirk does not
function correctly but instead needs model=3stack. Make this change
so that manual corrections to module-init-tools file(s) are not
required.
Reported-by: Lasse Havelund <lasse@havelund.org>
CC: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_cs46xx_codec_reset() bypassing the register cache, so as to not
clobber the cached register value during resume.
Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Disable the amplifiers for the headset outputs, and do not select
routings by default to the headset outputs.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes a division by zero error in the irq handler.
There is a small window between the hw_params() callback and when
runtime->frame_bits is set by ALSA middle layer. When another substream is
already running, if an interrupt is delivered during that window the irq
handler calls pcm_pointer() which does a division by zero. The patch below
makes the irq handler skip substreams that are initialized but not started
yet. Cc to Clemens Ladisch because he proposed an alternate fix.
For more information, please read the original thread in the linux-kernel
mailing list: http://lkml.org/lkml/2010/2/2/187
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
OSCSET calculation was not correct in case of 44.1KHz
sampling rate.
With small adjustment both 48 and 44.1 KHz calculation
now gives the correct value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In repeated playback the FIFOFLUSH bit remained set, and
never has been cleared.
Clear it during the setup phase.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide a sysfs file allowing userspace to inspect and change the
pmdown_time setting at runtime.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Make the pmdown_time a per-card setting rather than a global one,
initialised before the card initialisation runs. This allows cards
to override the default setting if it makes sense to do so (for
example, due to an unavoidable pop).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The usbmixer proc file contains mapping between ALSA control API and
USB mixer control units. The purpose of this file is for debugging
and a problem diagnostics.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Here's a patch that adds MIDI support through USB for one of the Access
Music synths, the VirusTI.
The synth uses standard USBMIDI protocol on its USB interface 3, although
it does signal "vendor specific" class. A magic string has to be sent on
interface 3 to enable the sending of MIDI from the synth (this string was
found by sniffing usb communication of the Windows driver). This is all
my patch does, and it works on my computer.
Please note that the synth can also do standard usb audio I/O on its
interfaces 2&3, which already works with the current snd-usb-audio driver,
except for the audio input from the synth. I'm going to work on it when I
have some time.
Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> (cosmetics, list terminator)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Extend the list of devices whose firmware does not expect more than one
USB MIDI packet in one USB packet.
bug report: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3752
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The MSI blacklist entry for ASUS mobo added in the commit
8ce28d6abf was based on the alsa-info
output wrongly posted. Fix the id to the right one now.
Reported-by: Sid Boyce <sboyce@blueyonder.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds rearranges parts of the initialization code and adds
suspend and resume callbacks.
This patch adds suspend and resume callbacks.
It also rearranges parts of the initialization code so it can be
used in both the first initialization (when the module is loaded we
also have to load default settings) and the resume callback (where
we have to restore the previous settings).
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the controls init code outside the init_hw() function because is must
not be called during resume.
This patch moves the code that initializes the card's controls with
default valued from the init_hw() function into a separated
set_mixer_defaults() function (one for each of the 16 supported
cards). This change is necessary because during resume we must
resurrect the hardware without losing the previous
settings. set_mixer_defaults() must be called only once when the
module is loaded.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements a simple cache for the firmware files when CONFIG_PM is defined.
This patch changes get_firmware(), free_firmware() and adds
free_firmware_cache(). The first two functions implement a very
simple cache and the latter is used to actually release all the stored
firmwares when the module is unloaded.
When CONFIG_PM is not enabled those functions act as before, that is
free_firmware() releases the firmware immediately and
free_firmware_cache() does nothing.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changes the way the firmware is passed through functions.
When CONFIG_PM is enabled the firmware cannot be released because the
driver will need it again to resume the card.
With this patch the firmware is passed as an index of the struct
firmware card_fw[] in place of a pointer. That same index is then used
to locate the firmware in the firmware cache.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add patch for the Conexant 5066 HDA codec to support the Lenovo IdeaPad U150
Signed-off-by: Greg Alexander <greigs@galexander.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the zero-division warning message with WARN_ON_ONCE() per the
advice by Linus. This shouldn't happen, but if it happens, it's
possible that the bug happens often due to buggy IRQs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM2000 is a low power, high quality handset receiver speaker
driver with Wolfson myZone™ Ambient Noise Cancellation (ANC). It
provides enhanced voice communication quality in a noisy environment
if the handset acoustics are designed appropriately.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Audio on Migo-R cannot work if CONFIG_SH_DMA_API=y, but compilation should not
break anyway.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch expands the omap3beagle sound soc for the
beagle board clone DevKit8000.
Signed-off-by: Thomas Weber <weber@corscience.de>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
USB devices tends to represent dB ranges in different way than ALSA expects.
Add possibility to override these values and add guessed values for
SoundBlaster MP3+.
Also rename 'Capture Input Source' control to 'Capture Source' for
SoundBlaster MP3+ and Extigy.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The serial interface (TDM/I2S) for the audio block have been
constantly enabled.
Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so
the interface is only enabled, when there is a need for it.
For example when only the analog loopback is enabled, there
is no need to keep the serial interface active.
I have added the persons who contributed to the Voice path
of twl4030 codec driver, so they might have the ability
to test this patch, and send an update for the Voice path,
if it is necessary
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable the bulk regulators at probe time so we can safely disable them
again when going to suspend without confusing the reference counter.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds sound support for Phytec PhyCORE / PhyCARD
modules in AC97 mode.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by
zero
for as-yet unknown reasons. A simple check for zero prevents it, though
the problem that causes it remains. Since the workaround is harmless and
won't affect anyone except victims of this bug, it should be safe;
moreover,
because this crash can be triggered by a user-mode application, there are
denial of service implications on the systems affected by the bug without
the patch.
Signed-off-by: Jody Bruchon <jody@nctritech.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
machine is compatible is an OF-specific call. It should have
the of_ prefix to protect the global namespace.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: Michal Simek <monstr@monstr.eu>
In particular, several occurances of funny versions of 'success',
'unknown', 'therefore', 'acknowledge', 'argument', 'achieve', 'address',
'beginning', 'desirable', 'separate' and 'necessary' are fixed.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Joe Perches <joe@perches.com>
Cc: Junio C Hamano <gitster@pobox.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
Use DEFINE_PCI_DEVICE_TABLE() to make PCI device ids go to
.devinit.rodata section, so they can be discarded in some cases,
and make them const.
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The previous commit caused a regression on HP laptops with 92HD83x/88x
codecs. The default polarity of mute-LED GPIO is inverted on these
devices.
Reference: Novell bnc#578190
https://bugzilla.novell.com/show_bug.cgi?id=578190
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have now a better mute-LED GPIO detection, and no need to assign the
values statically per model option.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge the mute-LED status callback function for both IDT 92HD7x and 8x
codecs to one function. Also it's changed to check all DACs, and called
in the initialization to sync with the current status.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The GPIO pin number for the mute LED control on HP laptops can be
determined more easily by checking the number of available GPIO pins
of the codec chip. On a small package with up to 3 GPIOs, GPIO 0 is
used while GPIO 3 is used for others.
This fixes the missing mute GPIO for some HP laptops with new codecs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Pandora's external DAC is connected to VSIM TWL4030 supply, so let's
start switching it too to save more power.
Also DAC got it's own DAPM handler.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Pandora's external DAC is using 256*Fs output from the TWL4030
codec, and TWL4030 needs to have APLL enabled for it's 256*Fs
output to function.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The volume register is from 0..0x7f and 0..0x1a range is mute.
Also, fix mute combining in wm_vol_put(). The wrong behaviour was
noticed by Peter Christensen.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This renames the interrupt name in /proc/interrupt.
HDA Intel -> hda_intel
This also eliminates space from the name, probably helping some
parsers.
Don't think anybody depends on this name in userspace
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My sound codec seems sometimes (very rarely) to omit interrupts (ALC268)
However, interrupt mode still works.
Thus if we get timeout, poll the codec once.
If we get 3 such polls in a row, then switch to polling mode.
This patch is maybe an bandaid, but this might be a workaround for hardware bug.
Signed-off-by: Maxim Levitsky <maximlevitsky@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I found that the sampling rate locking setting of the ice1712 sound driver
was only half-respected : when the driver was locked to, let's say, 44100Hz,
and a usermode app was requesting 48000Hz playback, the request was succesful
although the soundcard would continue to run at 44100Hz.
Here's a patch that will make those requests to fail.
Signed-off-by: Sebastien Alaiwan <sebastien.alaiwan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After hours of debugging, I finally found the reason why some source
and runtime combination does not work. The PTP (page table pages)
address must be aligned. I am not sure how much, but alignment to
PAGE_SIZE is sufficient. Also, use ALSA's page allocation routines
to ensure proper virtual -> physical address translation.
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- Add support for ALC665
- Add more ASUS model
- Modify common patch for ALC272 ALC273 ALC661 ALC662 ALC663 ALC665
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add new models ALC269VB_AMIC ALC269VB_DMIC
- Add alc269vb_laptop_dmic_setup
The record source index Dmic is 0x6 for ALC269VB.
- Change eeepc words for ALC269
- Modify init_verb tables of patch_alc269 patch_alc662 patch_alc882
- Modify common patch for ALC270 ALC269VB ALC275
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The module unloading path had several problems:
- it freed up the private structure twice
- it freed up the codec structure, which was allocated as part
of the private structure
- it did not freed up the reg_cache
- it did not unregistered the dais and the codec
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8912 is a DAC only device register compatible with the WM8904
CODEC with ADC portions omitted. Support it within the WM8904 driver
based on the configured I2C device name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Handle the output PGAs as part of the output powerup since they can
never be powered separately and reorder things so that we remove the
output shorts after both line and headphone outputs have been brought
up, minimising the opportunity for any issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
As well as disabling the biases of the CODEC the drop into BIAS_OFF will
also disable all the regulators powering the CODEC, allowing even greater
power savings on appropriately configured systems.
Since the regulator API does not currently provide notification when
regulators are disabled we assume that this always happens when we stop
using the regulators. Once 2.6.34 is merged this code can be optimised
to only sync the cache when power was actually removed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
While the regulators are disabled we cache all register writes.
Currently we assume that the regulator disable actually takes
effect, after the merge with the regulator tree in 2.6.34 the
regulator API will be able to notify us if the power is actually
removed (due to constraints or regulator sharing it may not be).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
At the minute the regulators are simply enabled for the entire
lifetime of the device.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Add a bit to the CODEC structure indicating if a cache sync is required.
By default this will be set if a cache only write is done to a soc-cache
register cache. This allows us to avoid syncing the cache back after
using cache only writes if there were no changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This patch fixes docking output support for IDT 92HD81/83/88 family codecs.
Typically one of ports 0xE or 0xF is used for docking output, while only
port 0xF is common on all the three codec families. We don't want the
pin to select the analog mixer here.
Signed-off-by: Charles Chin <Charles.Chin@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the soc-cache code will always write to the device, meaning
that we need the device to be powered and active at pretty much all
times the system is active. Allowing cache only writes lays some
groundwork for future enhancements to allow devices to be put into a
full off state when the audio subsystem is idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
ctrl_outl() has become void at some point, which breaks compilation of fsi.c.
Make writing functions void, as their output is anyway not evaluated, and use
__raw_writel and __raw_readl instead of deprecated ctrl_outl and ctrl_inl
respectively.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a cleanup for the dummy driver. The model kernel module parameter
is introduced to select the soundcard emulation.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Unconditionally save the register states when suspending and restore
them again at resume time. Register contents were not preserved over
suspend, and hence the driver takes false assumptions about them.
The clock must be enabled to access the register block.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
String constants that are continued on subsequent lines with \
are not good.
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dependency on MFD_WM8994 rather than I2C went awry.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Instead of padding with blanks and printing "number=0x a", print
"number=0x0a".
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8994 is a highly integrated ultra-low power hi-fi audio subsystem
designed for smartphones and other portable devices rich in multimedia
features. It provides advanced digital mixing facilities enabling low
power high quality interconnection of CPU, baseband and other audio
sources through flexible digital and analogue routing, and integrates
a class W headphone driver and stereo class D speaker drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use a two code correction for optimal performance.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Perform DC servo offset calibration using a series update sequence
rather than startup update sequence, tuning the configuration of the
WM8993 DC servo to make best use of this.
Also introduce currently unused data allowing us to correct for
any systematic errors in the DC servo calibration results and an
alternative startup path for the headphone output which performs
better with some chip revisions. The alternative setup sequence is
enabled for WM8993.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
String constants that are continued on subsequent lines with \
are not good.
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In case, if OPCLK is not used, and PLL is used for driving the codec, the
choice of PLL output frequency could result in a needlessly imprecise
system clock frequency. Use an iterative process to select a precise
configuration.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow TLV blocks that do not have any values; the smallest possible TLV
is an empty container or one where the information is only in the tag.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Creating a control with TLV_COMMAND access was not possible because
snd_ctl_new1() forgot to include it in the mask of allowable access
bits.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
wm8974 and wm8978 codec drivers control DAC and ADC oversampling rates in their
.set_clkdiv() methods, which is wrong, because these are simple boolean
switches and not clock dividers. Move these bits to sound controls. Also remove
manual configuration of the MCLK divider in wm8978, since it is configured
automatically.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tests showed, that bit 6 of the WM8978_POWER_MANAGEMENT_2 register of wm8978
affects codec clocks. Being useless for suspend / resume, it cannot be used in
bias-level control either. Remove this bit handling.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for audio on sh7722-based Migo-R boards, using SIU and wm8978
codec, recording via external microphone and playback via headphones are
implemented.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's more robust when references are provided in control names
rather than numid.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 761c9d45 (ASoC: Fix build of OMAP sound drivers) changes
CONFIG_MACH_OMAP3517EVM -> CONFIG_SND_OMAP_SOC_OMAP3517EVM in the
Makefile. Whereas the config option defined in Kconfig is
SND_OMAP_SOC_AM3517EVM. Because of this, ASoC driver for AM3517
was not getting compiled.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit e9ff5eb2 (Fixing infinite loop in resume path) uses wrong AIC23
register in resume function because of which register writes happen
on some non-existing registers.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In struct device_node, the phandle is named 'linux_phandle' for PowerPC
and MicroBlaze, and 'node' for SPARC. There is no good reason for the
difference, it is just an artifact of the code diverging over a couple
of years. This patch renames both to simply .phandle.
Note: the .node also existed in PowerPC/MicroBlaze, but the only user
seems to be arch/powerpc/platforms/powermac/pfunc_core.c. It doesn't
look like the assignment between .linux_phandle and .node is
significantly different enough to warrant the separate code paths
unless ibm,phandle properties actually appear in Apple device trees.
I think it is safe to eliminate the old .node property and use
phandle everywhere.
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
Acked-by: David S. Miller <davem@davemloft.net>
Tested-by: Wolfram Sang <w.sang@pengutronix.de>
Acked-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
This patch adds HP mute LED support for IDT 92HD81/3 family of the codecs.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The amp used for the mic input on HP Compaq F700 with Cxt5051 codec
has no multiple inputs, thus its index should be 0 instead of 1.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In hda_codec.c, it has define
"[HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 },",
it support up to device 9 for HDMI.
But in hda_intel.c, it only define AZX_MAX_PCMS as 8.
So if it have 4 hdmi codecs, when run azx_attach_pcm_stream(),
it will show error "Invalid PCM device number 8", and "... number 9",
and return "-EINVAL".
We should change the AZX_MAX_PCMS to 10.
Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also renames a few things to make volumes and switches match up in
alsamixer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The version isn't being updated or used, the kernel revision
tracking is enough.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Change the legacy default register configuration, which left some
internal components on.
Now we have either DAPM, or other ways to control these bits,
so there is no need to enable them by default.
The affected parts:
Disable ADCL and ADCR
Disable ARXL2 and ARXR2 analog PGA (playback)
Disable APLL by default
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fsi_master_xxx function should be protected by spin lock,
because it are used from both FSI-A and FSI-B.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow the override of vendor-id, subsystem-id, revision-id and chip name
via patch loading. Updated the document, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8978 codec from Wolfson Microelectronics is very similar to
wm8974, but is stereo and also has some differences in pin configuration
and internal signal routing. This driver is based on wm8974 and takes
the differences into account.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The board supports both GPIO sets for the AC97 bus and the analogue
outputs can be switched between this and the WM8580 so add some
comments saying what the setup the standard kernel expects is.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since, we have generic AC97 controller driver and all the machines
have moved to that, there is no need for old s3c2443-ac97.c to exist.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Switch to use s3c-ac97.c AC97 controller driver.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Switch to use s3c-ac97.c AC97 controller driver.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the common machine driver for SMDKs that
have a WM9713 codec attched to the AC97 controller.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the AC97 controller driver for Samsung SoCs that have one.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Sid Boyce reported that his machine locks up without enable_msi=0 option.
This looks like another ASUS mobo with Nvidia combo.
Reported-by: Sid Boyce <sboyce@blueyonder.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The code in pcm_lib updating runtime->hw_ptr_interrupt expects
that runtime->boundary is divisible with runtime->period_size.
Thanks are going to Clemens Ladisch for the notice.
Fix the runtime->boundary calculation using buffer_size * period_size
as base and find a least common multiple for 32bit platforms when
the expression might overflow.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Barry Song <Barry.Song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Barry Song <Barry.Song@analog.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr
update functions" commit:
"It is possible for the status/delay ioctls to be called when the sound
card's pointer register alreay shows a position at the beginning of the
new period, but immediately before the interrupt is actually executed.
(This happens regularly on a SMP machine with mplayer.) When that
happens, the code thinks that the position must be at least one period
ahead of the current position and drops an entire buffer of data."
Return back the hw_ptr_interrupt variable. The last interrupt pointer
is always computed from the latest hw_ptr instead of tracking it
separately (in this case all hw_ptr checks and modifications might
influence also hw_ptr_interrupt and it is difficult to keep it
consistent).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Sometimes after a suspend-resume cycle, the ALSA application
restarts the stream when resume fails and McASP fails to work
as the clock is not enabled. This patch corrects this bug.
Testes on TI DA850/OMAP-L138 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In azx_stream_start() and azx_stream_stop(),
it use azx_readb/azx_writeb to read/write SIE,
it just enable/disable 8 streams.
But according to the HDA spec, it support 30 streams,
and the new HDA controller will support more then 8
streams. So we should use azx_readl/azx_writel to
read/write SIE.
Signed-off-by: Wei Ni <wni@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make sure that capture DMA doesn't stay enabled after system resume
as that potentially prevents the processor from entering deep sleep
states.
Signed-off-by: Florian Zumbiehl <florz@florz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
tests show frequent suspend/resume(frequent poweroff/on ad1836 internal
components) maybe make ad1836 clock mode wrong sometimes after wakeup.
This patch reset/restore ad1836 clock mode while executing PM, then
ad1836 can always resume to right clock status.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Several SuperH platforms, including sh7722, sh7343, sh7354, sh7367 include
a Sound Interface Unit (SIU). This patch adds DAI and platform / DMA
drivers for this interface.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The COEF setup for model=auto seems problematic on some laptops,
resulting in the silent speaker output. Better to disable it for now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The output of COEF index/value in the proc file for Realtek codecs is
rather useless since the value varies together with the index.
Let's get rid of it again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remember to free the temporary register-cache.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Now one can choose speaker configuration in e.g. PulseAudio mixer
Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP pin (0x16) control has to be changed dynamically depending on
the master volume switch as well as the speaker pin (0x1a). Otherwise
the headphone still sounds with master off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the wrongly set up for SPDIF output on Conexant 5051 codec.
It must point to the audio out widget instead of a pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize the mic ports B & C on Conexant 5051 codec dynamically
according to the mic jack detection, instead of static init arrays.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for Toshiba Satellite M300 with Conexant 5051 codec.
Since the laptop has no port C connection and the pin reports always
the jack sense true, we need to ignore port-C unsol event.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor fixes for HP Compaq Presario F700 quirks with Cxt5051 codec:
- changed the capture mixer elements to the standard name.
- fixed the quirk name string without a space
- sorted the quirk list
- updated the documentation
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SPDIF-input pin 0x1c is muted by default in hardware. Unmute appropriate
pin to get captured samples instead zeros. Tested on Lenovo Thinkstation.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
This provides a small power saving when audio is inactive.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Currently ASoC always maintains the bias of the CODEC while the system
is active. With older mobile CODECs this is required since the outputs
are referenced to a non-zero voltage and enabling or disabling this
voltage without audible pops or clicks in the output takes too long to
do when starting or stopping audio.
As a result of features such as ground referenced outputs and class D
speaker drivers current generation devices are able to power on and off
much more quickly without these system level issues so provide a new
flag idle_bias_off in snd_soc_codec which will cause the core to turn
off the CODEC bias. The distinction between STANDBY and OFF is still
maintained. This is partly for consistency but also allows for
potential future extensions such as per-machine overrides or deferring
the bias removal.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
The same information is now visible via debugfs and with large modern
devices dumping everything to the console can be very resource
intensive, causing more harm than good.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O"
commit. New sleeping queue is introduced to separate user space and kernel
space wake_ups. runtime->nowake is renamed to twake (transfer wake).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Add possibility to configure the burst mode BCLK divider through platform
data structure.
The BCLK divider changes the actual speed of the serial bus in burst mode,
which is faster than the sampling frequency of the running stream.
In this way platforms can experiment with the optimal burst speed without
the need to modify the codec driver itself.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The BCLK divider was not configured in case of mode7.
This leads to unpredictable behavior when switching between FIFO modes.
Configure the BCLK divider depending on the fifo_mode (FIFO is in use,
or FIFO bypass).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the aut-mute setup on HP T5735 with ALC262 codec.
Instead of wrong amp, use pin control toggling for muting the speaker now.
Tested-by: Lee Trager <lee.trager@hp.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codecs disable widgets used for output pins and reserve as vendor-
spec widgets. Thus we need to check the widget type and pin cap before
actually sending SET_EAPD verbs in the auto-configuration mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC259 has a widget NID 0x21 for the output pin, but it wasn't handled
properly in alc268_new_analog_output().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the prefill number of samples as the same as the lower
threshold in mode7.
In this way the codec will read the same amount of data on
startup and during the running playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As snd_seq_timer_set_tick_resolution() is always called with the same
three fields of struct snd_seq_timer, it suffices to give that as the
only parameter.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd
need non-cached behavior more or less, even for the intermediate ring-
buffers.
Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback
that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jarkko Nikula