Clean up Conexant 5047 pareser code:
- Split mixer elements to separate arrays to reduce the duplicated
entires
- Fix mixer element names to the standard ones
- Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event
handler works fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the initial connections of output pins 0x13 and 0x1d for Conexant
5047 codec to point to the mixer amp properly.
Removed unneeded (doubly) verbs from arrays, also removed the unneeded
changing of widget 0x1c, which is now completely unused.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove superfluous verbs from cxt5047_toshiba_init_verbs[].
Also fix comments and minor coding style issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create "Capture Source" control dynamically for Conexant codecs.
If only one capture item is available, don't create such a control
since it's just useless.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of binding volumes, create a virtual master volume for Conexant
codecs. This allows separate HP and speaker volume controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated.
Code tested on HP HDX16 (not tested on HDX18 yet).
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added codec recognition of HP HDX platforms and added support of the
MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE
is needed to use event handling for mute control.
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will break any boards that don't register the AC97 controller
device due to using ASoC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the standard linked list for snd_monitor_file management.
Also, move the list deletion of shutdown_list element into
snd_disconnect_release() (for simplification).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_ctl_add_slave_uncached() function to add a slave element
with volatile controls. The values of normal slave elements are
supposed to be cachable, i.e. they are changed only via the put
callbacks. OTOH, when a slave element is volatile and its values may
be changed by other reason (e.g. hardware status change), the values
will get inconsistent.
The new function allows the slave elements with volatile changes.
When the slave is tied with this call, the native get callback is
issued at each time so that the values are always updated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pxa-regs.h and hardware.h are not intended for use directly in driver
code, remove those unnecessary references.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
1. Driver code where pxa_request_dma() is called will most likely
reference DMA registers as well, and it is really unnecessary
to include pxa-regs.h just for this. Move the definitions into
<mach/dma.h> and make relevant drivers include it instead of
<mach/pxa-regs.h>.
2. Introduce DMAC_REGS_VIRT as the virtual address base for these
DMA registers. This allows later processors to re-use the same
IP while registers may start at different I/O address.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
The timer callbacks are called in the protected status by the lock
of the timer instance, so there is no need for an extra lock in the
PCM substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up and improve snd_pcm_update_hw_ptr*() functions.
snd_pcm_update_hw_ptr() tries to detect the unexpected hwptr jumps
more strictly to avoid the position mess-up, which often results in
the bad quality I/O with pulseaudio.
The hw-ptr skip error messages are printed when xrun proc is set to
non-zero.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a driver for the SPI connected AK4104 S/PDIF transmitter
device. Its features are fairly simple, but as there is need to set up
certain bits in the IEC958 information, this better goes into a real
driver.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@sirena.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On the HT-Omega Claro (halo) sound cards, the headphone amplifier must
be enabled explicitly by setting a GPIO bit.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Removes numbers from the list of features/limitations and makes it
reflect recent changes to the code.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix headphone-detect regression with multiple HP jacks
ALSA: hda - Fix typos in slave controls in patch_sigmatel.c
Add support for true pause and unpause. Without this, mplayer will drop some
audio (less than one second, but still noticeable) when pausing playback.
Remove support for PM suspend and resume from the trigger function, since the
driver doesn't support PM anyway.
Optimize the delay after starting capture. Instead of delaying 1ms, the driver
now polls the hardware. The new delay is shorter by over 90% yet still
effective.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Upgrade the severity of some failure messages from debug level so
they're displayed by default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent set of S3C64xx patches re-added a lot of uses of DBG() that
had previously been removed - revert this so the standard pr_debug()
macro is used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Rob Maris <maris.rob@vdi.de>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add headset jack detection for SDP3430 boards using SoC jack
reporting interface. Headset detection on SDP3430 board is
achieved through TWL4030 GPIO_2 pin.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add a new device tree property for the SSI node: "fsl,ssi-asynchronous". If
defined, the SSI is programmed into asynchronous mode, otherwise it is
programmed into synchronous mode. In asynchronous mode, pin SRCK must be
connected to the same clock source as STFS, and pin SRFS must be connected to
the same signal as STFS. Asynchronous mode allows playback and capture to
use different sample sizes. It also technically allows different sample rates,
but the driver does not support that.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We now support the 64xx series as well as the 24xx series - make sure
people using Kconfig know this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_dapm_switch ends up ends up in dapm_new_mixer() (since a switch
is a special case of a mixer with only one input) but this wasn't
correctly handled in the code.
Also fix the coding style for the switch below while we're here.
Reported-by: Joonyoung Shim <dofmind@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A bit in PXA's SSCR0 register was erroneously named ADC but its name is
in fact ACS (audio clock select).
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enum type for selecting the desired ramp delay for the headset output.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Select the relevant DMA implementation when the
sound driver is selected.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add the initial code to support the S3C64XX I2S hardware using the
s3c-i2s-v2 core code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The S3C2412 I2S (IIS) interface is replicated on further Samsung SoC
parts in a broadly compatible way, so split the common code out into
a core called s3c-i2s-v2.[ch] so that the newer SoCs such as the
S3C6410 can make use of it.
As such, all the original s3c2412 functions are currently being left
with their original names, and will be renamed later in the series.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the Jive's WM8750 codec attached via the S3C2412 IIS.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Assign DACs to HP and speaker before mic-in/line-in shared outputs.
This improves the usability as it results in more intuitive mixer
names.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In stac92xx_auto_fill_dac_nids[], connect to the primary DAC if no
individual DAC is available for each pin. This ensures that the pin
works somehow at least.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create multiple "Headphone" and "Speaker" controls with non-zero index
numbers instead of "Headphone2", etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Improve the parser to pick up more intuitive control names for the
outputs judging from the pin type, instead of fixed names assigned
to channels.
Also, revive the multi-HP workaround since this change fixes the
problem with the multi-HP detection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent changes over the DAC detection mechanism in patch_sigmatel.c
breaks the HP detection on the machines with multiple HP jacks.
It's basically because of the workaround to support the multi-channel
output. Since the HP detection is more important feature, disable
the HP-swap workaroud temporarily.
Reference: Novell bnc#482052
https://bugzilla.novell.com/show_bug.cgi?id=482052
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Headphone Playback ..." appears twice in slave_vols[] and slave_sws[].
They should be "Headphone Playback2 ..."
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Noises can be heard on analog outputs of (some model of) Lenovo
Ideapad due to the hardware problem, and the only workaround right now
is to fix the sample rate to 44.1kHz.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The <mach/audio.h> file needs to be common to both ARCH_S3C2410 and
ARCH_S3C64XX as they share common driver code, so move it to <plat/audio.h>.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the IIS headers to their correct place.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ignore MIDI and PCM events in the interrupt handler until the device
gets initialized properly. Otherwise you may get kernel panic by the
access to uninitialized devices via hotplugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Considering the fact that most cpu_dai or codec_dai are using a same
'snd_soc_dai_ops' for several similar interfaces, 'ops' would be better
made a pointer instead, to make sharing easier and code a bit cleaner.
The patch below is rather preliminary since the asoc tree is being
actively developed, and this touches almost every piece of code,
(and possibly many others in development need to be changed as
well). Building of all codecs are OK, yet to every SoC, I didn't test
that.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mute speaker outputs on headphone insertion for machines that use
3stack-hp model.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When setting WM8510_MCLKDIV the pll was turned off.
When setting pll frequency you got twice the expected freq, because
the code calculated with postscaler of 8, but the hardware divide by 4.
Signed-off-by: Jonas Andersson <jonas@microbit.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add GPIO support to jack reporting framework in ASoC using gpiolib calls.
The gpio support exports two new functions: snd_soc_jack_add_gpios and
snd_soc_jack_free_gpios.
Client drivers using gpio feature must pass an array of jack_gpio pins
belonging to a specific jack to the snd_soc_jack_add_gpios function. The
framework will request the gpios, set the data direction and request irq.
The framework will update power status of related jack_pins when an event on
the gpio pins comes according to the reporting bits defined for each gpio.
All gpio resources allocated when adding jack_gpio pins can be released
using snd_soc_jack_free_gpios function.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent update enabled the model=sony-assamd for all ALC262 with
PCI SSID 104d:90xx. But this includes the VAIO VGN-AR* that has the
primary codec of STAC92xx and the secondary ALC262 as a slave
digital-only codec. For this device, the model=auto must be chosen
to work properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the mic input of HP dv6736 with Conexant 5051 codec chip.
This laptop seems have no mic-switching per jack connection.
A new model hp-dv6736 is introduced to match with the h/w implementation.
Reference: Novell bnc#480753
https://bugzilla.novell.com/show_bug.cgi?id=480753
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Only allow SND_SOC_DAIFMT_CBS_CFS for the playback DAI.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For consistency with 24-bit and 32-bit modes, don't send 16-bit stereo
in one 32-bit transfer. Use 2 slots instead on Zylonite. It should result
in exactly the same behaviour.
Now it is possible to use 16-bit single slot transfers in pxa-ssp, which
are needed for Magician to get two frame clock pulses per sample
(one for each channel).
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Tested-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the UDA1380's interpolator or decimator are set to be clocked from
the WSPLL (which syncs to the WSI signal), the DAI link must be running
to change the interpolator/decimator registers (which include volume
controls and digital mute setting).
* Queue work in the alsa PCM_START .trigger to flush registers
as soon as the link is running. This replaces the .prepare
and .digital_mute callbacks.
* Use the SILENCE override instead of MTM for muting and remove
its alsa control to avoid confusion.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's false positive, but annoying.
sound/pci/hda/hda_codec.c: In function ‘get_empty_pcm_device’:
sound/pci/hda/hda_codec.c:2772: warning: ‘dev’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Add probe_mask default for Toshiba laptop with ALC268
ALSA: hda - Add quirk for new HP xw series
ALSA: hda - Fix digital mic on dell-m4-1 and dell-m4-3
Allow more options to be set/reset via hwdep hint entry.
hp_detect, gpio_mask, gpio_dir, gpio_data, eapd_mask and eapd_switch
can be checked.
For example, to disable hp_detect on the fly,
# echo "hp_detect=0" > /sys/class/sound/hwC0D0/hints
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't create "Analog Loopback" controls as default since these controls
are usually more harmful than useful for normal users.
Only created when "loopback = yes" hint is given.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_get_hint() and snd_hda_get_bool_hint() helper functions
to retrieve a hint value.
Internally, the hint is stored in a pair of two strings, key and val.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This removes a misspelled comment and got rid of superfluous switch
case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When checking for the maximum queue length, we have to take into account
that MAX_QUEUE is measured in milliseconds (i.e., frames) while the unit
of urb_packs is whatever data packet interval the device uses (possibly
less than one frame when using high speed devices).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When storing the channel numbers used by a format, and if the device
happens to support 32 channels, the code would try to store 1<<32 in
a 32-bit value.
Since no valid format can have zero channels, we can use 1<<(channels-1)
instead of 1<<channels so that all the channel numbers that we test for
fit into 32 bits.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Two simple fixes:
1. Use the same pointer for the free_irq() and
the request_irq() calls.
2. A short name of card is appended with '2' or '3'
character depending on a detected chip. Remove
the '2' character from the short name.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't return a fatal error to the driver but continue to probe when
any error occurs at creating PCM streams for each codec.
It's often non-fatal and keeping it would help debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the codec probe instead of returning the error to the driver
when any error occurs at creating the control elements.
The control element conflict can be non-fatal in many cases,
especially if it comes from the digital-only codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the Toshiba probe_mask quirk for 2.6.29 kernel
(commit 38f1df27e3).
In the current tree, the digital-only codec is handled properly so
no codec conflict should occur.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When an ALC268 codec is set up as the digital-only (as found in Toshiba
laptops), it shouldn't contain any beep control that conflict with the
primary codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some Toshiba laptops have another ALC268 codec on slot#3 that conflicts
with the primary codec. The codec#3 is for the digital I/O, and should
be fixed by the driver, but it'd need a bunch of changes.
So, let's fix the probe problem temporarily by setting the default
probe_mask value.
Reference: kernel bugzilla #12735http://bugzilla.kernel.org/show_bug.cgi?id=12735
Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some systems it is desirable for control for DAPM pins to be provided
to user space. This is the case with things like GSM modems which are
controlled primarily from user space, for example. Provide a helper which
exposes the state of a DAPM pin to user space for use in cases like this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added support for scenarios where the Cirrus CS4270 audio codec is slave
to the bitclk and lrclk. Mixed setups are unsupported.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the copyright statements in two of the S3C24XX ASoC files
that have (c) when we require the full word.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Forgot to remove an unused variable.
sound/pci/hda/patch_realtek.c: In function ‘alc882_auto_init_analog_input’:
sound/pci/hda/patch_realtek.c:7018: warning: unused variable ‘vref’
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix num_dmuxes initialization for dell-m4-1 and dell-m4-3 models
of IDT 92HD71bxx codec, which was wrongly set to zero.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Model drivers assume that model_data is zeroed, so we better use
kzalloc() (like we did before when it was allocated together with the
card structure).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the Device IDs for MCP89 HD audio controller.
Removed the IDs of MCP7B cause this chipset had been cancelled.
Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the model=auto to STAC/IDT codecs to use the BIOS default setup
explicitly. It can be used to disable the device-specific model quirk
in the driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/soc/codecs/wm8753.c: In function 'wm8753_probe':
sound/soc/codecs/wm8753.c:1577: error: implicit declaration of function 'wm8753_add_controls'
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this sparse warning:
sound/pci/hda/hda_codec.c:1544:19: warning: incorrect type in assignment (different signedness)
sound/pci/hda/hda_codec.c:1544:19: expected unsigned long *vals
sound/pci/hda/hda_codec.c:1544:19: got long *<noident>
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this sparse warnings:
sound/pci/emu10k1/emu10k1_main.c:723:66: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:724:68: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:748:74: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:751:66: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:759:73: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:760:73: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:837:50: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:845:50: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:881:50: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:889:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:890:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:895:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:897:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:899:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:910:56: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:914:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:918:56: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:922:57: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:924:58: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:936:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:1073:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:1088:60: warning: incorrect type in argument 3 (different signedness)
sound/pci/emu10k1/emu10k1_main.c:1093:58: warning: incorrect type in argument 3 (different signedness)
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this sparse warning:
sound/drivers/vx/vx_uer.c:301:42: warning: incorrect type in argument 2 (different signedness)
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this sparse warning:
sound/usb/usx2y/usbusx2y.c:231:33: warning: do-while statement is not a compound statement
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Impact: Move declaration to header file.
Fix this sparse warning:
sound/usb/usx2y/usx2yhwdeppcm.c:739:5: warning: symbol 'usX2Y_hwdep_pcm_new' was not declared. Should it be static?
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Impact: Move variable to a more inner scope.
Fix this sparse warning:
sound/oss/sequencer.c:235:29: warning: symbol 'err' shadows an earlier one
sound/oss/sequencer.c:215:13: originally declared here
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Impact: Change signature of 'set_volume_stereo' and 'set_volume_mono'.
Fix this sparse warnings:
sound/oss/pss.c:545:42: warning: incorrect type in argument 2 (different signedness)
sound/oss/pss.c:546:42: warning: incorrect type in argument 3 (different signedness)
sound/oss/pss.c:554:59: warning: incorrect type in argument 2 (different signedness)
sound/oss/pss.c:560:59: warning: incorrect type in argument 2 (different signedness)
sound/oss/pss.c:566:59: warning: incorrect type in argument 2 (different signedness)
Signed-off-by: Hannes Eder <hannes@hanneseder.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will reduce the number of writes done on resume, allowing that to
complete faster (especially on systems with very slow I2C like the
current Samsung driver).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The base support for the only in-tree user, the GTA01, is out of tree
and will be updated separately.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch should be pure code motion, separating that out from the
functional changes to move to new style device registration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the pseudo device-locking using card->shutdown flag to avoid
the crash via clear/reconfig during operations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the upper limit 0dB to the volume of mixer amp 0x20 for
AD1984A HP laptops. The overloaded volume may damage the internal
speaker.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make user_pin overriding even the driver pincfg, e.g. the static / fixed
pin config table in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename from override_pin and cur_pin with user_pin and driver_pin,
respectively, to be a bit more intuitive.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sleeping for 2 seconds before checking for the iobox is not enough
on some systems.
this patch increases the timeout, but polls the card during that
time. it thus speeds up the module loading when the card has already
been initialized, while being more robust on systems, which require
a higher timeout than the predefined 2 seconds.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Incorrect variable was used to get the next sample which caused S2
to be stuck with the same value resulting in loud background noise.
Signed-off-by: Steve Chen <schen at mvista.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audiowerk2 driver snd-aw2 is bound to any saa7146 device as it does not
check subsystem ids. Many DVB devices are saa7146 based, so aw2 driver
grabs them as well.
According to http://lkml.org/lkml/2008/10/15/311 aw2 devices have the
subsystem ids set to 0, the saa7146 default.
Fix conflicts with DVB devices by checking for subsystem ids = 0
specifically.
Signed-off-by: Anssi Hannula <anssi.hannula@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
http://kisskb.ellerman.id.au/kisskb/buildresult/72115/:
| net/mac80211/ieee80211_i.h:327: error: syntax error before 'volatile'
| net/mac80211/ieee80211_i.h:350: error: syntax error before '}' token
| net/mac80211/ieee80211_i.h:455: error: field 'sta' has incomplete type
| distcc[19430] ERROR: compile net/mac80211/main.c on sprygo/32 failed
This is caused by
| # define mfp ((*(volatile struct MFP*)MFP_BAS))
in arch/m68k/include/asm/atarihw.h, which conflicts with the new "mfp" enum in
net/mac80211/ieee80211_i.h.
Rename "mfp" to "st_mfp", as it's a way too generic name for a global #define.
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Added the generic pincfg cache and save/restore functions.
Also introduced the pin-overriding via hwdep sysfs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the MIN_PACKS_URB symbol because other limits can force the
number of packets down to one, regardless of the value of this symbol,
and nobody has ever changed it anyway.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use -60 dB as the minimum value of the master volume mixer control.
While the DACs would support ranges down to about -120 dB, such
attenuations are not useful in practice.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HT-Omega Claro halo's ADC is an AK5385 instead of a WM8785, so we
should handle the ADC parameters as we do with the X-Meridian.
Using the code for the wrong ADC does not seem to have any audible
effects, and the Windows driver does it, but it is nonetheless a good
idea to run the AK5385 with an oversampling ratio that is not outside
the documented limits.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usage and comments make it clear values of 1/0 were intended
rather than -1/0
Noticed by sparse:
sound/pci/pcxhr/pcxhr.h💯20: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:101:22: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:102:24: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:103:21: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:104:25: error: dubious one-bit signed bitfield
sound/pci/pcxhr/pcxhr.h:105:20: error: dubious one-bit signed bitfield
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This avoids temporarily enabling the ouput stages during startup which
can cause audible effets in the output stages.
Reported-by: Fredrik Redgård <rik@svep.se>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the EEPROM was partially overwritten (which seems to happen before the OS is
booted), restore its entire contents by deducing it from the remaining
information.
This does not have any effect on the Linux driver, which works even with
incomplete information in the EEPROM, but it makes other drivers work again.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Under as yet unknown circumstances, the first word of the sound card's
EEPROM gets overwritten. When this has happened, we cannot rely on the
subsystem IDs that the kernel reads from the PCI configuration
registers. Instead, we read the IDs directly from the EEPROM and do the
ID matching manually.
Because the model-specific driver cannot determine the model before
calling oxygen_pci_probe(), that function now gets a get_model()
callback as parameter. The customizing of the model structure, which
was formerly done by the probe() callback, also has moved into
get_model().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When allocating resources, use a fixed name instead of reading it from
the model structure. This allows us to allocate the resources before
the actual model is known.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocate the model-specific data dynamically instead of including it in
the memory block of the card structure. This will allow us to determine
the actual model after the card creation.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the owner field out of the oxygen_model structure and make it
a parameter of oxygen_pci_probe(), because the actual owner module does
not depend on the card model. Furthermore, moving it out of the model
structure allows us to create the card structure before the actual model
is known.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 7e86c0e685 ("do not
overwrite EEPROM on Xonar D2/D2X") because it did not actually help with
the problem.
More user reports show that the overwriting of the EEPROM is not
triggered by using this driver but by installing Linux, and that the
installation of any other operating system (even one without any CMI8788
driver) has the same effect. In other words, the presence of this
driver does not have any effect on the occurrence of the error. (So
far, the available evidence seems to point to a BIOS bug.)
Furthermore, it turns out that the EEPROM chip is protected against
stray write commands by the command format and by requiring a separate
write-enable command, so the error scenario in the previous commit (that
SPI writes can be misinterpreted as an EEPROM write command) is not even
theoretically possible.
The mixer control that was removed as a consequence of the previous
commit can only be partially emulated in userspace, which also means it
cannot be seen be the in-kernel OSS API emulation, so it is better to
revert that change.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC268 can be configured as digital-only, e.g. for HDMI, on some
machines. Allow the parser to set up the digital-only mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the digital loopback/bypass support for twl4030 codec.
The digital loopback will let the digimic0 (routed in the TX1 capture path
inside of TWL4030) data to be routed back to the RX2 playback path
(I2S stereo). It can also route the analog capture date routed through the
TX1 back to RX2.
Effectively the digital loopback is routing the audio from the TX1 capture path
to the RX2 playback path.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the jack layer refers to card->longname as a part of
its input device name string. However, longname is often really long
and way too ugly as an identifier, such as,
"HDA Intel at 0xf8400000 irq 21".
This patch changes the code to use card->shortname instead.
The shortname string contains usually the h/w vendor and product
names but without messy I/O port or IRQ numbers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the WM8731 driver to use a more standard device registration
scheme where the device can be registered independantly of the ASoC
probe.
As a transition measure push the current manual code for registering
the WM8731 into the individual machine driver probes. This allows
separate patches to update the relevant architecture files with less
risk of merge issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a pure code motion patch intended to improve reviewability of a
following patch moving WM8731 to use more standard device registration.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a bit more idiomatic and makes identifying a configuration
based on the board type work better.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We have software control of the MCLK for the WM8731 so save a bit of
power by actively managing it within the machine driver, enabling it
only while the codec is active.
Once ASoC supports multiple boards and doesn't require the soc-audio
device the initial clock setup should be pushed down into the arch/arm
code but for now this reduces merge issues.
Tested-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8731 bias level configuration function was written slightly
obscurely - streamline the code a little and refresh the comments.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
WM8753 uses a tricky way to switch DAIs "on the fly", for that it
registers 2 dummy DAIs and substitutes them depending on mixer control.
List element of registered dummy DAIs should be preserved to allow
unregistering of DAIs on module unload.
Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since cs4232 and cs4236 drivers are merged, there is no reason to keep
snd-cs4236-lib module separately. Let's merge it into the main driver
as well.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed a typo of || and &&.
As it's in a disabled code section, there is no behavior change, though.
Reported-by: Jörg-Volker Peetz <jvpeetz@web.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
cs4232 and cs4236 driver merge to solve PnP BIOS detection.
Also, the patch adds recognition if the chip is cs4236b+
or earlier part. This unifies drivers for both cs4232
and cs4236+ chips. It allows to use the PnP BIOS
detection for the cs4236+ chips. Previously, only
the snd-cs4232 could be detected by the PnP BIOS.
The cs4232+ cards reports two separate PnP BIOS ids.
The patch adds search for the second id to find out
resources assigned to a control port.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CM6207 incorrectly advertises its 96 kHz playback setting as 48 kHz
in its USB device descriptor. This patch extends an existing workaround
in usbaudio.c to also cover the CM6207.
This resolves issue 0004249 in the ALSA bug tracker.
Signed-off-by: Joris van Rantwijk <jorispubl@xs4all.nl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The detection of non-continuous rates (given via rate tables) isn't
processed properly (e.g. for type II).
This patch fixes and simplifies the detection code.
Tested-by: Joris van Rantwijk <jorispubl@xs4all.nl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the snd_usbmidi_create_endpoints_midiman() function, which forgot to
set the out_interval member of the endpoint info structure for Midiman/
M-Audio devices. Since kernel 2.6.24, any non-zero value makes the
driver use interrupt transfers instead of bulk transfers. With EHCI
controllers, these random interval values result in unbearably large
latencies for output MIDI transfers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-by: David <devurandom@foobox.com>
Tested-by: David <devurandom@foobox.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Force speaker pin config with model=hp-dv5 model for cases when bios
doesn't set it up properly. All reported hp laptops using model=hp-dv5
model have speaker at pin 0x0d with same config, so it's safe to add
this within hp-dv5 model.
Reference: alsa-devel mailing list thread on
http://mailman.alsa-project.org/pipermail/alsa-devel/2009-February/014390.html
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 32e176c14d.
That commit caused a regression with suspend on Thinkpad SL300.
Reference: kernel bug#12711
http://bugzilla.kernel.org/show_bug.cgi?id=12711
Tested-by: Alexandre Rostovtsev <tetromino@gmail.com>
Acked-by: Rafael J. Wysocki <rjw@sisk.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix for the error when the audio module is unloaded. On unregistering
the platform_device, platform_device_release will free the platform
data.If platform data is static the kernel panics when it is freed.
Instead use the platform device helper function to add data.
This change has been tested on DM644x EVM, DM644x SFFSDR and DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: Only register AC97 bus if it's not done already
ALSA: hda - Add snd_hda_multi_out_dig_cleanup()
ALSA: hda - Add missing terminator in slave dig-out array
ALSA: hda - Change HP dv7 (103c:30f4) quirk from hp-m4 to hp-dv5 model
ALSA: hda - Register (new) devices at reconfig
ALSA: mtpav - Fix initial value for input hwport
ALSA: hda - add id for Intel IbexPeak integrated HDMI codec
ALSA: hda - compute checksum in HDMI audio infoframe
ALSA: hda - enable HDMI audio pin out at module loading time
ALSA: hda - allow multi-channel HDMI audio playback when ELD is not present
ASoC: Update SDP3430 machine driver for snd_soc_card
ALSA: hda - Add quirk for Asus z37e (1043:8284)
sound: Remove OSSlib stuff from linux/soundcard.h
ASoC: WM8990: Fix kcontrol's private value use in put callback
ASoC: TLV320AIC3X: Fix kcontrol's private value use in put callback
ASoC supports both explicit codec drivers for AC97 devices and a simple
driver which uses the standard ALSA AC97 framework for codec support.
When used with the generic AC97 codec support that will provide the
ad hoc AC97 device for drivers like touchscreens to attach to so the
core shouldn't do so.
Reported-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the CS4270 codec driver to allow applications to use the mixer to
control Digital Loopback, Soft Ramp, Zero Cross, Popguard, and Auto-Mute.
Soft Ramp, Zero Cross, and Auto-Mute are disabled by the driver when it first
initializes the hardware, but these features either don't work or interfere
with normal ALSA behavior. However, they can now be re-enabled by an
application if desired.
Remove CONFIG_SND_SOC_CS4270_HWMUTE and always allow ASoC to control the mute
bits. The driver previously and erroneously assumed that these bits
control only external muting circuitry, but they also control internal
muting circuitry, so they should always be used.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the support of multiple digital outputs via auto-probing for
Realtek ALC88x codecs. The multiple outputs are handled as slave
streams, so only one PCM stream (and the corresponding IEC958*
elements) is provided to control both digital outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the helper function snd_hda_multi_out_dig_cleanup() to clean up
the digital outputs with multi setup. This call is needed in cases
the codec supports multiple digital outputs as slaves. Otherwise the
slave widgets aren't properly cleaned up.
For a single digital output (e.g. in patch_conexant.c), this call isn't
needed.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Force model=auto for Acer AX1700-U3700A with ALC888 codec.
Since Acer devices are handlded as model=acer as default, the auto
parsing has to be specified explicitly.
(Maybe it's better rather to remove this default model=acer handling,
though.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change HP dv7 quirk: although reported to work with hp-m4 model
(https://bugzilla.novell.com/show_bug.cgi?id=445321), the original
report doesn't contain info about testing of internal microphone.
Recently I received a report about internal mic not working
(https://qa.mandriva.com/show_bug.cgi?id=44855#c193), this must be
related with the forced line in on pin 0x0e done with hp-m4 model. Thus
change the current quirk from STAC_HP_M4 to STAC_HP_DV5, later reported
to be fixed on a provided kernel with this change
(https://qa.mandriva.com/show_bug.cgi?id=44855#c196).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently the HP connector on X200 dock doesn't detect when a HP is connected
nor allows sound to be played using it. This patch fixes the problem by adding
a quirk for this specific model. It's possible that others have the same NID
(0x19) to report when dock HP is connected, but I don't have access to any.
Please Cc me in the reply since I'm not subscribed to alsa-devel@.
Signed-off-by: Aristeu Rozanski <aris@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add STAC_DELL_S14 quirk for new laptop series. Removed un-needed pins
in pin_nids for stac92hd83xxx. Also reorganized connection selection
code for the respective ports per quirk define.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS W5F needs the fixed codec-slots to probe to override the BIOS
problem.
Tested-by: Giovanni Moser Frainer <giovanni@redix.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some devices have broken BIOS and they don't set the codec probe-bit
properly after cleared by the driver. This makes the driver skipping
the necessary codec slots.
Since BIOS update isn't always easy, now the semantics of probe_mask
option is changed a bit. When it contains the bit 8 (0x100), the
lower bits are used to probe that slots regardless of codec-probe bits
returned by the hardware.
For example, probe_mask=0x103 will force to probe the codec slot #0
and #1.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After commit "ALSA: hda - Fix restore of pin configs at resume for
STAC/IDT codecs", the introduced stac_save_pin_cfgs function checks
already for pins == NULL case, saving then default pin configs from
machine with stac92xx_save_bios_config_regs. So we can remove the
extra checks when stac927x_brd_tbl[spec->board_config] == NULL.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The devices that have been newly added during reconfig must be
registered. Otherwise they won't be visible to user-space.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the initial value for input hwport. The old value (-1) may cause
Oops when an realtime MIDI byte is received before the input port is
explicitly given.
Instead, now it's set to the broadcasting as default.
Tested-by: Holger Dehnhardt <dehnhardt@ahdehnhardt.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Detect multiple digital-out pins in snd_hda_parse_pin_defconfig().
The dig_out_pin and dig_out_type fields become arrays.
The codec parser still doesn't use this multiple pins detection, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We found that enabling/disabling HDMI audio pin out at stream start/stop
time will kill the leading 500ms or so sound samples. Avoid this by enabling
pin out once and for ever at module loading time.
The leading ~500ms audio samples will still be lost when switching from
X-channel playback to Y-channel playback where X != Y. However there's no
much we can do about it: the audio infoframe has to change and it looks like
either G45 or YAMAHA requires some time to switch the configuration.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The YAMAHA AV-X1800 requires audio infoframe to include speaker-channel
mapping to play >2 channel HDMI audio. In theory that mapping should be
derived from its speaker configurations contained in its ELD. However we
currently cannot get ELD in console before the KMS functionalities are ready.
This is a more or less general issue at least in the near future. As a
workaround, we propose to allow playback of mult-channel audio when ELD
is not available.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch replaces "snd_soc_machine" structure by "snd_soc_card" in
SP3430 driver. This change is needed in SDP3430 driver to reflect
changes introduced by "ASoC: Rename snd_soc_card to snd_soc_machine" patch
(875065491f).
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the digital beep support for ALC268. It was missing in the
last patches.
However, ALC268 has a strange pin use for widget 0x1d, which could be
used as another purpose. So, the patch adds a check of the beep control
before creating the hook for input layer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With a postfix decrement the timeout will reach -1 rather than 0,
so the warning will not be issued.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
TLV320AIC3X volume controls are logarithmic. Export their dB ranges.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is a minor fix but helps to define dB ranges for volume controls.
Only DAC digital volume has full register value range from 0 to 127 but
ADC PGA gain and output stage volume controls don't.
For ADC PGA, maximum value is 119 and then it saturates to the same
gain value of 59.5 dB. For output stages, value 117 corresponds to -78.3 dB
and is muted for values 118 and above.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When checking for input amps on pins 0x0a, 0x0d and 0x0f, and
initializing them for 92hd71xxx codec models, we must skip nid 0x0f
for 4-port models too like with 5-port models, as it is unused
(nid 0x0f is vendor reserved in 4-port models).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a new sound quirk entry (model=ecs202) for an ecs motherboard
with IDT STAC9221 codec (1019:2950).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This removes the calls to pxa_gpio_mode from the pxa2xx-i2s driver.
Pin setup should be done during board init via pxa2xx_mfp_config
instead.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Acked-by: Eric Miao <eric.miao@marvell.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This machine driver enables sound functions on Mitac mio
a701 smartphone. Build upon ASoC v1, it handles :
- rear speaker
- front speaker
- microphone
- GSM
A global "Mio Mode" switch is not yet provided to cope with
audio path setup. As balance on audio chip line is no more
assured, an incorrect setup can produce a lot of heat and
even fry the battery behind the wm9713 and the speaker
amplifier.
It doesn't cope with :
- headset jack
- mio master mode
- master volume control
This driver is backported from ASoc v2, and amputated from
scenario setups and master volume control.
[Minor mods for terminology in comments -- broonie]
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use digital beep instead of analog pc-beep for AD codecs.
Create the beep mixer controls dynamically on demand.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the Freescale MPC8610 sound drivers, relocate all code from the _prepare
functions into the corresponding _hw_params functions. These drivers assumed
that the sample size is known in the _prepare function and not in the
_hw_params function, but this is not true.
Move the code in fsl_dma_prepare() into fsl_dma_hw_param(). Create
fsl_ssi_hw_params() and move the code from fsl_ssi_prepare() into it.
Turn off snooping for DMA operations to/from I/O registers, since that's not
necessary.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- make sport number handling more dynamic as not all
Blackfins have a linear sport map starting at 0
- indexes can be macroed away too
Signed-off-by: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Function wm899x_outpga_put_volsw_vu misuses the kcontrol's private value
by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it
as a pointer into struct soc_mixer_control after the commit
4eaa9819dc.
This is very similar fix than fix to TLV320AIC3X codec made by
Eero Nurkkala <ext-eero.nurkkala@nokia.com>. This fix is compile tested
only.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Function snd_soc_dapm_put_volsw_aic3x misuses the kcontrol's private value
by still accessing it as bitfields even SOC_SINGLE_VALUE constructs it
as a pointer into struct soc_mixer_control after the commit
4eaa9819dc.
This was causing arbitrary register writes when touching the controls
defined with SOC_DAPM_SINGLE_AIC3X.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For audio devices that do not have proper audio descriptors (e.g.,
Edirol UA-20), we use hardcoded parameters from our quirks list.
However, we must still read the maximum packet size from the standard
endpoint descriptor; otherwise, we might use packets that are too big
and therefore rejected by the USB core.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the SPDIF pin as slave digital out to enable concurrent
HDMI/SPDIF outputs for ASUS M3A-H/HDMI with ALC1200 codec.
Tested-by: Thomas Schneider <nailstudio@gmx.net>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds ALSA support for the AC97 controller found on Atmel
AVR32 devices.
Tested on ATSTK1006 + ATSTK1000 with a development board with a AC97
codec.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds ALSA support for the Audio Bistream DAC found on Atmel
AVR32 devices. The ABDAC is an Atmel IP which might show up on AT91
devices in the future, hence making a generic driver which can be
utilized by AT91 arch if needed.
Datasheet describing the ABDAC peripheral is available in the AT32AP7000
datasheet, http://www.atmel.com/dyn/products/datasheets.asp?family_id=682
Tested on ATSTK1006 + ATSTK1000 with a class D amplifier stage.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Impact: cleanup
snd_pcm_new takes a char *id argument, although it is not modifying
the string. it can therefore be declared as const char *id.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Indicates fixes affecting control messages and switching of input mode
on Audio 8 DJ and Audio 4 DJ.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove a duplicate control which causes an error when it is registered,
and causes later controls to not be registered. The device does not have
a fourth ground lift control.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the context of the Audio 4 DJ (when compared to Audio 8 DJ), hardware
input mode 0 is not used. Expose modes 1 (line) and 2 (phono) to the user
as modes 0 and 1 respectively.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not start the device with input mode undefined. Mimic the behaviour of
the Audio 8 DJ and start in phono input mode.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes a bug where an incorrect command was sent which had no effect on the
device.
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the ADATOut nid to a slave digital outs struct to allow output
via the DigOut pin.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use uneeded/wrong third parameter for stac92xx_parse_auto_config
in patch_stac92hd71bxx (no SPDIF in).
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Detect the number of connected ports and number of smuxes dynamically,
looking at pin configs, using new introduced functions
stac92hd71bxx_connected_ports and stac92hd71bxx_connected_smuxes. Also
use proper input mux configuration for 4port and 5port models.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current code for STAC92HD71Bx and STAC92HD75Bx doesn't consider pin
complexes 0x20 and 0x27. Also for 4 port models, nids 0x0e and 0x0f
are vendor reserved. This commit changes code so it'll consider the
additional pin complexes for models that have it, and avoid reserved
nids to be touched on 4 port models.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some fixes regarding snd-hda-intel workqueue:
- Use create_singlethread_workqueue() instead of create_workqueue()
as per-CPU work isn't required.
- Allocate workq name string properly
- Renamed the workq name to "hd-audio*" to be more obvious.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AFMT_S24_LE is set twice in return value
vi sound/core/oss/pcm_oss.c +640
#define AFMT_S24_LE 0x00008000
#define AFMT_S24_BE 0x00010000
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The wss_base is disuised parameter for one function.
It is converted to function parameter.
The code_type is only set but never read.
It is removed.
The midi_vol is set only to 0 so it does not work
as detection of change in midi volume. It is fixed.
The xport variable is alias to the port[dev]. Use
the port[dev] directly to increase readability.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A digital beep generator can be used via input layer.
Signed-off-by: Kusanagi Kouichi <slash@ma.neweb.ne.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PXA2xx/3xx SSP ports start from 1, not 0. Thus, the probe function
requested the wrong SSP port. Correcting this unveiled another bug
where ssp_init tries to request the already-requested SSP port again.
So this patch replaces the ssp_init/exit calls with their internals
from mach-pxa/ssp.c, leaving out the redundant ssp_request and the
unneeded IRQ request. Effectively, that leaves us with not much more
than enabling/disabling the SSP clock.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch splits set_dai_fmt into three variants (single interface,
dual interface playback only, dual interface capture only) so that
data input and output formats can be configured separately for dual
interface setups.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Tested-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this fix driver switches to WSPLL in uda1380_pcm_prepare
even if SYSCLK was chosen (uda1380_pcm_prepare modifies UDA1380_CLK
register to disable R00_DAC_CLK before flushing reg cache)
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Realtek ALC262 on the Tyan Thunder n6650W (S2915-E) mainboard has a
rather odd configuration template. As a result, the white AUX connector
can not be used. This rewrites the default config to more accurately
reflect the connector layout, colour and function.
Unfortunately the black CD_IN connector, which is suspected to be widget
0x1c refuses to switch into input (0x20), instead opting to remain on 0x0.
As such, no mixer controls are exposed for it. Autoswitching is implemented
between the front headphone output and back line output.
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This just updates my email address on some drivers I'd forgotten in a
previous patch.
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace printk calls with dev_xxx calls. Set the 'dev' field of the codec
and codec_dai structures so that these calls work.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix a oversight in the CS4270 codec driver that caused a build break.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
MIDI_TX IRQ seems always pending when any bytes on FIFO is available.
Thus, it's better to enable MPU_TX only when any bytres are really
stored in the substream, and disables immediately when the queue
becomes empty.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
omap_pcm_trigger is called also in interrupt context so CPU flags must
be restored when returning.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Volume-knob widgets have no widget selection although they have widget
connections. Thus, the connection list in the proc output shouldn't
contain the selection (*).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The missing module license generates warning
during module loading.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With a postfix decrement time will reach -1 rather than 0,
so the warning will not be issued.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update pandora board file for recent TWL4030 codec changes.
Also move output related snd_soc_dapm_nc_pin() calls to
omap3pandora_out_init(), where they belong.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Spruce up the documentation in the CS4270 codec. Use kerneldoc where
appropriate. Fix incorrect comments.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ess1688 driver uses the same port
for PCM audio (SB compatible) and OPL3
synthesis. It is not always right so allow to
choose a different port for OPL3 synthesis.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC codec drivers typically serve two masters: the I2C bus and ASoC itself.
When a codec driver registers with ASoC, a probe function is called. Most
codec drivers call ASoC first, and then register with the I2C bus in the ASoC
probe function.
However, in order to support multiple codecs on one board, it's easier if the
codec driver is probed via the I2C bus first. This is because the call to
i2c_add_driver() can result in the I2C probe function being called multiple
times - once for each codec. In the current design, the driver registers
once with ASoC, and in the ASoC probe function, it calls i2c_add_driver().
The results in the I2C probe function being called multiple times before the
driver can register with ASoC again.
The new design has the driver call i2c_add_driver() first. In the I2C probe
function, the driver registers with ASoC. This allows the ASoC probe function
to be called once per I2C device.
Also add code to check if the I2C probe function is called more than once,
since that is not supported with the current ASoC design.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
X86_PC is the only remaining 'sub' architecture, so we dont need
it anymore.
This also cleans up a few spurious references to X86_PC in the
driver space - those certainly should be X86.
Signed-off-by: Yinghai Lu <yinghai@kernel.org>
Signed-off-by: Ingo Molnar <mingo@elte.hu>
In the case of having a selector instead of mixer while initing input
sources, the case that happens with ALC889, we must select instead
of muting input. Problem was found while testing with hda-emu.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the analog loopback/bypass support for twl4030 codec.
Details for the implementation:
It seams that the analog loopback needs the DAC powered on on the channel,
where the loopback is selected. The switch for the DACs has been moved from
the DAPM_DAC to the "Analog XX Playback Mixer". In this way the DAC will be
powered while the audio playback is used or/and the loopback is enabled for
the channel.
The twl4030 codec powering has been reworked. Now the codec will be powered as
long as it does not receives the SND_SOC_BIAS_OFF event. The exceptions are
when the given change in the registers needs the codec power down/up cycle in
order to take effect. Otherwise the codec is on.
When the codec enters to STANDBY state and none of the loopback paths are
enabled, than the amplifiers, which are no in the DAPM path are forced to turn
off and the PLL is disabled. When playback/capture starts the disabled gains
are restored and the PLL is enabled.
When one of the loopback enabled in STANDBY mode, the disabled gains are
restored and the PLL is enabled also.
In short: the codec always goes to the lowest power state based on the
bias_level and the bypass_state.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
