Fix the codec_name field of the dai_link to match the actual device name
of the codec. Otherwise the card won't be instantiated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Commit f97d0c6d5f ("ASoC: AD1836: Add input gain control for ADC2") contained
a typo in the register name, causing a build error. This patch fixes it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
removing unnecessary if(ret) checks
This updated patch corrects a minor spelling problem in the commit message
and resolves two other (similar) issues found in wm8940.c by Jonathan Cameron.
Signed-off-by: Greg Dietsche <Gregory.Dietsche@cuw.edu>
Acked-by: Jonathan Cameron <jic23@cam.ac.uk>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://launchpad.net/bugs/792712
The original reporter states that sound from the internal speakers is
inaudible until using the model=auto quirk. This symptom is due to an
existing quirk mask for 0x102802b* that uses the model=dell quirk. To
limit the possible regressions, leave the existing quirk mask but add
a higher priority specific mask for the reporter's PCI SSID.
Reported-and-tested-by: rodni hipp
Cc: <stable@kernel.org> [2.6.38+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The AD1836 has a PGA for its second ADC. This patch adds a control for
adjusting the the gain of the PGA.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The control_type field is never used, so it can be removed. The
control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The AD183X codec devices are mostly register compatible and can easily be
supported by the same driver. The main difference between those devices
is the number of DACs and ADCs.
This patch adjusts the driver to allocate the controls, DAPM widgets and
routes for the DACs and ADCs dynamically based on the chip type.
The AD1836 is a bit special in that it supports different modes for its second
ADC, so it needs some special handling. Right now the driver hardcodes the mode
to the differential PGA mode.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The different ADC and DAC controls follow the same scheme, so add some helper
macros for declaring them.
This should make the code a bit more readable and also decreases the code size
a bit.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that the CODEC driver supports it defer configuration of the system
clock until bias management which is a much more idiomatic place to do
system power control and makes things a lot more happy when we're using
both interfaces.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This allows the card driver to use the bias level variable more easily in
multi component systems.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
It's redundant now thanks to the use of the generic trace infrastructure.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
If the only widgets active within a CODEC are supplies and micbiases we
are not passing audio, we are probably just doing microphone detection.
This will not generally require either fully accurate reference voltages
or much power so
If this turns out to be unsuitable for some systems we can provide a
facility to override this decision.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Allow more dynamic management of the device clocking by allowing BCLK to
be calculated when we set SYSCLK. This means that if the system is idle
when hw_params() runs then we don't try to use the SYSCLK used in that case
to set up the BCLK dividers, we can instead wait until a later point such
as bias level configuration. This makes it easier to manage low power modes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Avoids issues if someone does a read followed by restore and doesn't mask
out only the bits being updated.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
When the FLL locks on the WM8915 an interrupt is generated. For safety
error out if we don't get that interrupt when the IRQ output of the
WM8915 is hooked up. Since we *really* expect an interrupt but the
threaded IRQ handler may take a bit longer than expected to get
scheduled also dramatically increase the delay in this case.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The general concept of this change is to create a PCM device for each
pin widget instead of each converter widget. Whenever a PCM is opened,
a converter is dynamically selected to drive that pin based on those
available for muxing into the pin.
The one thing this model doesn't support is a single PCM/converter
sending audio to multiple pin widgets at once.
Note that this means that a struct hda_pcm_stream's nid variable is
set to 0 except between a stream's open and cleanup calls. The dynamic
de-assignment of converters to PCMs occurs within cleanup, not close,
in order for it to co-incide with when controller stream IDs are
cleaned up from converters.
While the PCM for a pin is not open, the pin is disabled (its widget
control's PIN_OUT bit is cleared) so that if the currently routed
converter is used to drive a different PCM/pin, that audio does not
leak out over a disabled pin.
We use the recently added SPDIF virtualization feature in order to
create SPDIF controls for each pin widget instead of each converter
widget, so that state is specific to a PCM.
In order to support this, a number of more mechanical changes are made:
* s/nid/pin_nid/ or s/nid/cvt_nid/ in many places in order to make it
clear exactly what the code is dealing with.
* We now have per_pin and per_cvt arrays in hdmi_spec to store relevant
data. In particular, we store a converter's capabilities in the per_cvt
entry, rather than relying on a combination of codec_pcm_pars and
the struct hda_pcm_stream.
* ELD-related workarounds were removed from hdmi_channel_allocation
into hdmi_instrinsic in order to simplifiy infoframe calculations and
remove HW dependencies.
* Various functions only apply to a single pin, since there is now
only 1 pin per PCM. For example, hdmi_setup_infoframe,
hdmi_setup_stream.
* hdmi_add_pin and hdmi_add_cvt are more oriented at pure codec parsing
and data retrieval, rather than determining which pins/converters
are to be used for creating PCMs.
This is quite a large change; it may be appropriate to simply read the
result of the patch rather than the diffs. Some small parts of the change
might be separable into different patches, but I think the bulk of the
change will probably always be one large patch. Hopefully the change
isn't too opaque!
This has been tested on:
* NVIDIA GeForce 400 series discrete graphics card. This model has the
classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM
audio to a PC monitor that supports audio.
* NVIDIA GeForce 520 discrete graphics card. This model is the new
1 codec n converters m pins m>n model. Tested stereo PCM audio to a
PC monitor that supports audio.
* NVIDIA GeForce 400 series laptop graphics chip. This model has the
classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM,
multi-channel PCM, and AC3 pass-through to an AV receiver.
* Intel Ibex Peak laptop. This model is the new 1 codec n converters m
pins m>n model. Tested stereo PCM, multi-channel PCM, and AC3 pass-
through to an AV receiver.
Note that I'm not familiar at all with AC3 pass-through. Hence, I may
not have covered all possible mechanisms that are applicable here. I do
know that my receiver definitely received AC3, not decoded PCM. I tested
with mplayer's "-afm hwac3" and/or "-af lavcac3enc" options, and alsa a
WAV file that I believe has AC3 content rather than PCM.
I also tested:
* Play a stream
* Mute while playing
* Stop stream
* Play some other streams to re-assign the converter to a different
pin, PCM, set of SPDIF controls, ... hence hopefully triggering
cleanup for the original PCM.
* Unmute original stream while not playing
* Play a stream on the original pin/PCM.
This was to test SPDIF control virtualization.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change won't store an entire hda_pcm_stream just to represent
the capabilities of a codec; a custom data-structure will be used. To
ease that transition, modify hdmi_eld_update_pcm_info to expect the
hda_pcm_stream to be pre-initialized with the codec's capabilities, and
to update those capabilities in-place based on the ELD.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A future change will significantly rework the generic implementation
in order to support codecs with a different number of pins and
converters. Isolate the more custom codec variants from this change by
duplicating the small portions of generic code they share. This
simplifies the later rework of that previously shared code, since we
don't have to consider the more custom codecs, and also prevents
support for those codecs from regressing.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SPDIF output controls apply to converter widgets. A future change
will create a PCM device per pin widget, and hence a set of SPDIF output
controls per pin widget, for certain HDMI codecs. To support this, we
need the ability to virtualize the SPDIF output controls. Specifically:
* Controls can be "unassigned" from real hardware when a converter is
not used for the PCM the control was created for.
* Control puts only write to hardware when they are assigned.
* Controls can be "assigned" to real hardware when a converter is picked
to support output for a particular PCM.
* When a converter is assigned, the hardware is updated to the cached
configuration.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, the data that backs the kcontrols created by
snd_hda_create_spdif_out_ctls is stored directly in struct hda_codec. When
multiple sets of these controls are stored, they will all manipulate the
same data, causing confusion. Instead, store an array of this data, one
copy per converter, to isolate the controls.
This patch would cause a behavioural change in the case where
snd_hda_create_spdif_out_ctls was called multiple times for a single codec.
As best I can tell, this is never the case for any codec.
This will be relevant at least for some HDMI audio codecs, such as the
NVIDIA GeForce 520 and Intel Ibex Peak. A future change will modify the
driver's handling of those codecs to create multiple PCMs per codec. Note
that this issue isn't affected by whether one creates a PCM-per-converter
or PCM-per-pin; there are multiple of both within a single codec in both
of those codecs.
Note that those codecs don't currently create multiple PCMs for the codec
due to the default HW mux state of all pins being to point at the same
converter, hence there is only a single converter routed to any pin, and
hence only a single PCM.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's perfectly valid for an ELD to contain no SADs. This simply means that
only basic audio is supoprted.
In this case, we still want to limit a PCM's capabilities based on the ELD.
History:
* Originally, ELD application was limited solely by sad_count>0, which
was used to check that an ELD had been read.
* Later, eld_valid was added to the conditions to satisfy.
This change removes the original sad_count>0 check, which when squashed
with the above two changes ends up replacing if (sad_count) with
if (eld_valid).
Signed-off-by: Stephen Warren <swarren@nvidia.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:
ALSA: usb - turn off de-emphasis in s/pdif for cm6206
ALSA: asihpi: Use angle brackets for system includes
ALSA: fm801: add error handling if auto-detect fails
ALSA: hda - Check pin support EAPD in ad198x_power_eapd_write
ALSA: hda - Fix HP and Front pins of ad1988/ad1989 in ad198x_power_eapd()
ALSA: 6fire: Don't leak firmware in error path
ASoC: Fix wm_hubs input PGA ZC bits
ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared
This reverts commit ed0bd2333c.
Since we reverted the TTY API change, we should revert the ASoC update
to it too.
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
CM6206: Turn off de-emphasis channel status bit in S/PDIF output.
Signed-off-by: Eric Lammerts <eric@lammerts.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update Makefile and Kconfig to build HDMI audio support for
OMAP4 SDP and Panda boards.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Add machine driver for HDMI audio on OMAP4 boards. This driver is
in charge of putting together the HDMI audio codec and the CPU DAI
and register the HDMI sound card with ALSA.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Addition of the HDMI CPU DAI driver for OMAP4. This driver is in
charge of configuring DMA settings for HDMI. Also, it finds
the HDMI video device and determines if audio playback can proceed.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
We should only call ssc_free() when ssc_request() succeeds or bad
things will happen.
Signed-off-by: Joachim Eastwood <joachim.eastwood@jotron.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the original code if auto detect failed and tea575x_tuner == 4
then we copy bogus information to chip->tea.card. I've changed the
autodetect code to cleanup and return -ENODEV on error instead.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In ad198x_power_eapd(), wrong pin NIDs are used for controlling EAPD for
HP and Front outputs of AD1988/AD1989. These are actually same with the
ones for AD1984 & co, port-A is 0x11 and port-D 0x12.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We only need to increase the detection rate to maximum if we're monitoring
for button presses as the response times needed for user interaction there
are much lower.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
One of the error paths in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload() neglects to free
the memory allocated for the firmware before returning, thus leaking the
memory.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revision 2 of the Speyside platform supplies a 32kHz clock on MCLK2 rather
than MCLK1.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
DCVDD and MICVDD are intended to be (and almost always are) generated by
on-board LDOs which are transparently controlled by the driver so we
shouldn't really be requesting them from the regulator API. If the driver
is updated to support external supply of these then we will need to change
the way we handle this.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This goto is after the call to clk_get, so it should go to the label that
includes a call to clk_put.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
@r exists@
expression e1,e2;
statement S;
@@
e1 = clk_get@p1(...);
... when != e1 = e2
when != clk_put(e1)
when any
if (...) { ... when != clk_put(e1)
when != if (...) { ... clk_put(e1) ... }
* return@p3 ...;
} else S
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Really this should be something the IRQ core can cope with for us but since
it doesn't currently do so (at least for threaded interrupts like this) do
so in the driver. This allows us to run with interrupt controllers that
only support edge triggered interrupts.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Current implementation set max98095->sysclk/max98088->sysclk to freq twice.
Set it once is enough, this patch removes the first assignment in case
we may set invalid clock frequency to max98095->sysclk/max98088->sysclk.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (45 commits)
ARM: 6945/1: Add unwinding support for division functions
ARM: kill pmd_off()
ARM: 6944/1: mm: allow ASID 0 to be allocated to tasks
ARM: 6943/1: mm: use TTBR1 instead of reserved context ID
ARM: 6942/1: mm: make TTBR1 always point to swapper_pg_dir on ARMv6/7
ARM: 6941/1: cache: ensure MVA is cacheline aligned in flush_kern_dcache_area
ARM: add sendmmsg syscall
ARM: 6863/1: allow hotplug on msm
ARM: 6832/1: mmci: support for ST-Ericsson db8500v2
ARM: 6830/1: mach-ux500: force PrimeCell revisions
ARM: 6829/1: amba: make hardcoded periphid override hardware
ARM: 6828/1: mach-ux500: delete SSP PrimeCell ID
ARM: 6827/1: mach-netx: delete hardcoded periphid
ARM: 6940/1: fiq: Briefly document driver responsibilities for suspend/resume
ARM: 6938/1: fiq: Refactor {get,set}_fiq_regs() for Thumb-2
ARM: 6914/1: sparsemem: fix highmem detection when using SPARSEMEM
ARM: 6913/1: sparsemem: allow pfn_valid to be overridden when using SPARSEMEM
at91: drop at572d940hf support
at91rm9200: introduce at91rm9200_set_type to specficy cpu package
at91: drop boot_params and PLAT_PHYS_OFFSET
...
Commit 9477c58e33 ("ALSA: hda - Reorganize controller quriks with bit
flags") changed the driver type compares into various quirk bits.
However, the check for AZX_DCAPS_NO_TCSEL got reverted: instead of
clearing TCSEL for chipsets that have that standard capability, it
cleared then when the NO_TCSEL bit was set.
This can lead to noise and repeated sounds - a weird "echo" behavior.
As the comment just above says: "Ensuring these bits are 0 clears
playback static on some HD Audio codecs". Which is definitely true at
least on my Core i5 Westmere system.
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
We want the default state of the HP_MUTE signal to be asserted, so that
the headphones are muted before the first audio playback. Without this,
the headphones are left unmuted until shortly after the first audio
playback completes.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Run the data through cpu_to_be16() so it's at least clear what we're up to.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Commit af46800 ("ASoC: Implement mux control sharing") introduced
function dapm_is_shared_kcontrol.
When this function returns true, the naming of DAPM controls is derived
from the kcontrol_new. Otherwise, the name comes from the widget (and
possibly a widget's naming prefix).
A bug in the implementation of dapm_is_shared_kcontrol made it return 1
in all cases. Hence, that commit caused a change in control naming for
all controls instead of just shared controls.
Specifically, a control is always considered shared because it is always
compared against itself. Solve this by never comparing against the widget
containing the control being created.
Equally, controls should never be shared between DAPM contexts; when the
same codec is instantiated multiple times, the same kcontrol_new will be
used. However, the control should no be shared between the multiple
instances.
I tested that with the Tegra WM8903 driver:
* Shared is now mostly 0 as expected, and sometimes 1.
* The expected controls are still generated after this change.
However, I don't have any systems that have a widget/control naming
prefix, so I can't test that aspect.
Thanks for Jarkko Nikula for pointing out how to fix this.
Reported-by: Liam Girdwood <lrg@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-next' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6: (57 commits)
regulator: Fix 88pm8607.c printk format warning
input: Add support for Qualcomm PMIC8XXX power key
input: Add Qualcomm pm8xxx keypad controller driver
mfd: Add omap-usbhs runtime PM support
mfd: Fix ASIC3 SD Host Controller Configuration size
mfd: Fix omap_usbhs_alloc_children error handling
mfd: Fix omap usbhs crash when rmmoding ehci or ohci
mfd: Add ASIC3 LED support
leds: Add ASIC3 LED support
mfd: Update twl4030-code maintainer e-mail address
mfd: Correct the name and bitmask for ab8500-gpadc BTempPullUp
mfd: Add manual ab8500-gpadc batt temp activation for AB8500 3.0
mfd: Provide ab8500-core enumerators for chip cuts
mfd: Check twl4030-power remove script error condition after i2cwrite
mfd: Fix twl6030 irq definitions
mfd: Add phoenix lite (twl6025) support to twl6030
mfd: Avoid to use constraint name in 88pm860x regulator driver
mfd: Remove checking on max8925 regulator[0]
mfd: Remove unused parameter from 88pm860x API
mfd: Avoid to allocate 88pm860x static platform data
...
* 'omap-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tmlind/linux-omap-2.6: (33 commits)
OMAP3: PM: Boot message is not an error, and not helpful, remove it
OMAP3: cpuidle: change the power domains modes determination logic
OMAP3: cpuidle: code rework for improved readability
OMAP3: cpuidle: re-organize the C-states data
OMAP3: clean-up mach specific cpuidle data structures
OMAP3 cpuidle: remove useless SDP specific timings
usb: otg: OMAP4430: Powerdown the internal PHY when USB is disabled
usb: otg: OMAP4430: Fixing the omap4430_phy_init function
usb: musb: am35x: fix compile error when building am35x
usb: musb: OMAP4430: Power down the PHY during board init
omap: drop board-igep0030.c
omap: igep0020: add support for IGEP3
omap: igep0020: minor refactoring
omap: igep0020: name refactoring for future merge with IGEP3
omap: Remove support for omap2evm
arm: omap2plus: GPIO cleanup
omap: musb: introduce default board config
omap: move detection of NAND CS to common-board-devices
omap: use common initialization for PMIC i2c bus
omap: consolidate touch screen initialization among different boards
...
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Cc: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.
Cc: Matti Aaltonen <matti.j.aaltonen@nokia.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
Commit 52ba67b ("ASoC: Force all DAPM contexts into the same bias state")
powers up all the DAPM contexts in a card if any DAPM context becomes
active. Unfortunately power down newer happens if per-card DAPM context
doesn't have any widgets.
Reason for this is that power state of per-card DAPM context without
widgets is never cleared and thus all the DAPM contexts remain permanently
active. Test for widgetless calling DAPM context in dapm_power_widgets()
doesn't work for per-card DAPM context since power change is never
originating from widgetless per-card DAPM context.
Fix this by pre-clearing power state flag of non-codec DAPM context at the
beginning of power sequence.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Codec output pin should be defined with SND_SOC_DAPM_OUTPUT.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's enough to include linux/delay.h just once in
sound/soc/codecs/wm8915.c, so remove the duplicate.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
at91sam9g20 is providing master clock to wm8731: not using a crystal but an
external MCLK. We can avoid conflict and save power using WM8731_SYSCLK_MCLK as
we do not need oscillator to be powered.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The crystal oscillator is only enabled if the WM8731_SYSCLK_XTAL master clock
is specified. Fix the connected() struct snd_soc_dapm_route function to take
this into account. Oscillator is not enabled on machine that need it otherwise.
Machine drivers have to make sure that they use the proper SYSCLK value.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We do not have to free a resource that is not allocated yet.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For cards that have two or more DAIs, snd_soc_resume's loop over all
DAIs ends up calling schedule_work(deferred_resume_work) once per DAI.
Since this is the same work item each time, the 2nd and subsequent
calls return 0 (work item already queued), and trigger the dev_err
message below stating that a work item may have been lost.
Solve this by adjusting the loop to simply calculate whether to run the
resume work immediately or defer it, and then call schedule work (or not)
one time based on that.
Note: This has not been tested in mainline, but only in chromeos-2.6.38;
mainline doesn't support suspend/resume on Tegra, nor does the mainline
Tegra ASoC driver contain multiple DAIs. It has been compile-checked in
mainline.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Introduce bit-flags indicating the necessary controller quirks, and
set them in pci driver_data field. This simplifies the checks in the
driver code and avoids the pci-id lookup in different places.
Also, this patch adds the PCI ID entry for AMD Hudson. AMD Hudson
requires a similar workaround like ATI SB while other generic ATI and
AMD controllers don't need but some ATI-HDMI quirks. So, we need a
different entry for Hudson.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed the wrong usage of snd_printdd() for debug prints of input
entries. It should be snd_printd() like others.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recently introduced NVIDIA GeForce GT 520 has 4 pins within a single
codec. Bump MAX_HDMI_PINS to accomodate this. Also bump MAX_HDMI_CVTS
to match it; this might be needed later too.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the PCM period size is set larger than 10 seconds, currently the
PCM core may abort the operation with DMA-error due to the fixed timeout
for 10 seconds. A similar problem is seen in the drain operation that
has a fixed timeout of 10 seconds, too.
This patch fixes the timeout length depending on the period size and
rate, also including the consideration of no_period_wakeup flag.
Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The comment does not reflect reality anymore since the multi-component
monster patch landed. Things are matched by names now, and not by
exporting and referencing a struct. Fix it to avoid confusion.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In the previous commit 'ASoC: davinci-pcm: convert to BATCH mode', the phase
offset of 2 was mentioned in the commit message but not well commented in the
source.
Add descriptive comments of the phase offset with and without ping-pong
buffers enabled.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The davinci-pcm driver's snd_pcm_ops pointer function currently calls into
the edma controller driver to read the current positions of the edma channels
to determine pos to return to the ALSA framework. In particular,
davinci_pcm_pointer() calls edma_get_position() and the latter has a comment
indicating that "Its channel should not be active when this is called" whereas
the channel is surely active when snd_pcm_ops.pointer is called.
The operation of davinci-pcm in capture and playback appears to follow close
the other pcm drivers who export SNDRV_PCM_INFO_BATCH except that davinci-pcm
does not report it's positions from pointer() using the last transferred
chunk. Instead it peeks directly into the edma controller to determine the
current position as discussed above.
Convert the davinci-pcm driver to BATCH mode: count the periods elapsed in the
prtd->period member and use its value to report the 'pos' to the alsa
framework in the davinci_pcm_pointer function.
There is a phase offset of 2 periods between the position used by dma setup
and the position reported in the pointer function. Either +2 in the dma
setup or -2 in the pointer function (with wrapping, both) accounts for this
offset -- I opted for the latter since it makes the first-time setup clearer.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extract functions that modify the prtd->period member in preparation for
conversion to BATCH mode playback.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The release of the dma channels was being performed in prepare and there was a
edma_resume call for the asp-channel only being executed on START, RESUME and
PAUSE_RELEASE.
The mcasp on da850evm with ping-pong buffers enabled was exhibiting an audible
glitch on every playback after the first. It was determined through trial and
error that the following two changes fix this problem:
1) Move the edma_start calls from prepare to trigger and 2) reverse the order
of starting the asp and ram channels.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on the registration of davinci-mcasp.1 in the davinci-evm platform
setup for da830 and dm6467, davinci-pcm can handle more than the currently
reported maximum channels of 2.
Increase the maximum channels to 384 to match the maximum reported by
davinci-mcasp.1.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Based on the data_type test in ping_pong_dma_setup, davinci-pcm is capable of
handling data of width up to and including 32bits.
"
if ((data_type == 0) || (data_type > 4)) {
printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type);
return -EINVAL;
}
"
Update the .format member of the snd_pcm_hardware instances it registers to
reflect this capability.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The setup of the pong channel uses EDMA_CHAN_SLOT instead of & 0x3f as the
setup of the ping channel does.
Make the setup of ping and pong symmetric. There is no functional change
introduced by this patch.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make use of the freshly introduced methods to re-use standard mixer
handling and add some controls that are hidden but implemented in a
standard conform way on M-Audio's FastTrack devices.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Original-code-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This quirk type will let the driver assume that there is a standard
mixer on a given interface, or that a specific mixer quirks will handle
the device.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to allow quirks functions to hook up to the standard feature
unit op tables, this patch exports a pointer to the struct that is used
internally.
That way, all the code handling the control can be kept private, and
external code can reference the symbol to re-use it.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch renames add_control_to_empty() to snd_usb_mixer_add_control()
and exports it, so the quirks functions can make use of it.
Also, as "struct mixer_build" is private to mixer.c, rewrite the
function to take an argument of type "struct usb_mixer_interface"
instead.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This change unifies the initial handling of a pin's state with the code to
update a pin's state after a hotplug (unsolicited response) event. The
initial probing, and all updates, are now routed through hdmi_present_sense.
The stored PD and ELDV status is now always derived from GetPinSense verb
execution, and not from the data in the unsolicited response. This means:
a) The WAR for NVIDIA codec's UR.PD values ("old_pin_detect") can be
removed, since this only affected the no-longer-used unsolicited
response payload.
b) In turn, this means that most NVIDIA codecs can simply use
patch_generic_hdmi instead of having a custom variant just to set
old_pin_detect.
c) When PD && ELDV becomes true, no extra verbs are executed, because the
GetPinSense that was previously executed by snd_hdmi_get_eld (really,
hdmi_eld_valid) has simply moved into hdmi_present_sense.
d) When PD && ELDV becomes false, there is a single extra GetPinSense verb
executed for codecs where old_pin_detect wasn't set, i.e. some NVIDIA,
and all ATI/AMD and Intel codecs. I doubt this will be a performance
issue.
The new unified code in hdmi_present_sense also ensures that eld->eld_valid
is not set unless eld->monitor_present is also set. This protects against
potential invalid combinations of PD and ELDV received from HW, and
transitively from a graphics driver.
Also, print the derived PD/ELDV bits from hdmi_present_sense so the kernel
log always displays the actual state stored, which will differ from the
values in the unsolicited response for NVIDIA HW where old_pin_detect was
previously set.
Finally, a couple of small tweaks originally by Takashi:
* Clear the ELD content to zero before reading it, so that if it's not
read (i.e. when !(PD && ELDV)) it's in a known state.
* Don't show ELD fields in /proc ELD files when the ELD isn't valid.
The only possibility I can see for regression here is a codec where the
GetPinSense verb returns incorrect data. However, we're already exposed
to that, since that data is used (a) from hdmi_add_pin to set up the
initial pin state, and (b) within snd_hda_input_jack_report to query
a pin's presence value. As such, I don't believe any HW has bugs here.
Includes-changes-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The microphone input on the back panel (pink connector)
stopped operating correctly after an upgrade from
2.6.35 to 2.6.38; the actual problem manifests itself
as a lack of microphone bias voltage (VREF_HIZ) on
node 0x17.
With AD1988_6STACK_DIG the maximum bias voltage (VREF_80)
is applied and the headset operates correctly.
Signed-off-by: Tony Vroon <tony@linx.net>
Tested-by: Doug Redlich <pbrigade@nxltech.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* git://git.kernel.org/pub/scm/linux/kernel/git/brodo/pcmcia-2.6:
pcmcia: Make struct pcmcia_device_id const, sound drivers edition
staging: pcmcia: Convert pcmcia_device_id declarations to const
pcmcia: Convert pcmcia_device_id declarations to const
pcmcia: Make declaration and uses of struct pcmcia_device_id const
pcmcia/sa1100: put sa11x0_pcmcia_hw_init[] to .devinit.data
Currently CODEC and platform drivers have their module reference count
incremented soc_probe_dai_link() whilst CPU DAI drivers have their reference
count incremented in soc_bind_dai_link().
CPU DAIs should have their reference count incremented in soc_probe_dai_link()
just like the CODEC and platform drivers.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit f0fba2ad (ASoC: multi-component - ASoC Multi-Component Support)
broke support for Raumfeld platforms as it didn't take into account the
different hardware features on individual devices.
In particular, Raumfeld speakers have no S/PDIF output, so the members
of the snd_soc_card struct must be set dynamically.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
This patch is preparation of cleanup suspend/resume patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver was using saved_xxx variable for suspend/resume.
OTOH, the start and stop of power/clock are controlled by
fsi_hw_startup/fsi_hw_shutdown in current FSI driver.
The all necessary registers value are set by fsi_hw_startup.
So, if fsi_hw_shutdown is called when "suspend" is generated,
and fsi_hw_startup is called at "resume",
the saved_xxx are not needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
FSIA/B ports is enabled by default when power-on,
and current FSI is supporting RuntimePM.
In addition, current fsi_module_init/kill doesn't care
simultaneous playback/recorde.
This mean FSI port control is not needed.
This patch remove fsi_module_init/kill
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fsi_stream_push/pop might be called in same time.
This patch protect it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pm_runtime_get/put_sync were used to access FSI register in fsi_dai_set_fmt
which is called when ALSA probe.
But this register value will disappear after pm_runtime_put_sync
if platform is supporting RuntimePM.
To solve this issue, this patch adds new variable for format,
and remove pm_runtime_get/put_sync from fsi_dai_set_fmt.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some variables on this driver were a unclear naming,
and were different unit (byte, frame, sample).
And some functions had wrong name
(ex. it returned "sample width" but name was "fsi_get_frame_width").
This patch tidy-up this issue, and the minimum unit become "sample".
Special thanks to Takashi YOSHII.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Using fsi_irq_enable/disable in fsi_port_start/stop is very natural.
This patch is preparation of cleanup suspend/resume patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver is using set_rate call back function which is for
master mode.
By this patch, it is used from fsi_set_master_clk.
This patch is preparation of cleanup suspend/resume patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is possible to create buff_len and period_len
from substream->runtime.
This patch is preparation of tidyup unclear variable naming patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix some logic failures in auto-mute handling in Conexant auto-parser.
Also, modify codes to be a bit more understandable.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'tty-next' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/tty-2.6: (48 commits)
serial: 8250_pci: add support for Cronyx Omega PCI multiserial board.
tty/serial: Fix break handling for PORT_TEGRA
tty/serial: Add explicit PORT_TEGRA type
n_tracerouter and n_tracesink ldisc additions.
Intel PTI implementaiton of MIPI 1149.7.
Kernel documentation for the PTI feature.
export kernel call get_task_comm().
tty: Remove to support serial for S5P6442
pch_phub: Support new device ML7223
8250_pci: Add support for the Digi/IBM PCIe 2-port Adapter
ASoC: Update cx20442 for TTY API change
pch_uart: Support new device ML7223 IOH
parport: Use request_muxed_region for IT87 probe and lock
tty/serial: add support for Xilinx PS UART
n_gsm: Use print_hex_dump_bytes
drivers/tty/moxa.c: Put correct tty value
TTY: tty_io, annotate locking functions
TTY: serial_core, remove superfluous set_task_state
TTY: serial_core, remove invalid test
Char: moxa, fix locking in moxa_write
...
Fix up trivial conflicts in drivers/bluetooth/hci_ldisc.c and
drivers/tty/serial/Makefile.
I did the hci_ldisc thing as an evil merge, cleaning things up.
TLV320AIC33, TLV320AIC34 and I believe others too in this family have some
hw bugs that cause that analogue and digital VDD supplies remain leaking
up to a few mA of current after certain use cases even the hw blocks inside
codec are driven to off.
Highest leakages occur after using the bypass paths inside codec but it
is possible to get smaller leakages just by toggling mute switches in
unused audio paths (i.e. no DAPM changes) while codec is on due another
active audio path.
While some cases are able to workaroud by making sure that e.g. output mixer
switches are muted before powering down the output stage this doesn't help
all the cases.
Therefore use the software reset command to clear possible leakage currents
since that works in every cases and affects only this codec instance. Only
drawback is that now cache sync is required everytime when codec bias comes
out from bias off state, not only when supply regulators were off.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
There is no need to sync first two registers from cache to hw after a reset.
First one is used to select page for register access and this driver is
normally accessing page 0 only. Second one does a software reset which is
obviously unneeded after hardware or previous software reset command.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Add model=asus quirk for Lenovo Ideapad U350 to make internal mic
work correctly.
Cc: stable@kernel.org (2.6.38+)
BugLink: http://bugs.launchpad.net/bugs/751681
Reported-by: Kent Baxley <kent.baxley@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 22de71b ("ASoC: core - allow ASoC more flexible machine name")
writes "(null)" to driver name string in struct snd_card if card->driver_name
is NULL. This causes segmentation faults with some user space ALSA utilities
like aplay and arecord.
Fix this by using the card->name if no driver name is specified.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/ieee1394/linux1394-2.6:
firewire: sbp2: parallelize login, reconnect, logout
firewire: sbp2: octlet AT payloads can be stack-allocated
firewire: sbp2: omit Scsi_Host lock from queuecommand
firewire: core: use non-reentrant workqueue with rescuer
firewire: optimize iso queueing by setting wake only after the last packet
firewire: octlet AT payloads can be stack-allocated
firewire: ohci: optimize find_branch_descriptor()
firewire: ohci: avoid separate DMA mapping for small AT payloads
firewire: ohci: do not start DMA contexts before link is enabled
The current PLL configuration code for the tlc320aic26 codec appears to assume a
hardcoded system clock of 12 MHz. Use the clock value provided by the DAI_OPS
API for the calculation.
Tested using a MityDSP-L138 platform providing a 24.576 MHz clock.
Signed-off-by: Michael Williamson <michael.williamson@criticallink.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
ATI and AMD chipsets seem not providing the proper position-buffer
information, and it also doesn't provide FIFO register required by
VIACOMBO fix. It's better to use LPIB for these.
Reported-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 447ee6a7cb.
The workaround introduced by this commit seems bogus.
The AMD chipsets don't provide proper position-buffer nor FIFO value
required by VIACOMBO fix.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Whenever we are doing a read or a write through the rbtree code, we'll
cache a pointer to the rbnode. To avoid looking up the register
everytime we do a read or a write, we first check if it can be found in
the cached register block, otherwise we traverse the rbtree and finally
cache the rbnode for future use.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch prepares the ground for the actual rbtree optimization patch
which will save a pointer to the last accessed rbnode that was used
in either the read() or write() functions.
Each rbnode manages a variable length block of registers. There can be no
two nodes with overlapping blocks. Each block has a base register and a
currently top register, all the other registers, if any, lie in between these
two and in ascending order.
The reasoning behind the construction of this rbtree is simple. In the
snd_soc_rbtree_cache_init() function, we iterate over the register defaults
provided by the driver. For each register value that is non-zero we
insert it in the rbtree. In order to determine in which rbnode we need
to add the register, we first look if there is another register already
added that is adjacent to the one we are about to add. If that is the case
we append it in that rbnode block, otherwise we create a new rbnode
with a single register in its block and add it to the tree.
In the next patch, where a cached rbnode is used by both the write() and the
read() functions, we also check if the register we are about to add is in the
cached rbnode (the least recently accessed one) and if so we append it in that
rbnode block.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The prototype of the inline dummy version of tegra_i2s_debug_add
was not consistent with the real version.
Reported-by: Rhyland-Klein <rklein@nvidia.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
MODULE_ALIAS is required so that the module will auto-load based on a
platform_device registration in the board file.
While we're at it, add some other MODULE_*.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver does no longer build since at least 2.6.30 and there is a
modern ALSA replacement for it. RIP, Rot In Pieces.
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the "Wave", "Wave Surround" or "Front" Playback Volume must be
changed to 70% (i.e. -12 dB) so that distortion won't occur when
increase Bass and Treble from 50% to 100%, so the maximum gain in
Bass and Treble are +12 dB.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Afer commit aa202455ee , none of realtek
codec has hardware volume control "PCM Playback Volume" and
"PCM Playback Switch".
As Virtual Master require all slave controls must have same number of step
and dB range.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Compare pin type enum to the pin type and not the array index.
Fixes bug#0005368.
Signed-off-by: Adrian Wilkins <adrian.wilkins@nhs.net>
Cc: <stable@kernel.org> (2.6.37 and later)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The davinci-i2s driver copies the platform data for playback and capture
sram sizes which is in turn used by davinci-pcm to allocate ping-pong
buffers.
Copy also the platform data in davinci-mcasp probe.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AK4641 connected via I2S and I2C, jack detection via GPIO.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A driver for the AK4641 codec used in iPAQ hx4700 and Glofiish M800
among others.
Signed-off-by: Harald Welte <laforge@gnumonks.org>
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes the input layer beep not working on some EeePC 1000 models by
adding the subsystem id into whitelist. Otherwise the corresponding ALSA
mixer is not enabled and stays muted, resulting in no console beep.
Signed-off-by: Madis Janson <madis@cyber.ee>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AMD 8111 southbridges contain a controller for MC'97 modem. Enable support
for this controller in intel8x0m driver.
Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just as for headphones and microphone jacks, this patch adds reporting
of HDMI jack status through the input layer.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The xrun_log function was augmented with the in_interrupt parameter whereas the
empty macro definition used when xrun logging is disabled was not.
Add a third parameter to the empty macro definition so as to not cause compiler
errors when xrun logging (CONFIG_SND_PCM_XRUN_DEBUG) is disabled.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Visual inspection shows that max98095_put_eq_enum() and
max98095_put_bq_enum() each have a possible NULL deref of 'pdata'.
This change moves the NULL check above the use.
Signed-off-by: Taylor Hutt <thutt@chromium.org>
Acked-by: Peter Hsiang <Peter.Hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When debugging pcm drivers I found the "period" or "hw" prefix printed
by either XRUN_DEBUG_PERIODUPDATE or XRUN_DEBUG_PERIODUPDATE events,
respectively to be very useful is observing the interplay between
interrupt-context updates and syscall-context updates.
Similarly, when debugging overruns with XRUN_DEBUG_LOG it is useful to
see the context of the last 10 positions.
Add an in_interrupt member to hwptr_log_entry which stores the value of
the in_interrupt parameter of snd_pcm_update_hw_ptr0 when the log entry
is created. Print a "[Q]" prefix when dumping the log entries if
in_interrupt was true.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB X-Fi S51 Pro volume and mute from the volume knob on the unit.
Compiled and tested with 2.6.39-rc7-git12
Signed-off-by: Mathieu Bouffard <mbouffard@strangequarks.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
That way, the class compliant MIDI interface is also handled.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Grant Diffey <gdiffey@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the interface can't report a clock's validity, assume that it's
valid.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Vicente Joel <vicentejoel@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This was a flaw in the reading of the spec tables - Native Instrument's
"Komplete Audio 6" device has no such extra controls.
This patch also fixes the device name in two comments.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just in case a prototype changes, we'll be warned. This also fixes a
sparse warning.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just defining it to nothing is dangerous as it can alter the code
execution flow, for example when used in as only function in a
conditional code block.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AMD Hudson controllers give noisy outputs when the buffer data is
rewritten on the fly as PulseAudio does. This seems fixed by the
snoop bit enabled just like ATI chipset.
Also, disable 64bit DMA as now, to be sure.
We can revisit this later.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a docking-station has a line-in jack, we can handle it also as
a detectable jack just like mic-in. This will improve the usability
of HP laptops with a docking-station.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In addition to the normal mic jack, the mic (or line-in) jack on the
docking-station is checked also as a candidate for auto-selection.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In addition to the normal external mic jack, check also the mic jack
on a docking-station as well, and select the input source appropriately.
The similar functionality was already implemented in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement the same functionality as Realtek's auto-mute mode control.
Now Conexant auto-parser can also mutes line-out and provide the enum
control for different automute behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enabling DEBUG_STRICT_USER_COPY_CHECKS causes the following
warning:
In file included from arch/x86/include/asm/uaccess.h:573,
from include/linux/poll.h:14,
from include/sound/pcm.h:29,
from include/sound/ac97_codec.h:31,
from sound/soc/soc-core.c:34:
In function 'copy_from_user',
inlined from 'codec_reg_write_file' at
sound/soc/soc-core.c:252:
arch/x86/include/asm/uaccess_64.h:65:
warning: call to 'copy_from_user_overflow' declared with
attribute warning: copy_from_user() buffer size is not provably
correct
presumably due to buf_size being signed causing GCC to fail to
see that buf_size can't become negative.
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit fafd217 ("ASoC: Store a list of widgets in a DAPM mux/mixer kcontrol")
changed the control private data type that is passed to snd_soc_cnew when
creating dapm mixer and mux controls. Commit did not update a few codec
drivers that are using their own put callbacks and thus are accessing a
wrong data type.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Don't want to upset the DSP.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
If we're setting the currently applied value for one of the DSP algorithm
configurations we can just skip all the handling as the control set is a
noop. This ensures we do not disrupt a running DSP.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
Also fix a left/right typo while we're at it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com.
Cc: stable@kernel.org
There are no signs of a dmic at node 0x0b, so the user is left with
an additional internal mic which does not exist. This commit removes
that non-existing mic.
Cc: stable@kernel.org (2.6.32+)
BugLink: http://bugs.launchpad.net/bugs/731706
Reported-by: James Page <james.page@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check the routing more exactly for avoiding the duplicated controls for
the very same effect for multiple capture routes in Conexant auto-parser.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the intermediate selector widget in the capture path provides the
boost volume, create the corresponding volume control.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to removing ARCH_S5P6442, we don't need to support
sound for S5P6442.
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Kukjin Kim <kgene.kim@samsung.com>
Added digital mute function in DAI1 and DAI2.
Signed-off-by: Jin Park <jinyoungp@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Moved the EX Limiter Mode from dapm widget to control, because it was not
required DAPM route.
Signed-off-by: Jin Park <jinyoungp@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixed invalid register definitions in mixer controls such as left
speaker mixer, left hp mixer and left rec mixer.
Signed-off-by: Jin Park <jinyoungp@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is an unfortunate difference in return values between spi_write()
and i2c_master_send() so we need an adaptor function to translate.
Reported-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Just for safety reason (for avoiding any possible regressions), don't
enable auto-parser as default for cxt5045 and 5051, as well as 5047.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove TEA575X_RADIO define from fm801.c.
Also update Kconfig help text to include all supported cards.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide real card and bus_info instead of hardcoded values.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
struct snd_card *card is present in struct snd_tea575x but never used.
Remove it.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
freq_fixup is a constant, no need to hold it in struct snd_tea575x and set in
each driver.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similarly like other Conexant codecs, now model=auto is supported for
cxt5047.
But the auto-parser mode isn't activated as default yet, since BIOS
pin-configs seem often broken on machines with this codec. User need
to pass model=auto explicitly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Extend the existing auto-parser for CX2064x for cxt5051 codec.
Now the auto-parser supports ADC-switching for this codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of keeping always EAPD on, turn on/off appropriately at jack
plugging in Conexant auto-parser mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The board support has never been merged for it as noticed
by Russell King <linux@arm.linux.org.uk>. So let's remove the
related dead code.
Cc: linux-fbdev@vger.kernel.org
Cc: alsa-devel@alsa-project.org
Cc: Paul Mundt <lethal@linux-sh.org>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Tomi Valkeinen <tomi.valkeinen@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Tony Lindgren <tony@atomide.com>
Current checks for cpu type were too restrictive leading
to failures for other silicons in same family.
The problem was found while testing audio playback on
AM37x and AM35x processors. But should exist on OMAP36xx
as well.
Signed-off-by: Sanjeev Premi <premi@ti.com>
cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
cc: Liam Girdwood <lrg@ti.com>
cc: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Allow disabling ALSA period wakeup interrupts.
This can only be done on OMAP2+ (2/3/4), since there
we can chain the DMA.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Fix the TWL6040 LINEGAIN volume control to match the TRM.
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Digital microphones can have some additional elements in their
audio path (like microphone bias). An input widget is required
for digital microphone CODEC driver to allow external connections
in machine drivers.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow ASoC machine drivers to register a driver name
and a longname. This allows user space to determine
the flavour of machine driver.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
receive_buf() was recently changed to return the number of bytes
received but the cx20442 driver wasn't updated to match the new API.
I don't have any hardware but since we don't actually appears to be
listening to the data at all just report that we accepted all the data
that was offered to us.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
Fix printk format warnings in wm8958-dsp2.c:
sound/soc/codecs/wm8958-dsp2.c:103: warning: format '%d' expects type 'int', but argument 4 has type 'size_t'
sound/soc/codecs/wm8958-dsp2.c:111: warning: format '%d' expects type 'int', but argument 3 has type 'size_t'
sound/soc/codecs/wm8958-dsp2.c:144: warning: format '%d' expects type 'int', but argument 5 has type 'size_t'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
It will be non-functional with the basises and clocks off anyway, if the
system needs microphone detection enabled over suspend then it should be
causing the CODEC to ignore suspend using the APIs for that to prevent
the biases being disabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
do_spi_write() is just an open coded copy of do_spi_write() so we can
delete it and just call spi_write() directly. Indeed, as a result of
recent refactoring all the SPI write functions are just very long
wrappers around spi_write() which don't add anything except for some
pointless copies so we can just use spi_write() as the hw_write
operation directly. It should be as type safe to do this as it is to do
the same thing with I2C and it saves us a bunch of code.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
snd_soc_4_12_spi_write() contains a byte swap. Since this code was written
for an Analog CODEC on a Blackfin reference board it appears that this is
done because while Blackfin is little endian the CODEC is big endian (as
are most CODECs).
Push this up into the generic 4x12 write function and use cpu_to_be16() to
do the byte swap so things are more regular and things work on both CPU
endiannesses.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Currently we'll force all registers to fit in 8 bits before passing
down to the I/O function. Looks like a cut'n'paste bug.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
sparse complains if 0 is used as a NULL pointer constant.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Lockdep complains about conflicts between isight->mutex,
ALSA's register_mutex, mm->mmap_sem, and pcm->open_mutex.
This can be fixed by moving the calls to isight_pcm_abort(),
snd_card_disconnect(), and fw_iso_resources_update() out of
isight->mutex. These functions are designed to be called
asynchronously; the mutex needs to protect only the device
streaming state modified by isight_start/stop_streaming().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Experiments have shown this driver to work now.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When aborting a PCM stream, the xrun is signaled only if the stream is
running. When disconnecting a PCM stream, calling snd_card_disconnect()
too early would change the stream into a non-running state and thus
prevent the xrun from being noticed by user space.
To prevent this, move the snd_card_disconnect() call after the xrun.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
which is needed to get the iSight to talk.
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the .header_size field when queueing packets to avoid a division by
zero.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After handling a received packet, we want to resubmit the same packet,
so do not increase the packet index too early.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix crashes in isight_pcm_abort() that happen when the driver tries to
access isight->pcm->runtime which does not exist when the device is not
open. Introduce a new field pcm_active to track this state.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds an experimental driver for the front and rear microphones of
the Apple iSight web camera.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Autodetect TEA575x tuner connection type during init. This allows tuner to
work out-of-the box.
tea575x_tuner module parameter remains functional to force tuner type.
Tested with SF256-PCP and SF64-PCR.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make declarations of struct pcmcia_device_id const.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Dominik Brodowski <linux@dominikbrodowski.net>
When queueing iso packets, the run time is dominated by the two
MMIO accesses that set the DMA context's wake bit. Because most
drivers submit packets in batches, we can save much time by
removing all but the last wakeup.
The internal kernel API is changed to require a call to
fw_iso_context_queue_flush() after a batch of queued packets.
The user space API does not change, so one call to
FW_CDEV_IOC_QUEUE_ISO must specify multiple packets to take
advantage of this optimization.
In my measurements, this patch reduces the time needed to queue
fifty skip packets from userspace to one sixth on a 2.5 GHz CPU,
or to one third at 800 MHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
We do not need slab allocations anymore in order to satisfy
streaming DMA mapping constraints, thanks to commit da28947e7e
"firewire: ohci: avoid separate DMA mapping for small AT payloads".
(Besides, the slab-allocated buffers that firewire-core, firewire-sbp2,
and firedtv used to provide for 8-byte write and lock requests were
still not fully portable since they crossed cacheline boundaries or
shared a cacheline with unrelated CPU-accessed data. snd-firewire-lib
got this aspect right by using an extra kmalloc/ kfree just for the
8-byte transaction buffer.)
This change replaces kmalloc'ed lock transaction scratch buffers in
firewire-core, firedtv, and snd-firewire-lib by local stack allocations.
Perhaps the most notable result of the change is simpler locking because
there is no need to serialize usages of preallocated per-device buffers
anymore. Also, allocations and deallocations are simpler.
Signed-off-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Increase the range of the Digital Capture Volume control to be 120 steps.
Each step is 0.75dB, and the range starts at -72dB, giving a max setting
of 18dB, which matches the latest datasheet, to the precision of the step
size.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use common functions to access TEA575x tuner - remove original read/write
functions and provide new pin manipulation functions instead.
Also convert the original triple implementation to a simple GPIO pin map.
Tested with SF256-PCP and SF64-PCR (added the GPIO pin for MO/ST signal
for them).
SF256-PCS untested (pin for MO/ST signal is a guess).
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use common functions to access TEA575x tuner - remove original read/write
functions and provide new pin manipulation functions instead.
Tested with SF64-PCE2 card.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement generic read/write functions to access TEA575x tuners. They're now
implemented 4 times (once in es1968 and 3 times in fm801).
This also allows mute to work on all cards.
Also improve tuner detection/initialization.
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Also fix the maximum value for the capture volume control.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Model the power supply for the digital core as a DAPM_SUPPLY widget. This allows
to cleanup the code a bit.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The SSM2603 is mostly register compatible with the SSM2602 and can be supported
by the current driver without any changes.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The SSM2604 is basically a lightweight variant of the SSM2602 with a compatible
register layout. Thus we can easily support both devices by the same driver,
by providing a slightly set of controls, widgets and routes.
Compared to the SSM2602 the SSM2604 has no microphone input and no headphone
output.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It is not required to have the codec powered at this stage and DAPM will power
the ADC and DAC down again after probe has run anyway.
Thus we avoid some unnecessary writes by this change.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some of the values in the default register cache did not represent the codecs
state after reset. This patch fixes it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Define POWER_OFF_ON_STANDBY cause trobles when trying to get some
sound from codec because code for bias setup was not compiled
(define wasn't defined). This define was removed in commit:
cc3202f5 but again introduced by commit: f0fba2ad1 which then
completely break codec functionality so remove it again.
Signed-off-by: Marek Belisko <marek.belisko@open-nandra.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Those are leftovers from a pre-multicomponent era.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are currently two controls which allow selecting the capture source, one
as a normal control, the other as part of a DAPM_MUX widget.
Remove the normal control.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Drop unused field from the coeff struct, precalculate the srate register at
compile-time and cleanup up the naming.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
reg_cache_size is supposed to be the number of elements in the register cache,
not the size in bytes.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The 'Mic Boost2' control's shift was off by one and thus was not working.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Annotate the i2c probe and remove functions with __devinit and __devexit.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we specifically want to write a block of data to the hw bypassing the
cache, then allow this to happen inside snd_soc_hw_bulk_write_raw().
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows to create DAPM routes depending on those widgets in the
codecs probe function. This is helpful when supporting similar codecs
with minor differences in the DAPM routing with the same driver.
Something similar has already been done for cards in commit
a841ebb9 (ASoC: Create card DAPM widgets early so they can be used in
callbacks).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AMD chipsets often behave pretty badly regarding the DMA position
reporting. It results in the bad quality audio recording.
Using position_fix=3 works well in general for them, so let's enable
it as default for AMD.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Actually the current code is perfectly sensible given the hardware.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The move over to exposing snd_soc_register_card() let the initialisation
of the driver data we use to find the card in PM operations go AWOL. Fix
this by setting the driver data when we register the card.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Control sharing is enabled when two widgets include pointers to the
same kcontrol_new in their definition. Specifically:
static const struct snd_kcontrol_new adcinput_mux =
SOC_DAPM_ENUM("ADC Input", adcinput_enum);
static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = {
SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux),
SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux),
};
This is useful when a single register bit or field affects multiple
muxes at once. The common case is to have separate control bits or
fields for each mux (channel). An alternative way of looking at this
is that the mux is a stereo (or even n-channel) mux, rather than
independant mono muxes.
Without this change, a separate kcontrol will be created for each
DAPM_MUX. This has the following disadvantages:
* Confuses the user/programmer with redundant controls that don't
map to separate hardware.
* When one of the controls is changed, ASoC fails to update the DAPM
logic for paths solely affected by the other controls impacted by
the same register bits. This causes some paths not to be correctly
powered up or down. Prior to this change, to work around this, the
user or programmer had to manually toggle all duplicate controls away
from the intended setting, and then back to it.
Control sharing implies that the control is named based on the
kcontrol_new itself, not any of the widgets that are affected by it.
Control sharing is implemented by: When creating kcontrols, if a
kcontrol does not yet exist for a particular kcontrol_new, then a new
kcontrol is created with a list of widgets containing just a single
entry. This is the normal case. However, if a kcontrol does already
exists for the given kcontrol_new, the current widget is simply added
to that kcontrol's list of affected widgets.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A future change will allow multiple widgets to be affected by the same
control. For example, a single register bit that controls separate muxes
in both the L and R audio paths.
This change updates the code that handles relevant controls to be able
to iterate over a list of affected widgets. Note that only the put
functions need significant modification to implement the iteration; the
get functions do not need to iterate, nor unify the results, since all
affected widgets reference the same kcontrol.
When creating the list of widgets, always create a 1-sized list, since
the control sharing is not implemented in this change.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Future changes will need reference to the kcontrol created for a given
kcontrol_new. Store the created kcontrol values now.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A future change will modify struct snd_soc_dapm_widget to store the
actual kcontrol pointers for each kcontrol_new in a field named
kcontrols. Rename the existing kcontrols field to enable this.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The i2s shutdown callback has the check whether it should be disabled reversed.
Currently it is disabled if another stream is still active, but kept enabled if
the last stream is closed. This patch fixes it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the DAPM debugfs entries before freeing the context's widgets, otherwise a
use after free situation might occur.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently debugfs entries for a DAPM widgets are only added in
snd_soc_dapm_debugfs_init. If a widget is added later (for example in the
dai_link's probe callback) it will not show up in debugfs.
This patch moves the creation of the widget debugfs entry to
snd_soc_dapm_new_widgets where it will be added after the widget has been
properly instantiated.
As a side-effect this will also reduce the number of times the DAPM widget list
is iterated during a card's instantiation.
Since it is possible that snd_soc_dapm_new_widgets is invoked form the codecs or
cards probe callbacks, the creation of the debugfs dapm directory has to be
moved before these are called.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the creation of the DAPM debugfs directory to snd_soc_dapm_debugfs_init
instead of having the same duplicated code in both codec and card DAPM setup.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Free the card's DAPM context when the card is removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Acked-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Renamed to Digial SRC Capture Switch for more correct representation.
Also fixed analog volume control on Lola161611 and lola881.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For assuring the synchronized state with the pause operation,
loop over the all linked streams and waits until all get ready
in a loop.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the refcounting for the exclusive SRC control.
Also, fixed the possible stall after PCM pause operations.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added granularity and sample_rate_min module options.
The former controls the h/w access granularity. As default, it's set
to the max value 32.
The latter controls the minimum sample rate in Hz, as default 16000.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a single BDL for both buffers instead of allocating for each.
Also a few tune-up to avoid the stream stalls in the PCM code and
the prelimianry work for SG-buffer support are added, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec proc file becomes a read only that shows the codec widgets
in a text form. A new proc file, codec_rw, is introduced instead for
accessing the Lola verb directly by reading and writing to it.
Also, regs proc file shows the contents of DSD, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a new driver for supporting Digigram Lola PCI-e boards.
Lola has a similar h/w design like HD-audio but with extended verbs.
Thus the driver is written similarly like HD-audio driver in the bus
part. The codec part is rather written in a fixed way specific to the
Lola board because of the verb incompatibility.
The driver provides basic PCM, supporting multi-streams and mixing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix NULL-dereference when try to use alt_playback since those codecs
which support multistreaming playback usually have more than 1 adc but
the driver should create alt_capture when spec->stream_analog_alt_capture
is also defined.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check of chained fixup list entry was done against the wrong element.
A stupid mistake during refactoring.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit c6b358748e.
It turned out that there are different pin configurations for this
PCI SSID, including multi-channel modes. And more proper fix for
allowing line-out mutes will come up in 2.6.40 tree, so we won't need
this fixup any more there.
Reported-by: Andrew Clayton <andrew@digital-domain.net>
Reported-by: Emmanuel Benisty <benisty.e@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI SSID is 1025:031c and the codec SSID is 1025:031d,
so the driver mistakes this for a SKU value, but looking at
the numbers, this is obviously wrong.
Cc: stable@kernel.org (2.6.38+)
BugLink: http://bugs.launchpad.net/bugs/761861
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PC Beep was not being reported as enabled on my EeePC 901:
SKU: enable_pcbeep=0x0
Signed-off-by: Daniel Cordero <danielcordero@lavabit.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support for the Terratec Aureon 7.1 USB which uses a
C-Media cm6206 and needs all the quirks already found in the past.
Signed-off-by: Wolfgang Breyha <wbreyha@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Auto-Mute Mode control is useful even when only two outputs
(e.g. HP and speaker) are available. Then user can enable/disable
the auto-mute behavior on the fly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not only supporting the line-out automute as additional feature
to the existing headphone automute, now the headphone jack can
mute the line-out alone even without the speaker outs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By popular demands, I add the functionality to mute / unmute the
line-out jacks per the headphone plug / unplug. For achieving this
and keeping the compatibility with the old behavior, the new mixer
enum "Auto-Mute Mode" is added. With this, user can control the
auto-mute behavior either disabled, speaker-only or lineout+speaker.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Yet another consolidation of auto-mute functions for the devices
controlling the output muts together with the master mixer switch,
typically found for ALC262 machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the common helper function and flags to support the auto-mute
per line-out jack detection, and also the mute of line-out jacks.
A few model-specific implementations are replaced with the common
helpers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models do mute on/off the connected mixer widget for the automatic
muting, instead of controlling the pin widget itself. This patch adds
the implementation of such type of auto-mute in the common helper
function, and reduces the redundant codes for each model preset.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are two entry points for the headphone automute functions for
Realtek, alc_automute_amp() and alc_automute_pin(). These call the
same function in the end, so we can basically consolidate these
with a flag in spec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Immediately after sending the last fix I realised that the CODEC DAI names
also don't correspond to the WM8994 driver. Update the DAI names to match.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This was typoed at some point in the multi-component merge, though the
driver was added along with that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In notify_aa_path_ctls(), adds 'rear mic' item and confirms the A-A
path control existing before notifying card that the A-A path volume
is muted if smart5.1 is enabled.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some crappy USB-audio devices give broken dB ranges, e.g. both min and max
are 0dB. This confuses the volume control that prefers dB expression such
as alsactl or PulseAudio. In such a case, it's much better not to expose
the broken dB information.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a helper function for searching DAPM widgets by name.
This allows to streamline functions which operate on widgets by name.
It also allows to get rid of copy'n'pasted code which was added to fallback to
widgets from other contexts if the widget was not found in the current context.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since we can now support multiple platforms allow machines to not specify
a platform in a DAI link. Since the rest of the code requires that we have
a struct device for all objects we do this by substituting in a dummy
device that we register automatically.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This allow application such as gstreamer and wine which use
snd_pcm_hw_params_set_buffer_time_near() won't fail any more
since sound chips require special containt power 2 period bytes
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the support of "Channel Mode" enum control to Realtek
auto-parser. When line-in or mic-in jacks are capable to output and
free DACs are available, the driver allows to switch to multi-channel
mode via "Channel Mode" enum switch, as already implemented in some
preset cases.
Not implemented in all Realtek codecs. Currently, ALC880, 882, 861,
662 and the compatible codecs are supported.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow alc662_dac_to_mix() and alc662_look_for_dac() to parse
down the selector widget that is found in ALC880-type codecs,
and rename them to alc_auto_*() accordingly.
This is for the next coming multi-io extensions.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For some motherboards with 5 or 6 audio jacks which had six or eight multiple
channels output, smart5.1 item is no useful and should be removed.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The workaround for AMD chipset via sync_write flag seems needed for
machines with Realtek codecs. So, it's better to activate it
generically in hda_intel.c from the beginning.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
EAPD power-down should be called also for normal shutup cases.
Let's move to there. This also fixes the compile warnings when
CONFIG_PM isn't set automatically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When bringing up audio low power modes boards may configure SYSCLK before
they actually start the FLL as we do much of the clocking setup prior to
the power up sequence.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
DMIC support is automatically disabled when none of the GPIOs are set up
to bring out the DMICCLK and DMICDAT pins at startup.
Note that there's no support for controlling DMIC routing except the power
control so the board DAPM configuration will need to manage DMIC enable and
disable if analogue mics (eg, a headset) also exist.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
We can safely divide these down to within the supported SYSCLK range.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The current davinci_mcasp_set_dai_fmt() sets bits ACLKX and ACLKR in the PDIR
register for the codec clock-master/frame-slave mode; however, this results in
the ACLKX and ACLKR pins being outputs according to SPRUFM1 [1] which
conflicts with "codec is clock master."
Similarly to the previous patch in this series, "fix _CBM_CFS hw_params" --
For codec clock-master/frame-slave mode (_CMB_CFS), clear bits ACLKX and ACLKR
in the PDIR register to set the pins as inputs and hence allow externally
sourced bit-clocks.
[1] http://www.ti.com/litv/pdf/sprufm1
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current davinci_mcasp_set_dai_fmt() sets bits ACLKXE and ACLKRE (CLKXM
and CLKRM as they are reffered to in SPRUFM1 [1]) for codec clock-slave/
frame-slave mode (_CBS_CFS) which selects internally generated bit-clock and
frame-sync signals; however, it does the same thing again for codec
clock-master/frame-slave mode (_CBM_CFS) in the very next case statement which
is incorrectly selecting internally generated bit-clocks in this mode.
For codec clock-master/frame-slave mode (_CBM_CFS), clear bits ACLKXE and
ACLKRE to select externally-generated bit-clocks.
[1] http://www.ti.com/litv/pdf/sprufm1
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current driver creates value for set/clr of PDIR using (x<<26) instead
of the #defines that are convieniently made available.
Update the driver to use the bitfield definitions of PDIR. There is no
functional change introduced by this patch.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current check for the number of tdm-slots specified by platform data is
always true (x >= 2 || x <= 32); therefore the else branch that warns of an
incorrect number of slots can never be taken.
Check that the number of tdm slots specified by platform data is between 2
and 32, inclusive.
Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: James Nuss <jamesnuss@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The FIFO of each port were always working though it was not used
in current FSI driver.
This patch add module/port clock control function for fixing it.
This patch is also caring suspend/resume.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Simon Horman <simon@horms.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI driver sets important settings when probing.
And it are not set again as long as driver is not bind again.
This mean FSI driver will lost it from register
if suspend/resume are happen.
This patch save important settings for suspend/resume.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Simon Horman <simon@horms.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If FSI port is clock master, it use set_rate function
which is callback from platform,
and it is not necessary to call it if FSI port is clock slave.
Current FSI driver called this callback if platform provide it.
This patch modify it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Simon Horman <simon@horms.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There needs to be a strong reason for overriding the Kconfig default.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The AMD chipset seems unstable in the normal operation mode, and it
seems requring more sensible access for each verb. Enabling sync_write
mode and allowing bus-reset is a sort of workaround for these chipset
stability issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_SOC_TEGRA_WM8903 is useful for many Tegra boards. To avoid the
ASoC tegra/Kconfig enumerating them all, instead have the Tegra machine
Kconfig select MACH_HAS_SND_SOC_TEGRA_WM8903 where appropriate, and have
SND_SOC_TEGRA_WM8903 depend on this.
[Redid ASoC diff so it applies. -- broonie]
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Olof Johansson <olof@lixom.net>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When SND_HDA_NEEDS_RESUME is not defined, the compiler identifies that
the following symbols are static but not used:
restore_shutup_pins
hda_cleanup_all_streams
Fix warnings by adding SND_HDA_NEEDS_RESUME guards.
Signed-off-by: Mike Waychison <mikew@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH.
Signed-off-by: Seth Heasley <seth.heasley@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This was reverted mistakenly in the recent update patch.
Fixed again.
Reported-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In addition to the currently supported analog capture path, the WM8903
also supports digital mics.
The analog and digital capture paths are exclusive; a mux is present to
select the capture source.
Logically, the mux exists to select the decimator's input, from either
the ADC or DMIC block outputs. However, the ADC power domain also
includes the DMIC interface. Consequently, this change represents the
mux as existing immediately before the ADC, and selecting between the
Input PGA and DMIC block outputs.
An alternative might be to represent the mux in its correct location,
and associate the ADC power enable controls with both the real ADC, and
a fake ADC for the DMIC?
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the equalizer and biquad filter controls.
Signed-off-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace calls to a variety of registration functions by updating
struct snd_soc_card snd_soc_tegra_wm8903 to directly point at the
various control/widget/map tables instead. The ASoC core now
performs any required registration based on these data fields.
(Applying Mark's TrimSlice review comments to the existing driver)
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
`type` parameter is not longer used in `snd_soc_codec_set_cache_io`,
so remove this line.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Not all widgets on a card are within the codec's DAPM context. Fix
snd_soc_dapm_get_pin_status to search all contexts when looking for a
widget.
This change is required when modifying tegra_wm8903 to use
snd_soc_card.widgets rather than calling snd_soc_dapm_new_controls; the
former adds the widgets to the card's DAPM context, whereas tegra_wm8903
uses the codec's DAPM context when calling snd_soc_dapm_new_controls.
By code inspection, I suspect this also applies to Samsung Speyside.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Card widgets are created in the card's DAPM context, not any codec's DAPM
context. Hence, w->codec==NULL. Instead, find the card from the widget
through the DAPM context of the widget, not the codec of the widget.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only the clock programming code needs to know whether the clocks changed,
and that is encapsulated within tegra_asoc_utils_set_rate(). The machine
driver's call to snd_soc_dai_set_sysclk(codec_dai, ...) is safe
irrespective of whether the clocks changed.
(Applying Mark's TrimSlice review comments to the existing driver)
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When the driver is not initialized/registered, nothing should be touching
these fields anyway, so there's no point clearing them out.
(Applying Mark's TrimSlice review comments to the existing driver)
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This machine driver is a platform driver, and hence will only be
instantiated on the correct machines. Hence, there is no need to
check the current machine during probe.
(Applying Mark's TrimSlice review comments to the existing driver)
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove "Front Playback Volume" and "Front Playback Switch" from emu10k1 only
for STAC9758/59
Since commit 7eae36fbd5
"Fix the confliction of 'Front' control",
the "Front Playback Volume" control created by commit
edf8e4565c
"emu10k1: Front channels via fxbus 8 and 9"
was removed
"Front Playback Volume" and "Surround Playback Volume" have same dB range
since I2S DAC of SB Live! and SB Live! Platinum does not has any hardware
volume control.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acer laptops with ALC271x needs a magic initialization for digital-mic
to make it working with mono streams (and PulseAudio).
Added a fix-up applied to Acer with ALC271x generically.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Ventana is identical to Harmony.
* Seaboard, Kaen, and Aebl are all pretty similar, mainly with slightly
different sets of GPIOs, and slightly different WM8903 pin connectivity.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Storing the struct in an array makes the assignments to the GPIO member a
little non-obvious, and is pointless when there's only a single GPIO.
(I thought I fixed this during the review cycle when first submitting this
driver, but I guess I overlooked that)
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The previous commit renames SND_TEGRA_SOC_HARMONY to SND_TEGRA_SOC_WM8903.
While we're breaking people's .config files, rename all Tegra/SOC-related
Kconfig variables to be more consistent with at least the core codec
variables. Note that there exist machines that name their variables both
ways.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Soon, this machine driver will be updated to handle a number of Tegra boards
using the WM8903 codec. Rename the file in advance to reflect this.
Fix the content of tegra_wm8903.c to match the rename; replace references
to Harmony board with something more generic.
* s/struct tegra_harmony/struct tegra_wm8903/
* s/harmony/machine/ # variable name
* Similar rename for some functions
* Similar comment fix
* Similar MODULE_DESCRIPTION fix
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Soon, this machine driver will be updated to handle a number of Tegra boards
using the WM8903 codec. Rename the platform device in advance to reflect this.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The audio driver will soon support more than just the Tegra Harmony board.
Rename the platform data header file and data type to reflect this.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch enables FSI driver autoloading on sh-mobile systems.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Reviewed-by: Simon Horman <horms@verge.net.au
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This new device by Native Instruments is also compliant to the USB
standard v2.0, but hides this detail at when connected.
It needs the same boot quirks than other models, and also has two
non-class-compliant mixer controls.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the newly introduced dapm_widgets, dpam_routes and controls fields of the
snd_soc_dai_driver struct to setup controls and DAPM.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use SND_SOC_DAPM_EVENT_OFF for determining whether the speaker should be turned
on or off instead of open coding it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch changes the qi_lb60 setup code to use gpio_request_array instead of
manually calling gpio_request and gpio_direction_output for each gpio.
Doing so makes the code a bit more compact.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the newly introduced dapm_widgets, dpam_routes and fields of the
snd_soc_card struct to setup DAPM.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit ce6120cc(ASoC: Decouple DAPM from CODECs) changed the signature of
snd_soc_dapm_widgets_new to take an pointer to a snd_soc_dapm_context instead of
a snd_soc_codec. The call to snd_soc_dapm_widgets_new in jz4740_codec_dev_probe
was not updated to reflect this change, which results in a compiletime warning
and a runtime OOPS.
Since the core code calls snd_soc_dapm_widgets_new after the codec has been
registered it can be dropped here.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Allow audio paths through the Speyside system to be kept active while the
system is suspended (for example, when on a voice call) by marking all the
external widgets and the DAI link to the WM1250-EV1 baseband module as
ignoring suspend.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Demonstrate the connection of a baseband to the system. We add a DAI for
the link to the baseband. This will become visible to the application
layer - audio should be started from the application layer using an
application such as this:
http://opensource.wolfsonmicro.com/~gg/bluetooth-pcm/bluetooth_pcm.c
which starts up audio as for CPU based playback and record up to the point
where data is streamed.
Due to non-availability of baseband simulation hardware we reuse the
configuration for the CPU link with the CODEC acting as clock master,
allowing signals to be observed with a scope. A more standard system
would have separate configuration for the baseband with its own ops
structure and operations. Normally the baseband would be clock master
as the baseband audio will be synchronised to the external telephony
network.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Pin switches enable direct control of the DAPM state from userspace,
enabling simple enabling and disabling of the path. This is especially
useful for outputs such as the speaker which are composed of several
physical devices as it allows them to be controlled as a group.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Speyside makes use of support the WM8915 has for detecting the polarity
of the microphone and ground connections on headsets, using a GPIO to
control the polarity of the ground connection and switching between the
two microphone bias supplies available on the device in order to do so.
As a result of this the detection support is more involved than for most
other CODECs, using a callback to configure the current polarity of the
jack and translate this into the board-specific connections required for
the current scenario.
On Android some additional work is required to hook this up to the
application layer as the Android HeadsetObserver monitors a custom
drivers/switch API rather than the standard Linux APIs. This can be
done by either updating HeadsetObserver or modifying the ALSA core to
report via drivers/switch as well.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Speyside includes a WM9081 configured as an external speaker driver taking
an analogue input from HPOUT2 on the WM8915 on the system. Add support for
this to the driver, using a prefix of "Sub" for the WM9081 controls to
ensure we avoid collisions with controls on the WM8915.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Dynamically enable and disable the FLL on the WM8915, configuring the
system clock to 256fs for 48kHz when the device is active but reverting
to using the input 32.768kHz clock directly at other times to support
features such as jack detection with minimal power consumption.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Provide widgets for the basic widgets connected directly to the WM8915
on Speyside - the headphones, speaker, digital and analogue microphones.
For the outputs this is just documentation, for the inputs this ensures
that the relevant microphone biases are enabled when they are in use.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This is minimal code required to get audio out of the Speyside audio
subsystem on the Wolfson Cragganmore 6410 reference platform. It sets
up the link between the CPU and AIF1 of the WM8915 on the system,
enabling audio playback via the headphone and speaker outputs of the
device (which require no further configuration except runtime). It
allows verification of basic functionality of the system.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This helps with things like setting up the initial state.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Since the WM8915 FLL is not tied to a particular audio interface move it
to a CODEC wide operation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
To get correct calibration, we can decrease the time
needed for the OSC to calibrate itself.
With this change we can save ~15ms in the OSC
calibration phase.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
The WM1250-EV1 Springbank audio I/O module for the Wolfson Glenfarclas
reference platform provides a simple audio I/O with an independant clock
domain, intended to simulate cellular modem and bluetooth subsystems
within the platform.
The card supports some limited GPIO based control but this is currently not
implemented.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The WM8915 is an ultra low power mobile CODEC designed for smartphones,
featuring a mixture of digital and analogue I/O with flexible mixing
options and advanced low power accessory detection functionality in a
compact package.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
card->num_rtd should be 0 after soc_romve_dai_link
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch add WM8580 PCM machine driver to support PCM audio
on SMDKC110, SMDKV210, SMDK6450, SMDK6440 boards.
Playback and Capture supports 8kHz sampling rates.
and It is tested on SMDKC110, SMDKV210, SMDK6450
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The output PGA was not being powered up in headphone and speaker paths,
removing the ability to offer volume control and mute with the output
PGA.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
According to the comments in include/linux/init.h:
"Pointers to __devexit functions must use __devexit_p(function_name), the
wrapper will insert either the function_name or NULL, depending on the config
options."
Fix this issue in codecs sn95031.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the inverted clocks handling for pcm cpu driver.
By using SND_SOC_DAIFMT_NB_NF, Audio noise can be generated on SMDK.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the possible dead lock shown below:
spin_lock
sst_get_stream_status
sst_period_elapsed
intel_sst_interrupt
handle_IRQ_event
handle_fasteoi_irq
do_IRQ
common_interrupt
spin_lock
sst_set_stream_status
sst_platform_pcm_trigger
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
free_irq and pm_runtime_disable should be called before
snd_soc_unregister_xxx
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Don't query connections for widgets have no connections
ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums
ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
ALSA: HDA: Fix dock mic for Lenovo X220-tablet
ASoC: format_register_str: Don't clip register values
ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare
ASoC: zylonite: set .codec_dai_name in initializer
The DAPM pin operations currently require that the specific DAPM context
that the pin being operated in is contained in be specified. With multi
component and especially with the addition of a per-card DAPM context
this isn't ideal as it means that things like disabling unused pins on
CODECs require looking up the CODEC DAPM context.
Fix this by falling back to matching a widget in any context if there isn't
a match in the current context. The code isn't ideal currently but will do
the job.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Currently we allow all DAPM contexts to determine their own bias level.
While this should in general work in most situations and will deliver the
lowest possible power it causes problems for our integration with the
card bias level as we're calling the card bias level functions for each
DAPM context even though they're card wide but don't say which CODEC
we're calling them for. Mitigate against this by forcing everything to
be in the same state.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Since we recently explicitly set the register for registerless widgets
to no register there is no longer any need to special case power updates
for them, we can allow them to be handled with the register compression
code as other widgets are.
As this is the only remaining user of dapm_generic_apply_power() and
dapm_update_bits() also remove those functions.
Noticed-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ssm2602 codec has a SPI interface as well as I2C, so add the simple
bit of glue to make it usable.
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow CODEC and card drivers to point to an array of controls from their
driver structure rather than explicitly calling snd_soc_add_controls().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the headphone and line out mixers and PGAs use the same logical
set of register bits and sequencing as the speaker mixer/PGA.
This allows ALSA controls for mute and volume on headphone and line out
to operate correctly.
Per conversation on alsa-devel, earlier datasheets indicated that the
POWER_MANAGEMENT_* register bits 0 and 1 were aliases to ANALOG_* register
bits 0 and 4, and hence only one copy of those bits was programmed.
However, later datasheets corrected this.
From: Dilan Lee <dilee@nvidia.com>
[swarren: Applied same change to headphone widgets]
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The connection lists are static and we can reuse the previous results
instead of querying via verb at each time. This will reduce the I/O
in the runtime especially for some codec auto-parsers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since we now set up the connections and mutes dynamically in the
auto-parser, all static initializations via alc662_init_verbs & co are
no longer needed. Let's drop them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
