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>
Instead of static init array, better to determine the connection and
the mute status of the pin/mixer/DAC route dynamically. This fixes the
uninitialized mixer 0x0f on ALC892.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In cases where there is only one internal mic connected to ADC 0x11,
alc275_setup_dual_adc won't handle the case, so we need to add the
ADC node to the array of candidates.
Cc: stable@kernel.org
BugLink: http://bugs.launchpad.net/bugs/752792
Reported-by: Vincenzo Pii
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The MCP7x hardware computes the audio infoframe channel count
automatically, but requires the audio driver to set the audio
infoframe checksum manually via the Nv_VERB_SET_Info_Frame_Checksum
control verb.
When audio starts playing, nvhdmi_8ch_7x_pcm_prepare sets the checksum
to (0x71 - chan - chanmask). For example, for 2ch audio, chan == 1
and chanmask == 0 so the checksum is set to 0x70. When audio playback
finishes and the device is closed, nvhdmi_8ch_7x_pcm_close resets the
channel formats, causing the channel count to revert to 8ch. Since
the checksum is not reset, the hardware starts generating audio
infoframes with invalid checksums. This causes some displays to blank
the video.
Fix this by updating the checksum and channel mask when the device is
closed and also when it is first initialized. In addition, make sure
that the channel mask is appropriate for an 8ch infoframe by setting
it to 0x13 (FL FR LFE FC RL RR RLC RRC).
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Acked-by: Stephen Warren <swarren@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add shutup callback to be called codec-specifically for avoiding pop
noises at suspend or shutdown. As a generic callback, just turn EAPD
off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, alc662_init_verbs[] is used for all ALC662-compatible chips,
but the EAPD controls for 0x15 in there is invalid for ALC892.
Also, since EAPDs should be set up in alc_auto_init_amp(), these static
elements aren't needed for auto-parser, too.
In this patch, the EAPD init verbs are split from alc662_init_verbs,
and applied only to static quirks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
initialize ret to invalid value so that when we reach the config error path in
soc_pcm_open, it will return the correct error code. without this patch, though
config error path is executed, soc_pcm_open will return 0 in
snd_pcm_open_substream and then cause double release of substream.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Suppose we have:
cpu_dai
channels_min = 1
channels_max = 1
codec_dai
channels_min = 2
channels_max = 2
This is a mismatch that should not happen, however according to the current
code, the result of runtime->hw will be:
channels_min = 2
channels_max = 1
We better spot it early. This patch checks this mismatch.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
module_name will be checked in register_sst_card.
It will fail to register sst card if it's not initialized.
Signed-off-by: Lu Guanqun <guanqun.lu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds the MAX98095 CODEC driver.
Signed-off-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC machine drivers that are their own platform_driver (as opposed to
those using the soc-audio platform_driver) need to explicitly set up
power-management operation callbacks.
To avoid cut/paste, snd_soc_pm_ops also needs to be exported.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current alc662 parser doesn't set the DAC for the mixer 0x0f
properly for ALC892, which has 4 DACs while ALC662 has 3.
Fixed by implementing alc662_mix_to_dac() more genericly with the
dynamic widget list.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some ALC272-quirks use alc662_dac_nids instead of alc272_dac_nids.
This patch fixes these entries. No functional change since the first
two elements are identical in both arrays.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SB Live! Platinum CT4760P is just a 4 channels sound card with STAC9721 and
Philips UDA1334 DAC.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
alc662 series only have 3 DAC, so it can only support 5stack-dig
instead of 6stack-dig.
[updated HD-Audio-Models.txt as well by tiwai]
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are many USB MIDI cables out there that have buggy
firmware that reports it can do more than 4 bytes in a
packet when they can only properly handle 4
This patch adds the ID of yet another one of those cables
Signed-off-by: Tarek Soliman <tarek@bashasoliman.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove some unneeded defintions
Use %pR to print resources
Make some data const
Consistent braces for else
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Define and use pcm_debug_name if CONFIG_SND_DEBUG
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow older non DMA capable cards to use MMAP by
emulating the DMA using read and write functions,
and getting rid of copy & silence callbacks that
were used only by older cards.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use the card drained status reporting for playback,
but allow it to persist for a few timer cycles before
signalling XRUN, to allow card to recover by itself.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clock source is neither capture nor playback,
so change 'Capture Clock' to 'Clock'.
Add spaces to control name string for consistency,
always 'PCM 0' , never 'PCM0'
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without the "thinkpad" quirk, the dock mic in
Lenovo X220 tablet edition won't work.
BugLink: http://bugs.launchpad.net/bugs/751033
Cc: stable@kernel.org
Tested-by: James Ferguson <james.ferguson@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is also in the old sysfs diagnostics but it's nice to have everything
in one place.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
