This means we don't need to walk through every single widget in the system
for each stream event which is a bit less silly.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In order to allow us to do smarter things with DAI links create DAPM
widgets which directly represent the DAIs in the DAPM graph. These are
automatically created from the DAIs as we probe the card with references
held in both directions between the widget and the DAI.
The widgets are not made available for direct instantiation by drivers,
they are created automatically from the DAIs. Drivers should be updated
to create stream routes using DAPM maps rather than by annotating AIF
and DAC widgets with streams.
In order to ease transition to this model from existing drivers we
automatically create DAPM routes between the DAI widgets and the existing
stream widgets which are started and stopped by the DAI widgets, though
the old stream handling mechanism is still in place. This also has the
nice effect of removing non-DAPM devices as any device with a DAI
acquires a widget automatically which will allow future simplifications
to the core DAPM logic.
The intention is that in future the AIF and DAI widgets will gain the
ability to interact such that we are able to manage activity on
individual channels independantly rather than powering up and down the
entire AIF as we do currently.
Currently we only generate these for CODECs, mostly as I have no systems
with non-CODEC DAPM to integrate with. It should be a simple matter of
programming to add the additional hookup for these.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Neater and avoids warnings when used in other places where const strings
are desired.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
In order to allow us to do something smarter than iterate through widgets
doing strcmp() to work out what to power up for stream events change the
interface used to generate them to be based on the combination of a DAI
and a stream direction rather than just a simple string identifying the
stream.
At some point we'll probably want a set of channels too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Let the caller fiddle with the widget after we're done in order to
facilitate further refactoring.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Everything now uses snd_soc_dapm_new_controls() instead so we don't need
to make it part of the external API.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
This fixes below build warning:
CC sound/soc/pxa/spitz.o
sound/soc/pxa/spitz.c: In function 'spitz_startup':
sound/soc/pxa/spitz.c:116: warning: passing argument 1 of 'spitz_ext_control' from incompatible pointer type
sound/soc/pxa/spitz.c:47: note: expected 'struct snd_soc_card *' but argument is of type 'struct snd_soc_codec *'
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This allows userspace control of final power off, allowing policy decisions
for register configuration retention.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Only enable and disable the FLL when explicitly told to, supporting some
additional use cases and making the driver behaviour more standard.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Make sure we update for any changes in cases where we reconfigure while
live (eg, for analogue bypass).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Machine kcontrols now use card instead of codec for thier "chip".
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Show the id we read when the id mismatch is detected.
This is useful for debugging.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow platform widgets to be visible in debugfs like codec widgets.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the addition of the non-CODEC control adds card controls like the
DAPM pin switch have been broken as they are expecting the private data
for the control to be the CODEC but it's now the card. Fix that for the
pin switches, an audit of other drivers is required.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Update the codec debugfs initialisation to use dev_warn() instead of
printk(KERN_WARNING).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently when DAPM widgets are power sequenced the stream_event()
completion callback is only called for the stream_event originator
DAPM context. Other components in the card may also be interested so
make sure they are also notified of any widget power events.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's useful to export the DAPM reset as a static function for future use
by other DAPM functions. e.g. The dynamic PCM query widgets resets the DAPM
graph before working out active paths.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The main role of the WM9081 is as a class D speaker amplifier so there
is no concern about pops. There are also very few registers and a fast
power up time so we can happily mark the driver as idle_bias_off.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
BugLink: https://bugs.launchpad.net/bugs/930842
The reporter states that audio is inaudible by default without muting
'External Amplifier'. Add a quirk to handle his SSID so that changing
the control is not necessary.
Reported-and-tested-by: Benjamin Carlson <elderbubba0810@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
driver.
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Merge tag 'asoc-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
A simple fix from Morimoto-san for the pointer() operation in the FSI
driver.
ALC5632 codec supports digital microphone. This patch adds DAPM widget.
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Conflicts:
sound/pci/hda/patch_realtek.c
Merged back the fix for Acer Aspire 6935 with ALC889 codec.
The fix commit was based on 3.2 kernel so that it can be applied to
stable kernel cleanly.
Since 3.2 kernel, the driver starts trying to assign the multi-io DACs
before the speaker, thus it assigns DAC2/3 for multi-io and DAC4 for
the speaker for a standard laptop setup like a HP, a speaker, a mic-in
and a line-in. However, on Acer Aspire 6935, it seems that the
speaker pin 0x14 must be connected with either DAC1 or 2; otherwise it
results in silence by some reason, although the codec itself allows
the routing to DAC3/4.
As a workaround, the connection list of each pin is reduced to be
mapped to either only DAC1/2 or DAC3/4, so that the compatible
assignment as in kernel 3.1 is achieved.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42740
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VT1705 codec has two ADCs where the secondary ADC has no MUX but only
a fixed connection to the mic pin. This confused the driver and it
tries always overriding the input-source selection by assumption of
the existing MUX for the secondary ADC, resulted in resetting the
input-source at each time PM (including power-saving) occurs.
The fix is simply to check the existence of MUX for secondary ADCs in
the initialization code.
Tested-by: Anisse Astier <anisse@astier.eu>
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Raise the ramp time to 50ms to cover corner cases, use the startup bias
generator, explicitly reset the ramp circuit when complete and reorder
things all of which should improve performance somewhat for systems that
are sensitive to noise from VMID.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The enable of the single ended line outputs on wm_hubs devices performs
better if the output is enabled prior to starting VMID. Since inactive
outputs are held at VMID anyway there is little cost to doing this for
unused outputs. Add callbacks into wm_hubs and keep track of which outputs
are really active so we can disable them once we're active.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the mixer path diagram input sources' attenuators logically
belong to the playback path and DMIC boost only affects capture.
Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This brings the TLVs in sync with the documentation and allows to properly
manipulate mic boost controls with alsamixer.
Signed-off-by: Paul Fertser <fercerpav@gmail.com>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch should fix output through speakers using HP mixer.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix I2S digital interface power for recording.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add DMIC switches and controls to ALC5632 codec.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is no Phone Mix<->Mono Mix route in datasheet.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Rename capture switches to common scheme XXX2REC.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove extra zero from volume step in DECLARE_TLV_DB_SCALE macro.
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The various devm_ functions allocate memory that is released when a driver
detaches. This patch uses these functions for data that is allocated in
the probe function of a platform device and is only freed in the remove
function.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Dong Aisheng <dong.aisheng@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix below build error which is introduced by
commit 022658 "ASoC: core: Add support for DAI and machine kcontrols".
CC [M] sound/soc/omap/n810.o
sound/soc/omap/n810.c: In function 'n810_set_input':
sound/soc/omap/n810.c:194: error: 'codec' undeclared (first use in this function)
sound/soc/omap/n810.c:194: error: (Each undeclared identifier is reported only once
sound/soc/omap/n810.c:194: error: for each function it appears in.)
sound/soc/omap/n810.c:188: warning: unused variable 'card'
make[3]: *** [sound/soc/omap/n810.o] Error 1
make[2]: *** [sound/soc/omap] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Correct spelling "memroy" to "memory" in
sound/soc/codecs/twl4030.c
Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
.. the number of the half-beast?
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Merge tag 'v3.3-rc3' as we've got several bugfixes in there which are
colliding annoyingly with development.
Linux 3.3-rc3
.. the number of the half-beast?
Conflicts:
sound/soc/codecs/wm5100.c
sound/soc/codecs/wm8994.c
current fsi_pointer() calculation was not correct for FSI driver.
This patch fix it up.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It's set unconditionally.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
This is usually not a use case dependant flag anyway.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Since we've already got logic to special case immediate teardown of the
stream we may as well use it if the pmdown_time has been set to zero by
the application layer instead of scheduling a work item with zero delay.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
The new HP laptops turns off the mute LED with VREF50 or VREF80, but
not in HIZ unlike the previous models. Since VREF50 (also 80) works
with the previous models, let's use VREF50 for all.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new ASoC dynamic PCM core needs to create PCMs and substreams that are
for use by internal ASoC drivers only and not visible to userspace for
direct IO. These new PCMs are similar to regular PCMs expect they have no
device nodes or procfs entries. The ASoC component drivers use them in exactly
the same way as regular PCMs for PCM and DAI operations.
The intention is that a dynamic PCM based driver will register both regular
PCMs and internal PCMs. The regular PCMs will be used for all IO with userspace
however the internal PCMs will be used by the driver to route digital audio
through numerous back end DAI links (with potentially a DSP providing different
hw_params, DAI formats based on the regular front end PCM params) to devices
like CODECs, MODEMs, Bluetooth, FM, DMICs, etc
This patch adds a new snd_pcm_new_internal() API call to create the internal PCM
without device nodes or procfs. It also adds adds a new internal flag to snd_pcm.
[fixed minor coding-style issues by tiwai]
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
that were sent. They collide with some -next work so I'd really like to
get them into 3.3-rc3 if possible to merge back up into the -next code.
All driver specific and unexciting in the grand scheme of things.
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Merge tag 'asoc-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
A few small WM8994 updates to go on top of the previous lot of things
that were sent. They collide with some -next work so I'd really like to
get them into 3.3-rc3 if possible to merge back up into the -next code.
All driver specific and unexciting in the grand scheme of things.
The VMID ramp rate is supposed to be 0x3, not 11b. Fix that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Make sure we generate enough BCLKs for I2S style modes by always having
a minimum of two channels worth of clocks for the BCLK.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The driver accidentally exchanged the left/right fields for stereo AC'97
mixer registers. This affected only the aux and CD inputs because the
line input bypasses the AC'97 codec and the mic input is mono; cards
without AC'97 (Xonar DS/DG/HDAV Slim, HG2PCI, HiFier) were not affected.
Reported-and-tested-by: Abby Cedar <abbycedar@yahoo.com.au>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.31+ <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
No practical impact but now we have the control type we may as well use
it for the slightly nicer sequencing.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This hardware requires same fixup for the node 0x0f like Asus A6Rp.
More information: https://bugzilla.redhat.com/show_bug.cgi?id=785417
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we're not implementing GPIO or microphone detection support
there's no users for the read() function any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
In patch_ca0132.c, the error returned from chipio_write() isn't checked
always. Also, the power-up/down sequence isn't tracked properly in some
error paths.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
the Android suspend ignore code for idle_bias_off CODECs. That one is
actually a regression fix as some of the new power savings that have
been introduced confused the suspend ignore code, making devices that
are active for non-audio reasons look like they are idle causing them to
be suspended instead of being kept active.
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Merge tag 'asoc-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
The only particularly remarkable change here is the one for handling of
the Android suspend ignore code for idle_bias_off CODECs. That one is
actually a regression fix as some of the new power savings that have
been introduced confused the suspend ignore code, making devices that
are active for non-audio reasons look like they are idle causing them to
be suspended instead of being kept active.
Make the mechanism used for WM8994 more like that for WM1811 and WM8958:
provide the logic to distinguish between headphone and headset and hard
code the reporting of sensible SND_JACK values. Should integration with
other detection mechanisms be required we can add appropriate callbacks
(though some integrations should be able to use the subsystem ones).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Otherwise we might get an error if the GPIO is configured as an input
since that makes the register volatile and a suspended device can't be
read from.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the default state that the runtime PM infrastructure expects so
instead just kick the runtime PM core to suspend us if we're not doing
anything (as is default).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix build breakage by using the correct API call to add kcontrols.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Looks like nobody is or will be using this code.
Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix some spelling mistakes in the header and remove the todo items. Most
todo items are now available as kcontrol options now anyway.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Allow for the operation of custom mixer and mux DAPM widgets that can call
snd_soc_dapm_mixer_update_power() and snd_soc_dapm_mux_update_power() directly
after updating their status. This is useful with complex DAPM Mixer operations
where we need to do additional work in addition to setting a few mixer register
bits.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In cases where we should enter STANDBY DAPM will power us up, otherwise
there is no need to power up and we can remain in OFF.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8994 has a different accessory detect architecture, call its setup
function too. We ignore the errors and the driver will check for chip type
so it's safe to call the setup functions for both architectures.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We can enable VMID independently of the bias in some use cases so we need
to ensure that the core device is powered up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
sparse complains that "spec->multiout.dac_nids" is a pointer.
sound/pci/hda/patch_realtek.c:2321:37: error: incompatible types for operation (>)
sound/pci/hda/patch_realtek.c:2321:37: left side has type unsigned short const [usertype] *dac_nids
sound/pci/hda/patch_realtek.c:2321:37: right side has type int
It was meant to be num_dacs instead of dac_nids.
Although the current code still works as expected (when num_dacs is zero,
dac_nids should be NULL, too), better to fix now, of course.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The header 'linux/moduleparam.h' is included twice in
'sound/isa/sb/emu8000_patch.c'. Once is enough.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently ASoC can only add kcontrols using codec and platform component device
handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for
SoC card machine drivers too. This allows the kcontrol to have a direct handle to
the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily
get it's private data.
This change makes snd_soc_add_controls() static and wraps it in the folowing
calls (card and dai are new) :-
snd_soc_add_card_controls()
snd_soc_add_codec_controls()
snd_soc_add_dai_controls()
snd_soc_add_platform_controls()
This patch also does a lot of small mechanical changes in individual codec drivers
to replace snd_soc_add_controls() with snd_soc_add_codec_controls().
It also updates the McBSP DAI driver to use snd_soc_add_dai_controls().
Finally, it updates the existing machine drivers that register controls to either :-
1) Use snd_soc_add_card_controls() where no direct codec control is required.
2) Use snd_soc_add_codec_controls() where there is direct codec control.
In the case of 1) above we also update the machine drivers to get the correct
component data pointers from the kcontrol (rather than getting the machine pointer
via the codec pointer).
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The recent changes in Realtek auto-parser added the new "Bass Speaker"
and "CLFE" mixer elements which should be tracked as vmaster slaves,
too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42720
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current FSI got each PortA/B parameter by porta_flags/portb_flags from platform.
And .set_rate function was shared for PortA/B.
This structure was not readable and not flexible.
This patch adds sh_fsi_port_info, and its own settings was added on each platform.
it is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
is_play should be kept as local valuable.
it prepare cleanup for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds struct fsi_stream_handler and defined fsi_pio_push/pop_handler.
these are controled by fsi_steam_xxx() function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fsi_get_current_fifo_samples() uses fsi_stream instead of is_play.
This is just prepare cleanup for DMAEngine support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch re-define fsi_is_play() and fsi_stream_is_play().
fsi_data_pop/push() function keeps direct value of "is_play" at this point,
but it will be removed soon.
This is just prepare cleanup for DMAEngine support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch modify fsi_pio_get_area() parameter to use
struct fsi_stream, and used it on fsi_fifo_data_ctrl().
This is just prepare cleanup for DMAEngine support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
fsi_fifo_init() is called only from fsi_hw_startup()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is preparation for DMAEngine support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Next transfer data size of "push" and "pop" had calculated on shared function.
But it was not readable code.
This patch divided it into for push, and for pop.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch gathered fsi_stream_xxx() functions in order to make it readable.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All DAPM routes are configured via device tree, and there is no need in
DAPM route structures in board file.
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is adding device tree support of headset autodetection on PAZ00 board.
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the standard logging macros and use dev_ variants where we can, also
reporting error codes whenever we report an error. These changes (the
error codes in particular) make it noticeably easier to figure out what
went wrong just from the basic dmesg output.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Since the dynamic pin power-control and the analog low-current mode
may lead to pop-noise, it's safer to set it off as default.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
VIA codecs have several different power-saving features, and one of
them is the analog low-current mode. But it turned out that the ALC
mode causes pop-noises at each on/off time on some machines. As a
quick workaround, disable the ALC when another power-saving feature,
the dynamic pin power-control, is turned off, too, since the dynamic
power-control is already exposed as a mixer enum element so that user
can turn it on/off freely.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The analog low-current mode must be enabled when the no stream is
running but the current detection checks it in a wrong way.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=741128
Cc: <stable@kernel.org> [v3.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of always writing AC_VERB_SET_POWER_STATE, check the current
power-state and don't write again if the value is already set.
This may reduce the click noise upon the dynamic power-state change
(e.g. in analog-input mixer).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The second line output mixer has the controls for the line input bypasses
in the opposite order.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Use the dai_link's dai_fmt attribute to setup the DAI format instead of doing
this manually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This typo caused the wrong codec's nid to be checked for wcaps type.
As a result, sometimes speakers would duplicate the output sent to
HDMI output.
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/924320
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For master mode, set Output [X|A|V]SP_SCLK Sourcing Mode to MCLK Mode.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If we get an indeterminate impedance with both headset polarities then
give up and report the accessory as a headphone rather than continually
retrying.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
.. several days delayed. No reason, I just didn't think of it.
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Merge tag 'v3.3-rc2' into for-3.4
A reasonable amount of new development is causing fiddly merge conflicts
between different resource management changes (mostly fixing bugs in
resource management due to noticing things while doing enhancements in
the same area).
Linux 3.3-rc2
.. several days delayed. No reason, I just didn't think of it.
If cs_automic is called twice (like it is during init) while the mic
is present, it will over-write the last_input with the new one,
causing it to switch back to the automic input when the mic is
unplugged. This leaves the driver in a state (cur_input, last_input,
and automix_idx the same) where the internal mic can not be selected
until it is rebooted without the mic attached.
Check that the mic hasn't already been switched to before setting
last_input.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements device tree support for Tegra boards with ALC5632
codec.
Signed-off-by: Leon Romanovsky <leon@leon.nu>
Acked-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For later wm_hubs devices we have much less need to keep the biases up
even when using single ended line outputs so flag idle_bias_off for
everything except the WM8993 and WM8994.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the same pm_ops for all system suspend and resume paths. This isn't
ideal for suspend to disk with older CODECs as we'll suspend and then
resume the CODEC before powering off all of which takes a long time due
to VMID ramps but it's very simple to implement and for modern CODECs the
overhead should be minimal.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
If an idle_bias_off device is in any state other than off then it is still
active for some reason (typically a low power function such as accessory
detection). This wasn't an issue when the feature was implemented as we
always went to _ON for any active function, subsequent power improvements
have changed things.
With the modern way of doing things we should overhaul the infrastructure
to allow devices to explicitly take references for these functions but
that's a much more invasive change and will require driver updates to
deploy, this will bring the framework into line with the existing driver
set before we do that work.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Now we've switched over to regmap the ASoC level cache sync will be
ineffectual and potentially harmful as there is no longer an ASoC level
cache.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a codec has both a front and a rear Line In, two controls both
named "Line Jack" will be created, which causes parsing to fail.
While a long term solution might be to name the jacks differently,
this extra check is consistent with what is already being done in many
auto-parsers, and will also protect against other cases when two
inputs have the same label.
BugLink: https://bugs.launchpad.net/bugs/923409
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IN1L/R is routed to both line output mixers, we don't route IN1 to LINEOUT1
and IN2 to LINEOUT2.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
The runtime PM core no longer requires any callbacks so don't provide
empty ones for it any more.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The Mute Switch should be called Playback Switch to match the volume
and the regmap core has been updated so we don't need to mark all the
registers as non-volatile.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Wolfram Sang <w.sang@pengutronix.de>
Add a driver supporting the volume control and the mute pin. Shdn pin
and DAPM are not taken care of yet.
Signed-off-by: Wolfram Sang <w.sang@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
It turned out that other ASUS laptops require the similar fix to
enable the VREF on the pin 0x0f for the secret output amp, not only
ASUS A6Rp. Moreover, it's required even when the pin is being used
as the output. Thus, writing a fixed value doesn't work always.
This patch applies the VREF-fix for all ASUS laptops with ALC861/660
in a fixup function that checks the current value and turns on only
the VREF value no matter whether input or output direction is set.
The automute function is modified as well to keep the pin VREF upon
muting/unmuting via pin-control; otherwise the pin VREF is reset at
plugging/unplugging a jack.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=42588
Cc: <stable@kernel.org> [v3.2+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Openmoko GTA01 machine has been removed from the machine ID database,
so we need to remove references to it as well.
Without that fix we have:
sound/soc/samsung/neo1973_wm8753.c: In function ‘neo1973_wm8753_init’:
sound/soc/samsung/neo1973_wm8753.c:325:2: error: implicit declaration of function ‘machine_is_neo1973_gta01’
Signed-off-by: Denis 'GNUtoo' Carikli <GNUtoo@no-log.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
