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>
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>
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
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 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>
