ALSA: hda - fix indices on boost volume on Conexant
After the recent patch "ALSA: hda - use both input paths on Conexant
auto parser" suddenly we can have more than one "Mic Boost", this
happened on Acer Aspire One 722. Therefore we must add the possibility
to put an index on this "Mic Boost" just as we do for the other
"Mic Boost" earlier in the same function.
BugLink: https://bugs.launchpad.net/bugs/1059523 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the loopback timer handler is running, calling del_timer() (for STOP
trigger) will not wait for the handler to complete before deactivating the
timer. The timer gets rescheduled in the handler as usual. Then a subsequent
START trigger will try to start the timer using add_timer() with a timer pending
leading to a kernel panic.
Serialize the calls to add_timer() and del_timer() using a spin lock to avoid
this.
Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@linux.intel.com> Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: hda - Fix hang caused by race during suspend.
There was a race condition when the system suspends while hda_power_work
is running in the work queue. If system suspend (snd_hda_suspend)
happens after the work queue releases power_lock but before it calls
hda_call_codec_suspend, codec_suspend runs with power_on=0, causing the
codec to power up for register reads, and hanging when it calls
cancel_delayed_work_sync from the running work queue.
The call chain from the work queue will look like this:
hda_power_work <<- power_on = 1, unlock, then power_on cleard by suspend
hda_call_codec_suspend
hda_set_power_state
snd_hda_codec_read
codec_exec_verb
snd_hda_power_up
snd_hda_power_save
__snd_hda_power_up
cancel_delayed_work_sync <<-- cancelling executing wq
Fix this by waiting for the work queue to finish before starting suspend
if suspend is not happening on the work queue.
ALSA: hda/realtek - Fix detection of ALC271X codec
In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define()
always after fixup"), alc_auto_parse_customize_define was moved after
detection of ALC271X.
The problem is that detection of ALC271X relies on spec->cdefine.platform_type,
and it's set on alc_auto_parse_customize_define.
Move the alc_auto_parse_customize_define and its required fixup setup
before the block doing the ALC271X and other codec setup.
BugLink: https://bugs.launchpad.net/bugs/1006690 Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com> Reviewed-by: David Henningsson <david.henningsson@canonical.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Sat, 6 Oct 2012 14:33:52 +0000 (16:33 +0200)]
Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Additional updates for v3.7
A couple more updates for 3.7, enhancements to the ux500 and wm2000
drivers, a new driver for DA9055 and the support for regulator bypass
mode. With the exception of the DA9055 this has all had a chance to
soak in -next (the driver was added on Friday so should be in -next
today).
This patch adds support for Dialog semiconductor's DA9055 audio codec.
This has been tested on DA9055 EVB with Samsung SMDK6410 board.
Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel Mack [Thu, 27 Sep 2012 23:36:44 +0000 (01:36 +0200)]
ALSA: ASoC: add DT bindings for CS4271
Apart from pure matching, the bindings also support setting the the
reset gpio line.
Signed-off-by: Daniel Mack <zonque@gmail.com> Cc: Alexander Sverdlin <subaparts@yandex.ru> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Thu, 27 Sep 2012 17:35:24 +0000 (18:35 +0100)]
ASoC: wm_hubs: Ensure volume updates are handled during class W startup
In some circumstances we may need to flush volume updates to the device
after switching to class W mode. Do this unconditionally to ensure that
these situations are handled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
Charles Keepax [Thu, 27 Sep 2012 12:21:48 +0000 (13:21 +0100)]
ASoC: wm5110: Adding missing volume update bits
The volume update bits were being set on all but one input and one output.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
Mark Brown [Fri, 7 Sep 2012 04:57:11 +0000 (12:57 +0800)]
ASoC: dapm: Allow regulators to bypass as well as disable when idle
Allow regulators managed via DAPM to make use of the bypass support that
has recently been added to the regulator API by setting a flag
SND_SOC_DAPM_REGULATOR_BYPASS. When this flag is set the regulator will
be put into bypass mode before being disabled, allowing the regulator to
fall into bypass mode if it can't be disabled due to other users.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to cs4270 datasheet, there is no reference to mono mode.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Acked-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 24 Sep 2012 07:58:04 +0000 (10:58 +0300)]
dmaengine: Add flags parameter to dmaengine_prep_dma_cyclic()
With this parameter added to dmaengine_prep_dma_cyclic() the API will be in
sync with other dmaengine_prep_*() functions.
The dmaengine_prep_dma_cyclic() function primarily used by audio for cyclic
transfer required by ALSA, we use the from audio to ask dma drivers to
suppress interrupts (if DMA_PREP_INTERRUPT is cleared) when it is supported
on the platform.
Merge tag 'asoc-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v3.7
Lots and lots of driver specific cleanups and enhancements but the only
substantial framework feature this time round is the compressed API
binding:
- Addition of ASoC bindings for the compressed API, used by the mid-x86
drivers.
- Lots of cleanups and API refreshes for CODEC drivers and DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:32 +0000 (13:46 +0300)]
ASoC: twl4030: Support for DT booted kernel
When the kernel has been booted with DT blob the platform data is NULL for
the driver.
We need to construct the pdata based on the DT information for runtime use.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:29 +0000 (13:46 +0300)]
ASoC/mfd: twl4030: Remove set_hs_extmute callback from platform data
We no longer have users for the set_hs_extmute callback which has been
replaced by hs_extmute_gpio so the codec driver can handle the external
mute if it is needed by the board.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:28 +0000 (13:46 +0300)]
ARM: OMAP/ASoC: Zoom2: Let the codec to handle the hs_extmute GPIO
Remove the use of set_hs_extmute callback and let the codec driver to
handle the extmute GPIO.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:27 +0000 (13:46 +0300)]
ASoC: twl4030: Move hs_extmute GPIO handling to driver
The external mute (if it is in use) is handled by a GPIO line. Prepare to
remove the set_hs_extmute callback and replace it with:
hs_extmute_gpio: the GPIO number to use for external mute
When the users of set_hs_extmute has been converted the callback can be removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:26 +0000 (13:46 +0300)]
Input: twl4030-vibra: Support for DT booted kernel
Add support when the kernel has been booted with DT blob. In this case the
pdata is NULL, we need to reach up to the core node and check if the codec
part has been enabled to determine if we need to coexist with the codec or
not.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Dmitry Torokhov <dmitry.torokhov@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:25 +0000 (13:46 +0300)]
mfd: twl4030-audio: Add DT support
Support for loading the twl4030 audio module via devicetree.
Sub devices for codec and vibra will be created as mfd devices once the
core MFD driver is loaded when the kernel is booted with a DT blob.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:23 +0000 (13:46 +0300)]
mfd: twl4030-audio: Get audio MCLK via twl-core API instead of pdata
twl-core has API to get the boot time configured HFCLK rate which has the
same rate as the audio MCLK.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:22 +0000 (13:46 +0300)]
mfd: twl-core: Add API to query the HFCLK rate
CFG_BOOT register's HFCLK_FREQ field hold information about the used HFCLK
frequency.
Add possibility for users to get the configured rate based on this
register.
This register was configured during boot, without it the chip would not
operate correctly, so we can trust on this information.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:21 +0000 (13:46 +0300)]
mfd: twl4030-audio: Rearange and clean-up the probe function
To facilitate the device tree support the probe function need to be rearanged.
Small cleanup in the APLL frequency selection part as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:20 +0000 (13:46 +0300)]
mfd: twl4030-audio: Convert to use devm_kzalloc
To clean up the module probe and remove functions.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Mon, 10 Sep 2012 10:46:19 +0000 (13:46 +0300)]
mfd: twl4030-audio: Clean up MODULE_* and platform_driver part
Place the MODULE_* lines in the same block and add MODULE_DESCRIPTION.
Rearange the platform_driver structure at the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:58 +0000 (15:05 +0300)]
ASoC: omap-pcm: Convert to use dmaengine
Original author: Russell King <rmk+kernel@arm.linux.org.uk>
Switch the omap-pcm to use dmaengine.
Certain features are not supported by after dmaengine conversion:
1. No period wakeup mode
DMA engine has no way to communicate this information through
standard channels.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:57 +0000 (15:05 +0300)]
ASoC: OMAP: mcbsp, mcpdm, dmic, hdmi: Set dma_data at startup time
Set the dma_data for the stream (snd_soc_dai_set_dma_data) at dai_startup
time so omap-pcm will have access to the needed information regarding to
the DMA channel earlier.
This is needed for the clean dmaengine support.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:56 +0000 (15:05 +0300)]
ASoC: omap-pcm, omap-dmic: Change the use of omap_pcm_dma_data->data_type
Instead of the OMAP DMA data type definition the data_type will be used to
specify the number of bits the DMA word should be configured or 0 in case
when based on the stream's format the omap-pcm can decide the needed DMA
word size.
This feature is needed for the omap-hdmi where the sDMA need to be
configured for 32bit word type regardless of the audio format used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:55 +0000 (15:05 +0300)]
ASoC: OMAP: mcbsp, mcpdm, dmic: Let omap-pcm to pick the dma_type
omap-pcm can figure out the correct dma_type based on the stream's format.
In this way we can get rid of the plat/dma.h include from these drivers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:54 +0000 (15:05 +0300)]
ASoC: omap-mcpdm: Use platform_get_resource_* to get resources
Get the needed resources in a correct way and avoid using defines for them.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:53 +0000 (15:05 +0300)]
ARM: OMAP4: hwmod_data: Add resource names to McPDM memory ranges
To help the driver to get the correct memory range to access McPDM
registers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:52 +0000 (15:05 +0300)]
ASoC: omap-pcm: Prepare to configure the DMA data_type based on stream properties
Based on the format of the stream the omap-pcm can decide alone what data
type should be used with by the sDMA.
Keep the possibility for OMAP dai drivers to tell omap-pcm if they want to
use different data type. This is needed for the omap-hdmi for example which
needs 32bit data type even if the stream format is S16_LE.
The check if (dma_data->data_type) is safe at the moment since omap-pcm
does not support 8bit samples (OMAP_DMA_DATA_TYPE_S8 == 0x00).
The next step is to redefine the meaning of dma_data->data_type to unblock
this limitation.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:51 +0000 (15:05 +0300)]
ASoC: OMAP: Remove sync_mode from omap_pcm_dma_data struct
The omap-pcm platform driver no longer needs this parameter to select
between ELEMENT and PACKET mode. The selection is based on the configured
packet_size.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:50 +0000 (15:05 +0300)]
ASoC: omap-pcm: Select sDMA synchronization based on packet_size
Since we only have element or packet synchronization we can use the
dma_data->packet_size to select the desired mode:
if packet_size is 0 we use ELEMENT mode
if packet_size is not 0 we use PACKET mode for sDMA synchronization.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:49 +0000 (15:05 +0300)]
ASoC: omap-mcbsp: Use sDMA packet mode instead of frame mode
When McBSP is configured in threshold mode we can use sDMA packet mode in
all cases.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:48 +0000 (15:05 +0300)]
dmaengine: omap-dma: Add support to suppress interrupts in cyclic mode
When requested (DMA_PREP_INTERRUPT is cleared in flags) disable all DMA
interrupts for the channel. In this mode user space does not expect
periodic reports from kernel about the progress of the audio stream.
PulseAudio for example support this type of mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Vinod Koul <vinod.koul@linux.intel.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:47 +0000 (15:05 +0300)]
dmaengine: Pass flags via device_prep_dma_cyclic() callback
Change the parameter list of device_prep_dma_cyclic() so the DMA drivers
can receive the flags coming from clients.
This feature can be used during audio operation to disable all audio
related interrupts when the DMA_PREP_INTERRUPT is cleared from the flags.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com> Acked-by: Shawn Guo <shawn.guo@linaro.org> Acked-by: Vinod Koul <vinod.koul@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:45 +0000 (15:05 +0300)]
dmaengine: omap: Add support for pause/resume in cyclic dma mode
The audio stack used omap_stop_dma/omap_start_dma to pause/resume the DMA.
This method has been used for years on OMAP based products.
We only allow pause/resume when the DMA has been configured in cyclic mode
which is used by the audio stack.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Vinod Koul <vinod.koul@linux.intel.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Fri, 14 Sep 2012 12:05:44 +0000 (15:05 +0300)]
dmaengine: omap: Support for element mode in cyclic DMA
When src_maxburst/dst_maxburst is set to 0 by the users of cyclic DMA
(mostly audio) indicates that we should configure the omap DMA to element
sync mode instead of packet mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Vinod Koul <vinod.koul@linux.intel.com> Tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eric Millbrandt [Thu, 20 Sep 2012 14:36:44 +0000 (10:36 -0400)]
ASoC: fsl: pcm030-audio-fabric use snd_soc_register_card
Convert pcm030-audio-fabric to use the new snd_soc_register_card api
instead of the older method of registering a separate platform device.
Create the dai_link to the mpc5200_psc_ac97 platform using the device tree.
Convert the pcm030-audio-fabric driver to a platform-driver and add a
remove function.
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 20 Sep 2012 13:32:15 +0000 (16:32 +0300)]
ASoC: twl6040: Convert to use DAI DAPM widgets
Use DAPM mapping for stream events and give unique names for the streams.
This change also fixes the following warning:
twl6040-codec twl6040-codec: Failed to create Capture debugfs file
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 20 Sep 2012 13:32:02 +0000 (16:32 +0300)]
ASoC: twl4030: Convert to use DAI DAPM widgets
Use DAPM mapping for stream events and give unique names for the streams.
This change also fixes the following warning:
twl4030-codec twl4030-codec: Failed to create Capture debugfs file
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
DMA Position in Buffer (DPIB) should be used for
ring buffer management, while LPIB register provides
information on the number of samples transfered on
the link. The difference between the two pieces of
information corresponds to hardware/DMA buffering.
This patch reports this difference in runtime->delay, and
removes the use of the COMBO mode on recent Intel hardware.
Credits to Takashi Iwai for an initial patch.
[rebased to for-next branch and replaced snd_printk() with
snd_printdd() by tiwai]
SSYNC bits are typically used to start multiple
streams synchronously. It makes sense to use them
for a single stream for a more predictable startup
sequence. The transfers only start once the DMA and
FIFOs are ready. This results in a better correlation
between timestamps and number of samples played.
Credits to Kar Leong Wang for suggesting this
improvement.
ALSA: hda/via - don't report presence on HPs with no presence support
If headphone jack can't detect plug presence, and we have the jack in
the jack table, snd_hda_jack_detect will return the plug as always
present (as it'll be considered as a phantom jack). The problem is that
when this happens, line out pins will always be disabled, resulting in
no sound if there are no headphones connected.
This was reported as a no sound problem after suspend on
http://bugs.launchpad.net/bugs/1052499, since the bug doesn't manifests
on first initialization before the phantom jack is added, but on resume
we reexecute the initialization code, and via_hp_automute starts
reporting HP always present with the jack now on the table.
Timur Tabi [Thu, 20 Sep 2012 18:57:27 +0000 (13:57 -0500)]
ASoC: wm8960: remove 'dres' field from platform data structure
The 'dres' field (discharge resistance for headphone outputs) is no longer
used in the driver, so remove it.
It was used in the original version of the driver when entering standby
from off, but we stopped using it when we switched from having a single
startup sequence to having separate cap and capless sequences.
Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA: hda - use both input paths on Conexant auto parser
On the Thinkpad W520 - and probably several other machines with
Conexant 506x chips - the Dock Mic and Mic are connected to the
same two selector nodes. This patch will make Dock Mic take one
selector node and Mic take the other, when possible.
Without the patch, both paths would take the first selector,
leading to the normal Mic's volume being controlled by
"Dock Mic Boost".
(On other machines, this could instead fixup similar problems between
Mic and Line In, for example.)
BugLink: https://bugs.launchpad.net/bugs/1037642 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lee Jones [Tue, 21 Aug 2012 09:06:44 +0000 (10:06 +0100)]
ASoC: Ux500: Minor coding layout changes
Includes removal of duplicate debug print affirming entry into
the probe function, an unnecessary line break of a coding line
<80 chars and a white space change (unintentional tab).
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Signed-off-by: Lee Jones <lee.jones@linaro.org>
Lee Jones [Fri, 27 Jul 2012 07:50:05 +0000 (08:50 +0100)]
ASoC: codecs: Enable AB8500 CODEC for Device Tree
We continue to allow the AB8500 CODEC to be registered via the AB8500
Multi Functional Device API, only this time we extract its configuration
from the Device Tree binary.
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org>
Lee Jones [Thu, 26 Jul 2012 16:07:26 +0000 (17:07 +0100)]
ASoC: Ux500: Enable ux500 MSP driver for Device Tree
Register both parts of the MSP driver from Device Tree so that they
are probed when Device Tree is enabled. Also, as there is platform
data involved, we ensure that there is allocated memory to place the
configuration into and that the correct information is extracted from
the DT binary.
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org>
Lee Jones [Thu, 26 Jul 2012 15:48:34 +0000 (16:48 +0100)]
ASoC: Ux500: Enable MOP500 driver for Device Tree
Here we ensure that the MOP500 audio driver will be probed during a
Device Tree boot. We also parse the sound node to link together the
codec, dma and the CPU-side Digital Audio Interface.
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org>
Lee Jones [Fri, 14 Sep 2012 15:16:08 +0000 (16:16 +0100)]
ASoC: Ux500: Move MSP pinctrl setup into the MSP driver
In the initial submission of the MSP driver msp1 and msp3's associated
pinctrl mechanism was passed back to platform code using a plat_init()
call-back routine, but it has no place in platform code. The MSP driver
should set this up for the appropriate ports. Instead we use a use_pinctrl
identifier which is passed from platform_data/Device Tree which indicates
which ports should use pinctrl.
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com> Acked-by: Linus Walleij <linus.walleij@linaro.org> Signed-off-by: Lee Jones <lee.jones@linaro.org>
ALSA: usb-audio: Avoid unnecessary EP setups in prepare
The recent fix for USB suspend breakage moved the code to set up EP
from hw_params to prepare, but it means also the EP setup might be
called multiple times unnecessarily because the prepare callback can
be called multiple times without starting the stream (e.g. OSS
emulation).
This patch adds a new flag to struct snd_usb_substream indicating
whether the setup of EP is required, and do it only when necessary,
i.e. right after hw_params or suspend.
Move interface and endpoint configuration from hw_params to prepare
callback. During system suspend/resume when the USB device power isn't
cycled the interface and endpoint configuration need to be set before
audio playback can continue. Resume involves another call to prepare
but not to hw_params, moving it here allows a playing stream to continue
after resume.
ALSA: usb-audio: Don't require hw_params in endpoint.
Change the interface to configure an endpoint so that it doesn't require
a hw_params struct. This will allow it to be called from prepare
instead of hw_params, configuring it after system resume.
Mark Brown [Thu, 13 Sep 2012 03:46:58 +0000 (11:46 +0800)]
ASoC: wm8960: Support shared LRCLK
If the LRCLK is shared and the WM8960 is clock master then we should
enable the LRCM bit to tell the device that it should drive LRCLK when
either ADC or DAC is enabled rather than separately driving the two
LRCLKs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eric Millbrandt [Thu, 13 Sep 2012 21:43:12 +0000 (17:43 -0400)]
ASoC: fsl: mpc5200 add missing information to snd_soc_dai_driver
Add missing dai_driver information to avoid these runtime errors
[ 16.433788] asoc: error - multiple DAI f0002c00.i2s registered with no name
[ 16.453551] Failed to register DAI
[ 16.461222] mpc5200-psc-i2s: probe of f0002c00.i2s failed with error -22
[ 16.475242] asoc: error - multiple DAI f0002000.ac97 registered with no name
[ 16.488087] mpc5200-psc-ac97 f0002000.ac97: Failed to register DAI
[ 16.502222] mpc5200-psc-ac97: probe of f0002000.ac97 failed with error -22
Signed-off-by: Eric Millbrandt <emillbrandt@dekaresearch.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: Avoid recalculating the bitmask for SOC_ENUM controls
For ENUM controls the bitmask is calculated based on the number of items.
Currently this is done each time the control is accessed. And while the
performance impact of this should be negligible we can easily do better. The
roundup_pow_of_two macro performs the same calculation which is currently done
manually, but it is also possible to use this macro with compile time constants
and so it can be used to initialize static data. So we can use it to initialize
the mask field of a ENUM control during its declaration.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
projects / karo-tx-linux.git / log