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>
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>
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>
* 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>
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>
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>
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>
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>
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>
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>
This patch fixes to avoid compile error when ASoC codec doesn't use I2C
nor SPI on snd_soc_hw_bulk_write_raw().
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'x86-fixes-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/linux-2.6-tip:
x86, UV: Fix kdump reboot
x86, amd-nb: Rename CPU PCI id define for F4
sound: Add delay.h to sound/soc/codecs/sn95031.c
x86, mtrr, pat: Fix one cpu getting out of sync during resume
x86, microcode: Unregister syscore_ops after microcode unloaded
x86: Stop including <linux/delay.h> in two asm header files
wordsize is used as the textual width of a register address.
regsize is used as the textual width of a register value.
The assignments to these values were swapped. In the case of WM8903, which
has 8-bit register addresses and 16-bit register values, this caused the
register values to be clipped to 2 digits instead of the full 4.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch firstly restructurizes the code a bit by getting rid of continuous
checking for machine type in spitz_mic_bias().
Then the patch properly requests the MIC GPIO in the spitz_init() and frees it
in spitz_exit().
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_jack_gpio has a name field. Use that name when registering the IRQ,
since this is far more informative than the codec driver name. This shows
up in /proc/interrupts.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
pxa2xx_pcm_hw_free frees dma channel and sets prtd->dma_ch to -1,
but does not set prtd->params to NULL, so if pxa2xx_pcm_hw_params will
be called immediately, it leaves prtd->dma_ch initialized with -1,
and it results in oops in __pxa2xx_pcm_prepare. This bug is triggered
via SDL.
This patch adds check for prtd->dma_ch to __pxa2xx_pcm_prepare and
cleans prtd->params, so now it works properly.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.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: pcm: fix infinite loop in snd_pcm_update_hw_ptr0()
ALSA: HDA: Add dock mic quirk for Lenovo Thinkpad X220
ALSA: ens1371: fix Creative Ectiva support
ALSA: firewire-speakers: fix hang when unplugging a running device
ASoC: Fix CODEC device name for Corgi
ALSA: hda - Fix pin-config of Gigabyte mobo
ASoC: imx: fix burstsize for DMA
ASoC: imx: set watermarks for mx2-dma
ASoC: twl6040: Return -ENOMEM if create_singlethread_workqueue fails
ASoC: tlv320dac33: Restore L/R DAC power control register
ASoC: Explicitly say registerless widgets have no register
ASoC: tlv320dac33: Fix inconsistent spinlock usage
