Currently everytime we get the following message on boot:
fsl-ssi-dai 202c000.ssi: could not get baud clock: -2
This is not really useful information to get on every boot, so make it a debug
message instead.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since commit aafa85e71a (ASoC: fsl_ssi: Add DAI master mode support for SSI on
i.MX series) master mode is supported, so update the comments and code to
reflect that.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Latest evaluation of the the device has given some patch file additions
for improved performance.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Since the GPIO jacks are only supported if gpiolib is built and fail to
compile otherwise add a build depedency. This is unlikely to have any
practical impact outside of coverage testing.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Symbol MFD_MC13783 always selected by MFD_MC13XXX, so no need
to keep additional symbol.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Lee Jones <lee.jones@linaro.org>
The prefix for the codec driver can be used during dual identical
codec usecases. However, dapm adds prefix twice for codec DAI widget
in snd_soc_dapm_add_route API.
This change is to avoid double prefix addition for codec DAI widget
and is needed while using identical dual codecs.
Signed-off-by: Songhee Baek <sbaek@nvidia.com>
Signed-off-by: Arun Shamanna Lakshmi <aruns@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently compressed audio streams are statically routed from the /dev
to the DAI link. Some DSPs can route compressed data to multiple BE DAIs
like they do for PCM data.
Add support to allow dynamically routed compressed streams using the existing
DPCM infrastructure. This patch adds special FE versions of the compressed ops
that work out the runtime routing.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC compressed code needs to call the internal DPCM APIs in order to
dynamically route compressed data to different DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since the soc generic dmaengine pcm driver allows using the defualt settings,
so the pcm->config maybe NULL.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
A few more updates for v3.14 since the last set, highlights include:
- Lots of DMA updates from Lars-Peter
- Improvements to the constraints handling code from Lars-Peter
- A very helpful conversion of the TWL4030 driver to regmap from Peter
- A new driver for the Freescale ESAI controller from Nicolin Chen
- Conversion of some of the drivers to use params_width()
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Merge tag 'asoc-v3.14-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: More updates for v3.14
A few more updates for v3.14 since the last set, highlights include:
- Lots of DMA updates from Lars-Peter
- Improvements to the constraints handling code from Lars-Peter
- A very helpful conversion of the TWL4030 driver to regmap from Peter
- A new driver for the Freescale ESAI controller from Nicolin Chen
- Conversion of some of the drivers to use params_width()
When we plug a 3-ring headset on some Dell machines, the headset
mic can't be detected, after apply this patch, the headset mic
can work well on all those machines.
On the machine with the Subsytem ID 0x10280610, if we use
ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, the headset mic can be
detected and work well, but the sound can't be outputed via
headphone anymore, use ALC269_FIXUP_DELL3_MIC_NO_PRESENCE
can fix this problem.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Tested-by: David Chen <david.chen@canonical.com>
Tested-by: Cyrus Lien <cyrus.lien@canonical.com>
Tested-by: Shawn Wang <shawn.wang@canonical.com>
Tested-by: Chih-Hsyuan Ho <chih.ho@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
range_min is the lowest address in the virtual register range. This is
the first register with address 0, not the first register of page 1.
Currently all writes to page 1 are mapped to page 0, so the codec fails
to operate.
Fixes: 4d208ca429 (ASoC: tlv320aic32x4: Convert to direct regmap API usage)
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org (v3.13 if the fix misses -final)
Make it easier for generic code to work with set_sysclk() by distinguishing
between the operation not being supported and an error as is done for
other operations like set_dai_fmt()
Signed-off-by: Mark Brown <broonie@linaro.org>
soc_widget_read API returns the register data and it is possible
that a register can contain 0xffffffff. Thus, change the prototype
of soc_widget_read to return only the error code and pass the reg
data through pointer argument.
Signed-off-by: Arun Shamanna Lakshmi <aruns@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The Samsung dmaengine ASoC driver is used with two different dmaengine drivers.
The pl80x, which properly supports residue reporting and the pl330, which
reports that it does not support residue reporting. So there is no need to
manually set the NO_RESIDUE flag. This has the advantage that a proper (race
condition free) PCM pointer() implementation is used when the pl80x driver is
used. Also once the pl330 driver supports residue reporting the ASoC PCM driver
will automatically start using it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The pl330 driver properly reports that it does not have residue reporting
support, which means the PCM dmanegine driver is able to figure this out on its
own. So there is no need to set the flag manually. Removing the flag has the
advantage that once the pl330 driver gains support for residue reporting it will
automatically be used by the generic dmaengine PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The dmaengine framework now exposes the granularity with which it is able to
report the transfer residue for a certain DMA channel. Check the granularity in
the generic dmaengine PCM driver and
a) Set the SNDRV_PCM_INFO_BATCH if the granularity is per period or worse.
b) Fallback to the (race condition prone) period counting if the driver does
not support any residue reporting.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently we have two different snd_soc_platform_driver structs in the generic
dmaengine PCM driver. One for dmaengine drivers that support residue reporting
and one for those which do not. When registering the PCM component we check
whether the NO_RESIDUE flag is set or not and use the corresponding
snd_soc_platform_driver. This patch modifies the driver to only have one
snd_soc_platform_driver struct where the pointer() callback checks the
NO_RESIDUE flag at runtime. This allows us to set the NO_RESIDUE flag after the
PCM component has been registered. This becomes necessary when querying whether
the dmaengine driver supports residue reporting from the dmaengine driver itself
since the DMA channel might only be requested after the PCM component has been
registered.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
It's a bug that writing to the platform data directly, for it should
be constant. So just copy it before writing.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of open-coding the intersecting of two rate masks (and getting slightly
wrong for some of the corner cases) use the new snd_pcm_rate_mask_intersect()
helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
A bit of special care is necessary when creating the intersection of two rate
masks. This comes from the special meaning of the SNDRV_PCM_RATE_CONTINUOUS and
SNDRV_PCM_RATE_KNOT bits, which needs special handling when intersecting two
rate masks. SNDRV_PCM_RATE_CONTINUOUS means the hardware supports all rates in a
specific interval. SNDRV_PCM_RATE_KNOT means the hardware supports a set of
discrete rates specified by a list constraint. For all other cases the supported
rates are specified directly in the rate mask.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
SNDRV_PCM_RATE_CONTINUOUS means that all rates (possibly limited to a certain
interval) are supported. There is no need to manually set other rate bits.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Daniel Glöckner <daniel-gl@gmx.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
SNDRV_PCM_RATE_CONTINUOUS means that all rates (possibly limited to a certain
interval) are supported. There is no need to manually set other rate bits.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll
end up with the rate_max field of the runtime hardware set to 0. (Note that it
is still possible for the components to constrain the supported sample rates
using other methods, e.g. setting a list constraint) If rate_max is 0 this means
that the sound card doesn't support any rates at all, which is not the desired
result. So initialize rate_max to UINT_MAX. For symmetry reasons also set
rate_min to 0.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are three files in oss for which I could not find an easy way to
replace interruptible_sleep_on_timeout with a non-racy version. This
patch instead just adds a private implementation of the function, now
named oss_broken_sleep_on, and changes over the remaining users in
sound/oss/ so we can remove the global interface.
[fixed coding style warnings by tiwai]
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The use of interruptible_sleep_on_timeout in the dmasound driver
is questionable and we want to kill off all sleep_on variants.
This replaces the calls with wait_event_interruptible_timeout
where possible, to wait for a particular event instead of blocking
in a racy way. In the sq_write function, the easiest solution is
an open-coded prepare_to_wait loop.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sleep_on is known to be racy and going away because of this. All instances
of interruptible_sleep_on and interruptible_sleep_on_timeout in the midibuf
driver can trivially be replaced with wait_event_interruptible and
wait_event_interruptible_timeout.
[fixed coding style warnings by tiwai]
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Interruptible_sleep_on is racy and we want to remove it. This replaces
the use in the vwsnd driver with an open-coded prepare_to_wait
loop that fixes the race between concurrent open() and close() calls,
and also drops the global mutex while waiting here, which restores
the original behavior that was changed during the BKL removal.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to remove all sleep_on variants from the kernel because they are
racy. In case of the pinnacle driver, we can replace
interruptible_sleep_on_timeout with wait_event_interruptible_timeout
by changing the meaning of a few flags used in the driver so they
are cleared at wakeup time, which is a somewhat more appropriate
way to do the same, although probably still racy.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new vmaster hook, update_tpacpi_mute_led(), calls the original
vmaster hook, but I forgot to save the original hook function but keep
calling the updated one, which of course results in a stupid endless
loop. Fixed now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No code change, just a cosmetic cleanup to keep entries ordered by the
device ID within a block of unique vendor IDs.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Creative Live! Cam Vista IM (VF0420) reports rate of 16kHz while working
at 8kHz. The patch adds its USB ID to the existing quirk.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some AIO (All In One) models with the codec alc668
(Vendor ID: 0x10ec0668) on it, when we plug a headphone into the jack,
the system will switch the output to headphone and set the speaker to
automute as well as change the speaker Pin-ctls from 0x40 to 0x00,
this will bring loud noise to the headphone.
I tried to disable the corresponding EAPD, but it did not help to
eliminate the noise.
According to Takashi's suggestion, we use amp operation to replace the
pinctl modification for the automute, this really eliminate the noise.
BugLink: https://bugs.launchpad.net/bugs/1268468
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove asoc_simple_get_card_info macro and use snd_soc_card_set_drvdata
and snd_soc_card_get_drvdata instead.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Apply the codec->power_filter to the FG nodes in general for reducing
hackish set_power_state ops override in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This codec driver fails to probe because it has a higher regmap
range_max value than max_register. This patch sets the range_max to the
max_register value as described in the for struct regmap_range_cfg:
"@range_max: Address of the highest register in virtual range."
Fixes: 4d208ca429 (ASoC: tlv320aic32x4: Convert to direct regmap API usage)
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org (v3.13 if the fix misses -final)
Some old AD codecs don't like the independent HP handling, either it
contains a single DAC (AD1981) or it mandates the mixer routing
(AD1986A). This patch removes the indep_hp flag for such codecs.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68081
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCI devices with DMA masks smaller than 32bit should enable
CONFIG_ZONE_DMA. Since the recent change of page allocator, page
allocations via dma_alloc_coherent() with the limited DMA mask bits
may fail more frequently, ended up with no available buffers, when
CONFIG_ZONE_DMA isn't enabled. With CONFIG_ZONE_DMA, the system has
much more chance to obtain such pages.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=68221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The failures of buffer preallocations at driver initializations aren't
critical but it's still helpful to inform, so that user can know that
something doesn't work as expected.
For example, the recent page allocator change triggered regressions,
but developers didn't notice until recently because the driver didn't
complain.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch implements a device-tree-only CPU DAI driver for Freescale ESAI
controller that supports:
- 12 channels playback and 8 channels record.
[ Some of the inner transmitters and receivers are sharing same group of
pins. So the maxmium 12 output or 8 input channels are only valid if
there is no pin conflict occurring to it. ]
- Independent (asynchronous mode) or shared (synchronous mode) transmit and
receive sections with separate or shared internal/external clocks and frame
syncs, operating in Master or Slave mode.
[ Current ALSA seems not to allow CPU DAI drivers to configure DAI format
separately for PLAYBACK and CAPTURE. So this first version only supports
the case that uses the same DAI format for both directions. ]
- Various DAI formats: I2S, Left-Justified, Right-Justified, DSP-A and DSP-B.
- Programmable word length (8, 16, 20 or 24bits)
- Flexible selection between system clock or external oscillator as input
clock source, programmable internal clock divider and frame sync generation.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add controls to enable/disable the headphone short circuit protection of
the headphone outputs.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Occasionally, on playback stream ringbuffer wraparound, the EMU20K1
hardware will momentarily return 0 instead of the proper current(loop)
address. This patch handles that case, fixing the problem of playback
position corruption and subsequent loss of buffered sound data, that
occurs with some common buffering layout patterns(e.g. multiple
simultaneous output streams with differently-sized or
non-power-of-2-sized buffers).
An alternate means of fixing the problem would be to read the ca
register continuously, until two sequential reads return the same
value; however, that would be a more invasive change, has performance
implications, and isn't necessary unless there are also issues with the
value not being updated atomically in regards to individual bits or
something similar(which I have not encountered through light testing).
I have no EMU20K2 hardware to confirm if the issue is present there,
but even if it's not, this change shouldn't break anything that's not
already broken.
Signed-off-by: Sarah Bessmer <aotos@fastmail.fm>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the dt is used and the cinfo is NULL, the -ENOMEM should be return.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
If the dt is not used, the cinfo will be always NULL before using it.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For those platforms using DAI master mode like I2S, it's better to pre-set
a default slot number so that there's no need for these common cases to set
the slot number from its machine driver any more.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add the missing clk_disable_unprepare() before return from
fsl_ssi_probe() in the request irq error handling case.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some DMA cores might add additional restrictions on which in memory audio
formats can be supported by the compound sound card. If the PCM component
specifies a set of formats it supports (by setting the formats field to non 0)
take these into account when calculating the format set for the compound sound
card.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
single_open requires a function with signature
'int (*)(struct seq_file *, void *)'. This patch fixes the warning by
fixing the wrong return type of fsl_ssi_stats_show.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of keeping a separate snd-page-alloc module, merge into the
core snd-pcm module, as we don't need to keep it as an individual
module due to the drop of page reservation.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Nowadays we have CMA for obtaining the contiguous memory pages
efficiently. Let's kill the old kludge for reserving the memory pages
for large buffers. It was rarely useful (only for preserving pages
among module reloading or a little help by an early boot scripting),
used only by a couple of drivers, and yet it gives too much ugliness
than its benefit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Broadwell and Haswell have the same behavior on display audio. So this patch
defines is_haswell_plus() to include codecs for both Haswell and its successor
Broadwell, and apply all Haswell fix-ups to Broadwell.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds codec ID (0x80862808) and module alias for Broadwell
display codec.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds the device ID for Intel Broadwell display HD-Audio controller,
and applies Haswell properties to this device.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_SOC_DMAENGINE_PCM was renamed to CONFIG_SND_DMAENGINE_PCM
recently. And yet we don't have to select it since
CONFIG_SND_GENERIC_DMAENGINE_PCM selects the dependency by itself, so
just rip it off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Florian Meier <florian.meier@koalo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move the start of booting the DSP to earlier in the DAPM process, and
move the final starting of the DSP to later in the DAPM process. This
allows us to overlap some of the processing with other components of the
system being brought up.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move the ADSP2 boot proceedure into a work structure in preparation for
running it asynchronously with the reset of the audio path bring up.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The normal trigger function can now be used for AC97. Drop AC97 trigger
function.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch defines the appropriate register values for different
oparation modes and IP versions. We have to handle DMA/FIQ, AC97,
DEBUG-IRQs and offline/online configuration support.
With this patch we cleanup some driver code that was not reference
manual conform and try to cleanup the whole trigger function to seperate
the actual register values from the enable/disable logic, which is now
hidden in fsl_ssi_config helpers.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a struct 'fsl_ssi_rxtx_reg_val' which holds register
values necessary to enable rx/tx. Based on those preset register values,
the added configuration functions will cleanly enable/disable different
parts of the SSI IP while supporting online/offline configuration.
Different operating modes can be setup directly as different register
values in fsl_ssi_reg_val.
These functions and structs will help to cleanup and simplify the
trigger function to support many different IP versions (online/offline
configuration) and different operating modes.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fabio Estevam