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: 4d208ca429ad (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>
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>
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>
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: 4d208ca429ad (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)
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>
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 driver defines ADAU1701_SEROCTL_WORD_LEN_16 as 0x10 while it should be b10,
so 0x2. This patch fixes it.
Reported-by: Magnus Reftel <magnus.reftel@lockless.no>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Connect the DAPM graph through each BE DAI link to the componnent(s) on the
other side of the BE DAI link. This allows the graph to be walked on
both sides of the link when graph changes are made.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Provide a quick way to tell if a DAI is a dummy DAI or a regular DAI.
This is for internal DAPM usage only and is used to determine whether to
insert a DAI link connection into the DAPM graph.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some BE DAIs can be "dummy" (when the DSP is controlling the DAI) and as such
wont have set a minimum number of playback or capture channels required for BE
DAI registration (to establish supported stream directions).
Force machine drivers to explicitly set whether they support playback and capture
stream directions for every BE DAIs.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the platform driver has no ops, the platform function
bespoke_trigger() is no more called.
The problem was introduced by the commit c5914b0aaea6494aaa9e415cbd32f8b7eb604af0
"ASoC: pcm: Check for ops before deferencing them"
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow PCMs that do not impose any restrictions on the supported formats to set
the formats field to 0, Instead of assuming that this means that the PCM does
not support any formats (which doesn't make much sense), assume that it supports
all formats. This brings the behavior of DPCM closer to that of non-DPCM.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have the same code for initializing the runtime pcm on both the playback and
the capture path. Factor this out into a common helper function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
ffs() returns the bit position from 1, while the ssm2158 driver code
assumes it being 0-based. Also, the bit mask computation of the two
channel slots are incorrect; it must have worked just casually.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Try to get the device's module clock if the dt has no clocks and
system-clock-frequency properties.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is a quick fix for the below two issues when building spdif as modules.
1) If modprobing modules in order: (Step 1) snd-soc-fsl-spdif -> (Step 2)
snd-soc-imx-spdif -> (Step 3) snd-soc-spdif-tx/rx, we will fail to create
imx-spdif card and dai link unless we rmmod snd-soc-imx-spdif and modprobe
it again due to the execution platform_driver_unregister() in probe() when
meeting -EPROBE_DEFER at Step 2.
2) After "imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok",
'rmmod snd-soc-imx-spdif' would cause kernel dump with warning:
WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90()
sysfs: can not remove 'dapm_widget', no directory
This should be caused by disordered resourse releasing of the whole link.
And trying to unregister the card and then CODEC dev can't fix this issue.
Thus this patch just provides a simple fix to these two bugs by using the
snd-soc-dummy in the core instead of seperate snd-soc-spdif-tx/rx so that
there's no need to handle the registering and unregistering of CODEC or
CODEC dai any more.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
From "ASoC: make snd_soc_dai_link more symmetrical", can we see that
the name of CPU DAI maybe omitted. If the DAI name is omitted, try to
use the component name instead.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds SRC support to Renesas sound driver.
SRC converts sampling rate between codec <-> cpu.
It needs special codec chip,
or very simple DA/AD converter to use it.
This patch was tested via ak4554 codec,
and supports Gen1 only at this point.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.
This patch cleanup it with focusing about SRC and Gen2 part.
ssi clock which is calculated from rsnd_ssi_master_clk_start()
should have flexibility since Renesas sound has
SRC (= Sampling Rate Converter).
But current implementation is using runtime->rate directly.
This patch tidyup rsnd_ssi_master_clk_start() parameter
as preparation of future SRC support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.
This patch cleanup it with focusing about SRC and Gen2 part.
rsnd_scu_set_hpbif() is renamed to rsnd_scu_rate_ctrl(),
since its naming doesn't indicate the function meaning.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
INT_ENABLE is needed only Gen2.
rsnd_mod_write() do nothing on Gen1, but it is confusable.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.
This patch cleanup it with focusing about SRC and Gen2 part.
SRC_CTRL/BUSIF_MODE are used for transfer start.
This patch adds rsnd_scu_transfer_start() and merge these
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound has SRC (= Sampling Rate Converter),
but, the HW implementation depends on its generation.
It was part of SRU on Gen1, and SCU on Gen2.
This SCU needs DMA transfer to use it.
Current rsnd driver is using it as DMA transfer buffer
(= no rate convert), and Gen1 is only supported at this point.
This patch cleanup it with focusing about SRC and Gen2 part.
rsnd_scu_set_route() is needed only Gen1.
This patch renames it to rsnd_scu_set_route_if_gen1()
and it adds comment to rsnd_reg member
in order to clarify it is used for Gen1.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This driver is assuming that
RBGA is used as source clock of 44.1kHz category, and
RBGB is used as source clock of 48kHz category.
This patch clarifies the variable name.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use correct register name which appears in the datasheet
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
