All CODEC input and output widgets are either in the DAPM routing table or
manually marked as non-connected. This means the card is fully routed and we can
let the core take care of disconnecting non-connected pins.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Most of the ASoC s6000 code is architecture independent. This patch makes it
possible to select the platform when COMPILE_TEST is enabled.
The only architecture dependent code is the PCM driver which will still only be
selected if XTENSA_VARIANT_S6000 is enabled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform_driver remove callback return type is int not void.
Fixes the following warning:
sound/soc/s6000/s6000-i2s.c:604:19: warning: incorrect type in initializer (different base types)
sound/soc/s6000/s6000-i2s.c:604:19: expected int ( *remove )( ... )
sound/soc/s6000/s6000-i2s.c:604:19: got void ( static [toplevel] *<noident>)( ... )
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Based on the dma_slave_caps's addr_widths queried from the dma driver
prepare the hw.formats mask to include only formats which is supported by
the DMA engine.
In case the dma driver does not implement the slave_caps the default
assumption is that it supports 1, 2 and 4 bytes widths.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds Realtek ALC286 codec driver.
ALC286 is a dual mode codec, which can run as HD-A or I2S mode.
It is controlled by HD-A verb commands via I2C protocol.
The following is the I/O difference between ALC286 and general I2S codecs.
1. A HD-A verb command contains three parts, NID, VID, and PID.
And an I2S command contains only two parts: address and data.
2. Not only the register address is written, but the read command also
includes the entire write command.
3. rt286 uses different registers for read and write the same bits.
We map verb command to regmap structure. However, we read most registers from
cache to prevent the asymmetry read/write issue in rt286.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Gustaw Lewandowski <gustaw.lewandowski@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since MODULE_LICENSE is missing the module load fails,
so add this for module.
Signed-off-by: Praveen Diwakar <praveen.diwakar@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Fix a long standing bug in the read register routing of adau1701.
The bytes arrive in the buffer in big-endian, so the result has to be
shifted before and-ing the bytes in the loop.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
This patch adds ASoC support for SiRF SoCs USP interface.
Features include:
1. Only support slave mode.
2. Support I2S and DSP_A mode.
3. Support S16_LE, S24_LE and S24_3LE formats.
4. Support stereo and mono mode.
5. The biggest Support is 192Khz sample rate.
Signed-off-by: Rongjun Ying <rongjun.ying@csr.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The TAS5086 has two power domains, DVDD and AVDD. Enable them both as
long as the codec is in use.
Also, switch on the power to identify the chip at device probe level,
and switch it off again afterwards. The codec level will take care for
power handling later.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are two call sites for snd_soc_post_component_init(), one passes 0 and the
other 1 for the 'dailess' parameter of snd_soc_post_component_init(). Depending
on whether 'dailess' is 0 or 1 snd_soc_post_component_init() runs different code
at the beginning and the end of the function. The patch moves this conditional
code out of snd_soc_post_component_init() and into the call sites. This removes
the need for snd_soc_post_component_init() to know whether it is called for a
DAI link or a aux dev.
Also do the initialization of rtd->card when the rtd struct is allocated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently in snd_soc_instantiate_card() we only check if the aux dev exists, but
do not yet assign it to its rtd. This means that we need to lookup the aux dev
again in soc_probe_aux_dev(). This patch changes the behavior to assign the aux
dev to the rtd in soc_check_aux_dev() (and renames it to soc_bind_aux_dev()).
This simplifies the implementation a bit and also removes the need for
soc_post_component_init() to know about the specific CODEC that was assigned to
the rtd. The later is necessary for componentization as the code should work for
all types of components not just CODECs. This new behavior is also more in sync
with how soc_bind_dai_link()/soc_probe_link_dais() works.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
soc_find_matching_codec() works in the same way as soc_find_codec() except that
it only works for auxdevs. It can easily be replaced by the generic
soc_find_codec().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec is already initialized in soc_bind_dai_link(), so there is no need to
do it again in soc_dai_link_init(). Removing the rtd->codec initialization from
soc_dai_link_init() also removes the need for soc_dai_link_init() to know about
the CODEC at all.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a minimum support of Realtek ALC5670 codec.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
R-Car sound uses Audio DMAC and Audio DMAC peri peri.
Audio DMAC peri peri transfers data inside circuit.
DMA transfer needs source / destination address,
and destination address can be set via dmaengine_slave_config().
The source address can be set when starting DMAEngine.
Because Audio DMAC peri peri always ignores its value,
current driver always used same source address for
Audio DMAC / Audio DMAC peri peri
(Audio DMAC peri peri source / destination address
is always fixed value)
But, This is not good match for DT booting.
This patch properly uses DMA start address
for Audio DMAC / Audio DMAC peri peri.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current R-Car sound SSI/SRC/DVC selection has feature limit.
(It is assuming that SSI/SRC are using same index number)
So that enabling SSI/SRC flexible selection,
this patch modifies DMA settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Refactor the function to facilitate the migration to
multiple codecs.
Fix a trailing space in the header as well.
No functional change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since multiple codecs DAI will be usable in the future, remove
explicit unique codec_dai and cpu_dai parameters.
Replace them with snd_soc_pcm_runtime pointer that will contain
every instances.
No functionale change.
Signed-off-by: Benoit Cousson <bcousson@baylibre.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
I received a report this morning from one of the Novena developers that
the behaviour of the iMX6 ASoC codec driver (using imx-pcm-dma.c) was
sub-optimal under high system load.
While there are issues relating to system load remaining, upon reviewing
the ASoC imx-pcm-dma.c driver, it was noticed that it not using the
residue support, because SDMA doesn't support it. This has the effect
that SDMA has to make multiple calls into the ASoC and ALSA code, one
for each period.
Since ALSA's snd_pcm_elapsed() does not need to be called multiple times
and it is entirely sufficient to call it once to update ALSA with the
current buffer position via the pointer method, we can do better here.
We can also avoid stopping the DMA entirely, just like real cyclic DMA
implementations behave. While this means that we replay some old samples,
this is a nicer behaviour than having audio stop and restart.
The changes to achieve this are relatively minor - imx-sdma.c can track
where the DMA is to the nearest descriptor boundary - it does this
already when deciding how many callbacks to issue. In doing this,
buf_tail always points at the descriptor which will complete next.
The residue is defined by the bytes remaining to the end of the buffer,
when the buffer is viewed as a single block of memory [start...end].
So, when we start out, there's a full buffer worth of residue, and this
counts down as we approach the end of the buffer, eventually becoming
zero at the end, before returning to the full buffer worth when we
wrap back to the start.
Moving the walking of the descriptors into the interrupt handler means
that we can update the BD_DONE flag at interrupt time, thus avoiding
a delayed tasklet stopping the cyclic DMA.
This means that the residue can be calculated from (total descriptors -
buf_tail) * descriptor size. This is what the change below does. We
update imx-pcm-dma.c to remove the NO_RESIDUE flag since we now provide
the residue.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
This renames all the local <mach/gpio.h> headers in the S5P platforms
to <mach/gpio-samsung.h> indicating a scope local to this platform,
and cuts the implicit inclusion of <mach/gpio.h> from <linux/gpio.h>
by removing the use of NEED_MACH_GPIO_H from all S5P variants.
Acked-by: Alexandre Courbot <acourbot@nvidia.com>
Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
Peach-pi board has MAX98091 CODEC. Extend snow machine driver to support
this board.
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Reviewed-by: Doug Anderson <dianders@chromium.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The MAX98091 CODEC is the same as MAX98090 CODEC, but with an extra
microphone. Existing driver for MAX98090 CODEC already has support
for MAX98091 CODEC. Adding proper compatible string so that MAX98091
CODEC can be specified from device tree.
Signed-off-by: Wonjoon Lee <woojoo.lee@samsung.com>
Signed-off-by: Doug Anderson <dianders@chromium.org>
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The implementation of rt5677_pll_calc() has the same logic of rl6231_pll_calc().
The only difference is the lower boundary checking for freq_in.
This patch calls rl6231_pll_calc() instead of open-coded.
The k_bp of struct rt5677_pll_code is always false, thus also remove the
code to check pll_code.k_bp.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In case of S24_LE/U24_LE modes we expect 24bits on the bus while the samples
are stored and transferred in memory on 32bits (lower 3 bytes of the 4
bytes).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Tested-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The codec need to be configured to 24bit mode in case of S24_LE format.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Correct the hw_params callback to configure the codec correctly in case of
S24_3LE format since in case of S24_3LE the codec has been configured to
16bit format mode.
S24_LE is not defined as supported format for the codec.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The cs4265_volatile_register reutrns a bool. The function now returns
true or false vs 1 and 0.
Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current DVC can be enabled only when playback,
but, this came from misunderstanding.
It is not correct.
DVC <-> DMA relationship is...
Playback: MEM -> DMAC -> SRC -> DVC -> DMACp -> SSI
Capture: SSI -> DMACp -> SRC -> DVC -> DMAC -> MEM
DVC can be used for both Playback/Capture
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current R-Car sound driver is using DMAEngine directly,
but, ASoC is requesting to use common DMA transfer method,
like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops.
It is difficult to switch at this point, since Renesas
driver is also supporting PIO transfer.
This patch uses dmaengine_prep_dma_cyclic() instead
of dmaengine_prep_slave_single().
It is used in requested method,
and is good first step to switch over.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Sound data needs to be sent to R-Car sound SSI when playback.
But, there are 2 interfaces for it.
1st is SSITDR/SSIRDR which are mapped on SSI.
2nd is SSIn_BUSIF which are mapped on SSIU.
2nd SSIn_BUSIF is used when DMA transfer,
and it is always used if sound data came from via SRC.
But, we can use it when SSI+DMA case too.
(Current driver is assuming 1st SSITDR/SSIRDR for it)
2nd SSIn_BUSIF can be used as FIFO.
This is very helpful/useful for SSI+DMA.
But DMA address / DMA ID are not same between 1st/2nd cases.
This patch care about these settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current FSI driver is using DMAEngine directly,
but, ASoC is requesting to use common DMA transfer method,
like snd_dmaengine_pcm_trigger() or dmaengine_pcm_ops.
It is difficult to switch at this point, since Renesas
driver is also supporting PIO transfer.
This patch uses dmaengine_prep_dma_cyclic() instead
of dmaengine_prep_slave_single().
It is used in requested method,
and is good first step to switch over.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
fsi PIO/DMA handler are using each own pointer update method,
but these can be share.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current fsi driver is using SNDRV_DMA_TYPE_CONTINUOUS
for snd_pcm_lib_preallocate_pages_for_all().
But, it came from original dma-sh7760.c,
and no longer needed.
This patch exchange its parameter, and removed
original dma mapping and un-needed
dma_sync_single_xxx() from driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Avoid creating duplicate directories by prefixing codecs and platforms
with their separate identifiers. This avoids snd-soc-dummy (which can
appear both as a dummy platform and a dummy codec on the same card)
from clashing.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Permit ALSA to run without hardware interrupts from the audio interface.
Instead, ALSA will use a kernel timer to decide when to check the buffer
state, resulting in a lighter workload for the CPU.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
There is no hardware restriction requiring a minimum of 8 periods, or
a minimum of 2048 bytes in a period. Let's drop these values so that
userspace has more flexibility in choosing these parameters.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
We still see the occasional timeout waiting for busy to clear. As the
spec is contradictory, and we know that the current implementation
doesn't work, try an alternative interpretation from the spec. This
one appears to work - I have yet to find any issue with it during my
testing over several months.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
The spec requires that the mute bits must be set while the channel
is disabled. Ensure that this is the case by providing a helper
which ensures that the appropriate mute bit is set while the enable
bit is clear.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since we wish to disable capture inputs for some formats, we need to
ensure that we clear the enable bits in our cached record control
register. This seems to have been missed, resulting in the register
only accumulating enable bits.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add a KIRKWOOD_RECCTL_ENABLE_MASK definition to complement the existing
PLAYCTL definition, and make use of it where we wish to clear both
enable bits.
Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk>
Tested-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
Also add const to array
text data bss dec hex filename
10946 2904 3528 17378 43e2 sound/soc/codecs/wm0010.o-before
10891 2840 3512 17243 435b sound/soc/codecs/wm0010.o-after
Signed-off-by: Fabian Frederick <fabf@skynet.be>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let memory subsystem handle the error logging.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Acked-by: Paul Handrigan <Paul.Handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Kernel should not enable/disable speakers and digital microphone whenever
jack is inserted/removed. This is more use-case than kernel specific
decision. For instance one may want to play VoIP ring tones using both
speakers and headphone but play music only from one of them.
Because of above reason remove "Ext Spk" and "Int Mic" update when jack
state is changed. Also this update was illogical anyway: "Ext Spk" was
enabled when jack was inserted and disabled when jack was removed.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It turned out there is no need to enable microphone detection in MAX98090
codec. Headset microphone is anyway detected by a GPIO signal from another
chip and headset button presses cannot be detected either because a signal
needed for it is not connected.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Pass actual jack type bitmask to snd_soc_jack_new() in order to report
also microphone detections and not only headphone. While at it change also
jack name and pass also SND_JACK_LINEOUT type.
Reported-by: Jin Yao <yao.jin@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Headset jack has only mono microphone input.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Mic detect GPIO is active low when headset microphone is detected. Found
both by debugging and checking the schematics.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move "MICBIAS" as a supply widget to "Headset Mic" instead of keeping it
between input pin "IN34" and "Headset Mic".
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Cirrus Logic CS4265 Stereo I2C CODEC.
Signed-off-by: Paul Handrigan <paul.handrigan@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use clk_prepare_enable/clk_disable_unprepare to make the driver
work properly with common clock framework.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use clk_prepare_enable/clk_disable_unprepare to make the driver
work properly with common clock framework.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use dmaengine instead of legacy s3c24xx DMA API for s3c24xx and s3c2412
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As added in previosu patch along with stream to piep conversion si required for
compressed audio too
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Merrifield DSP used various pipelines to identify the streams and processing
modules. Add these defination in the pcm driver and also add a table for device
entries to firmware pipeline id conversion
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixed coding style issues of "Missing Blank line after declaration"
Signed-off-by: Shahina Shaik <sharab.shaik@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixed Coding style issues of lines over 80 characters.
Signed-off-by: Shahina Shaik <sharab.shaik@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixed a brace coding style issue in the tlv320aic32x4.c
Signed-off-by: Shahina Shaik <sharab.shaik@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Tis will be used to add table based support for pcm front ends in subsequent
patches
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This will be used to update current driver as well as in support for the mrfld
patches
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit 31ee2bfd72 ("ASoC: fsl: select SND_SOC_IMX_PCM_DMA
where needed") started selecting SND_SOC_IMX_PCM_DMA and
SND_SOC_IMX_PCM_FIQ for two drivers when building for i.MX.
This has turned out too aggressive, as FIQ is only available
for i.mx2 through i.mx5, but not i.mx6 or vybrid.
Further, two more drivers have become user-selectable in the
meantime, and they both depend on DMA for the imx platform
as well.
This changes the selection of FIQ to depend on the TZIC or
AVIC interrupt controllers that actually export the imx
specific FIQ interfaces, and adds the missing select statements
for SAI and ESAI.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Both codecs used by snow, max98090 and max98095 require the use
of I2C, so we can only select this driver if I2C is there, otherwise
we get a build error like:
codecs/max98090.c:2494:1: warning: data definition has no type or storage class [enabled by default]
module_i2c_driver(max98090_i2c_driver);
^
codecs/max98095.c:2443:1: warning: data definition has no type or storage class [enabled by default]
module_i2c_driver(max98095_i2c_driver);
^
This adds one more I2C dependency to the hundreds we already
have.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The platform field in the snd_soc_dapm_widget and snd_soc_dapm_context structs
is now unused can be removed. New code that wants to get the platform for a
widget or dapm context should use snd_soc_dapm_to_platform(w->dapm) or
snd_soc_dapm_to_platform(dapm).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DAI DAPM context was added in commit be09ad90 ("ASoC: core: Add platform DAI
widget mapping") and the only user was removed again in commit ae10e7e8f ("ASoC:
core: Only add platform DAI widgets once."). Now that we have a per component
DAPM context it is unlikely that we'll need the DAI DAPM context again.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds stream_event() and seq_notifier() callbacks similar to those
found in the snd_soc_codec_driver and snd_soc_platform driver struct to the
snd_soc_component_driver struct. This is meant to unify the handling of these
callbacks across different types of components and will eventually allow their
removal from the CODEC and platfrom driver structs.
The new callbacks are slightly different from the old ones in that they take a
snd_soc_component as a parameter rather than a snd_soc_dapm_context. This was
done since otherwise casting from the DAPM context to the component would
typically be the first thing to do in the callback. And the interface becomes
slightly cleaner by passing a snd_soc_component to all callbacks in the
snd_soc_component_driver struct.
The patch also already removes the stream_event() callback from the
snd_soc_codec_driver and snd_soc_platform_driver structs as it is currently
unused.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_soc_platform dapm field is not accessed outside of the ASoC core. Switch
it over to using the snd_soc_component DAPM context.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds full DAPM support at the component level. Previously there was
only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor
MUXs) for platforms. Having DAPM support at the component level will allow all
types of components to use DAPM and also help in consolidating the DAPM support
between CODECs and platforms.
Since the DAPM context is directly embedded into the snd_soc_codec and
snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of
drivers moving the field just right now is not possible without causing code
churn. The approach this patch takes is to add two new fields to the component
struct. One field which is the pointer to the actual DAPM context used by the
component and one DAPM context that will be used as the default if no other
context was specified. For CODECs and platforms the pointer is initialized to
point to the CODEC or platform DAPM context. All generic code when referencing
a component's DAPM struct will go via the pointer. This will make it possible to
eventually seamlessly move the DAPM context from snd_soc_codec and
snd_soc_platform struct over once all direct references have been eliminated.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently the DAPM code directly looks at the CODEC driver struct to get a
handle to the set_bias_level() callback. This patch adds a new set_bias_level()
callback to the DAPM context struct. The DAPM code will use this new callback
instead of the CODEC callback. For CODECs the new callback is set up to call the
CODEC specific set_bias_level callback(). Not looking directly at the CODEC
driver struct will allow non CODEC DAPM contexts to implement a set_bias_level()
callback.
This is also similar to how the seq_notifier() and stream_event() callbacks are
currently handled.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently only pins in CODEC DAPM contexts are automatically marked as
non-connected if the card has the fully_routed flag set. This makes sense since
widgets which qualify for auto-disconnection are only found in CODEC DAPM
contexts. But with componentisation this is going to change, so consider all
widgets for auto-disconnection.
Also it is probably faster to walk the widgets list only once rather than once
for each CODEC.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Split snd_soc_component_register() into snd_soc_component_initialize() and
snd_soc_component_add(). Using a 2-stage registration approach has the advantage
that it is possible to modify the component after it has been initialized, but
before it is made visible to the system. This e.g. allows CODECs or platforms to
overwrite some of the default settings made in snd_soc_component_initialize().
Similar snd_soc_component_unregister() is split into two steps as well,
snd_soc_component_delete(), which removes the component from the system, and
snd_soc_component_cleanup(), which frees all the resources allocated by the
component.
Furthermore this patch makes sure that if a component is visible on two list
(e.g. the component list and the CODEC list) it is added or removed to both
lists atomically.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The component struct already has a name and id field which are initialized to
the same values as the same fields in the CODEC and platform structs. So remove
them from the CODEC and platform structs and used the ones from the component
struct instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move the name_prefix from the CODEC struct to the component struct. This will
eventually allow to specify prefixes for all types of components. It is also
necessary to make the DAPM code component type independent (i.e. a DAPM context
does not need to know whether it belongs to a CODEC or a platform or something
else).
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
One-bit signed bitfields have two possible values: 0 and -1. This sometimes
leads to unexpected results (e.g. foo.bar = 1; foo.bar == 1 => false) which is
why it is recommended to make one-bit bitfields unsigned.
This fixes the following sparse warnings:
sound/soc/sh/fsi.c:267:25: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:268:22: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:269:20: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:270:28: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:271:26: error: dubious one-bit signed bitfield
sound/soc/sh/fsi.c:272:25: error: dubious one-bit signed bitfield
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
dmaengine_prep_slave_single() expects a enum dma_transfer_direction and not a
enum dma_data_direction. Since the integer representations of both DMA_TO_DEVICE
and DMA_MEM_TO_DEV aswell as DMA_FROM_DEVICE and DMA_DEV_TO_MEM have the same
value the code worked fine even though it was using the wrong type.
Fixes the following warning from sparse:
sound/soc/sh/rcar/core.c:227:49: warning: mixing different enum types
sound/soc/sh/rcar/core.c:227:49: int enum dma_data_direction versus
sound/soc/sh/rcar/core.c:227:49: int enum dma_transfer_direction
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
omap_pcm_platform_register() is declared in omap-pcm.h and defined in
omap-pcm.c. To make sure that the function signature matches for both omap-pcm.c
should include omap-pcm.h
Fixes the following warning from sparse:
sound/soc/omap/omap-pcm.c:235:5: warning: symbol
'omap_pcm_platform_register' was not declared. Should it be static?
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The cs42xx8_of_match table is not used outside of the driver, hence it can and
should be made static.
Fixes the following warning from sparse:
sound/soc/codecs/cs42xx8.c:425:27: warning: symbol 'cs42xx8_of_match' was
not declared. Should it be static?
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
SOC_ENUM_SINGLE_DECL() already adds the const qualifier, so there is no need to
manually specify it.
Fixes the following warnings from sparse:
sound/soc/codecs/tlv320aic31xx.c:253:1: warning: duplicate const
sound/soc/codecs/tlv320aic31xx.c:255:1: warning: duplicate const
sound/soc/codecs/tlv320aic31xx.c:257:1: warning: duplicate const
sound/soc/codecs/tlv320aic31xx.c:260:1: warning: duplicate const
sound/soc/codecs/tlv320aic31xx.c:262:1: warning: duplicate const
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Now the DSP is capable of reporting the delay, report it to upper layers
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The argument was called mad_substream which is no longer apt as older driver
is not used anymore so rename as arg
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This helps us to handle pcm and compress ops seperately and per dai
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Connect "Headset Mic" to "MICBIAS1" supply widget of RT5640 in order to
enable bias voltage for headset microphones.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DSP initialization complete message IPC_IA_FW_INIT_CMPLT is a large message
carrying firmware details in mailbox. Read and show those details during
init in order to be able to get that information to QA reports.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When SSC work as master, it will generate the frame sync signal.
On old SoCs, it only supports frame sync length less or equal to
16bits, on newer SoCs, it supports frame sync length extension,
which can support frame size larger than 16 bits.
So, add this to make it supports playback 24/32 bits audio clips.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
On the wm5110 CODEC both the left and right channel must be powered
when an output is being used as a mono output, although no audio is
routed to the right output channel. This patch adds additional DAPM
routes to link the right channel to the left in the case where an output
is marked as mono. Audio must always be brought in on the left channel
for mono operation.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch replaces the string "micbias1" to "MICBIAS1".
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
we need to release dapm widget list after dpcm_path_get in
soc_dpcm_runtime_update. otherwise, there will be potential memory
leak. add dpcm_path_put to fix it.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Passing unsigned int pointers as u32 ponters may be dangerous on 64-bit
system.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Do not fail if the exact BLCK rate can not be produced, just print a
warning. Check that sysclk frequency is set before implicitly setting
the BCLK divider.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixes build with SND_DAVINCI_SOC or SND_OMAP_SOC alone and adds build
dependecy to SND_DAVINCI_SOC or SND_OMAP_SOC.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
of_get_child_by_name expects of_node_put be called when done.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
max98090.c doesn't free the threaded interrupt it requests. This causes
an oops when doing "cat /proc/interrupts" after snd-soc-max98090.ko is
unloaded.
Fix this by requesting the interrupt by using devm_request_threaded_irq().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Cc: Stable <stable@vger.kernel.org> # 3.10+
Signed-off-by: Mark Brown <broonie@linaro.org>
We now have a generic helper function to cast from a DAPM context to a CODEC.
Make use of it in the places which previously open-coded it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In DIT (S/PDIF) mode, program the transmitted user bits to reflect the
configured sample rate, along with some other details.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Document the newly added regulators to the DT binding document.
Also, "static const char const *x" is not identical to "static const
char * const x", which sparse now complains about. Fix it.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The chip has two power supplies, VA and VDD. Enable them both as long
as the codec is in use.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current dma name search loop didn't care about SSI index
This patch fixes it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ad32d0c7b0
(ASoC: rsnd: add rsnd_gen_dma_addr() for DMAC addr)
added rsnd_gen_dma_addr() to calculate DMA addr,
but, it is necessary only for Gen2.
This patch ignores Gen1 case.
Kernel will be panic without this patch.
Special thanks to Simon
Reported-by: Simon Horman <horms@verge.net.au>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Simon Horman <horms+renesas@verge.net.au>
Signed-off-by: Mark Brown <broonie@linaro.org>
Building the pxa/mmp audio driver without support for the mmp
sram driver enabled results in this link error:
sound/built-in.o: In function `mmp_pcm_free_dma_buffers':
:(.text+0x3e734): undefined reference to `sram_get_gpool'
sound/built-in.o: In function `mmp_pcm_new':
:(.text+0x3e7c0): undefined reference to `sram_get_gpool'
The sram driver is cannot be manually enabled and needs to
be turned on by selecting MMP_SRAM from each module that
needs it, which is what this patch does.
Ideally, MMP should move over to the generic SRAM support, but
for the moment, we can avoid the build error.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
wm1811_micd_stop takes the accdet_lock mutex, and is called from two
places, one of which is already holding the accdet_lock. This obviously
causes a lock up.
This patch fixes this issue by removing the lock from wm1811_micd_stop
and ensuring that it is always locked externally.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
do_div() requires that the first parameter is a 64-bit integer,
which but clkrate was defined as an unsigned long. This caused
the following warnings:
CC sound/soc/fsl/fsl_ssi.o
sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_set_bclk':
sound/soc/fsl/fsl_ssi.c:593:3: warning: comparison of distinct pointer types lacks a cast
sound/soc/fsl/fsl_ssi.c:593:3: warning: right shift count >= width of type
sound/soc/fsl/fsl_ssi.c:593:3: warning: passing argument 1 of '__div64_32' from incompatible pointer type
include/asm-generic/div64.h:35:17: note: expected 'uint64_t *' but argument is of type 'long unsigned int *'
Signed-off-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit a0b148b4 (ASoC: wm8985: Use devm_regulator_bulk_get()) removed the last
user of pointer wm8985 to struct wm8985_priv. Thus remove it. Detected by
Coverity CID 1222150.
Signed-off-by: Christian Engelmayer <cengelma@gmx.at>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add support for the output sample rates 96kHz and 192kHz.
Tested with a Cubox-i imx6 system and an Onkyo TX-SR607 receiver.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
1) Apply better indentations
2) Drop braces for single statement.
3) Use simpler ternary to reduce code.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Transmitters and receivers may support a 192kHz sample rate.
Tested with a Cubox-i imx6 system and an Onkyo TX-SR607 receiver.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Mark Brown <broonie@linaro.org>
Controls for shaping the ultrasonic frequency response were introduced
in this commit:
commit 720630c002ffc7b0fa2ed5b3f4bfb36fd8f87ca6
ASoC: wm5102: Add controls to allow shaping of ultrasonic response
However, they mistakenly used snd_kcontrol_chip instead of
snd_soc_kcontrol_codec, which has replaced it now the framework is
moving to componentisation. This patch fixes this.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit 432481220 (ASoC: fsl-ssi: Use regmap) removed struct ccsr_ssi.
Unfortunately, the structure is still used. This causes
mpc85xx_smp_defconfig and mpc85xx_defconfig builds to fail with
sound/soc/fsl/fsl_dma.c:926:50:
error: invalid use of undefined type 'struct ccsr_ssi'
dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0);
ound/soc/fsl/fsl_dma.c:927:50:
error: invalid use of undefined type 'struct ccsr_ssi'
dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0);
Fix by using constants, similar to original commit.
Cc: Markus Pargmann <mpa@pengutronix.de>
Signed-off-by: Guenter Roeck <linux@roeck-us.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
Index of dma name should use -1, not +1 when capture case.
Thank you Dan.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the SigmaDSP module is built-in, but the I2C core is build as a module
we'll get a undefined reference:
sound/built-in.o: In function `sigma_action_write_i2c':
:(.text+0x5d8d4): undefined reference to `i2c_master_send'
This can happen if a audio driver that is using the regmap SigmaDSP interface is
built into the kernel, but core I2C support is build as a module. To fix this
split the SigmaDSP module into three modules, one module providing the core
infrastructure and two small modules implementing the regmap and I2C interfaces.
This allows e.g. the core infrastructure and regmap support to be built into the
kernel while I2C support can still be build as a module.
Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add controls to allow custom shaping of the ultrasonic response. This
custom shaping can be turned on/off at runtime, although, it should be
noted that settings will not affect a currently open audio stream,
they will be applied when the next audio stream is started.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The calculation code does
u64 = (u32 - u32) * 100000;
The 64 bits are of no help here as the type is casted only after the
multiplication, and therefore the result may overflow, possibly causing
inoptimal or wrong clock setup in an unfortunate case (the maximum
result value of the first substraction is currently 47999).
Fix the code to cast before multiplication.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We should not copy the return value into this val since it's supposed to
get the value of the register not the success result of regmap_read().
Thus fix it.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Commit c9e065c27f ("ASoC: dapm: Make sure to always update the DAPM graph
in _put_volsw()") stopped updating register values in those cases where
initial after boot state of kcontrol appears to not change but where
register value still needs update because it is not in sync with the
kcontrol state.
Fix this by doing snd_soc_test_bits() unconditionally as it was before but
by using separate flags for kcontrol and register state changes. This allow
both DAPM graph to be updated when disabling auto-muted control and update
register if it is out-of-sync in respect of kcontrol state.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the SigmaDSP module is built-in, but the I2C core is build as a module
we'll get a undefined reference:
sound/built-in.o: In function `sigma_action_write_i2c':
:(.text+0x5d8d4): undefined reference to `i2c_master_send'
This can happen if a audio driver that is using the regmap SigmaDSP interface is
built into the kernel, but core I2C support is build as a module. To fix this
split the SigmaDSP module into three modules, one module providing the core
infrastructure and two small modules implementing the regmap and I2C interfaces.
This allows e.g. the core infrastructure and regmap support to be built into the
kernel while I2C support can still be build as a module.
Fixes: dab464b60 ("ASoC: Add ADAU1361/ADAU1761 audio CODEC support")
Reported-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
From e7a94bb7fb871c73cc85712d89c1f48d0271c1be Mon Sep 17 00:00:00 2001
From: Arnd Bergmann <arnd@arndb.de>
Date: Thu, 5 Jun 2014 12:31:28 +0200
Subject: [PATCH] ASoC: MMP audio needs sram support
Building the pxa/mmp audio driver without support for the mmp
sram driver enabled results in this link error:
sound/built-in.o: In function `mmp_pcm_free_dma_buffers':
:(.text+0x3e734): undefined reference to `sram_get_gpool'
sound/built-in.o: In function `mmp_pcm_new':
:(.text+0x3e7c0): undefined reference to `sram_get_gpool'
The sram driver is cannot be manually enabled and needs to
be turned on by selecting MMP_SRAM from each module that
needs it, which is what this patch does.
Ideally, MMP should move over to the generic SRAM support, but
for the moment, we can avoid the build error.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Cc: Eric Miao <eric.y.miao@gmail.com>
Cc: Haojian Zhuang <haojian.zhuang@gmail.com>
Cc: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
At this time, majority of changes come from ASoC world while we got a
few new drivers in other places for FireWire and USB. There have been
lots of ASoC core cleanups / refactoring, but very little visible to
external users.
ASoC
- Support for specifying aux CODECs in DT
- Removal of the deprecated mux and enum macros
- More moves towards full componentisation
- Removal of some unused I/O code
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers
- Several drivers exposed directly in Kconfig for use with simple-card
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651 and
ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and ADAU1781,
and Realtek RT5677
HD-audio:
- Clean up Dell headset quirks
- Noise fixes for Dell and Sony laptops
- Thinkpad T440 dock fix
- Realtek codec updates (ALC293,ALC233,ALC3235)
- Tegra HD-audio HDMI support
FireWire-audio:
- FireWire audio stack enhancement (AMDTP, MIDI), support for incoming
isochronous stream and duplex streams with timestamp synchronization
- BeBoB-based devices support
- Fireworks-based device support
USB-audio:
- Behringer BCD2000 USB device support
Misc:
- Clean up of a few old drivers, atmel, fm801, etc
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Merge tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into next
Pull sound updates from Takashi Iwai:
"At this time, majority of changes come from ASoC world while we got a
few new drivers in other places for FireWire and USB. There have been
lots of ASoC core cleanups / refactoring, but very little visible to
external users.
ASoC:
- Support for specifying aux CODECs in DT
- Removal of the deprecated mux and enum macros
- More moves towards full componentisation
- Removal of some unused I/O code
- Lots of cleanups, fixes and enhancements to the davinci, Freescale,
Haswell and Realtek drivers
- Several drivers exposed directly in Kconfig for use with
simple-card
- GPIO descriptor support for jacks
- More updates and fixes to the Freescale SSI, Intel and rsnd drivers
- New drivers for Cirrus CS42L56, Realtek RT5639, RT5642 and RT5651
and ST STA350, Analog Devices ADAU1361, ADAU1381, ADAU1761 and
ADAU1781, and Realtek RT5677
HD-audio:
- Clean up Dell headset quirks
- Noise fixes for Dell and Sony laptops
- Thinkpad T440 dock fix
- Realtek codec updates (ALC293,ALC233,ALC3235)
- Tegra HD-audio HDMI support
FireWire-audio:
- FireWire audio stack enhancement (AMDTP, MIDI), support for
incoming isochronous stream and duplex streams with timestamp
synchronization
- BeBoB-based devices support
- Fireworks-based device support
USB-audio:
- Behringer BCD2000 USB device support
Misc:
- Clean up of a few old drivers, atmel, fm801, etc"
* tag 'sound-3.16-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (480 commits)
ASoC: Fix wrong argument for card remove callbacks
ASoC: free jack GPIOs before the sound card is freed
ALSA: firewire-lib: Remove a comment about restriction of asynchronous operation
ASoC: cache: Fix error code when not using ASoC level cache
ALSA: hda/realtek - Fix COEF widget NID for ALC260 replacer fixup
ALSA: hda/realtek - Correction of fixup codes for PB V7900 laptop
ALSA: firewire-lib: Use IEC 61883-6 compliant labels for Raw Audio data
ASoC: add RT5677 CODEC driver
ASoC: intel: The Baytrail/MAX98090 driver depends on I2C
ASoC: rt5640: Add the function "get_clk_info" to RL6231 shared support
ASoC: rt5640: Add the function of the PLL clock calculation to RL6231 shared support
ASoC: rt5640: Add RL6231 class device shared support for RT5640, RT5645 and RT5651
ASoC: cache: Fix possible ZERO_SIZE_PTR pointer dereferencing error.
ASoC: Add helper functions to cast from DAPM context to CODEC/platform
ALSA: bebob: sizeof() vs ARRAY_SIZE() typo
ASoC: wm9713: correct mono out PGA sources
ALSA: synth: emux: soundfont.c: Cleaning up memory leak
ASoC: fsl: Remove dependencies of boards for SND_SOC_EUKREA_TLV320
ASoC: fsl-ssi: Use regmap
ASoC: fsl-ssi: reorder and document fsl_ssi_private
...
We have in the past added 'depends on I2C' for some of the PXA boards
after hitting randconfig build bugs. I have seens a couple of new
bugs in this area during the linux-next cycle for 3.16, after it
became possible to build some more PXA machines with I2C disabled.
To shut this up for good, this adds the dependency to every board
that uses I2C as the interface to the codec. I have gone through
all board files and verified that they all either use AC97 or
I2C, and this annotates the latter. Some of these already enable
I2C from mach-pxa/Kconfig, but since that can change it's better
to be explicit here.
The link error that can result otherwise happens when CONFIG_I2C
is set to 'm' and the codec driver is built-in as a result of being
selected by the platform specific glue.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The commit [e1d4d3c8: ASoC: free jack GPIOs before the sound card is
freed] introduced snd_soc_card remove callbacks to a few drivers, but
they are implemented with a wrong argument type. The callback should
receive snd_soc_card pointer instead of snd_soc_pcm_runtime.
Fixes: e1d4d3c854 ('ASoC: free jack GPIOs before the sound card is freed')
Acked-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the same change as commit fb6b8e7144 "ASoC: tegra: free jack
GPIOs before the sound card is freed", but applied to all other ASoC
machine drivers where code inspection indicates the same problem exists.
That commit's description is:
==========
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, guard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
==========
Note that I have not even compile-tested this in most cases, since most
of the drivers rely on specific mach-* support I don't have enabled, and
don't support COMPILE_TEST. Testing by the relevant board maintainers
would be useful.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A quite large set of SoC updates this cycle. In no particular order:
- Multi-cluster power management for Samsung Exynos, adding support for
big.LITTLE CPU switching on EXYNOS5
- SMP support for Marvell Armada 375 and 38x
- SMP rework on Allwinner A31
- Xilinx Zynq support for SOC_BUS, big endian
- Marvell orion5x platform cleanup, modernizing the implementation and
moving to DT.
- _Finally_ moving Samsung Exynos over to support MULTIPLATFORM, so
that their platform can be enabled in the same kernel binary as most
of the other v7 platforms in the tree. \o/ The work isn't quite complete,
there's some driver fixes still needed, but the basics now work.
New SoC support added:
- Freescale i.MX6SX
- LSI Axxia AXM55xx SoCs
- Samsung EXYNOS 3250, 5260, 5410, 5420 and 5800
- STi STIH407
Plus a large set of various smaller updates for different platforms. I'm
probably missing some important one here.
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Merge tag 'soc-for-3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc into next
Pull part one of ARM SoC updates from Olof Johansson:
"A quite large set of SoC updates this cycle. In no particular order:
- Multi-cluster power management for Samsung Exynos, adding support
for big.LITTLE CPU switching on EXYNOS5
- SMP support for Marvell Armada 375 and 38x
- SMP rework on Allwinner A31
- Xilinx Zynq support for SOC_BUS, big endian
- Marvell orion5x platform cleanup, modernizing the implementation
and moving to DT.
- _Finally_ moving Samsung Exynos over to support MULTIPLATFORM, so
that their platform can be enabled in the same kernel binary as
most of the other v7 platforms in the tree. \o/
The work isn't quite complete, there's some driver fixes still
needed, but the basics now work.
New SoC support added:
- Freescale i.MX6SX
- LSI Axxia AXM55xx SoCs
- Samsung EXYNOS 3250, 5260, 5410, 5420 and 5800
- STi STIH407
plus a large set of various smaller updates for different platforms.
I'm probably missing some important one here"
* tag 'soc-for-3.16' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (281 commits)
ARM: exynos: don't run exynos4 l2x0 setup on other platforms
ARM: exynos: Fix "allmodconfig" build errors in mcpm and hotplug
ARM: EXYNOS: mcpm rename the power_down_finish
ARM: EXYNOS: Enable mcpm for dual-cluster exynos5800 SoC
ARM: EXYNOS: Enable multi-platform build support
ARM: EXYNOS: Consolidate Kconfig entries
ARM: EXYNOS: Add support for EXYNOS5410 SoC
ARM: EXYNOS: Support secondary CPU boot of Exynos3250
ARM: EXYNOS: Add Exynos3250 SoC ID
ARM: EXYNOS: Add 5800 SoC support
ARM: EXYNOS: initial board support for exynos5260 SoC
clk: exynos5410: register clocks using common clock framework
ARM: debug: qcom: add UART addresses to Kconfig help for APQ8084
ARM: sunxi: allow building without reset controller
Documentation: devicetree: arm: sort enable-method entries
ARM: rockchip: convert smp bringup to CPU_METHOD_OF_DECLARE
clk: exynos5250: Add missing sysmmu clocks for DISP and ISP blocks
ARM: dts: axxia: Add reset controller
power: reset: Add Axxia system reset driver
ARM: axxia: Adding defconfig for AXM55xx
...
It is not an error to have no cache so we shouldn't return an error code
and cause our callers to fail, just silently do nothing instead. Thanks
to Jarkko for identify the problematic commit.
Reported-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Reported-by: Fabio Estevam <festevam@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch adds the function "get_clk_info" to RL6231 shared support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch adds the function of the PLL clock calculation to RL6231 shared
support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The patch adds the RL6231 class device shared support for RT5640, RT5645 and
RT5651. The function of the DMIC clock calculation can be shared by RL6231
shared support.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since we cannot make sure the 'reg_size' will always be none zero here,
and then if 'reg_size' equals to zero, the kzalloc() will return ZERO_SIZE_PTR,
which equals to ((void *)16).
So this patch fix this with just doing the 'reg_size' zero check before calling
kzalloc().
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The mono output PGA input only has four possible sources, so
omit the rest.
Signed-off-by: Matt Reimer <mreimer@sdgsystems.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Eukrea-i.MX51 board was converted to use DT, ie we no longer have a
MACH_EUKREA_MBIMXSD51_BASEBOARD symbol.
Transformation of other boards planned for the near future, so this
patch removes all these dependencies and restricts build of this
driver to ARCH_MXC.
Signed-off-by: Alexander Shiyan <shc_work@mail.ru>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch replaces the ssi specific functions write_ssi, read_ssi and
write_ssi_mask by standard regmap function calls.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reorder all variables in struct fsl_ssi_private to have groups that make
sense together. The patch also updates the struct documentation.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The baudclock may be used and set by different streams.
Allow only the first stream to set the bitclock rate. Other streams have
to try to get to the correct rate without modifying the bitclock rate
using the SSI internal clock modifiers.
The variable baudclk_streams is introduced to keep track of the active
streams that are using the baudclock. This way we know if the baudclock
may be set and whether we may enable/disable the clock.
baudclock enable/disable is moved to hw_params()/hw_free(). This way we can
keep track of the baudclock in those two functions and avoid a running
clock while it is not used. As hw_params()/hw_free() may be called
multiple times for the same stream, we have to use baudclk_streams
variable to know whether we may enable/disable the clock.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In i2s master mode the fsl_ssi driver depends on someone calling
.set_tdm_slot correctly. In this mode though only a DC value of
2 is allowed, so set it in this case and no longer depend on
.set_tdm_slot.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The baudclock_locked variable is only used in functions which
are serialized anyway from the core. No need to have a lock
around the variable, so remove it.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The fsl_ssi driver uses the .set_sysclk callback to configure the
bitclock for master mode. This is unnecessary since the bitclock
is known in hw_params. This patch configures the bitclock from .hw_params.
.set_dai_sysclk now sets a bitclock frequency which is preferred over
the default calculated bitclock frequency.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The simple soundcard binding has its own way for specifying the dai
format. To be able to use this binding we have to make the fsl,mode
property optional. As the property is used in existing devicetrees
keep the option around for compatibility reasons.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Introduce a SoC data struct which contains the differences between
the different SoCs this driver supports. This makes it easy to support
more differences without having to introduce a new switch/case each
time.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use card PM ops from ASoC core instead of defining custom PM ops here since
we are calling anyway common suspend/resume callbacks.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify byt-rt5640.c and haswell.c machine drivers by using
devm_snd_soc_register_card(). Remove also needless dev_set_drvdata()
from byt_rt5640_probe() since snd_soc_register_card() does it too.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A few files contain duplicate headers. This patch removes the second entry of
duplicate in each file under question.
There is no functional changes.
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Cc: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Stored DSP DMA pointer must be cleared before starting the stream since
PCM pointer callback sst_byt_pcm_pointer() can be called before pointer is
updated. In that case last position of previous stream was wronly returned.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The new value argument needs proper shift to match the mask bit fields.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Baytrail and Haswell SST IPC don't stop the kernel thread in error and
cleanup path thus leaving orphan kernel thread behind in such a case.
Also while at it, fix one error path in sst-haswell-ipc.c that doesn't free
hsw->msg.
[Jarkko: I edited the commit log a little]
Signed-off-by: Imre Deak <imre.deak@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add machine driver and ACPI probing for Baytrail SST with MAX98090 codec.
Jack detect code from Kevin Strasser <kevin.strasser@intel.com>, GPIO
resolving from Mika Westerberg <mika.westerberg@linux.intel.com> and fixes
and cleanups from Liam Girdwood <liam.r.girdwood@linux.intel.com>.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For some unknown reason the parameters for snd_soc_test_bits() were in wrong
order:
It was:
snd_soc_test_bits(codec, val, mask, reg); /* WRONG!!! */
while it should be:
snd_soc_test_bits(codec, reg, mask, val);
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
WM8804 can run with PLL frequencies of 256xfs and 128xfs for
most sample rates. At 192kHz only 128xfs is supported. The
existing driver selects 128xfs automatically for some lower
samples rates. By using an additional mclk_div divider, it
is now possible to control the behaviour. This allows using
256xfs PLL frequency on all sample rates up to 96kHz. It
should allow lower jitter and better signal quality. The
behavior has to be controlled by the sound card driver,
because some sample frequency share the same setting. e.g.
192kHz and 96kHz use 24.576MHz master clock. The only
difference is the MCLK divider.
Signed-off-by: Daniel Matuschek <daniel@matuschek.net>
Tested-by: Florian Meier <florian.meier@koalo.de>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a ASoC machine driver to support the EVAL-ADAU1X81 board
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a ASoC machine driver to support the EVAL-ADAU1X61 board
connected to a Analog Devices BF5XX evaluation board.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1381 and ADAU1781 audio
CODECs. The device is a low-power, 24-bit stereo audio CODEC with multiple
analog inputs and outputs, two digital microphone inputs and an I2S interface.
The device can be controlled either using I2C or SPI. The main difference
between the two variants is that the ADAU1781 has a freely programmable SigmaDSP
processor, while the ADAU1381 has a fixed function wind noise reduction filter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the Analog Devices ADAU1361 and ADAU1761 CODECs.
The device is a a low-power, 24-bit stereo audio CODEC with multiple analog
input and outputs, one digital microphone input and an I2S interface. The device
can be controlled either via I2C or SPI. The main difference between the two
variants is that the ADAU1761 has a built-in SigmaDSP, while the ADAU1361 has
not.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ADAU1X61 and ADAU1X81 are very similar in the digital domain, but are quite
different in the analog domain. This patch adds support for the common parts of
the ADAU1X61 and ADAU1X81 CODECs.
The patch also restores some of the alphabetical order in the Makfile and
Kconfig.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have defines for adsp messages best to consistently use them.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since commit e5d80e82e3 (ASoC: sgtl5000: Convert to use regmap directly) a
kernel oops is observed after a suspend/resume sequence.
The kernel oops happens inside sgtl5000_restore_regs() as codec->reg_cache is no
longer a valid pointer.
Add the remaining register entries into sgtl5000_reg_defaults[] and remove
sgtl5000_restore_regs() completely, which allows suspend/resume to work fine and
make the code simpler.
Tested on a im53-qsb board.
Reported-by: Shawn Guo <shawn.guo@freescale.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Shawn Guo <shawn.guo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
commit 8c5178fca4 ("ALSA: Add params_width() helpers") introduces
a helper to get the sample width. Updating Samsung related sound
drivers to use this helper.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Having the binary ones complement operator in the new bitmak value makes the
code hard to read.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Documentation has been updated to reflect the change.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
If master clock is provided through device tree, then update
the master clock frequency during set_sysclk.
Documentation has been updated to reflect the change.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add "depends on I2C" to shut up the build errors from randconfig.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make including the omap-pcm.h outside sound/soc/omap more convenient.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Allow jack GPIO pins be defined also using GPIO descriptor-based interface
in addition to legacy GPIO numbers. This is done by adding two new fields to
struct snd_soc_jack_gpio: idx and gpiod_dev.
Legacy GPIO numbers are used only when GPIO consumer device gpiod_dev is
NULL and otherwise idx is the descriptor index within the GPIO consumer
device.
New function snd_soc_jack_add_gpiods() is added for typical cases where all
GPIO descriptor jack pins belong to same GPIO consumer device. For other
cases the caller must set the gpiod_dev in struct snd_soc_jack_gpio before
calling snd_soc_jack_add_gpios().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch does basic GPIO descriptor conversion to soc-jack. Even the GPIOs
are still passed and requested using legacy GPIO numbers the driver
internals are converted to use GPIO descriptor API.
Motivation for this is to prepare soc-jack so that it will allow registering
jack GPIO pins using both GPIO descriptors and legacy GPIO numbers.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Failure to terminate this match table can lead to boot failures
depending on where the compiler places the match table.
Signed-off-by: Stephen Boyd <sboyd@codeaurora.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DMAC src/dst addr needs to be set from driver when DT case.
(It was set from SoC/DMAEngine code when non-DT case)
This patch adds rsnd_gen_dma_addr() to set DMAC src/dst addr.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting to use DMAEngine.
But, DMA slave channel name "tx", "rx" is not enough
in DT case.
Becuase, it has many ports and path combination.
This patch adds rsnd_dma_of_name() to find
DMA channel name, for example
memory to SSI0 is "mem_ssi0",
SSI0 to memory is "ssi0_mem",
SSI0 to SRC0 is "ssi0_src0",
SRC0 to SSI0 is "src0_ssi0",
SRC0 to DVC0 is "src0_dvc0"...
Renesas sound want to use PIO transfer mode for some reasons.
It will be PIO tranfer mode if device node doesn't have
DMA settings.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver uses many modules (= SSI/SRC/DVC),
and each module had own name.
But, each module name can be used as several purpose,
like clock name, DMA name etc...
This patch uses common name for each module.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Renesas sound driver is supporting Gen1/Gen2.
SRC probe can return error if it was unknown
generation.
Now, rsnd_src_non_ops is not needed.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
DT DMA support needs struct platform_device pointer,
and it can get struct device pointer from platform_device.
Save platform_device instead of device.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Driver needs to call of_node_put() after of_get_chile_by_name()
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
snd_soc_jack_add_gpios() schedules a work queue item to poll the GPIO to
generate an initial jack status report. If sound card initialization
fails, that work item needs to be cancelled, so it doesn't run after the
card has been freed. Specifically, freeing the card calls
snd_jack_dev_free() which calls snd_jack_dev_disconnect() which sets
jack->input_dev = NULL, and input_dev is used by snd_jack_report(), which
is called from the work queue item.
snd_soc_jack_free_gpios() cancels the work item. The Tegra ASoC machine
drivers do call this function in the platform driver remove() callback.
However, this happens after the sound card is freed, at least when the
card is freed due to errors late during snd_soc_instantiate_card(). This
leaves a window where the work item can execute after the card is freed.
In next-20140522, sound card initialization does fail for unrelated
reasons, and hits the problem described above.
To solve this, fix the Tegra ASoC machine drivers to clean up the Jack
GPIOs during the snd_soc_card's .remove() callback, which is executed
before the overall card object is freed. also, gGuard the cleanup call
based on whether we actually setup up the GPIOs in the first place.
Ideally, we'd do the cleanup in a struct snd_soc_dai_link .fini/remove
function to match where the GPIOs get set up. However, there is no such
callback.
This change fixes all Tegra machine drivers. By code inspection, I
believe some non-Tegra machine drivers have the same issue. I'll send a
patch for that separately, once this is reviewed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This makes sure a format string can never get processed into the worker
thread name from the device name.
Signed-off-by: Kees Cook <keescook@chromium.org>
Acked-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Some platforms require that the codecs mclk is a fixed multiplication
factor of the audio stream rate. Add a optional property to the
binding to hold this factor and implement a hw_params() function to
make use of it.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the clock divisor and multiplier NI, MI values for audio
sampling frequencies 44100 and 48000 Hz and PCLK 19.2 MHz. This is useful
for the Odroid X2/U2 boards when the codec works in master mode and its
MCLK clock is fed from the I2S CDCLK output.
Signed-off-by: Chen Zhen <zhen1.chen@samsung.com>
[s.nawrocki@samsung.com: edited the commit description]
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Ensure i2s->op_clk is not used when clk_get() for this clock fails.
This prevents working with an incorrectly configured clock in some
conditions.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' into asoc-linus
ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
# gpg: Signature made Wed 14 May 2014 12:40:27 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' into asoc-linus
ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
# gpg: Signature made Wed 14 May 2014 12:49:57 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' into asoc-linus
ASoC: Core fixes for v3.15
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
# gpg: Signature made Wed 14 May 2014 12:59:19 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg: aka "Mark Brown <broonie@debian.org>"
# gpg: aka "Mark Brown <broonie@kernel.org>"
# gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg: aka "Mark Brown <broonie@linaro.org>"
# gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
clk_get in probe function can be safely replaced with devm_clk_get.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replaced snd_soc_register_component with its devres equivalent,
devm_snd_soc_register_component.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replaced snd_soc_register_platform with devm_snd_soc_register_platform
in samsung_asoc_dma_platform_register(). This makes the function
samsung_asoc_dma_platform_unregister() redundant. This is removed and
all its users are updated.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replace snd_soc_register_card with devm_snd_soc_register_card.
With this change, we can delete the empty remove functions.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
The DAPM routes for this board are complete, hence we can let the core take care
of disconnecting non-connected pins rather than doing it manually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for setting source clock of the "Core CLK"
of the IIS Multi Audio Interface.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As we are moving the mmp platform towards multiplatform support,
we have to stop including platform header files.
This changes the pxa-ssp sound driver file to no longer depend
on mach/hardware.h and mach/dma.h. The code using the definitions
from those headers is actually gone already, the only thing
that was still being used was the pxa_dma_desc typedef, which
we can easily work around by using the normal 'struct pxa_dma_desc'
name.
The pxa2xx-dma driver still uses this header, so we include it
explicitly there, which is ok because that is only used on pxa,
not on mmp.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Xia Kaixu <kaixu.xia@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Let the ALC5623 codec be instantiated from DT. Add a simple binding
for the additional control register and the jack detect register.
Also, add a prompt to the Kconfig entry for this CODEC, so that it can
be selected. Since kirkwood-t5325.c will no longer be used, we need to
be able to enable the CODEC in the mvebu_v5_defconfig etc.
Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Acked-by: Jason Cooper <jason@lakedaemon.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
fsl_ssi_set_dai_sysclk will be called from fsl_ssi_hw_params in the
next patch. Move up to avoid forward declaration and to keep the next patch
more readable. No functional change.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the fsl-ssi unit is used in i2s slave mode, it is possible that the
SSI unit starts transmitting data on the wrong channel. This happens
because the SSI does not synchronize with the left-right-clock by
default.
This patch enables transmit enable synchronization.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
There are some variables defined in struct fsl_ssi_private that describe
states that are also described by other variables.
This patch adds some helper functions that return exactly the same
information based on available variables. This helps to clean up struct
fsl_ssi_private and remove them from the probe function.
It also removes some not really used variables (new_binding, name).
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reorder the probe function to be able to move the second imx-specific
block to the seperate imx probe function. The patch also removes some
comments/variables/code that are not used anymore or could be simply
replaced by other variables.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Simplify dma DT property handling. fsl,ssi-dma-events is not used
anymore. It passes invalid data to imx_pcm_dma_params_init_data() which
copies some data into an imx dma struct. This struct is never used in
imx-dma or imx-sdma because of generic OF DMA handling. The
"fsl,ssi-dma-events" is not used anywhere in dts files.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move imx specific probe code to a seperate function. It reduces the
size of the probe() function and makes the code and error handling
easier to understand.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of creating a name using string manipulation functions, we can
simply use the device name for the DAI driver struct.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Move all code that is only used for debugging to a seperate file. This
makes it easier to see what functions are used for debugging only.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The bits we have to clear when disabling are different when the other
stream is still active.
This patch fixes the calculation of new register values after disabling
one stream. It also adds a more detailed description of the new register
value calculation.
Signed-off-by: Markus Pargmann <mpa@pengutronix.de>
Tested-By: Michael Grzeschik <mgr@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to go via the CODEC to get a pointer to the card. This will help to
eventually remove the card field from the snd_soc_codec struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Using a proper do-while loop here instead of a open-coded goto loop is both
cleaner and shorter.
Also fixes the following warnings from smatch:
sound/soc/codecs/ad1980.c:213 ad1980_reset() info: loop could be replaced with if statement.
sound/soc/codecs/ad1980.c:212 ad1980_reset() info: ignoring unreachable code.
sound/soc/codecs/ad1980.c:215 ad1980_reset() info: ignoring unreachable code.
While we are at it also change retry_cnt to unsigned int, using u16 for a
on-stack loop counter doesn't make that much sense.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
In some cases the pcm stream is closed while context has been
scheduled to be restored, causing a null pointer deref panic.
Cancel work to ensure stream does not get freed while work is
still active/pending.
Also, restoring the pcm context can be safely skipped after the
stream has been stopped. Check if pcm stream is still running
before restoring stream context to help pending work finish
more quickly in stream close path.
Signed-off-by: Kevin Strasser <kevin.strasser@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This looks like a copy-paste bug, fix it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALSA SoC core marks widgets as connected by default when they are
initialized in snd_soc_dapm_new_control() so there is no need to call
snd_soc_dapm_enable_pin() from machine driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
ALSA SoC core takes care of calling snd_soc_dapm_sync() at the end
snd_soc_instantiate_card() so there is no need to call it from machine
driver init functions.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
rtd->codec does not necessarily point to the CODEC instance for which the
callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use
dai->codec instead.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Microphone bias level configuration register can configure voltage between
2.2 V and 2.8 V but doesn't manage is voltage on or off. Microphone bias
on/off state is controlled by "MICBIAS" DAPM widget.
Therefore there is no need to update bias voltage conditionally depending on
jack state each time when codec goes to SND_SOC_BIAS_ON state and setting
can be moved to max98090_probe() as driver currently doesn't support other
levels than 2.8 V.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
BCL is being configured in two places producing a warning message.
Make sure we only configure BCLK once and when we are master.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add ACPI ID for MAX98090 and ACPI 5 I2C device probing support.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure the cache is fully flushed at resume time.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Make sure we reset codec and clear any IRQs on resume. This matches
the init sequence in probe.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reset needs to wait 20ms before other codec IO is performed. This wait
was not being performed. Fix this by making sure the reset register is not
restored with the cache, but use the manual reset method in resume with
the wait.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
TLV for digital sidetone volume is wrong, this fix matches it to the
datasheet.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As we already have a memeber struct snd_sst_params.codec to fill this.
so removing duplicate instance
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds the support to implement drain_notify in Intels mfld driver
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This reverts commit 0cac6fc3eb.
This comiit was dropped from rev2 and would not be required as it renames the
platform ops as well which is not required.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
I managed to drop a change to stream ID setting from commit 49fee17816
("ASoC: Intel: Only export one Baytrail DAI") leading to non-working
simultaneous capture-playback since after one DAI conversion
rtd->cpu_dai->id + 1 will be the same for both playback and capture.
Use substream->stream + 1 like it was in original Liam's patch.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Use the %pap printk specifier to print resource_size_t variables. This
fixes warnings on platforms where resource_size_t has a different size
than int.
Signed-off-by: Laurent Pinchart <laurent.pinchart+renesas@ideasonboard.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We don't need more than one DAI for Baytrail SST. Usage becomes also more
straightforward by grouping playback and capture streams under the same PCM
device.
[Jarkko: I made Liam's sst-baytrail-pcm.c change a few lines smaller and
squashed together with my byt-rt5640.c change]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Prepare for single Baytrail DAI playback/capture link by accessing PCM data
using stream ID instead of rtd->dev. Now rtd->dev is unique for playback
and capture since they are exported as separate DAIs but not once converted
to single DAI.
[Jarkko: Separated from another commit with updated commit log]
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The return statement was not indented correctly. I lined up the
condition a bit as well.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel. It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be. Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Core fixes for v3.15
A few things here:
- Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
have audio paths which shouldn't be present causing spurious powerups
and potential audible issues for users.
- Ensure the suspend->off transition doesn't have spurious transitions
to prepare added to the sequence.
- Fix incorrect skipping of PCM suspension for active audio streams.
- Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
this and Timur no longer has the boards that he was using.
Adding missing pm ops so that audio playback works across
suspend and resume cycle.
Signed-off-by: Tushar Behera <tushar.behera@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Additionally to the ak4642 pll frequencies the ak4648 also supports 13MHz,
19.2MHz and 26MHz. This adds support for these frequencies.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Currently unused, this is done to let the driver distinguish between
the different supported codec types in later patches.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Lars-Peter Clausen