CODEC type (MAX98090/MAX98091) can be specified from device-tree file,
it can also be obtained from the CODEC during runtime.
Add an explicit check to figure out if both are matching, else print
a message warning about the same.
Signed-off-by: Tushar Behera <tushar.b@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We should save/restore relevant I2S registers regardless of
the dai->active flag, otherwise some settings are being lost
after system suspend/resume cycle. E.g. I2S slave mode set only
during dai initialization is not preserved and the device ends
up in master mode after system resume.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
The newly introduced sirf-usp driver defines sirf_usp_pcm_{suspend,resume}
functions only when PM_RUNTIME is enabled, but also uses them when that
is disabled and only PM_SLEEP is turned on, resulting in this error:
../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_suspend':
../sound/soc/sirf/sirf-usp.c:308:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_suspend' [-Werror=implicit-function-declaration]
sirf_usp_pcm_runtime_suspend(dev);
^
../sound/soc/sirf/sirf-usp.c: In function 'sirf_usp_pcm_resume':
../sound/soc/sirf/sirf-usp.c:319:3: error: implicit declaration of function 'sirf_usp_pcm_runtime_resume' [-Werror=implicit-function-declaration]
ret = sirf_usp_pcm_runtime_resume(dev);
^
cc1: some warnings being treated as errors
To fix that, this patch changes the #ifdef to CONFIG_PM, which
is enabled when at least one of PM_SLEEP or PM_RUNTIME are enabled.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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>
In case of _3LE/_3BE formats the samples are stored in 3 consecutive bytes
without padding it to 4 bytes. This means that the DMA needs to be able to
support 3 bytes word length in order to read/write the samples from memory
correctly. Originally the code treated 24 bits physical length samples as
they were 32 bits which leads to corruption when playing or recording audio.
The hw.formats field has already been prepared to exclude formats not
supported by the DMA engine in use, which means that only on platforms where
3 bytes is supported by the DMA will be able to use this format.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.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>
This small patch completes #defines for Control/Status Register,
adds comments for the missing ones there and on the Interrupt Mask
Register and additionally corrects "#define ICE1712_SERR_LEVEL 0x04 -> 0x08",
according to documentation.
Signed-off-by: Konstantinos Tsimpoukas <kostaslinuxxx@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For HSW/BDW display HD-A controller, hda_set_bclk() is defined to set BCLK
by programming the M/N values as per the core display clock (CDCLK) queried from
i915 display driver.
And the audio driver will also set BCLK in azx_first_init() since the display
driver can turn off the shared power in boot phase if only eDP is connected
and M/N values will be lost and must be reprogrammed.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The "list" field has been omitted from struct azx, but its
initialization remained mistakenly in hda_tegra.c, which leads to a
compile error:
sound/pci/hda/hda_tegra.c: In function 'hda_tegra_create':
sound/pci/hda/hda_tegra.c:481:22: error: 'struct azx' has no member
named 'list'
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 9a34af4a33 ('ALSA: hda - Move more PCI-controller-specific stuff from generic code')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
'private_data' is not used in the function. Remove it.
Signed-off-by: Sachin Kamat <sachin.kamat@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
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 recent commit [6194b99d: ALSA: hda - Kill the rest of snd_print*()
usages] changed the callback map_slaves(), but one call was forgotten
to be replaced due to the cast, which leads to kernel Oops due to
invalid function. This patch replaces it with a proper function.
Fixes: 6194b99de9 ('ALSA: hda - Kill the rest of snd_print*() usages')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
