Running HPDET while the headphone outputs are enabled can disrupt the
operation of HPDET. In order to avoid this HPDET needs to disable the
headphone outputs and ASoC needs to not enable them while HPDET is
running.
Do the ASoC side of this by storing the enable state in the core driver
structure and only writing to the device if a flag indicating that the
accessory detection side is in a state where it can have the headphone
output stage enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
These functions were initially added to be able to support some oddball dma
drivers, but all users have been updated to deal with the situation without the
help of snd_dmaengine_pcm_{set,get}_data, so these two functions can be removed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the mxs_dma_data struct, which gets passed to the dmaengine driver, is
allocated in the pcm driver's open callback. The mxs_dma_data struct has exactly
one field which is initialized from the the same field in the mxs_pcm_dma_params
struct. The mxs_pcm_dma_params struct gets passed to the pcm driver from the dai
driver. Instead of taking this indirection embed the mxs_dma_data struct
directly in the mxs_pcm_dma_params struct. This allows us to simplify the pcm
driver quite a bit, since we don't have to care about memory managing the
mxs_dma_data struct anymore.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the imx_dma_data struct, which gets passed to the dmaengine driver, is
allocated and constructed in the pcm driver from the data stored in the
dma_params struct. The dma_params struct gets passed to the pcm driver from the
dai driver. Instead of going this route of indirection embed the dma_data struct
directly into the dma_params struct and let the dai driver fill it in. This
allows us to simplify the imx-pcm-dma driver quite a bit, since it doesn't have
care about memory managing the imx_dma_data struct anymore.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dma filter parameters are only used within filter callback, so there is no
need to allocate them on the heap and keep them around until the PCM has been
closed.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently the ep93xx_dma_params struct which is passed to the dmaengine driver
is constructed at runtime from the ep93xx_pcm_dma_params that gets passed to the
ep93xx PCM driver from one of the ep93xx DAI drivers. The ep93xx_pcm_dma_params
struct is almost identical to the ep93xx_dma_params struct. The only missing
field is the 'direction' field, which is computed at runtime in the PCM driver
based on the current substream. Since we know in advance which
ep93xx_pcm_dma_params struct is being used for which substream at compile time,
we also already know which direction to use at compile time. So we can easily
replace all instances of ep93xx_pcm_dma_params with their ep93xx_dma_params
counterpart. This allows us to simplify the code in the ep93xx pcm driver quite
a bit.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Ryan Mallon <rmallon@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We want to get rid of snd_dmaengine_pcm_{set,get}_data(). All instances of
snd_dmaengine_pcm_get_data() in the atmel pcm driver can easily be replaced with
snd_soc_dai_get_dma_data().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Stephen Warren <swarren@wwwdotorg.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If a PCM driver using the dmaengine PCM helper functions doesn't need to do
anything special in its pcm_close callback, snd_dmaengine_pcm_close can be used
directly for as the pcm_close callback and there is no need to wrap it in a
custom function.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The generic dmaengine based PCM driver code takes care of setting this
constraint, there is no need of doing it manually in the ux500 driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Ola Lilja <ola.o.lilja@stericsson.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The omap PCM driver provides a set_threshold callback which gets called by the
PCM driver when either playback or capture is started. The only DAI driver which
sets this callback is the mcbsp driver. This patch removes the callback from the
PCM driver and moves the invocation of the omap_mcbsp_set_threshold() function
to the mcbsp hw_params callback since this is the only place where the threshold
size can change. Doing so allows us to use the default dmaengine PCM trigger
callback in the omap PCM driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tegra114 requires different PLL rates. Modify the code to know about
this.
On Tegra114 only for now, use regular clk_get() rather than clk_get_sys()
to retrieve clocks. This assumes that the clocks will be represented in
device tree. We can assure that from the start of any Tegra114 audio
support. For older chips, I'll add the required clocks properties to the
device trees this kernel cycle, and switch this code to only support the
"new_clocks" path next cycle.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tegra114's AHUB shares a design with Tegra30, with the followin changes:
* Supports more (10 vs. 4) bi-directional FIFO channels into RAM.
* Requires a separate block of registers to support the above.
* Supports more attached clients, i.e. new audio multiplexing and
de-multiplexing modules.
* Is affected by more clocks due to the above.
This change fully defines the device tree binding changes required to
represent these changes, and minimally extends the driver to support
the new hardware, without exposing any of the new FIFO channels.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since imx-sgtl5000 is only used on DT platform, it makes more sense to
use cpu_of_node rather than cpu_dai_name to match cpu dai.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This ioctl is defined as IOW, so pass the argument as const.
Signed-off-by: Hans Verkuil <hans.verkuil@cisco.com>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
The recent fix for the independent HP reduced the availability of the
side surround output, because there are only 4 DACs for 7.1 and a HP
outputs. Adjust the badness tables for VIA so that 7.1 outputs are
activated for the cost of missing independent HP.
Once when we implement the dynamic DAC switching to multiple outputs,
this conflicts will be eased in future...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The lack of independent HP mode shouldn't be too bad, but currently
its badness is set a bit too high. Let's lower it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The standard badness values don't seem to fit to all preferences.
Some configuration prefer the side output over the headphone, some
want the speaker over the surround, etc.
This patch moves the badness table pointers into hda_gen_spec, so that
the codec driver can override them.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge back for-linus branch for the badness table adjustment for VIA codecs
* for-linus:
ALSA: hda - Fix DAC assignment for independent HP
ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
ALSA: hda - Fix typo in checking IEC958 emphasis bit
ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
ALSA: snd-usb: mixer: propagate errors up the call chain
ALSA: usb: Parse UAC2 extension unit like for UAC1
ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
The dma-sh7760 currently fails with the following compile error:
sound/soc/sh/dma-sh7760.c:346:2: error: unknown field 'pcm_ops' specified in initializer
sound/soc/sh/dma-sh7760.c:346:2: warning: initialization from incompatible pointer type
sound/soc/sh/dma-sh7760.c:347:2: error: unknown field 'pcm_new' specified in initializer
sound/soc/sh/dma-sh7760.c:347:2: warning: initialization makes integer from pointer without a cast
sound/soc/sh/dma-sh7760.c:348:2: error: unknown field 'pcm_free' specified in initializer
sound/soc/sh/dma-sh7760.c:348:2: warning: initialization from incompatible pointer type
sound/soc/sh/dma-sh7760.c: In function 'sh7760_soc_platform_probe':
sound/soc/sh/dma-sh7760.c:353:2: warning: passing argument 2 of 'snd_soc_register_platform' from incompatible pointer type
include/sound/soc.h:368:5: note: expected 'struct snd_soc_platform_driver *' but argument is of type 'struct snd_soc_platform *'
This is due the misnaming of the snd_soc_platform_driver type name and 'ops'
field. The issue was introduced in commit f0fba2a("ASoC: multi-component - ASoC
Multi-Component Support").
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
Since ASoC does not yet really have the framework features needed to
support propagating sample rates through the device well yet implement
basic support for the ISRCs equivalent to that we currently have for the
ASRCs. The user can opt for 8kHz or 16kHz as the rate for the DSP blocks
in addition to the main audio rate, these being the primary use cases.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that the device state does not diverge from the state we have set
in the register map in order to make the behaviour clearer.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some system designs have been identified which repurpose portions of the
speaker driver circuits for other functions which will require that they
not be managed using DAPM. Prepare for this by factoring out the creation
of the speaker widgets into the core driver, the widgets will be replaced
by dummy ones when the additional functions are enabled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some systems use the audio CODEC to clock a DAI with multiple data lines
in parallel, meaning that bit clocks are only required for a smaller number
of channels than data is sent for. In some cases providing the extra bit
clocks can take the other devices on the audio bus out of spec.
Support such systems by allowing a maximum number of channels to be
specified.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The generic parser should evaluate the availability of the independent
HP when specified. Otherwise a DAC without the direct connection to
the corresponding pin may be assigned for the HP, but the driver
doesn't check it at all. The problem was actually seen on some
machines with VT1708s or equivalent codec, where DAC0 is assigned to
HP although it can be connected only via aamix.
This patch adds the badness evaluation for the independent HP to make
it working properly.
Reported-by: Lydia Wang <LydiaWang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have a "Headset Mic" name, let's use it for some devices
we know for sure has a headset mic jack.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headset mic jacks, i e TRRS style jacks with Headphone Left,
Headphone Right, Mic and GND signals, are becoming increasingly
common and are now being shipped by several manufacturers.
Unfortunately, the HDA specification does not give us any hint
of whether a Mic pin belongs to such a jack or not, but it would
still be helpful for the user to know (especially if there is one
TRS Mic jack and one TRRS headset jack).
This new fixup causes the first (non-dock, non-internal) mic to
be a headset mic jack. The algorithm can be later refined if needed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current DSP loader code abuses snd_hda_lock_devices() for ensuring
the DSP loader not conflicting with the other normal operations. But
this trick obviously doesn't work for the PM resume since the streams
are kept opened there where snd_hda_lock_devices() returns -EBUSY.
That means we need another lock mechanism instead of abuse.
This patch provides the new lock state to azx_dev. Theoretically it's
possible that the DSP loader conflicts with the stream that has been
already assigned for another PCM. If it's running, the DSP loader
should simply fail. If not -- it's the case for PM resume --, we
should assign this stream temporarily to the DSP loader, and take it
back to the PCM after finishing DSP loading. If the PCM is operated
during the DSP loading, it should get an error, too.
Reported-and-tested-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
