It turns out the devices from Playback Design need the delay quirk
after usb_set_interface from clocks.c as well. Make it a proper
quirks function and factor out the code to quirks.c.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Charge Pump needs the DSP clock to work properly, without it the
bypass to HP/LINEOUT is not working properly. This requirement is not
mentioned in the datasheet but has been confirmed by Mark Brown from
Wolfson.
Signed-off-by: Alban Bedel <alban.bedel@avionic-design.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
There's already a device revision stored in the core data structure,
don't duplicate it in the CODEC driver.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
For playback add the codec-side delay to the timestamp, for capture
subtract it. This brings the timestamps in line with the time that
was recently added to the delay reporting.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A test for CONFIG_SND_SOC_UX500_AB5500 was added in v3.5. But there
never was a corresponding Kconfig symbol so this test has always
evaluated to true. And since AB5500 support was removed in v3.5 it
appears safe to remove this test and a few lines of code.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The snd_pcm_hardware structs for playback and capture in the ux500 PCM are
identical, so remove one of them and use the same snd_pcm_hardware struct for
both playback and capture. Also move the defines used to initialize the
snd_pcm_hardware fields from ux500_pcm.h to ux500_pcm.c since that's the only
place where they are used.
Also drop the assignment of the snd_pcm_hardware struct to runtime->hw since
that is what the call to snd_soc_set_runtime_hwparams() right above it already
does, so the second assignment is redundant.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The usb_control_msg() function expects __u16 types and performs
the endianness conversions by itself.
However, in three places, a conversion is performed before it is
handed over to usb_control_msg(), which leads to a double conversion
(= no conversion):
* snd_usb_nativeinstruments_boot_quirk()
* snd_nativeinstruments_control_get()
* snd_nativeinstruments_control_put()
Caught by sparse:
sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:512:38: expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:512:38: got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types)
sound/usb/mixer_quirks.c:543:35: expected unsigned short [unsigned] [usertype] value
sound/usb/mixer_quirks.c:543:35: got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:543:56: expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:543:56: got restricted __le16 [usertype] <noident>
sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types)
sound/usb/quirks.c:502:35: expected unsigned short [unsigned] [usertype] value
sound/usb/quirks.c:502:35: got restricted __le16 [usertype] <noident>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct pin configs for the Acer AC700. Most importantly indicate
that SPDIF is connected, it routes to HDMI out.
Similar to Aspire models, chain in the DMIC fixup and allow it to be
applied to this codec (ALC269VB) as well.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Channel size settings will be made at the end of
davinci_mcasp_hw_params() routine and thus overwrite frame
format settings made for DIT mode. This patch fixes this issue
by taking op_mode into account. Tested with official PSP 3.2
kernel and sii9022a HDMI transmitter.
Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com>
Tested-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AFSX won't be used in DIT mode. The related pins are AHCLKX and
the data pins.
Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com>
Acked-by: Vaibhav Bedia <vaibhav.bedia@ti.com>
Tested-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 453807f3 ("ASoC: ep93xx: Use ep93xx_dma_params instead of
ep93xx_pcm_dma_params") introduced a small compile error by not updating the
name of the 'dma_port' field to 'port'. This patch fixes it.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the common DAI DMA data struct for fsl/imx, this allows us to use the common
helper function to configure the DMA slave config based on the DAI DMA data.
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 DSP in the CA0132 codec adds a variable latency to audio depending
on what processing is being done. Add a new patch op to return that
latency for capture and playback streams. The latency is determined
by which blocks are enabled and knowing how much latency is added by
each block.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a new codec PCM ops, get_delay(), to obtain the codec/stream-
specific PCM delay count. When it's NULL, nothing changes.
This new feature was requested for CA0132, which has significant
delays in the path depending on the running DSP code.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
changed is not initialized in path_power_down_sync, but it is expected
to be false in case no change happened in the loop. So set it to
false.
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This reverts commit 6ab317419c.
The commit [6ab317419c: ALSA: hda - Allow power_save_controller option
override DCAPS] changed the behavior of power_save_controller so that
it can override the driver capability. This assumed that this option
is rarely changed dynamically unlike power_save option. Too naive.
It turned out that the user-space power-management tool tries to set
power_save_controller option to 1 together with power_save option
without knowing what's actually doing. This enabled forcibly the
runtime PM of the controller, which is known to be broken om many
chips thus disabled as default.
So, the only sane fix is to revert this commit again. It was intended
to ease debugging/testing for runtime PM enablement, but obviously we
need another way for it.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171
Reported-and-tested-by: Nikita Tsukanov <keks9n@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rename "Digitial In" to "Digital In". This function is only used for
proc output, so should not cause any problems to change.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Added the device ID to the modalias list and assinged ALC662 patches
for it
* Added 4 port support for the device ID 0671 in alc662_parse_auto_config
Signed-off-by: Rainer Koenig <Rainer.Koenig@ts.fujitsu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some clocks might be read-only, e.g., external clocks (see also
UAC2 4.7.2.1).
In this case, setting the sample frequency will always fail
(even if the rate is equal to the current clock rate),
therefore do not write, but read the value and compare to the
requested rate.
If the clock is read only, avoid reading it twice.
If it doesn't match, return -ENXIO since the clock is invalid for
this configuration.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Show the error code returned from the USB subsystem in
the debug messages.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a module param to disable auto clock selection.
This is provided for users that expect the audio stream to
fail when the clock source is invalid (e.g., the word clock
was unintentionally disconnected).
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If a selector is available on a device, it may be pointing to a
clock source which is currently invalid.
If there is a valid clock source which can be selected, switch
to it.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the check that parse_audio_format_rates_v2() do after
receiving the clock source entity ID directly into the find
function and add a validation flag to the function.
This patch does not introduce any logic flow change.
It is provided to allow introducing automatic clock switching
easier later. By moving this uac_clock_source_is_valid callsite,
2 additional callsites can be avoided.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace the endianness conversions with the kernel-wide swabbing macros
in get/set_sample_rate_v2.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Correct spelling of snd_usb_endpoint_implict_feedback_sink in all
occurances.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Minor style fix, following a general code style in the kernel.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge branch 'fix/samsung' of
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into
asoc-component to resolve trivial conflict
Conflicts:
sound/soc/samsung/i2s.c
Use the common DAI DMA data struct for tegra, this allows us to use the common
helper function to configure the DMA slave config based on the DAI DMA data.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the common DAI DMA data struct for omap, this allows us to use the common
helper function to configure the DMA slave config based on the DAI DMA data.
For omap-dmic and omap-mcpdm also move the DMA data from a global variable to
the driver state struct.
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>
This patch adds a common DMA data struct which can be used by DAI drivers to
communicate their DMA configuration requirements to the DMA pcm driver. Having
a common data structure for this allows us to implement common functions on top
of them, which can be used by multiple platforms.
This patch also introduces a new function to initialize certain fields of a
dma_slave_config struct from the common DAI DMA data struct.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Usually device_fc should be set to false for audio DMAs. Initialize it in a
common place so drivers don't have to do this manually.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The tegra dmaengine driver does not support pausing and resuming a DMA stream.
The tegra PCM driver still claims to support pause and resume though and
implements them by stopping and restarting the stream. This is not what an
application using pause/resume would expect. Usually applications have support
for working around PCMs which do not support suspend and resume, so don't set
the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME flags for the tegra PCM and
use the default snd_dmaengine_pcm_trigger callback.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Tested-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Just for cleaning up, introduce a new function get_sample_rate_v2()
for replacing two identical calls in set_sample_rate_v2().
No functional change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, a new platform device is created for secondary device
by calling platform_device_register_resndata and then the drvdata
is set for this device.
The following patch has been added to driver core:
"driver core: fix possible missing of device probe".
This results in the added device getting probed immediately but
the drvdata for the secondary device is not yet set.
This patch removes the platform_device_register_resndata call and
instead calls platform_device_alloc, platform_set_drvdata and
platform_device_add which fixes the above issue.
Signed-off-by: Prathyush K <prathyush.k@samsung.com>
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes a possible crash in case drvdata for the secondary
device is not set.
Signed-off-by: Prathyush K <prathyush.k@samsung.com>
Signed-off-by: Padmavathi Venna <padma.v@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The C-Media CM6631 USB receiver doesn't respond to changes in sample rate
while the interface is active. The same behavior is observed in other UAC2
hardware like the VIA VT1731.
Reset the interface after setting the sampling frequency on sample rate
changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is
used. Otherwise, the device will try to use the sample rate of the previous
stream, causing distorted sound on sample rate changes.
The reset is performed for all UAC2 devices, as it should not affect a
standards compliant device, but it is only necessary for C-Media CM6631,
VIA VT1731 and possibly others.
Failure to read sample rate from the device is not handled as an error in
set_sample_rate_v2(), as (permanent or intermittent) failure to read sample
rate isn't essential for a successful sample rate set.
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch let ELD debug message show 'pin_eld->monitor_present' which reflects
the real pin response to verb GET_PIN_SENSE.
'eld->monitor_present' should not be used here because 'eld' is a temp
structure now and so its "monitor_present" is not set.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0.
Otherwise it will be returned uninitialized as non-zero after ELD info is got
successfully. Thus hdmi_present_sense() will always assume ELD info is invalid
by mistake, and /proc file system cannot show the proper ELD info.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Cc: stable@vger.kernel.org
Acked-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Turing on the headphone amp interferes with the impedance measurement
used to detect a TRRS style headset microphone. Delay the HP turn on
until 500ms after the jack is detected, allowing the mic detection
state machine to run to completion.
Signed-off-by: Chih-Chung Chang <chihchung@chromium.org>
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Used kmemdup instead of replicating it's behaviour with kmalloc followed
by memcpy.
Patch found using coccinelle.
Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Used kmemdup instead of replicating it's behaviour with kmalloc followed
by memcpy.
Patch found using coccinelle.
Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Kconfig symbol SND_SOC_OF_SIMPLE got removed in commit
f0fba2ad1b ("ASoC: multi-component - ASoC
Multi-Component Support"). But that commit missed one instance. Remove
it now, together with the prompt it has effectively hidden ever since.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When clearing the walked flags there is no need to clear all paths, we
only need to clear the paths we actually walked. This means we can split
dapm_clear_walk() into input and output versions and rather than going
through all DAPM paths we can recurse down the path until we encounter
paths we have not yet walked.
This reduces the number of operations we need to perform and improves
cache locality.
[Pulled out of the vendor tree that the patch was originally generated
for by me, any bugs were introduced in that process -- broonie]
Signed-off-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
We already clear the walked state in dapm_widget_power_check(), no need
to do it again.
Signed-off-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is the equivalent of commit af46800 "ASoC: Implement mux control
sharing", but applied to mixers instead of muxes.
This allows a single control to affect multiple mixer widgets at once,
which is useful when there is a single set of register bits that affects
multiple mixers in HW, for example both the L and R mixers of a stereo
path.
Without this, you either:
1) End up with multiple controls that affect the same register bits, but
whose DAPM state falls out of sync with HW, since the DAPM state is only
updated for the specific control that is modified, and not for other
paths that are affected by the register bit(s).
2) False paths through DAPM, since you end up merging unconnected stereo
paths together into a single widget which hosts the single control, and
then branching back out again, thus conjoining the enable states of the
two input paths.
Now that the kcontrol creation logic is split out into a separate
function, dapm_create_or_share_mixmux_kcontrol(), also use that to
replace most of the body of dapm_new_mux(). This should produce no
functional change, but simply eliminates some mostly duplicated code.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
while at it, update the copyright timeline too
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we have regular register mapped controls we should be splitting
the control sets for ADSP1 and ADSP2 as the register maps are not
identical. Do that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When a new stream is being opened it is necessary to cancel any delayed
power down of the audio.
[Fixed unused variable -- broonie]
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The ASoC core does no not modify the driver of a platform. Making it const
allows ASoC platform drivers to declare the snd_soc_platform_driver struct as
const.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this, modules will fail to link against those symbols.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
siu_dai.c is using snd_soc_register_dais(),
even though array size of siu_i2s_dai is 1.
OTOH, new API snd_soc_register_component() uses properly
snd_soc_register_dai() (henceforth dai()) or
snd_soc_register_dais() (henceforth dais()) via num_dai.
Then, cpu_dai_name will be "siu-i2s-dai" if dais() was used,
and it will be "siu-pcm-audio" if dai() was used.
Therefore this patch fixup migor_dai :: cpu_dai_name too.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All drivers are using snd_soc_register_component()
instead of snd_soc_register_dai[s]()
snd_soc_[un]register_dai[s]() are no longer needed
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_soc_register_dai() uses fmt_single_name(), and
snd_soc_register_dais() uses fmt_multiple_name()
for dai->name which is used for name based matching.
This patch uses properly snd_soc_register_dai() it it was single driver,
and uses snd_register_dais() if it were multiple drivers.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Compress core added metadata apis in 9727b4, so add same in ASoC
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
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Merge tag 'arizona-extcon-asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/misc into asoc-arizona
ASoC/extcon: arizona: Fix interaction between HPDET and headphone outputs
This patch series covers both ASoC and extcon subsystems and fixes an
interaction between the HPDET function and the headphone outputs - we
really shouldn't run HPDET while the headphone is active. The first
patch is a refactoring to make the extcon side easier.
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>
The driver never uses snd_dmaengine_pcm_get_data(), so there is no need to use
snd_dmaengine_pcm_set_data().
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 driver never uses snd_dmaengine_pcm_get_data(), so there is no need to use
snd_dmaengine_pcm_set_data().
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>
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>
Used PTR_RET function instead of IS_ERR and PTR_ERR.
Patch found using coccinelle.
Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.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>
It can be 0 or 1 return value of snd_soc_update_bits_locked() when it is
success. So just check return value is negative.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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>
