As the ssi is not the only cpu dai, there are esai, spdif, sai.
and imx_pcm_dma can be used by all of them. Especially ESAI need
a larger DMA buffer size. So Add dedicated DMA buffer for each cpu
dai.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Specifying unstable coefficients for the low/high pass filters can have
a severe impact on the audio. This patchs adds a stability check on the
coefficients written to the low/high pass filter block to prevent this.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Specifying unstable coefficients for the EQ can have a severe impact on
the audio. This patchs adds a stability check on the coefficients
written to the EQ, for this it is necessary to merge the mode control
and the coefficients as some coefficients may only be unstable with a
certain mode setting so it is ideal if these are always updated in sync.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is a similar patch to "move RT5650 muxes to rt5650_specific_
dapm_widgets" patch. The purpose is to silence the "has no paths"
warnings.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should use "HiFi Playback" and "HiFi Capture".it will fix below err
cht-bsw-max98090: ASoC: no sink widget found for AIF1 Playback
cht-bsw-max98090: ASoC: Failed to add route ssp2 Tx -> direct ->
AIF1 Playback
cht-bsw-max98090: ASoC: no source widget found for AIF1 Capture
cht-bsw-max98090: ASoC: Failed to add route AIF1 Capture -> direct ->
ssp2 Rx
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
dmi_check_system() takes "const struct dmi_system_id *", so make the
dmi_system_id table const.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make it obvious that unexpected value read from ID register is printed in
hexadecimal.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The DAPM lock must be held when accessing the DAPM graph status through
sysfs or debugfs, otherwise concurrent changes to the graph can result in
undefined behaviour.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
TLV size calculation was incorrectly calculated. Fix this according to
include/sound/tlv.h
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit 02aa78abec ("ASoC: DAPM: Add APIs to create individual DAPM
controls.") added locking to the snd_soc_dapm_new_control function but
did not update the call to snd_soc_dapm_new_control in the auto-disable
mux code, this appears to be because the patches were sent at fairly
similar times.
This patch change the call in the auto-disable mux code to use the new
snd_soc_dapm_new_control_unlocked function instead.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It was a busy development cycle at this time, as you can see a wide
range of changes in diffstat. There are no big changes but many
refactoring and improvements. Here we go some highlights:
* ALSA core:
- Procfs codes were cleaned up to use seq_file
- Procfs can be opt out via Kconfig (only for EXPERT)
- Two types of jack API were unified finally; now both kctl and input
jack devs are handled via a single function call.
* HD-audio
- Continued code restructuring for the future ASoC driver; now HDA
controller driver is split to a core helper module.
- Preliminary codes for Skylake audio support in HDA core.
- Proper i915 gfx power well management for SKL & co
- Enabled runtime PM as default for Intel HDMI/DP codecs
- Newer Tegra chip supports
- More quirks for Dell headsets, Alienware (with CA0132), etc.
- A couple of DRM ELD helper API functions
* ASoC
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring
- Big refactoring, cleanup and enhancement for the Wolfson ADSP driver
- Cleanup series for TI TAS2552 and R-CAR drivers
- Fixes and improvements on RT56xx codecs
- Support for TI TAS571x power amplifiers
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs
- Support for x86 systems with RT5650 and Qualcomm Storm
- Support for Mediatek AFE (Audio Front End) unit
- Other various small fixes to ASoC codec drivers
* Firewire
- Enhanced to allow non-blocking streams to use timestamp
synchronization
- Improve support for DM1500 and BeBoBv3
* Misc
- Cleanup of old pci API functions over all PCI sound drivers
- Fix long-standing regression of the old powermac i2c setup
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Merge tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"It was a busy development cycle at this time, as you can see a wide
range of changes in diffstat. There are no big changes but many
refactoring and improvements. Here we go some highlights:
ALSA core:
- Procfs codes were cleaned up to use seq_file
- Procfs can be opt out via Kconfig (only for EXPERT)
- Two types of jack API were unified finally; now both kctl and input
jack devs are handled via a single function call.
HD-audio:
- Continued code restructuring for the future ASoC driver; now HDA
controller driver is split to a core helper module.
- Preliminary codes for Skylake audio support in HDA core.
- Proper i915 gfx power well management for SKL & co
- Enabled runtime PM as default for Intel HDMI/DP codecs
- Newer Tegra chip supports
- More quirks for Dell headsets, Alienware (with CA0132), etc.
- A couple of DRM ELD helper API functions
ASoC:
- Support for loading ASoC topology maps from firmware, intended to
be used to allow self-describing DSP firmware images to be built
which can map controls added by the DSP to userspace without the
kernel needing to know about individual DSP firmwares
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring
- Big refactoring, cleanup and enhancement for the Wolfson ADSP
driver
- Cleanup series for TI TAS2552 and R-CAR drivers
- Fixes and improvements on RT56xx codecs
- Support for TI TAS571x power amplifiers
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs
- Support for x86 systems with RT5650 and Qualcomm Storm
- Support for Mediatek AFE (Audio Front End) unit
- Other various small fixes to ASoC codec drivers
Firewire:
- Enhanced to allow non-blocking streams to use timestamp
synchronization
- Improve support for DM1500 and BeBoBv3
Misc:
- Cleanup of old pci API functions over all PCI sound drivers
- Fix long-standing regression of the old powermac i2c setup"
* tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (533 commits)
ALSA: pcm: Fix pcm_class sysfs output
ALSA: hda-beep: Update authors dead email address
ASoC: wm_adsp: Move DSP Rate controls into the codec
ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case
ALSA: hda: provide default bus io ops extended hdac
ALSA: hda: add hda link cleanup routine
ALSA: hda: add hdac_ext stream creation and cleanup routines
ASoC: rsrc-card: remove unused ret
ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
ASoC: mediatek: Add machine driver for rt5650 rt5676 codec
ASoC: mediatek: Add machine driver for MAX98090 codec
ASoC: mediatek: Add AFE platform driver
ASoC: rsnd: remove io from rsnd_mod
ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working()
ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx()
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx()
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx()
ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr()
ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA
...
Pull clkdev updates from Russell King:
"This series addresses some breakage in clkdev caused by a previous
patch set from the clk tree which introduced per-user clk structures.
This basically renamed the existing 'struct clk' to 'struct clk_hw',
and introduced a new 'struct clk'.
This change will break anyone using clk_add_alias() with the common
clk code enabled. Thankfully, the intersection of users of
clk_add_alias() and those using the common clk code is practically
zero, but this is something which should be fixed to keep the code
sane.
The problem is that clk_add_alias() does this:
r = clk_get(...);
l = clkdev_alloc(r, ...);
clk_put(...);
which causes the alias to store a pointer to 'r', which has been
freed.
The original patch set tried to work around this problem incorrectly -
at clk_get() time, it tried to convert the struct clk to a struct
clk_hw, and then creating a new struct clk from that. Clearly, if the
original struct clk has been freed, then we have a use-after-free bug.
We have other places in the tree which do something similar, so this
series also addresses those locations too.
This series addresses this problem by converting clkdev to store and
use the clk_hw pointer. This allows clk_get() to only have to create
it's per-user struct clk from the clk_hw. We can also get to the
desired clk_hw at clk_add_alias() or clk lookup creation time, when
the struct clk is "alive".
We also perform some cleanups of the code:
- replacing looped calls to clkdev_add() with clkdev_add_table()
- replacing open-coded lookup allocation (which should have been
using clkdev_alloc()) and subsequent clkdev_add() with
clkdev_create()
- replacing open-coded clk_add_alias() with clk_add_alias()"
* 'for-linus-clk' of git://ftp.arm.linux.org.uk/~rmk/linux-arm:
clk: s2mps11: use clkdev_create()
ASoC: migor: use clkdev_create()
ARM: omap2: use clkdev_add_alias()
ARM: omap2: use clkdev_create()
ARM: orion: use clkdev_create()
ARM: lpc32xx: convert to use clkdev_add_table()
SH: use clkdev_add_table()
clkdev: add clkdev_create() helper
clkdev: const-ify connection id to clk_add_alias()
clkdev: get rid of redundant clk_add_alias() prototype in linux/clk.h
clkdev: drop __init from clkdev_add_table()
clk: update clk API documentation to clarify clk_round_rate()
clkdev: use clk_hw internally
This adds call to runtime suspend in dev remove. It fixs the problem that
suspend is not called in the case of CONFIG_PM=n. It also fixs build
warning when CONFIG_PM=n.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
These were sitting on my laptop.
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Merge tag 'asoc-v4.2-3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: A couple more updates for v4.2
These were sitting on my laptop.
There's a bunch of additional updates and fixes that came in since my
orignal pull request here, including DT support for rt5645 and fairly
large serieses of cleanups and improvements to tas2552 and rcar.
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Merge tag 'asoc-v4.2-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Further updates for v4.2
There's a bunch of additional updates and fixes that came in since my
orignal pull request here, including DT support for rt5645 and fairly
large serieses of cleanups and improvements to tas2552 and rcar.
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
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Merge tag 'asoc-v4.2' into asoc-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
# gpg: Signature made Mon 08 Jun 2015 18:48:37 BST using RSA key ID 5D5487D0
# gpg: Oops: keyid_from_fingerprint: no pubkey
# 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>"
The rate controls are codec-specific, it's not possible to
generically say what the range or the meaning of each control
is (or even if they exist at all) - that depends on the
particular codec.
This is currently being handled for Arizona codecs by putting
an Arizona-specific table of controls inside the wm_adsp driver.
This creates a dependency between wm_adsp and arizona.c, and is an
awkward solution if the ADSP is used in another family of codecs
Fix this by moving the Arizona-specific rate controls into the
Arizona codec drivers.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix trivial typo.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
'047000278da3a17f8("ASoC: rsrc-card: cleanup for DPCM")'
cleanuped rsrc-card driver, but then, unused ret was left.
Below warning happen without this patch
${LINUX}/sound/soc/sh/rcar/rsrc-card.c: In function 'rsrc_card_startup':
${LINUX}/sound/soc/sh/rcar/rsrc-card.c:78:6: warning: unused variable \
'ret' [-Wunused-variable]
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the DPCM based machine driver with rt5650 and rt5676
Signed-off-by: Nicolas Boichat <drinkcat@chromium.org>
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is the DPCM based platform driver of AFE (Audio Front End) unit.
Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many path
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. rsnd_mod_to_io() is no longer needed. Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship.
This patch checks module working status via io instead of mod
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from snd_kcontrol
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_src_xxx()
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_ssi_xxx()
and related function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_dma_xxx()
and related function
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This patch removes rsnd_mod_to_io() from rsnd_get_adinr()
and its related function
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. Then, interrupt handler can't use rsnd_mod_to_io().
This patch adds SSI/SRC/DMA common interrupt handler frame
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This means we can't call rsnd_mod_to_io() any more.
This patch adds struct rsnd_dai_stream to each function as parameter.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. In such case, mod <-> io is no longer 1:1
relationship. This means we can't use rsnd_mod_to_io() in SSI/SRC/DMA
interrupt handler. In such case, we need to check all io in interrupt
handler, and then, "priv" is needed.
This patch adds rsnd_priv pointer in rsnd_mod for prepare it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths
if it supports MIXer. Then, we don't need to re-call each mod function
that had been called. This patch count each mod status.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsrc-card driver is based on simple-card driver which is caring about
CPU / Codec connection. OTOH, rsrc-card is used for DPCM system.
FE portion is constituted by CPU and dummy Codec, and BE is constituted
by dummy CPU and Codec in DPCM system.
Because of this, current rsrc-card is doing pointless method. It works well
if FE/BE was 1:1, but not good for multi FE/BE.
This patch cleanups rsrc-card driver for DPCM. and this is prepare for
MIX support for Renesas sound driver.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is prepare for DPCM cleanup
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rsrc-card is assuming 1 FE (= CPU), 1 BE (= codec) on card.
But, it will support multi FE/BE card. This is prepare for it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current dai_link name is using "cpu_dai_name + codec_dai_name",
but one of them is always "snd-soc-dummy-dai" when DPCM.
This patch uses "fe.xxx" for cpu, "be.xxx" for codec.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
'a9e1ac1a9e4585b5("ASoC: rsnd: spin lock for interrupt handler")'
added spin lock under interrupt handler to solve HW restart issue.
OTOH, current rsnd driver calls snd_pcm_period_elapsed() from
rsnd_dai_pointer_update(). but, it will be called under spin lock
if SSI was PIO mode.
If it was called under spin lock, it will call
snd_pcm_update_state() -> snd_pcm_drain_done().
Then, it calls rsnd_soc_dai_trigger() and will be dead-lock.
This patch doesn't call rsnd_dai_pointer_update() under spin lock
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PIO is used only for checking data path / codec settings. And underrun
is very normal when PIO mode. Let's don't care about under/over run
error when PIO case. Otherwise, 1) too many HW restart happens, 2) some
sounds which need much data transfer can't play since it falls into
error detection method which was created for DMA transfer
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Developing a driver for an Asus X205TA laptop I get these dmesg
errors:
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC1 Swap Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC2 Swap Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC3 Swap Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 ADC Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 L Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC1 R Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 L Mux has no paths
rt5645 i2c-10EC5648:00: ASoC: mux RT5650 IF1 DAC2 R Mux has no paths
so, move these muxes to the rt5650_specific_dapm_widgets[] list.
Signed-off-by: Michele Curti <michele.curti@gmail.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Constify the ACPI device ID array, it doesn't need to be writable at
runtime.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime. Also drop the unneeded RT5670_INIT_REG_LEN
define.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Constify the ACPI device ID array and the register map, no need to have
them writable at runtime. Also drop the unneeded RT5640_INIT_REG_LEN
define.
Signed-off-by: Mathias Krause <minipli@googlemail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current code has duplicate code for 16000, 32000 and 48000 sample rates.
get_srate() returns negative error code for unsupported rate, so we can
remove the duplicate code in the swith cases by calling get_srate() first.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In order to make TI button interrupt working max98090 codec
Need provide mic bias all the time as long as mic is present
so SHDN and micbias pin are forced on.we also need set max98090
codec bias close or lower than TI bias.We set them in bios/coreboot
kernel reads them from device property
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some tiny improvements, cutting 180 bytes off the generated code.
- use strchr() for single-character needle
- compute index using pointer subtraction instead of two strlen()
calls
- factor out the common check for whether the initial part of
kctl->id.name (before the space) is identical to w->name.
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When running in master mode the bclk divider must be configured to generate a
sane bitclock. Pick the smallest fs multiplicator, which can hold all
transmitted bits.
Signed-off-by: Julian Scheel <julian@jusst.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the enum of "DAC Polarity" should be wm8960_enum[1].
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Add mclk-fs ratio property per dai-link sub node. This will
allow to manage several codecs with different ratio.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fixed below error/warnings
sound/built-in.o: In function `rt5645_i2c_probe':
>> rt5645.c:(.text+0xe38f5): undefined reference to
>> `devm_regmap_init_i2c'
sound/built-in.o: In function `rt5645_i2c_driver_exit':
>> rt5645.c:(.exit.text+0x60e): undefined reference to `i2c_del_driver'
sound/built-in.o: In function `rt5645_i2c_driver_init':
>> rt5645.c:(.init.text+0x1a90): undefined reference to
>> `i2c_register_driver'
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since hp-detect is optional, use devm_gpiod_get_optional instead.
In additional, it should return error if devm_gpiod_get_optional fails.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This was spotted by Fernando Lopez-Lezcano <nando@ccrma.Stanford.EDU>
while he tried to compile a -RT kernel with this driver enabled.
"make C=2" would also warn about this. This is is based on his patch.
Reported-by: Fernando Lopez-Lezcano <nando@ccrma.Stanford.EDU>
Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It is currently possible to have CONFIG_SND_ATMEL_SOC_SSC=y with either
CONFIG_SND_ATMEL_SOC_PDC=m or CONFIG_SND_ATMEL_SOC_DMA=m. This results in a
driver that compiles but does not link with this kind of error:
sound/built-in.o: In function `atmel_ssc_set_audio':
(.text+0x87d90): undefined reference to `atmel_pcm_pdc_platform_register'
sound/built-in.o: In function `atmel_ssc_put_audio':
(.text+0x8879a): undefined reference to `atmel_pcm_pdc_platform_unregister'
Introduce new config options SND_ATMEL_SOC_SSC_PDC and
SND_ATMEL_SOC_SSC_DMA which should be used by the board drivers and the
correct logic to properly select the SND_ATMEL_SOC_PDC and
SND_ATMEL_SOC_DMA states.
Reported-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
After discussing with the Kconfig maintainer, we found a better fiw
allowing to keep each driver as modules.
This reverts commit 0ef9dc139d.
Signed-off-by: Alexandre Belloni <alexandre.belloni@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Modify the RT5645 driver to parse platform data from device tree. This is
missing from previous patch in sound/soc/codecs/rt5645.c, that was present
in v3.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch corrected pcm_delay calculation in BSW sst driver
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch separates the IO function from the rt286. It is prepared to share
for new chips that support the same IO function.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit ea178d1456 ("ASoC: tas2552: Make the enable-gpio really optional")
makes enable-gpio optional. devm_gpiod_get_optional() is the better
function for optional gpio, so let's switch to use it.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds some debugfs nodes to get information
about the currently running firmware.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now that we have a codec_probe stage initialization in the wm_adsp
driver, we can make the wm_adsp driver create its own ALSA controls
instead of having that responsibility pushed to every codec driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the only init function in wm_adsp is called by the
codec driver early in its probe before the codec has been
registered with SOC.
This patch adds stubs for the codec_probe and codec_remove stages
and calls them from WM5102 and WM5110 codec drivers. This allows us
to hang anything that needs setup during the codec probe stage off
these functions without further modification of the codec drivers.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
My static checker complains that:
sound/soc/fsl/imx-wm8962.c:196 imx_wm8962_probe() warn:
we tested 'ret' before and it was 'false'
The intent was that we use "ret" to check imx_audmux_v2_configure_port().
Fixes: 8de2ae2a7f ('ASoC: fsl: add imx-wm8962 machine driver')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Otherwise, Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Move gpio to gpio_desc and use gpiod APIs in codec driver.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
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Merge tag 'asoc-v4.2' into asoc-rt5645
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
Register RT5650_TDM_CTRL_4(0x7A) is readable and used for mixer
setting. It should be added in rt5645_readable_register function.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rt5645_enable_push_button_irq uses snd_soc_dapm_*_unlocked
functions, so it needs to lock the required dapm mutex.
Signed-off-by: Nicolas Boichat <drinkcat@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds apq8016 machine driver support. This patch is tested on
DB410c and msm8916-mtp board for both hdmi and analog audio
features.
Acked-by: Kenneth Westfield <kwestfie@codeaurora.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Find the configured DMA controller by asking for a DMA channel in the
probe phase and releasing it right after. The controller device can be
found via the dma_chan struct and the controller is recognized from
the compatible property of its device node. The patch assumes EDMA if
there is no device node.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
According to the schematics the external speaker is differential, so the
HPLCOM and HPRCOM pins of the CODEC should be connected to it.
Otherwise the routing looks complete, so add the missing routes and set the
fully_routed flag of the card instead of manually marking the unused inputs
and outputs as not connected. This makes the code a bit shorter and
cleaner.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes special casing the EPROBE_DEFER error handling in the
driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch moves static allocation of snd_soc_card to dynamic allocation,
the reason to do this is to avoid holding up any dangling pointers
in this static structures. And I see no use for having this struct as static
given that the card->name is also populated dynamically from dt.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
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Merge tag 'asoc-v4.2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- Support for loading ASoC topology maps from firmware, intended to be
used to allow self-describing DSP firmware images to be built which
can map controls added by the DSP to userspace without the kernel
needing to know about individual DSP firmwares.
- Lots of refactoring to avoid direct access to snd_soc_codec where
it's not needed supporting future refactoring.
- Big refactoring and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
The set of supported sample rates depends on the master clock supplied
to the codec. Allow the machine driver to set the required master clock
in hw_params().
Signed-off-by: Sergej Sawazki <ce3a@gmx.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In tas2552_sw_shutdown() tas_data is used while the rest of the driver uses
tas2552 when dealing with the 'struct tas2552_data'
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Double semicolon was added by the following commit:
ea178d1456 ASoC: tas2552: Make the enable-gpio really optional
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The module can not be loaded again after it has been removed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Align the numbers in the header file to the same column.
At the same time change the wrapping of CFG_2 register write in the probe
function to be uniform with the other calls.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Correct the bit definition so the code will change the bits what it
supposed to change. Also rename the register define to
TAS2552_BOOST_APT_CTRL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Do not write to DOUT Tristate register at probe time, specially not write
data which is defined to be used in Output Data Register.
Fix the defines for the Output Data Register and correct the register write
at probe time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
'DIN source' enum can be used to select the DIN Source (muted, left, right
or average of left and right channels).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Do not restrict the sampling rate to 44.1/48KHz. The pll_clk clock should
be (sampling rate * 512) in all cases.
Correct the J.D calculation (the D part was incorrectly calculated).
Restore PLL enable status after we are done with the configuration.
Implement hardware constraint handling towards the pll_clkin:
if D != 0 (in J.D) then 1.1MHz <= pll_clkin <= 9.2MHz needs to be checked.
If the PLL setup does not met with this constraint, fall back to BCLK as
reference clock, if BCLK fails, use the internal 1.8MHz clock.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The default state when serializers are in inactive slots is Hi-Z.
In some cases, there are no additional components driving the data
lines to a safe state so they might have noise.
While in inactive slots, the McASP AXR pins configured as outputs
can be driven low through the serializer pin drive mode setting
(DISMOD) to prevent such noise.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of hard wiring the WCLK frequency at probe time do it runtime.
The hard wired 88_96KHz was not even setting the correct bits since it was
defined as (1 << 6) which will change the I2S_OUT_SEL bit and will leave
the amplifier configured for 8KHz.
At the same time clean up and fix the CFG3 register bits.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Configure the word length based on the params_width of the stream.
Also configure the clock per frame value which is used when tas2552 is bus
master.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Certain sequence need to be followed in order to have smooth power up and
power down performance.
Execute this sequence via DAPM_POST widget.
Remove patching the RESERVED_0D register at probe time since it has to be
handled every time when we stop or start the amplifier.
In order to be able to execute the sequence at the correct time, the driver
need to request to ignore the pmdown time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The last parameter for DECLARE_TLV_DB_SCALE() is to tell if the gain will
be muted or not when it is set to raw 0. IN this case it is not muted.
The PGA_GAIN is in 0-4 bits in the register. Fix the offset in the
SOC_SINGLE_TLV() for this.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The strings should be: 'static const char * const tas2552_input_texts[]'
SOC_DAPM_ENUM should have "Route" in place of xname and no need to have it
as an array.
Also align the parameters.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
TDM support is achieved using DSP transfer mode and setting a programmable
offset which specifies where data begins with respect to the frame sync.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use names from the datasheet for the definitions.
Correct the data format definitions since they were not correct.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DSP_A mode require one bit delay from the FS, DSP_B is without data delay.
When checking the requested format, also match the bit and fs inversion
flag along with the format since it is not possible to change inversion.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of hard wiring the PLL_CLKIN and PDM_CLK to be sourced from BCLK add
proper clock configuration via the set_dai_sysclk callback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Prevents frequent panic on boot, if the irq handler rt5645_irq
gets called before the workqueue rt5645_jack_detect_work is
initialized.
Signed-off-by: Nicolas Boichat <drinkcat@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
sound/soc/zte/zx296702-i2s.c:428:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
MCLK is one of the possible source for the pll_clkin frequency. Make this
clear by renaming the variable.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The function name and parameters of:
tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown)
implies that if sw_shutdown is 1 we should be entering to the software
shutdown mode.
The code can be simplified as well within the function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Initialize the cfg1_reg to 0 and set the mute bit only when it is needed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the _MASK postfix of the bit definitions, collect the CFG1 bit
definition in one place and correct the bit shifts at the same time.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The PDM clock can be selected via bit0-1.
PDM_DATA_ES bit is at bit2.
The code were trying to select BCLK as PDM reference clock but instead
it was selecting PLL and set the DATA_ES bit to 1.
Selecting the PLL output as reference clock as default does make sense,
but the driver should not change the PDM data edge.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SOC_DAPM_SINGLE("Playback AMP", ..) should not be under kcontrols. It
causes kernel crash (NULL pointer) when the mixers are listed.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
If the card is not part of any card the tas_data->codec is NULL since it is
set only during snd_soc_codec_driver.probe, which is not yet called.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Do not fail the probe if the enable-gpio is not specifiedbut handle
deferred probe case.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/zte/zx296702-spdif.c:191:1-4: WARNING: end returns can be simpified
Simplify a trivial if-return sequence. Possibly combine with a
preceding function call.
Generated by: scripts/coccinelle/misc/simple_return.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/zte/zx296702-spdif.c:361:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the following error:-
All error/warnings (new ones prefixed by >>):
>
> sound/built-in.o: In function `soc_tplg_dapm_widget_create':
> >> :(.text+0x25a90): undefined reference to `snd_soc_dapm_new_control'
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The topology core parses the FW topology file for known block types and
instanciates any common ALSA/ASoC objects that it discovers. The core
also passes any block that is does not understand to client component
drivers for enumeration.
The core exports some APIs to client drivers in order to load and unload
firmware topology data as use case require.
Currently the core deals with the following object types :-
o kcontrols. This includes TLV, enumerated and bytes controls.
o DAPM widgets. All types with any associated kcontrol.
o DAPM graph.
o FE PCM. FE PCM capabilities and configuration can be defined.
o BE DAI Link. BE DAI link capabilities and configuration can be defined.
o Codec <-> codec style links capabilities and configuration.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DM_INH = 1 (stereo downmix prohibited) and CA = 0x00 (Channel
Allocation: FR, FL) is an invalid combination according to the
HDMI Compliance Test 7.31 "Audio InfoFrame".
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ti.com>
Acked-by: Mark Brown <broonie@kernel.org>
There is a constraint in the OMAP4 HDMI IP that requires to use
the 8-channel code when transmitting more than two channels.
The constraint doesn't apply for OMAP5 so don't force the channel
allocation in the sound driver as it can be done specifically for
OMAP4 later in the hdmi4 core.
Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Tomi Valkeinen <tomi.valkeinen@ti.com>
Acked-by: Mark Brown <broonie@kernel.org>
If an ASoC component device does not have a device tree node, use its
parent's node instead, when looking for a matching DAI based on a
device tree reference.
This allows video device drivers to register a separate child device
for their ASoC side audio functionality. [And MFDs in general --
broonie]
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The WM8997 and WM5102 codecs need to boost DVFS for higher sample rates.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In theory the ADSP driver should not need to know anything
about the codec it is part of. But the WM5102 needs DVFS
control based on ADSP clocking speed. This was being handled
by bundling part of the knowledge of this into the ADSP driver.
This change moves this handling out of the ADSP driver and
into the WM5102 driver.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The WM5102 and WM8997 codecs have an internal dynamic clock booster.
When this booster is active, the DCVDD voltage must be increased.
If all the currently active audio paths can run with the root SYSCLK
we can disable the booster, allowing us to turn down DCVDD voltage
to save power.
Previously this was being done by having the booster enable bit set
as a side-effect of the LDO1 regulator driver, which is unexpected
behaviour of a regulator and not compatible with using an external
regulator. [Originally this was documented as a feature of the internal
LDO -- broonie]
This patch exports functions to handle the booster enable and
DCVDD voltage, with each relevant subsystem flagging whether it can
currently run without the booster. Note that these subsystems are
stateless and none of them are nestable, so there's no need for
reference counting, we only need a simple boolean for each subsystem
of whether their current condition could require the booster or will
allow us to turn the codec down to lower operating power.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch is a fixup to correct dependencies in patch 9bae4880ac
("ASoC: qcom: move ipq806x specific bits out of lpass driver.")
Originally this change-set was suggested by Arnd on mailing list.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The change fixes the following compilation problem:
sound/soc/soc-dapm.c: In function 'dapm_kcontrol_data_alloc':
sound/soc/soc-dapm.c:388:4: error: implicit declaration of function
'snd_soc_dapm_new_control' [-Werror=implicit-function-declaration]
data->widget = snd_soc_dapm_new_control(widget->dapm,
^
sound/soc/soc-dapm.c:387:17: warning: assignment makes pointer
from integer without a cast [enabled by default]
data->widget = snd_soc_dapm_new_control(widget->dapm,
^
sound/soc/soc-dapm.c: At top level:
sound/soc/soc-dapm.c:3269:1: error: conflicting types for
'snd_soc_dapm_new_control'
snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
^
In addition to the fix add static qualifier to
snd_soc_dapm_new_control() function to silence checkpatch.
Fixes: 02aa78abec ("ASoC: DAPM: Add APIs to create individual DAPM controls.")
Signed-off-by: Vladimir Zapolskiy <vz@mleia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
replace of_property_read_u32 with device_property_read_u32
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It needs free pcm runtime modules before unloading firmware, here
add hsw_pcm_suspend() to handle this procedure:
suspends firmware ==> frees runtime modules ==> unloads firmware.
This fixes the broadwell module unload failed issue.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add check runtime module pointers before freeing them, and clear
them to NULL after freed.
With this implemented, we can avoid NULL pointer dereference or
double free errors.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 01f202c7b4.
We shouldn't leave the device as suspended state after module freed,
it is not good to do runtime suspend at driver free, here revert
this fixing, and replace it with the procedure:
suspends firmware ==> frees runtime modules ==> unloads firmware.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 506c148ee5.
We still need this hsw_pcm_free_modules(), we plan to remove the
runtime modules at both fw_unload(D0->D3) and snd_soc_sst_haswell_pcm
module removing.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We don't need compress offload feature for broadwell broadwell machine,
here remove the non exist dependency.
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Also drop the unnecessary comparison in the set_bias_level() callback that
checks if the device is already at the target level. The core already takes
care of this and will not call the callback if the device is already at the
target level.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level() and replace all
other manual access to codec->dapm with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to codec->dapm
with snd_soc_codec_get_dapm().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The dapm field of the snd_soc_codec struct is eventually going to be
removed, in preparation for this replace all manual access to
codec->dapm.bias_level with snd_soc_codec_get_bias_level().
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
