Hi,
Changes since v3:
- Fix the single clock source handling and typo
Changes since v2:
- DT binding:
- use proper (?) patch subject for the binding docuemtn patch
- drop pll4 and pll15 from DT - driver should check the rate via
clk_get_parent. If it is not available (as it is not currently) then use the
match_data provided rates.
- add simple explanation for the clocking setup
- Use descriptive names for clocks: cpb/ivi-mcasp-auxclk and cpb/ivi-codec-scki
- dt_binding_check shows no errors/warnings
- ASoC machine driver:
- Try to read the PLL4/15 rate with clk API (parent of the two clock divider)
if it is not available then use the match_data provided numbers.
- Support for single PLL setup
Changes since v1:
- Fixed DT binding documentation errors
- Rebased on ASoC head and updated the driver to compile and work
This series adds support for the analog audio setup on the j721e EVM.
The audio setup of the EVM is:
Common Processor Board (CPB): McASP10 <-> pcm3168a
Infotainment Expansion Board (IVI): McASP0 <-> 2x pcm3168a
Both CPB and IVI wired in parallel serializer setup.
The first patch adds the stream_name for McASP driver as it is needed in
multicodec (and would be needed in DPCM) setup for proper DAPM handling.
The second patch adds two DT schema, one for the cpb and one for the cpb+ivi
card.
Regards,
Peter
---
Peter Ujfalusi (3):
ASoC: ti: davinci-mcasp: Specify stream_name for playback/capture
ASoC: dt-bindings: Add documentation for TI j721e EVM (CPB and IVI)
ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)
.../bindings/sound/ti,j721e-cpb-audio.yaml | 95 ++
.../sound/ti,j721e-cpb-ivi-audio.yaml | 150 +++
sound/soc/ti/Kconfig | 8 +
sound/soc/ti/Makefile | 2 +
sound/soc/ti/davinci-mcasp.c | 3 +
sound/soc/ti/j721e-evm.c | 896 ++++++++++++++++++
6 files changed, 1154 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml
create mode 100644 Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml
create mode 100644 sound/soc/ti/j721e-evm.c
--
Peter
Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki.
Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki
The ASRC not only supports ideal ratio mode, but also supports
internal ratio mode.
For internal rato mode, the rate of clock source should be divided
with no remainder by sample rate, otherwise there is sound
distortion.
Add function fsl_asrc_select_clk() to find proper clock source for
internal ratio mode, if the clock source is available then internal
ratio mode will be selected.
With change, the ideal ratio mode is not the only option for user.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/1593525367-23221-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The following build warnings are seen with 'make dt_binding_check':
Documentation/devicetree/bindings/sound/simple-card.example.dts:209.46-211.15: Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@0: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:213.37-215.15: Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@1: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:250.42-261.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@0: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:263.42-288.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:270.32-272.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@0: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:273.23-275.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@1: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:276.23-278.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@2: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:279.23-281.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@3: node has a unit name, but no reg or ranges property
Documentation/devicetree/bindings/sound/simple-card.example.dts:290.42-303.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@2: node has a unit name, but no reg or ranges property
Fix them all.
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20200630223020.25546-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
intel,keembay-i2s has two register regions:
- I2S registers
- I2S gen configuration
Describe these regions accordingly to fix the following warning seen
with 'make dt_binding_check':
Documentation/devicetree/bindings/sound/intel,keembay-i2s.example.dt.yaml: example-0: i2s@20140000:reg:0: [538181632, 512, 539623588, 4] is too long
Signed-off-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/20200630224459.27174-1-festevam@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio support on the board is using pcm3168a codec connected to McASP10
serializers in parallel setup.
The pcm3168a SCKI clock is coming via the j721e AUDIO_REFCLK2 pin.
In order to support 48KHz and 44.1KHz family of sampling rates the parent clock
for AUDIO_REFCLK2 needs to be changed between PLL4 (for 48KHz) and PLL15 (for
44.1KHz). The same PLLs are used for McASP10's AUXCLK clock via different
HSDIVIDER.
Generic card can not be used for the board as we need to switch between
clock paths for different sampling rate families and also need to change
the slot_width between 16 and 24 bit audio.
The audio support on the Infotainment Expansion Board consists of McASP0
connected to two pcm3168a codecs with dedicated set of serializers to each.
The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin.
It is extending the audio support on the CPB.
Due to the fact that the same PLL4/15 is used by both domains (CPB/IVI)
there are cross restriction on sampling rates.
The IVI side is represented as multicodec setup.
PCMs available on a plain CPB (no IVI addon):
hw:0,0 - cpb playback (8 channels)
hw:0,1 - cpb capture (6 channels)
When the IVI addon is present, additional two PCMs will be present:
hw:0,2 - ivi multicodec playback (16 channels)
hw:0,3 - ivi multicodec capture (12 channels)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20200630125843.11561-4-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio support on the Common Processor Board board is using
pcm3168a codec connected to McASP10 serializers in parallel setup.
The Infotainment board plugs into the Common Processor Board, the support
of the extension board is extending the CPB audio support by adding
the two codecs on the expansion board.
The audio support on the Infotainment Expansion Board consists of McASP0
connected to two pcm3168a codecs with dedicated set of serializers to each.
The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20200630125843.11561-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The adc_vol_tlv volume-control has a range from -17.625 dB to +30 dB,
not -176.25 dB to + 300 dB. This wrong scale is esp. a problem in userspace
apps which translate the dB scale to a linear scale. With the logarithmic
dB scale being of by a factor of 10 we loose all precision in the lower
area of the range when apps translate things to a linear scale.
E.g. the 0 dB default, which corresponds with a value of 47 of the
0 - 127 range for the control, would be shown as 0/100 in alsa-mixer.
Since the centi-dB values used in the TLV struct cannot represent the
0.375 dB step size used by these controls, change the TLV definition
for them to specify a min and max value instead of min + stepsize.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20200628155231.71089-5-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The Lenovo Miix 2 10 has a keyboard dock with extra speakers in the dock.
Rather then the ACL5672's GPIO1 pin being used as IRQ to the CPU, it is
actually used to enable the amplifier for these speakers
(the IRQ to the CPU comes directly from the jack-detect switch).
Add a quirk for having an ext speaker-amplifier enable pin on GPIO1
and replace the Lenovo Miix 2 10's dmi_system_id table entry's wrong
GPIO_DEV quirk (which needs to be renamed to GPIO1_IS_IRQ) with the
new RT5670_GPIO1_IS_EXT_SPK_EN quirk, so that we enable the external
speaker-amplifier as necessary.
Also update the ident field for the dmi_system_id table entry, the
Miix models are not Thinkpads.
Fixes: 67e03ff3f3 ("ASoC: codecs: rt5670: add Thinkpad Tablet 10 quirk")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723
Link: https://lore.kernel.org/r/20200628155231.71089-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The RT5670_PWR_ANLG1 register has 3 bits to select the LDO voltage,
so the correct mask is 0x7 not 0x3.
Because of this wrong mask we were programming the ldo bits
to a setting of binary 001 (0x05 & 0x03) instead of binary 101
when moving to SND_SOC_BIAS_PREPARE.
According to the datasheet 001 is a reserved value, so no idea
what it did, since the driver was working fine before I guess we
got lucky and it does something which is ok.
Fixes: 5e8351de74 ("ASoC: add RT5670 CODEC driver")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200628155231.71089-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The default mode for SSP configuration is TDM 4 slot and so far we were
using this for the bus format on cht-bsw-rt56732 boards.
One board, the Lenovo Miix 2 10 uses not 1 but 2 codecs connected to SSP2.
The second piggy-backed, output-only codec is inside the keyboard-dock
(which has extra speakers). Unlike the main rt5672 codec, we cannot
configure this codec, it is hard coded to use 2 channel 24 bit I2S.
Using 4 channel TDM leads to the dock speakers codec (which listens in on
the data send from the SSP to the rt5672 codec) emiting horribly distorted
sound.
Since we only support 2 channels anyways, there is no need for TDM on any
cht-bsw-rt5672 designs. So we can simply use I2S 2ch everywhere.
This commit fixes the Lenovo Miix 2 10 dock speakers issue by changing
the bus format set in cht_codec_fixup() to I2S 2 channel.
This change has been tested on the following devices with a rt5672 codec:
Lenovo Miix 2 10
Lenovo Thinkpad 8
Lenovo Thinkpad 10 (gen 1)
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: stable@vger.kernel.org
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723
Link: https://lore.kernel.org/r/20200628155231.71089-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Vsense slot configuration based on the device tree. Adding this
property enables the slot programming to be moved to the tdm_set_slot
callback. This in affect sets the slots for the Isense and Vsense and
enabling this these modes are now based on whether these features were
powered on or not.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Link: https://lore.kernel.org/r/20200626154143.20351-3-dmurphy@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When building on allyesconfig kernel for a NO_DMA=y platform (e.g.
Sun-3), CONFIG_SND_SOC_QCOM_COMMON=y, but CONFIG_SND_SOC_QDSP6_AFE=n,
leading to a link failure:
sound/soc/qcom/common.o: In function `qcom_snd_parse_of':
common.c:(.text+0x2e2): undefined reference to `q6afe_is_rx_port'
While SND_SOC_QDSP6 depends on HAS_DMA, SND_SOC_MSM8996 and SND_SOC_SDM845
don't, so the following warning is seen:
WARNING: unmet direct dependencies detected for SND_SOC_QDSP6
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && HAS_DMA [=n]
Selected by [y]:
- SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y]
- SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && CROS_EC [=y] && I2C [=y] && SOUNDWIRE [=y]
Until recently, this warning was harmless (from a compile-testing
point-of-view), but the new user of q6afe_is_rx_port() turned this into
a hard failure.
As the QDSP6 driver itself builds fine if NO_DMA=y, and it depends on
QCOM_APR (which in turns depends on ARCH_QCOM || COMPILE_TEST), it is
safe to increase compile testing coverage. Hence fix the link failure
by dropping the HAS_DMA dependency of SND_SOC_QDSP6.
Fixes: a212008925 ("ASoC: qcom: common: set correct directions for dailinks")
Fixes: 6b1687bf76 ("ASoC: qcom: add sdm845 sound card support")
Fixes: a6f933f63f ("ASoC: qcom: apq8096: Add db820c machine driver")
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://lore.kernel.org/r/20200629122443.21736-1-geert@linux-m68k.org
Signed-off-by: Mark Brown <broonie@kernel.org>
When the cml_rt1011_rt5682_dailink[].codecs pointer is overridden by
a quirk with a devm allocated structure and the probe is deferred,
in the next probe we will see an use-after-free condition
(verified with KASAN). This can be avoided by using statically allocated
configurations - which simplifies the code quite a bit as well.
KASAN issue fixed.
[ 23.301373] cml_rt1011_rt5682 cml_rt1011_rt5682: sof_rt1011_quirk = f
[ 23.301875] ==================================================================
[ 23.302018] BUG: KASAN: use-after-free in snd_cml_rt1011_probe+0x23a/0x3d0 [snd_soc_cml_rt1011_rt5682]
[ 23.302178] Read of size 8 at addr ffff8881ec6acae0 by task kworker/0:2/105
[ 23.302320] CPU: 0 PID: 105 Comm: kworker/0:2 Not tainted 5.7.0-rc7-test+ #3
[ 23.302322] Hardware name: Google Helios/Helios, BIOS 01/21/2020
[ 23.302329] Workqueue: events deferred_probe_work_func
[ 23.302331] Call Trace:
[ 23.302339] dump_stack+0x76/0xa0
[ 23.302345] print_address_description.constprop.0.cold+0xd3/0x43e
[ 23.302351] ? _raw_spin_lock_irqsave+0x7b/0xd0
[ 23.302355] ? _raw_spin_trylock_bh+0xf0/0xf0
[ 23.302362] ? snd_cml_rt1011_probe+0x23a/0x3d0 [snd_soc_cml_rt1011_rt5682]
[ 23.302365] __kasan_report.cold+0x37/0x86
[ 23.302371] ? snd_cml_rt1011_probe+0x23a/0x3d0 [snd_soc_cml_rt1011_rt5682]
[ 23.302375] kasan_report+0x38/0x50
[ 23.302382] snd_cml_rt1011_probe+0x23a/0x3d0 [snd_soc_cml_rt1011_rt5682]
[ 23.302389] platform_drv_probe+0x66/0xc0
Fixes: 629ba12e99 ("ASoC: Intel: boards: split woofer and tweeter support")
Suggested-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Fred Oh <fred.oh@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200625191308.3322-12-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Speaker amplifier feedback is not modeled as being dependent on any
active output. Even when there is no playback happening, parts of the
graph, specifically the IV sense->speaker protection->output remains
active and this prevents the DSP from entering low-power states.
This patch suggests a machine driver level approach where the speaker
pins are enabled/disabled dynamically depending on stream start/stop
events. DPAM graph representations show the feedback loop is indeed
disabled and low-power states can be reached.
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200625191308.3322-8-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
During the bring-up of new platforms, or to take care of specific
hardware reworks, it's useful to add a kernel parameter to override
the default DMI-based quirks.
For example, adding the following line in a .conf file in
/etc/modprobe.d/ will change the default quirk and log the changes if
dynamic debug is enabled.
options snd_soc_sof_sdw quirk=0x802
[ 735.025785] sof_sdw sof_sdw: Overriding quirk 0x10 => 0x802
[ 735.025787] sof_sdw sof_sdw: quirk realtek,jack-detect-source 2
[ 735.025790] sof_sdw sof_sdw: quirk SOF_RT715_DAI_ID_FIX enabled
Tested on ICL RVP with add-on board instead of default codec.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200625191308.3322-6-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
To fix compilation warnings:
- struct 'snd_soc_pcm_runtime' declared inside parameter list will not
be visible outside of this definition or declaration
- struct 'soc_enum' declared inside parameter list will not be visible
outside of this definition or declaration
Declares the missing structure prototypes.
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20200625153543.85039-3-tzungbi@google.com
Signed-off-by: Mark Brown <broonie@kernel.org>
