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[PATCH] ALSA: hda/realtek: Enable headset mic on Lenovo M70 Gen5
by Bin Li 17 Jan '24

17 Jan '24

Lenovo M70 Gen5 is equipped with ALC623, and it needs ALC283_FIXUP_HEADSET_MIC quirk to make its headset mic work. Signed-off-by: Bin Li <bin.li(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b68c94757051..7057e888d373 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10231,6 +10231,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), + SND_PCI_QUIRK(0x17aa, 0x334b, "Lenovo ThinkCentre M70 Gen5", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), -- 2.34.1

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[PATCH alsa-ucm-conf v4] sof-soundwire: Add basic support for cs42l43
by Charles Keepax 17 Jan '24

17 Jan '24

cs42l43 is a codec device, add basic support for it. Including a dual channel DMIC input, stereo headphones, and a mono headset microphone. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No changes just a resend of v4. Thanks, Charles ucm2/codecs/cs42l43/init.conf | 14 ++++++++ ucm2/sof-soundwire/cs42l43-dmic.conf | 26 +++++++++++++++ ucm2/sof-soundwire/cs42l43.conf | 46 +++++++++++++++++++++++++++ ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 4 files changed, 87 insertions(+), 1 deletion(-) create mode 100644 ucm2/codecs/cs42l43/init.conf create mode 100644 ucm2/sof-soundwire/cs42l43-dmic.conf create mode 100644 ucm2/sof-soundwire/cs42l43.conf diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf new file mode 100644 index 0000000..10a6e1f --- /dev/null +++ b/ucm2/codecs/cs42l43/init.conf @@ -0,0 +1,14 @@ +# cs42l43 specific control settings + +LibraryConfig.remap.Config { + ctl.default.map { + "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Decimator 3 Switch'".vindex.0 0 + "name='cs42l43 Decimator 4 Switch'".vindex.1 0 + } + "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Decimator 3 Volume'".vindex.0 0 + "name='cs42l43 Decimator 4 Volume'".vindex.1 0 + } + } +} diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf new file mode 100644 index 0000000..d5f895d --- /dev/null +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -0,0 +1,26 @@ +# Use case Configuration for sof-soundwire card + +SectionDevice."Mic" { + Comment "Microphones" + + ConflictingDevice [ + "Headset" + ] + + EnableSequence [ + cset "name='cs42l43 DP1TX1 Input' 'Decimator 3'" + cset "name='cs42l43 DP1TX2 Input' 'Decimator 4'" + ] + + DisableSequence [ + cset "name='cs42l43 DP1TX1 Input' 'None'" + cset "name='cs42l43 DP1TX2 Input' 'None'" + ] + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId},4" + CaptureMixer "default:${CardId}" + CaptureMixerElem "cs42l43 PDM2" + } +} diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf new file mode 100644 index 0000000..1702cf0 --- /dev/null +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -0,0 +1,46 @@ +# Use case Configuration for sof-soundwire card + +SectionDevice."Headphones" { + Comment "Headphones" + + EnableSequence [ + cset "name='cs42l43 Headphone L Input 1' 'DP5RX1'" + cset "name='cs42l43 Headphone R Input 1' 'DP5RX2'" + ] + + DisableSequence [ + cset "name='cs42l43 Headphone L Input 1' 'None'" + cset "name='cs42l43 Headphone R Input 1' 'None'" + ] + + Value { + PlaybackPriority 200 + PlaybackPCM "hw:${CardId},0" + PlaybackVolume "cs42l43 Headphone Digital Volume" + JackControl "Headphone Jack" + } +} + +SectionDevice."Headset" { + Comment "Headset Microphone" + + EnableSequence [ + cset "name='cs42l43 ADC1 Input' 'IN1'" + cset "name='cs42l43 Decimator 1 Mode' 'ADC'" + + cset "name='cs42l43 DP1TX1 Input' 'Decimator 1'" + cset "name='cs42l43 DP1TX2 Input' 'Decimator 1'" + ] + + DisableSequence [ + cset "name='cs42l43 DP1TX1 Input' 'None'" + cset "name='cs42l43 DP1TX2 Input' 'None'" + ] + + Value { + CapturePriority 200 + CapturePCM "hw:${CardId},4" + CaptureMixerElem "cs42l43 Decimator 1" + JackControl "Headset Mic Jack" + } +} diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 18b1bf8..35a262c 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -47,7 +47,7 @@ DefineRegex { If.hs_init { Condition { Type RegexMatch - Regex "(rt5682|rt700|rt711|rt713(-sdca)?)" + Regex "(cs42l43|rt5682|rt700|rt711|rt713(-sdca)?)" String "${var:HeadsetCodec1}" } True.Include.hs_init.File "/codecs/${var:HeadsetCodec1}/init.conf" -- 2.30.2

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[PATCH 0/7] ASoC: qcom: volume fixes and codec cleanups
by Johan Hovold 17 Jan '24

17 Jan '24

To reduce the risk of speaker damage the PA gain needs to be limited on machines like the Lenovo Thinkpad X13s until we have active speaker protection in place. Limit the gain to the current default setting provided by the UCM configuration which most user have so far been been using (due to a bug in the configuration files which prevented hardware volume control [1]). Included is also a related fix for the LPASS WSA macro driver, which was changing the digital gain setting behind the back of user space and which can result in excessive (or too low) digital gain. There are further Qualcomm codec driver that appear to manipulate various gain settings, but on closer inspection this turned out to be effectively dead code which can be removed. Johan [1] https://github.com/alsa-project/alsa-ucm-conf/pull/382 Johan Hovold (7): ASoC: qcom: sc8280xp: limit speaker volumes ASoC: codecs: lpass-wsa-macro: fix compander volume hack ASoC: codecs: lpass-wsa-macro: drop dead mixer-path gain hack ASoC: codecs: lpass-rx-macro: drop dead mixer-path gain hack ASoC: codecs: wcd9335: drop dead gain hacks ASoC: codecs: wcd934x: drop dead gain hacks ASoC: codecs: msm8916-wcd-digital: drop dead gain hacks sound/soc/codecs/lpass-rx-macro.c | 16 +--- sound/soc/codecs/lpass-wsa-macro.c | 19 +--- sound/soc/codecs/msm8916-wcd-digital.c | 26 +----- sound/soc/codecs/wcd9335.c | 115 ++++--------------------- sound/soc/codecs/wcd934x.c | 102 +++++----------------- sound/soc/qcom/sc8280xp.c | 8 +- 6 files changed, 53 insertions(+), 233 deletions(-) -- 2.41.0

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[PATCH v2 0/3] ASoC: qcom: volume fixes and codec cleanups
by Johan Hovold 17 Jan '24

17 Jan '24

To reduce the risk of speaker damage the PA gain needs to be limited on machines like the Lenovo Thinkpad X13s until we have active speaker protection in place. Limit the gain to the current default setting provided by the UCM configuration which most user have so far been using (due to a bug in the configuration files which prevented hardware volume control [1]). Included is also a related fix for the LPASS WSA macro driver, which was changing the digital gain setting behind the back of user space and which can result in excessive (or too low) digital gain. There are further Qualcomm codec drivers that similarly appear to manipulate various gain settings, but on closer inspection it turns out that they only write back the current settings. Tests reveal that these writes are indeed needed for any prior updates to take effect (at least for the WSA and RX macros). Johan [1] https://github.com/alsa-project/alsa-ucm-conf/pull/382 Changes in v2 - keep the volume register write on power-on in lpass-wsa-macro - drop the other patches removing volume register writes on DAPM events - only drop the constant-zero gain offsets in wcd9335 Johan Hovold (3): ASoC: qcom: sc8280xp: limit speaker volumes ASoC: codecs: lpass-wsa-macro: fix compander volume hack ASoC: codecs: wcd9335: drop unused gain hack remnant sound/soc/codecs/lpass-wsa-macro.c | 7 ------- sound/soc/codecs/wcd9335.c | 4 ---- sound/soc/qcom/sc8280xp.c | 8 +++++--- 3 files changed, 5 insertions(+), 14 deletions(-) -- 2.41.0

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[GIT PULL] ASoC fixes for v6.8-merge-window
by Mark Brown 16 Jan '24

16 Jan '24

The following changes since commit 67508b874844b80ac49f70b78d67036c28b9fe7e: ASoC: pxa: sspa: Don't select SND_ARM (2023-12-29 17:41:15 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.8-merge-window for you to fetch changes up to ab09fb9c629ed3aaea6a82467f08595dbc549726: ASoC: SOF: ipc4-loader: remove the CPC check warnings (2024-01-15 14:04:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.8 A bunch of small fixes that come in during the merge window, mainly fixing issues from some core refactoring around dummy components that weren't detected until things reached mainline. The TAS driver changes are a little larger than normal for a device ID addition due to some shuffling around of where things are registered and DT updates but aren't really any more substantial than normal. ---------------------------------------------------------------- AngeloGioacchino Del Regno (2): ASoC: mediatek: mt8195: Remove afe-dai component and rework codec link ASoC: mediatek: sof-common: Add NULL check for normal_link string ChiYuan Huang (2): ASoC: codecs: rtq9128: Fix PM_RUNTIME usage ASoC: codecs: rtq9128: Fix TDM enable and DAI format control flow Kai Vehmanen (1): ASoC: SOF: ipc4-loader: remove the CPC check warnings Kuninori Morimoto (1): ASoC: audio-graph-card2: fix index check on graph_parse_node_multi_nm() Mark Brown (1): rtq9128: Fix pm runtime and TDM usage Nícolas F. R. A. Prado (1): ASoC: mediatek: mt8192: Check existence of dai_name before dereferencing Peter Ujfalusi (3): ASoC: Intel: bxt_da7219_max98357a: Fix kernel ops due to COMP_DUMMY change ASoC: Intel: bxt_rt298: Fix kernel ops due to COMP_DUMMY change ASoC: SOF: icp3-dtrace: Revert "Fix wrong kfree() usage" Rander Wang (1): ASoC: SOF: ipc4-pcm: remove log message for LLP Shenghao Ding (4): ASoC: dt-bindings: move tas2563 from tas2562.yaml to tas2781.yaml ASoC: tas2562: move tas2563 from tas2562 driver to tas2781 driver ASoC: tas2781: Add tas2563 into header file for DSP mode ASoC: tas2781: Add tas2563 into driver .../devicetree/bindings/sound/tas2562.yaml | 2 - .../devicetree/bindings/sound/ti,tas2781.yaml | 78 +++++++++++++++++----- include/sound/tas2781.h | 9 +-- sound/soc/codecs/rtq9128.c | 73 +++++++++++--------- sound/soc/codecs/tas2562.c | 3 - sound/soc/codecs/tas2781-i2c.c | 8 ++- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 6 ++ sound/soc/intel/boards/bxt_rt298.c | 3 +- sound/soc/mediatek/common/mtk-dsp-sof-common.c | 2 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 3 +- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 33 +-------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 41 ++++++++---- sound/soc/sof/ipc3-dtrace.c | 3 +- sound/soc/sof/ipc4-loader.c | 11 ++- sound/soc/sof/ipc4-pcm.c | 4 +- 16 files changed, 161 insertions(+), 120 deletions(-)

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[PATCH 00/33] spi: get rid of some legacy macros
by Uwe Kleine-König 16 Jan '24

16 Jan '24

Hello, In commit 8caab75fd2c2 ("spi: Generalize SPI "master" to "controller"") some functions were renamed. Further some compat defines were introduced to map the old names to the new ones. In this series all drivers still using the old names are changed to use the new ones and then in patch #32 the compat defines are dropped. Variables and struct members of type pointer to struct spi_controller that were named "master" (matching the old name of spi_controller) were renamed to "ctlr", which slightly increased the count of touched lines, but is nice for consistency. Patch #18 and #19 touch the same driver, otherwise the patches #1 - #31 are pairwise independent and could be applied by their respective maintainers. The alternative is to let all patches go via the spi tree. Mark, what's your preference here? Patch #33 updates the documentation. (This one is a bit fuzzy, because it also does some s/master/host/ which doesn't match the remainder of the series). Also patch #18 is an improvement I noticed while touching this driver that doesn't match the series' topic. I still kept it included here. If you should not like it, it can just be dropped. Best regards Uwe Uwe Kleine-König (33): fpga: ice40-spi: Follow renaming of SPI "master" to "controller" ieee802154: ca8210: Follow renaming of SPI "master" to "controller" iio: adc: ad_sigma_delta: Follow renaming of SPI "master" to "controller" Input: pxspad - follow renaming of SPI "master" to "controller" Input: synaptics-rmi4 - follow renaming of SPI "master" to "controller" media: mgb4: Follow renaming of SPI "master" to "controller" media: netup_unidvb: Follow renaming of SPI "master" to "controller" media: usb/msi2500: Follow renaming of SPI "master" to "controller" media: v4l2-subdev: Follow renaming of SPI "master" to "controller" misc: gehc-achc: Follow renaming of SPI "master" to "controller" mmc: mmc_spi: Follow renaming of SPI "master" to "controller" mtd: dataflash: Follow renaming of SPI "master" to "controller" mtd: rawnand: fsl_elbc: Let .probe retry if local bus is missing net: ks8851: Follow renaming of SPI "master" to "controller" net: vertexcom: mse102x: Follow renaming of SPI "master" to "controller" platform/chrome: cros_ec_spi: Follow renaming of SPI "master" to "controller" spi: bitbang: Follow renaming of SPI "master" to "controller" spi: cadence-quadspi: Don't emit error message on allocation error spi: cadence-quadspi: Follow renaming of SPI "master" to "controller" spi: cavium: Follow renaming of SPI "master" to "controller" spi: geni-qcom: Follow renaming of SPI "master" to "controller" spi: loopback-test: Follow renaming of SPI "master" to "controller" spi: slave-mt27xx: Follow renaming of SPI "master" to "controller" spi: spidev: Follow renaming of SPI "master" to "controller" staging: fbtft: Follow renaming of SPI "master" to "controller" staging: greybus: spi: Follow renaming of SPI "master" to "controller" tpm_tis_spi: Follow renaming of SPI "master" to "controller" usb: gadget: max3420_udc: Follow renaming of SPI "master" to "controller" video: fbdev: mmp: Follow renaming of SPI "master" to "controller" wifi: libertas: Follow renaming of SPI "master" to "controller" spi: fsl-lib: Follow renaming of SPI "master" to "controller" spi: Drop compat layer from renaming "master" to "controller" Documentation: spi: Update documentation for renaming "master" to "controller" .../driver-api/driver-model/devres.rst | 2 +- Documentation/spi/spi-summary.rst | 74 +++++++++---------- drivers/char/tpm/tpm_tis_spi_main.c | 4 +- drivers/fpga/ice40-spi.c | 4 +- drivers/iio/adc/ad_sigma_delta.c | 14 ++-- drivers/input/joystick/psxpad-spi.c | 4 +- drivers/input/rmi4/rmi_spi.c | 2 +- drivers/media/pci/mgb4/mgb4_core.c | 14 ++-- .../media/pci/netup_unidvb/netup_unidvb_spi.c | 48 ++++++------ drivers/media/usb/msi2500/msi2500.c | 38 +++++----- drivers/media/v4l2-core/v4l2-spi.c | 4 +- drivers/misc/gehc-achc.c | 8 +- drivers/mmc/host/mmc_spi.c | 6 +- drivers/mtd/devices/mtd_dataflash.c | 2 +- drivers/mtd/nand/raw/fsl_elbc_nand.c | 3 +- drivers/net/ethernet/micrel/ks8851_spi.c | 4 +- drivers/net/ethernet/vertexcom/mse102x.c | 2 +- drivers/net/ieee802154/ca8210.c | 2 +- .../net/wireless/marvell/libertas/if_spi.c | 2 +- drivers/platform/chrome/cros_ec_spi.c | 8 +- drivers/spi/spi-ath79.c | 4 +- drivers/spi/spi-bitbang.c | 64 ++++++++-------- drivers/spi/spi-butterfly.c | 6 +- drivers/spi/spi-cadence-quadspi.c | 7 +- drivers/spi/spi-cavium.c | 6 +- drivers/spi/spi-cavium.h | 2 +- drivers/spi/spi-davinci.c | 6 +- drivers/spi/spi-fsl-lib.c | 14 ++-- drivers/spi/spi-geni-qcom.c | 2 +- drivers/spi/spi-gpio.c | 2 +- drivers/spi/spi-lm70llp.c | 6 +- drivers/spi/spi-loopback-test.c | 4 +- drivers/spi/spi-oc-tiny.c | 6 +- drivers/spi/spi-omap-uwire.c | 4 +- drivers/spi/spi-slave-mt27xx.c | 2 +- drivers/spi/spi-xilinx.c | 4 +- drivers/spi/spi-xtensa-xtfpga.c | 2 +- drivers/spi/spi.c | 2 +- drivers/spi/spidev.c | 2 +- drivers/staging/fbtft/fbtft-core.c | 4 +- drivers/staging/greybus/spilib.c | 66 ++++++++--------- drivers/usb/gadget/udc/max3420_udc.c | 2 +- drivers/video/fbdev/mmp/hw/mmp_spi.c | 26 +++---- include/linux/spi/spi.h | 20 +---- include/linux/spi/spi_bitbang.h | 2 +- include/media/v4l2-common.h | 6 +- 46 files changed, 249 insertions(+), 267 deletions(-) base-commit: 8d04a7e2ee3fd6aabb8096b00c64db0d735bc874 -- 2.43.0

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[PATCHv3 RESEND 00/10] ASoC: fsl-asoc-card: compatibility integration of a generic codec use case for use with S/PDIF controller
by Elinor Montmasson 16 Jan '24

16 Jan '24

Hello, This is the v3 of the series of patch aiming to make the machine driver "fsl-asoc-card" compatible with use cases where there is no real codec driver. It proposes to use the "spdif_receiver" and "spdif_transmitter" drivers instead of the dummy codec. This is a first step in using the S/PDIF controller with the ASRC. The five first patches add compatibility with the pair of codecs "spdif_receiver" and "spdif_transmitter" with a new compatible, "fsl,imx-audio-generic". Codec parameters are set with default values. Consequently, the driver is modified to work with multi-codec use cases. It can get 2 codecs phandles from the device tree, and the "fsl_asoc_card_priv" struct now has 2 "codec_priv" to store properties of both codecs. It is fixed to 2 codecs as only "fsl,imx-audio-generic" uses multiple codecs at the moment. However, the driver now uses for_each_codecs macros when possible to ease future implementations of multi-codec configurations. The remaining patches add configuration options for the device tree. They configure the CPU DAI when using "fsl,imx-audio-generic". These options are usually hard-coded in "fsl-asoc-card.c" for each audio codec. Because the generic codec could be used with other CPU DAIs than the S/PDIF controller, setting these parameters could be required. This series of patch was successfully built for arm64 and x86 on top of the latest for-next branch of the ASoC git tree on the 14th of December. These modifications have also been tested on an i.MX8MN evaluation board, with a linux kernel RT v6.1.26-rt8. We also have a few questions, following remarks made by Krzysztof Kozlowski in a previous email for patch 10/10: > >>> The compatible list for this generic sound card currently: > >>> @@ -48,6 +51,8 @@ The compatible list for this generic sound card > >>> currently: > >>> > >>> "fsl,imx-audio-nau8822" > >>> > >>> + "fsl,imx-audio-generic" > >> > >> Generic does not look like hardware specific. > > > > Even if our end goal is to use it with the S/PDIF controller, this new > > support can be used with different hardware that doesn't > > require a codec. Thus, we don't really want to specify "spdif" in it. > > > > Is this compatible string not suitable ? > > Should we rename it to something else, like "fsl,imx-audio-no-codec" ? > > Maybe Mark or Rob will help here, but for me "imx-audio" is just way too > generic. * Which generic name should we use ? Or how should we change it? > Also, you add several new properties, so I really expect either > converting old binding to DT schema first or adding new device in DT > schema format. * fsl-asoc-card.txt currently follows the old dt-bindings format. Should we update it to DT schema format in this patch series before adding my new properties? Best regards, Elinor Montmasson Changelog: v2 -> v3: * when the bitmaster or framemaster are retrieved from the device tree, the driver will now compare them with the two codecs possibly given in the device tree, and not just the first codec. * improve driver modifications to use multiple codecs for better integration of future multi-codec use cases: * use for_each_codec macros when possible. * "fsl_asoc_card_priv" struct now has 2 "codec_priv" as the driver can currently retrieve 2 codec phandles from the device tree. * fix subject of patch 10/10 to follow the style of the subsystem and previous commits of the file. * v2 patch series at: https://lore.kernel.org/alsa-devel/20231027144734.3654829-1-elinor.montmass… v1 -> v2: * replace use of the dummy codec by the pair of codecs "spdif_receiver" / " spdif_transmitter". * adapt how dai links codecs are used to take into account the possibility for multiple codecs per link. * change compatible name. * adapt driver to be able to register two codecs given in the device tree. * v1 patch series at: https://lore.kernel.org/alsa-devel/20230901144550.520072-1-elinor.montmasso… Elinor Montmasson (10): ASoC: fsl-asoc-card: add support for dai links with multiple codecs ASoC: fsl-asoc-card: add second dai link component for codecs ASoC: fsl-asoc-card: add compatibility to use 2 codecs in dai-links ASoC: fsl-asoc-card: add new compatible for a generic codec use case ASoC: fsl-asoc-card: set generic codec as clock provider ASoC: fsl-asoc-card: add dts property "cpu-slot-width" ASoC: fsl-asoc-card: add dts property "cpu-slot-num" ASoC: fsl-asoc-card: add dts properties "cpu-sysclk-freq" ASoC: fsl-asoc-card: add dts properties "cpu-sysclk-dir-out" ASoC: bindings: fsl-asoc-card: add compatible for generic codec .../bindings/sound/fsl-asoc-card.txt | 28 +- sound/soc/fsl/fsl-asoc-card.c | 299 +++++++++++------- 2 files changed, 218 insertions(+), 109 deletions(-) -- 2.25.1

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[PATCH v5] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 16 Jan '24

16 Jan '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loadspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v5: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime --- sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 918 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 97 ++++ 4 files changed, 1030 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 59f9742e9ff4..7008a23f87f4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1847,6 +1848,18 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -664,6 +665,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..b74808dd79cf --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,918 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/of.h> +#include <sound/pcm_params.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0xc000: /* Page 0 ~ 127. */ + case 0xfe80 ... 0xfeff: /* Page 253. */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8001: + /* Only reset register was volatiled. + * Software reset. Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + /* Keep in valid area, out of range value don't care. */ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +static int tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val = 0, ret; + + if (!ucontrol) { + dev_err(tas_dev->dev, "%s, wrong parameter.\n", __func__); + return -EINVAL; + } + /* Read current volume from the device. */ + ret = regmap_read(map, mc->reg, &val); + if (ret < 0) { + dev_err(tas_dev->dev, "%s, get digital vol error %x.\n", + __func__, ret); + return ret; + } + ucontrol->value.integer.value[0] = + tasdevice_clamp(val, mc->max, mc->invert); + + return ret; +} + +static int tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + int val, ret; + + if (!ucontrol) { + dev_err(tas_dev->dev, "%s, wrong parameter.\n", __func__); + return -EINVAL; + } + val = tasdevice_clamp(ucontrol->value.integer.value[0], + mc->max, mc->invert); + + ret = regmap_write(map, mc->reg, val); + if (ret < 0) { + dev_dbg(tas_dev->dev, "%s, Put vol %d into %x %x.\n", + __func__, val, mc->reg, ret); + return ret; + } + + return 1; +} + +static int tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask = 0; + int ret, val = 0; + + if (!ucontrol) { + dev_err(tas_dev->dev, "%s, wrong parameter.\n", __func__); + return -EINVAL; + } + /* Read current volume from the device. */ + ret = regmap_read(map, mc->reg, &val); + if (ret < 0) { + dev_err(tas_dev->dev, "%s get AMP vol from %x with %d.\n", + __func__, mc->reg, ret); + return ret; + } + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = (val & mask) >> mc->shift; + ucontrol->value.integer.value[0] = tasdevice_clamp(val, mc->max, + mc->invert); + + return ret; +} + +static int tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct regmap *map = tas_dev->regmap; + unsigned char mask; + int val, ret; + + if (!ucontrol) { + dev_err(tas_dev->dev, "%s, wrong parameter.\n", __func__); + return -EINVAL; + } + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], mc->max, + mc->invert); + ret = regmap_update_bits(map, mc->reg, mask, val << mc->shift); + if (ret < 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + + return 1; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, dvc_tlv), +}; + +static void tas2783_apply_calib(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *reg_start; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s, slaver doesn't exist.\n", + __func__); + return; + } + if ((tas_dev->sdw_peripheral->id.unique_id < TAS2783_ID_MIN) || + (tas_dev->sdw_peripheral->id.unique_id > TAS2783_ID_MAX)) { + dev_err(tas_dev->dev, "%s, error unique_id.\n", + __func__); + return; + } + + reg_start = (u8 *)(cali_data + (tas_dev->sdw_peripheral->id.unique_id + - TAS2783_ID_MIN) * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + reg_start + i, 4); + if (ret != 0) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +/* Update the calibrate data, including speaker impedance, f0, etc, into algo. + * Calibrate data is done by manufacturer in the factory. These data are used + * by Algo for calucating the speaker temperature, speaker membrance excursion + * and f0 in real time during playback. + * In case of no or valid calibrated data, dsp will still works with default + * calibrated data inside algo. + */ +static int tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_dev->tm; + unsigned int attr = 0, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_dev->cali_data.total_sz = 128; + + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, tas_dev->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_dev->cali_data.total_sz, + tas_dev->cali_data.data); + dev_dbg(tas_dev->dev, "cali get %lx bytes result:%ld\n", + tas_dev->cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return 0; + } + + tmp_val = (unsigned int *)tas_dev->cali_data.data; + + crc = crc32(~0, tas_dev->cali_data.data, 84) ^ ~0; + + if (crc == tmp_val[21]) { + /* Date and time of calibration was done. */ + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_dev->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2783_apply_calib(tas_dev, tmp_val); + } else { + dev_dbg(tas_dev->dev, "CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[21]); + tas_dev->cali_data.total_sz = 0; + } + + return 0; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret, value_sdw; + + mutex_lock(&tas_dev->codec_lock); + + if (!fmw || !fmw->data) { + /* No firmware binary, devices will work in ROM mode. */ + dev_err(tas_dev->dev, + "Failed to read %s, no side-effect on driver\n", + tas_dev->rca_binaryname); + ret = -EINVAL; + goto out; + } + buf = (unsigned char *)fmw->data; + + img_sz = le32_to_cpup((__le32 *)&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_err(tas_dev->dev, "Size not matching, %d %u", + (int)fmw->size, img_sz); + ret = -EINVAL; + goto out; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int)*5, p->length); + } else + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + + if (ret != 0) { + dev_dbg(tas_dev->dev, "Load FW fail: %d.\n", ret); + goto out; + } + offset += sizeof(unsigned int)*5 + p->length; + } + /* Select left/right channel based on unique id. */ + value_sdw = 0x1a; + value_sdw += ((tas_dev->sdw_peripheral->dev_num & 1) << 4); + dev_dbg(tas_dev->dev, "%s dev_num = %u", __func__, + tas_dev->sdw_peripheral->dev_num); + regmap_write(map, TASDEVICE_REG(0, 0, 0x0a), value_sdw); + + tas2783_calibration(tas_dev); + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "Unable to configure port\n"); + return ret; + } + + dev_dbg(dai->dev, "%s fomrat: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "Mute or unmute %d.\n", mute); + + if (mute) { + /* Echo channel can't be shutdown while tas2783 must keep + * working state while playback is on. + */ + if (direction == SNDRV_PCM_STREAM_CAPTURE + && tas_dev->pstream == true) + return 0; + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, TAS2783_SDCA_CTL_FU_MUTE, 0), + 1); + ret |= regmap_write(map, TASDEVICE_REG(0, 0, 0x02), 0x1a); + tas_dev->pstream = false; + } else { + /* FU23 Unmute, 0x40400108. */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, TAS2783_SDCA_CTL_FU_MUTE, 0), + 0); + ret |= regmap_write(map, TASDEVICE_REG(0, 0, 0x02), 0x0); + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + tas_dev->pstream = true; + } + + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2783_reset(struct tasdevice_priv *tas_dev) +{ + struct regmap *map = tas_dev->regmap; + int ret; + + ret = regmap_write(map, TAS2873_REG_SWRESET, 1); + if (ret) { + dev_err(tas_dev->dev, "Reset failed.\n"); + return; + } + usleep_range(1000, 1050); +} + +static int tasdevice_component_probe(struct snd_soc_component *component) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + + tas_dev->component = component; + + dev_dbg(tas_dev->dev, "%s was called.\n", __func__); + + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_component_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + } + + tas2783_reset(tas_dev); + /* tas2783-8[9,...,f].bin was copied into /lib/firmware/ */ + scnprintf(tas_dev->rca_binaryname, 64, "tas2783-%01x.bin", + tas_dev->sdw_peripheral->id.unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, + tas_dev, tasdevice_rca_ready); + if (ret) { + dev_dbg(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", + __func__, tas_dev->sdw_peripheral->id.unique_id, ret); + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + + pm_runtime_enable(tas_dev->dev); + + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + + return ret; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (tas_priv->hw_init) + return 0; + + if (tas_priv->first_hw_init) { + regcache_cache_only(tas_priv->regmap, false); + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + /* sw reset */ + regmap_write(tas_priv->regmap, 0x8001, 0x01); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } else + tas_priv->first_hw_init = true; + /* Mark Slave initialization complete */ + tas_priv->hw_init = true; + + return 0; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + mutex_destroy(&tas_dev->codec_lock); +} + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + int ret; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, + &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(dev, "Failed %d of devm_regmap_init_sdw.", ret); + } else + ret = tasdevice_init(tas_dev); + + if (ret < 0) + tasdevice_remove(tas_dev); + + return ret; +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + tasdevice_remove(tas_dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {}, +}; + +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SET_SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + SET_RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..703b2c0b78a6 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,97 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* Unique id start */ +#define TAS2783_ID_MIN 0x08 +/* Unique id end */ +#define TAS2783_ID_MAX 0x0F + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 + +#define TAS2783_MAX_CALIDATA_SIZE 252 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct calibration_data cali_data; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct mutex codec_lock; + struct regmap *regmap; + struct device *dev; + struct tm tm; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

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UCM is not supported for this HDA model (HDA NVidia at 0xfc080000 irq 130) - how to disable HDA Nvidia?
by GitHub issues - opened 15 Jan '24

15 Jan '24

alsa-project/alsa-ucm-conf issue #384 was opened from motomyslen: I upgraded my system and all microphones stopped working. Before that I had loaded custom UCM configs to make my microphone working. Upgrades which can be related to the issue: - `alsa-ucm-conf:all 1.2.6.3-1ubuntu1.4 1.2.6.3-1ubuntu1.9` - `nvidia-driver-510:amd64 510.108.03-0ubuntu0.22.04.1 525.147.05-0ubuntu0.22.04.1` After OS upgrade I loaded again all latests UCM configs from the repo. I also tried older which I had but the error is the same. First issue: ``` alsaucm reload; ALSA lib parser.c:2021:(parse_master_file) Incompatible syntax 6 in HDA-Intel.conf ALSA lib main.c:1412:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -22 alsaucm: error failed to open sound card hw:0: Invalid argument ``` I fixed it by changing syntax from 6 to 4 in HDA-Intel.conf. It worked for me before OS upgrade. Next issue: ``` $ alsaucm reload; ALSA lib parser.c:243:(error_node) UCM is not supported for this HDA model (HDA NVidia at 0xfc080000 irq 130) ALSA lib main.c:1412:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -6 alsaucm: error failed to open sound card hw:0: No such device or address ``` I tried to disable HDA Nvidia module from loading by adding `snd_hda_codec_hdmi` to `/etc/modprobe.d/blacklist.conf` but it didn't work. How to solve this issue? I don't need audio through HDMI. It would be nice to have it working without any hacks but disabling it is enough for me. Can I somehow disable HDA Nvidia in UCM? I can't downgrade Nvidia driver because I need that. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/13942996… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/384 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Qcom volume fixes
by GitHub pull_request - edited 15 Jan '24

15 Jan '24

alsa-project/alsa-ucm-conf pull request #382 was edited from jhovold: Fix the Qualcomm sc8280xp and x1e80100 configurations which accidentally left the Speaker and Headphones mixer controls undefined so that the sound server falls back to software mixing. Also set the default volume levels in machine specific BootSequences so that the user can override the defaults. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/382 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/382.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0