January 2025 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/9] ALSA: usb-audio: More fixes / improvements on PCM
by Takashi Iwai 22 Apr '25

22 Apr '25

Hi, this is a patch set for USB-audio to fix and improve the PCM stream handling. The primary target is the enhancement of the low-latency playback that was introduced recently for dealing with the small period size (that is used typically by JACK). There are a few other fixes that are relevant with it. The current patches are found in topic/usb-audio branch of sound.git tree, too. Takashi === Takashi Iwai (9): ALSA: usb-audio: Restrict rates for the shared clocks ALSA: usb-audio: Fix possible race at sync of urb completions ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback ALSA: usb-audio: Disable low-latency playback for free-wheel mode ALSA: usb-audio: Disable low-latency mode for implicit feedback sync ALSA: usb-audio: Check available frames for the next packet size ALSA: usb-audio: Add spinlock to stop_urbs() ALSA: usb-audio: Improved lowlatency playback support ALSA: usb-audio: Avoid killing in-flight URBs during draining sound/usb/card.h | 11 ++- sound/usb/endpoint.c | 230 +++++++++++++++++++++++++++++++------------ sound/usb/endpoint.h | 13 ++- sound/usb/pcm.c | 157 ++++++++++++++++++++++------- 4 files changed, 305 insertions(+), 106 deletions(-) -- 2.26.2

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ALSA 1.2.8 release
by Jaroslav Kysela 15 Apr '25

15 Apr '25

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf tinycompress Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7.2_v1.2.8 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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Ask snd_seq_set_client_midi_version
by Correo Alternativo 17 Feb '25

17 Feb '25

Hi, I'm trying out MIDI 2.0 and found a problem, when using snd_seq_set_client_midi_version With SND_SEQ_CLIENT_LEGACY_MIDI I get the following sequence of events when connecting a midi device (using the method snd_seq_event_input or snd_seq_ump_event_input) (timidity command:timidity --interface=A --buffer-fragments=2,8 --output-mode=s1l --sampling-freq=44100): SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START But with SND_SEQ_CLIENT_UMP_MIDI_1_0 and SND_SEQ_CLIENT_UMP_MIDI_2_0 I don't get the SND_SEQ_EVENT_PORT_START events, I only get the following: SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM I don't know if I'm asking this in the right place, what mistake am I making? Just by modifying that it stops working.

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[PATCH V2 00/23] ASoC: amd: acp70: add soundwire and acp pdm support
by Vijendar Mukunda 11 Feb '25

11 Feb '25

This patch series includes below changes - Add Audio IO support for ACP7.0 and ACP7.1 platforms for SoundWire IO and ACP PDM controller combination. - Add common driver modules for ACP7.0 and ACP7.1 platforms (acp pci driver, Soundwire dma driver, pdm platform driver). - Add SoundWire generic machine driver changes for legacy stack (No DSP enabled) for ACP7.0 & ACP7.1 platforms. - Add SoundWire machines for ACP7.0 & ACP7.1 platforms. Changes since v1: - Change the code sequence for handling SoundWire wake interrupts. - Add new patch to restore host wake interrupt mask - Add new patch to enable ACP PME during ACP init sequence. - Change ACP PCI driver PM ops code sequence. Vijendar Mukunda (23): ASoC: amd: add register header file for ACP7.0 platform ASoC: amd: acp70: add acp pci driver for ACP7.0 and ACP7.1 platforms ASoC: amd: acp70: add acp init and de-init functions ASoC: amd: acp70: add logic for scanning acp child devices ASoC: amd: acp70: create platform devices for acp child nodes ASoC: amd: acp70: enable driver build for ACP7.0 platform ASoC: amd: acp70: add acp pdm platform driver ASoC: amd: acp70: add acp pdm driver dma ops and dai ops ASoC: amd: acp70: add acp soundwire dma driver ASoC: amd: update ACP7.0 KConfig option description ASoC: amd: acp70: add soundwire dma driver dma ops ASoC: amd: acp70: add acp ip interrupt handler ASoC: amd: acp70: add acp pdm driver pm ops ASoC: amd: acp70: add pm ops support for soundwire dma driver ASoC: amd: acp70: add acp driver pm ops support ASoC: amd: acp70: enable wake capability for acp pci driver ASoC: amd: acp70: add soundwire wake interrupt handling ASoC: amd: acp70: enable soundwire host wake irq mask ASoC: amd: acp70: enable acp pme enable during acp init sequence ASoC: amd: acp70: create a device node for soundwire machine driver ASoC: amd: acp: add machine driver changes for ACP7.0 and ACP7.1 platforms ASoC: amd: acp: add RT711, RT714 & RT1316 support for ACP7.0 platform ASoC: amd: acp: amd-acp70-acpi-match: Add rt722 support include/sound/acp70_chip_offset_byte.h | 484 +++++++++++++ sound/soc/amd/Kconfig | 12 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 + sound/soc/amd/acp/acp-sdw-mach-common.c | 34 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 132 ++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 11 + sound/soc/amd/acp70/Makefile | 9 + sound/soc/amd/acp70/acp70-pdm-dma.c | 463 +++++++++++++ sound/soc/amd/acp70/acp70-sdw-dma.c | 586 ++++++++++++++++ sound/soc/amd/acp70/acp70.h | 288 ++++++++ sound/soc/amd/acp70/pci-acp70.c | 833 +++++++++++++++++++++++ sound/soc/amd/mach-config.h | 1 + 14 files changed, 2871 insertions(+), 1 deletion(-) create mode 100644 include/sound/acp70_chip_offset_byte.h create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/acp70/Makefile create mode 100644 sound/soc/amd/acp70/acp70-pdm-dma.c create mode 100644 sound/soc/amd/acp70/acp70-sdw-dma.c create mode 100644 sound/soc/amd/acp70/acp70.h create mode 100644 sound/soc/amd/acp70/pci-acp70.c -- 2.34.1

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[GIT PULL] ASoC fixes for v6.14-merge-window
by Mark Brown 29 Jan '25

29 Jan '25

The following changes since commit fee89ddd76e45841a2b01d87b481bc02483f4572: ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled() (2025-01-16 15:20:41 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-merge-window for you to fetch changes up to 6c1bb4031729871fa203983bd77bed1ee3c61347: ASoC: codec: es8316: "DAC Soft Ramp Rate" is just a 2 bit control (2025-01-27 16:22:37 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 A bunch of fixes that came in during the merge window, plus a few new device IDs. The i.MX changes are a little large since they add some new quirk data as well as device IDs, and the audio graph card change for picking the correct endpoint for links is large due to updating a number of call sites. ---------------------------------------------------------------- Alexander Boehm (1): ASoC: SOF: imx8m: add SAI2,5,6,7 Arnd Bergmann (1): ASoC: fsl_asrc_m2m: select CONFIG_DMA_SHARED_BUFFER Claudiu Beznea (1): ASoC: da7213: Initialize the mutex Daniel Baluta (2): ASoC: SOF: imx: Add mach entry to select cs42888 topology ASoC: amd: acp: Fix possible deadlock Detlev Casanova (1): ASoC: rockchip: i2s_tdm: Re-add the set_sysclk callback Fabio Estevam (1): ASoC: dt-bindings: ti,pcm1681: Fix the binding title Geert Uytterhoeven (2): ASoC: renesas: SND_SIU_MIGOR should depend on DMADEVICES firmware: cs_dsp: FW_CS_DSP_KUNIT_TEST should not select REGMAP Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add DMI quirk for Vexia Edu Atla 10 tablet 5V Iuliana Prodan (3): dt-bindings: arm: imx: Add board revisions for i.MX8MP, i.MX8QM and i.MX8QXP ASoC: SOF: imx8: Add entries for new 8QM and 8QXP revisions ASoC: SOF: imx8m: Add entry for new 8M Plus revision Ivaylo Dimitrov (1): ASoC: audio-graph-card2: use correct endpoint when getting link parameters Luoxi Li (1): ASoC: use to_platform_device() instead of container_of() Mario Limonciello (1): ASoC: acp: Support microphone from Lenovo Go S Mark Brown (2): ASoC: fsl: two fixes for asrc memory to memory Add SOF support for new board revisions Ondrej Jirman (1): ASoC: codec: es8316: "DAC Soft Ramp Rate" is just a 2 bit control Shengjiu Wang (2): ASoC: fsl_asrc_m2m: only handle pairs for m2m in the suspend ASoC: fsl_asrc_m2m: return error value in asrc_m2m_device_run() Zhang Yi (1): ASoC: codecs: ES8326: Improved PSRR Documentation/devicetree/bindings/arm/fsl.yaml | 3 ++ .../devicetree/bindings/sound/ti,pcm1681.yaml | 2 +- drivers/firmware/cirrus/Kconfig | 6 +-- sound/soc/amd/acp/acp-i2s.c | 1 + sound/soc/amd/yc/acp6x-mach.c | 28 ++++++++++ sound/soc/codecs/da7213.c | 2 + sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8326.c | 4 +- sound/soc/codecs/rt5514.c | 3 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_asrc_m2m.c | 16 +++--- sound/soc/generic/audio-graph-card2.c | 62 ++++++++++------------ sound/soc/intel/boards/bytcr_rt5640.c | 17 +++++- sound/soc/renesas/Kconfig | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 31 ++++++++++- sound/soc/sof/imx/imx8.c | 24 ++++++++- sound/soc/sof/imx/imx8m.c | 52 +++++++++++++++++- sound/soc/sof/imx/imx8ulp.c | 3 +- sound/soc/sof/intel/bdw.c | 3 +- sound/soc/sof/intel/byt.c | 3 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 6 +-- 22 files changed, 205 insertions(+), 68 deletions(-)

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[PATCH V3 0/6] soundwire: amd: code improvements and new platform support
by Vijendar Mukunda 28 Jan '25

28 Jan '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Changes since v2: - Rebase the patch series after dropping the patch which adds conditional check for SoundWire manager device resume when wake event is asserted. Changes since v1: - Drop white line insertion in the code. - Add a conditional check for SoundWire manager device resume when wake event is asserted. - Add new patch to set ACP_PME_EN during runtime suspend sequence - Split the existing implementation of host wake interrupt mask enablement and setting device power state in to two separate patches and refactor set device power state logic. - Drop PME status clear logic. Vijendar Mukunda (6): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: set device power state during suspend/resume sequence soundwire: amd: set ACP_PME_EN during runtime suspend sequence soundwire: amd: add soundwire host wake interrupt enable/disable sequence drivers/soundwire/amd_manager.c | 150 +++++++++++++++++++++++++++++- drivers/soundwire/amd_manager.h | 26 ++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 173 insertions(+), 5 deletions(-) -- 2.34.1

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[PATCH 00/34] address all -Wunused-const warnings
by Arnd Bergmann 27 Jan '25

27 Jan '25

From: Arnd Bergmann <arnd(a)arndb.de> Compilers traditionally warn for unused 'static' variables, but not if they are constant. The reason here is a custom for C++ programmers to define named constants as 'static const' variables in header files instead of using macros or enums. In W=1 builds, we get warnings only static const variables in C files, but not in headers, which is a good compromise, but this still produces warning output in at least 30 files. These warnings are almost all harmless, but also trivial to fix, and there is no good reason to warn only about the non-const variables being unused. I've gone through all the files that I found using randconfig and allmodconfig builds and created patches to avoid these warnings, with the goal of retaining a clean build once the option is enabled by default. Unfortunately, there is one fairly large patch ("drivers: remove incorrect of_match_ptr/ACPI_PTR annotations") that touches 34 individual drivers that all need the same one-line change. If necessary, I can split it up by driver or by subsystem, but at least for reviewing I would keep it as one piece for the moment. Please merge the individual patches through subsystem trees. I expect that some of these will have to go through multiple revisions before they are picked up, so anything that gets applied early saves me from resending. Arnd Arnd Bergmann (31): powerpc/fsl-soc: hide unused const variable ubsan: fix unused variable warning in test module platform: goldfish: remove ACPI_PTR() annotations i2c: pxa: hide unused icr_bits[] variable 3c515: remove unused 'mtu' variable tracing: hide unused ftrace_event_id_fops Input: synaptics: hide unused smbus_pnp_ids[] array power: rt9455: hide unused rt9455_boost_voltage_values efi: sysfb: don't build when EFI is disabled clk: ti: dpll: fix incorrect #ifdef checks apm-emulation: hide an unused variable sisfb: hide unused variables dma/congiguous: avoid warning about unused size_bytes leds: apu: remove duplicate DMI lookup data iio: ad5755: hook up of_device_id lookup to platform driver greybus: arche-ctrl: move device table to its right location lib: checksum: hide unused expected_csum_ipv6_magic[] sunrpc: suppress warnings for unused procfs functions comedi: ni_atmio: avoid warning for unused device_ids[] table iwlegacy: don't warn for unused variables with DEBUG_FS=n drm/komeda: don't warn for unused debugfs files firmware: qcom_scm: mark qcom_scm_qseecom_allowlist as __maybe_unused crypto: ccp - drop platform ifdef checks usb: gadget: omap_udc: remove unused variable isdn: kcapi: don't build unused procfs code cpufreq: intel_pstate: hide unused intel_pstate_cpu_oob_ids[] net: xgbe: remove extraneous #ifdef checks Input: imagis - remove incorrect ifdef checks sata: mv: drop unnecessary #ifdef checks ASoC: remove incorrect of_match_ptr/ACPI_PTR annotations spi: remove incorrect of_match_ptr annotations drivers: remove incorrect of_match_ptr/ACPI_PTR annotations kbuild: always enable -Wunused-const-variable Krzysztof Kozlowski (1): Input: stmpe-ts - mark OF related data as maybe unused arch/powerpc/sysdev/fsl_msi.c | 2 + drivers/ata/sata_mv.c | 64 +++++++++---------- drivers/char/apm-emulation.c | 5 +- drivers/char/ipmi/ipmb_dev_int.c | 2 +- drivers/char/tpm/tpm_ftpm_tee.c | 2 +- drivers/clk/ti/dpll.c | 10 ++- drivers/comedi/drivers/ni_atmio.c | 2 +- drivers/cpufreq/intel_pstate.c | 2 + drivers/crypto/ccp/sp-platform.c | 14 +--- drivers/dma/img-mdc-dma.c | 2 +- drivers/firmware/efi/Makefile | 3 +- drivers/firmware/efi/sysfb_efi.c | 2 - drivers/firmware/qcom/qcom_scm.c | 2 +- drivers/fpga/versal-fpga.c | 2 +- .../gpu/drm/arm/display/komeda/komeda_dev.c | 8 --- drivers/hid/hid-google-hammer.c | 6 +- drivers/i2c/busses/i2c-pxa.c | 2 +- drivers/i2c/muxes/i2c-mux-ltc4306.c | 2 +- drivers/i2c/muxes/i2c-mux-reg.c | 2 +- drivers/iio/dac/ad5755.c | 1 + drivers/input/mouse/synaptics.c | 2 + drivers/input/touchscreen/imagis.c | 4 +- drivers/input/touchscreen/stmpe-ts.c | 2 +- drivers/input/touchscreen/wdt87xx_i2c.c | 2 +- drivers/isdn/capi/Makefile | 3 +- drivers/isdn/capi/kcapi.c | 7 +- drivers/leds/leds-apu.c | 3 +- drivers/mux/adg792a.c | 2 +- drivers/net/ethernet/3com/3c515.c | 3 - drivers/net/ethernet/amd/xgbe/xgbe-platform.c | 8 --- drivers/net/ethernet/apm/xgene-v2/main.c | 2 +- drivers/net/ethernet/hisilicon/hns_mdio.c | 2 +- drivers/net/wireless/intel/iwlegacy/4965-rs.c | 15 +---- drivers/net/wireless/intel/iwlegacy/common.h | 2 - drivers/platform/goldfish/goldfish_pipe.c | 2 +- drivers/power/supply/rt9455_charger.c | 2 + drivers/regulator/pbias-regulator.c | 2 +- drivers/regulator/twl-regulator.c | 2 +- drivers/regulator/twl6030-regulator.c | 2 +- drivers/rtc/rtc-fsl-ftm-alarm.c | 2 +- drivers/scsi/hisi_sas/hisi_sas_v1_hw.c | 2 +- drivers/scsi/hisi_sas/hisi_sas_v2_hw.c | 2 +- drivers/spi/spi-armada-3700.c | 2 +- drivers/spi/spi-img-spfi.c | 2 +- drivers/spi/spi-meson-spicc.c | 2 +- drivers/spi/spi-meson-spifc.c | 2 +- drivers/spi/spi-orion.c | 2 +- drivers/spi/spi-pic32-sqi.c | 2 +- drivers/spi/spi-pic32.c | 2 +- drivers/spi/spi-rockchip.c | 2 +- drivers/spi/spi-s3c64xx.c | 2 +- drivers/spi/spi-st-ssc4.c | 2 +- drivers/staging/greybus/arche-apb-ctrl.c | 1 + drivers/staging/greybus/arche-platform.c | 9 +-- drivers/staging/pi433/pi433_if.c | 2 +- drivers/tty/serial/amba-pl011.c | 6 +- drivers/tty/serial/ma35d1_serial.c | 2 +- drivers/usb/gadget/udc/omap_udc.c | 10 +-- drivers/video/fbdev/sis/init301.c | 3 +- kernel/dma/contiguous.c | 2 +- kernel/trace/trace_events.c | 4 ++ lib/checksum_kunit.c | 2 + lib/test_ubsan.c | 2 +- net/sunrpc/cache.c | 10 +-- scripts/Makefile.extrawarn | 1 - sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/samsung/aries_wm8994.c | 2 +- 69 files changed, 121 insertions(+), 169 deletions(-) -- 2.39.2 Cc: Michael Ellerman <mpe(a)ellerman.id.au> Cc: Christophe Leroy <christophe.leroy(a)csgroup.eu> Cc: Damien Le Moal <dlemoal(a)kernel.org> Cc: Jiri Kosina <jikos(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Corey Minyard <minyard(a)acm.org> Cc: Peter Huewe <peterhuewe(a)gmx.de> Cc: Jarkko Sakkinen <jarkko(a)kernel.org> Cc: Tero Kristo <kristo(a)kernel.org> Cc: Stephen Boyd <sboyd(a)kernel.org> Cc: Ian Abbott <abbotti(a)mev.co.uk> Cc: H Hartley Sweeten <hsweeten(a)visionengravers.com> Cc: Srinivas Pandruvada <srinivas.pandruvada(a)linux.intel.com> Cc: Len Brown <lenb(a)kernel.org> Cc: "Rafael J. Wysocki" <rafael(a)kernel.org> Cc: John Allen <john.allen(a)amd.com> Cc: Herbert Xu <herbert(a)gondor.apana.org.au> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Ard Biesheuvel <ardb(a)kernel.org> Cc: Bjorn Andersson <andersson(a)kernel.org> Cc: Moritz Fischer <mdf(a)kernel.org> Cc: Liviu Dudau <liviu.dudau(a)arm.com> Cc: Benjamin Tissoires <benjamin.tissoires(a)redhat.com> Cc: Andi Shyti <andi.shyti(a)kernel.org> Cc: Michael Hennerich <michael.hennerich(a)analog.com> Cc: Peter Rosin <peda(a)axentia.se> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Jonathan Cameron <jic23(a)kernel.org> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Markuss Broks <markuss.broks(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)foss.st.com> Cc: Lee Jones <lee(a)kernel.org> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Shyam Sundar S K <Shyam-sundar.S-k(a)amd.com> Cc: Iyappan Subramanian <iyappan(a)os.amperecomputing.com> Cc: Yisen Zhuang <yisen.zhuang(a)huawei.com> Cc: Stanislaw Gruszka <stf_xl(a)wp.pl> Cc: Kalle Valo <kvalo(a)kernel.org> Cc: Sebastian Reichel <sre(a)kernel.org> Cc: Tony Lindgren <tony(a)atomide.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Cc: Xiang Chen <chenxiang66(a)hisilicon.com> Cc: "Martin K. Petersen" <martin.petersen(a)oracle.com> Cc: Neil Armstrong <neil.armstrong(a)linaro.org> Cc: Heiko Stuebner <heiko(a)sntech.de> Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Cc: Vaibhav Hiremath <hvaibhav.linux(a)gmail.com> Cc: Alex Elder <elder(a)kernel.org> Cc: Jiri Slaby <jirislaby(a)kernel.org> Cc: Jacky Huang <ychuang3(a)nuvoton.com> Cc: Helge Deller <deller(a)gmx.de> Cc: Christoph Hellwig <hch(a)lst.de> Cc: Robin Murphy <robin.murphy(a)arm.com> Cc: Steven Rostedt <rostedt(a)goodmis.org> Cc: Masami Hiramatsu <mhiramat(a)kernel.org> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: Kees Cook <keescook(a)chromium.org> Cc: Trond Myklebust <trond.myklebust(a)hammerspace.com> Cc: Anna Schumaker <anna(a)kernel.org> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-ide(a)vger.kernel.org Cc: openipmi-developer(a)lists.sourceforge.net Cc: linux-integrity(a)vger.kernel.org Cc: linux-omap(a)vger.kernel.org Cc: linux-clk(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: dmaengine(a)vger.kernel.org Cc: linux-efi(a)vger.kernel.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-fpga(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-input(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-arm-kernel(a)lists.infradead.org Cc: netdev(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-wireless(a)vger.kernel.org Cc: linux-rtc(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-amlogic(a)lists.infradead.org Cc: linux-rockchip(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: greybus-dev(a)lists.linaro.org Cc: linux-staging(a)lists.linux.dev Cc: linux-serial(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: iommu(a)lists.linux.dev Cc: linux-trace-kernel(a)vger.kernel.org Cc: kasan-dev(a)googlegroups.com Cc: linux-hardening(a)vger.kernel.org Cc: linux-nfs(a)vger.kernel.org Cc: linux-kbuild(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-sound(a)vger.kernel.org

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USB-Audio: ALC4082 - Add support for ASUS ROG Crosshair X870E Hero (U…
by GitHub pull_request - opened 27 Jan '25

27 Jan '25

alsa-project/alsa-ucm-conf pull request #503 was opened from Fabcien: …SB 0b05:1b7c) https://rog.asus.com/motherboards/rog-crosshair/rog-crosshair-x870e-hero/sp… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/503 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/503.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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No optical S/PDIF output on Debian 12 Gnome with ASUS ROG STRIX X670E-A
by GitHub issues - opened 24 Jan '25

24 Jan '25

alsa-project/alsa-ucm-conf issue #500 was opened from BGnoinski: I am currently running into an issue with my Gnome desktop where audio from applications like Firefox is not working. I have run `alsa-info.sh` and the contents [have been uploaded here](http://alsa-project.org/db/?f=d18e76790182eaa8b63e03f435397660b5879b0… I know that debian is running an old version of the `alsa-ucm-conf` package (1.2.8) so I have updated the ucm and ucm2 confs with: ``` #!/bin/bash curl -L -o alsa-ucm-conf.tar.gz https://github.com/alsa-project/alsa-ucm-conf/archive/refs/heads/master.tar… tar xvzf alsa-ucm-conf.tar.gz -C /usr/share/alsa --strip-components=1 --wildcards "*/ucm" "*/ucm2" ``` I can see that my boards identifier(not sure if that's the correct terminology) exists in the ucm2 USB-Audio conf ``` grep -Hrn '0b05:1a52' /usr/share/alsa/ /usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf:58: # 0b05:1a52 ASUS ROG Strix X670E-F & Z790-E Gaming Wifi ``` Even though the comment doesn't have my board specifically in it, looking at the uploaded package I see: ``` Bus 003 Device 003: ID 0b05:1a52 ASUSTek Computer, Inc. USB Audio ``` So my understanding is that should be fine. Even when I look at the [motherboard specs](https://rog.asus.com/us/motherboards/rog-strix/rog-strix-x670e-a-gam… I see that it says it's `ROG SupremeFX 7.1 Surround Sound High Definition Audio CODEC ALC4080` so I don't know if this is different from the Realtek ALC4080 Maybe? And the odd part is if I run `aplay -D plughw:Audio,3 /usr/share/sounds/alsa/Front_Right.wav ` I get the following output: ``` Playing WAVE '/usr/share/sounds/alsa/Front_Right.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono ``` And I do in fact hear audio coming from my speakers. Looking in the sound settings it appears as if I am able to select the correct output, and when audio _should_ be playing the little blue line does go up. ![Image](https://github.com/user-attachments/assets/9e865f37-8153-4bdf-946d-3a428b561321) So it seems like I am just missing something obvious to tie all of the components together to get desktop applications working with the correct output. Any help would be greatly appreciated. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/500 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ucm2: Intel: avs: Add UCM files for HDA configuration
by GitHub pull_request - opened 23 Jan '25

23 Jan '25

alsa-project/alsa-ucm-conf pull request #499 was opened from amadeuszslawinski-intel: Add configs for devices using HDA codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/499 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/499.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Qualcomm: Add SM8750 MTP HiFi config
by GitHub pull_request - opened 22 Jan '25

22 Jan '25

alsa-project/alsa-ucm-conf pull request #498 was opened from krzk: Add UCM2 configs for the Qualcomm SM8750-MTP Board to handle: - on-board Speakers - Headphones speakers - Top and Bottom on-board Microphones Not yet implemented/tested are Headphone Microphones over USB-C port. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/498 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/498.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[RESEND PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 22 Jan '25

22 Jan '25

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 58c3381725fb..41a4bd0cd0a7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23206,25 +23206,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23234,9 +23251,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> base-commit: c4b9570cfb63501638db720f3bee9f6dfd044b82 -- 2.34.1

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[PATCH V2 3/7] soundwire: amd: add conditional for check for device resume
by Vijendar Mukunda 21 Jan '25

21 Jan '25

Add a conditional check to resume SoundWire manager device, when the SoundWire manager instance is in the suspended state. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 60be5805715e..6831b3729db5 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -850,7 +850,10 @@ static void amd_sdw_update_slave_status(u32 status_change_0to7, u32 status_chang static void amd_sdw_process_wake_event(struct amd_sdw_manager *amd_manager) { dev_dbg(amd_manager->dev, "SoundWire Wake event reported\n"); - pm_request_resume(amd_manager->dev); + if (pm_runtime_suspended(amd_manager->dev)) { + dev_dbg(amd_manager->dev, "Device is in suspend state\n"); + pm_request_resume(amd_manager->dev); + } writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_8TO11); } -- 2.34.1

2 1

[PATCH] ALSA: hda/realtek: Enable Mute LED on HP Laptop 14s-fq1xxx
by Sebastian Wiese-Wagner 20 Jan '25

20 Jan '25

This HP Laptop uses ALC236 codec with COEF 0x07 controlling the mute LED. Enable existing quirk for this device. Signed-off-by: Sebastian Wiese-Wagner <seb(a)fastmail.to> --- 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 46623cd615a4..d3c9ed963588 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10415,6 +10415,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x887a, "HP Laptop 15s-eq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x887c, "HP Laptop 14s-fq1xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x888a, "HP ENVY x360 Convertible 15-eu0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8895, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED), -- 2.48.1

1 0

[PATCH] ASoC: codecs: ES8326: Improved PSRR
by Zhang Yi 20 Jan '25

20 Jan '25

Modified configuration to improve PSSR when ES8326 is working Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b06eead7e0f6..066d92b54312 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -911,7 +911,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x1f); - regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x08); + regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x0d); queue_delayed_work(system_wq, &es8326->jack_detect_work, msecs_to_jiffies(400)); es8326->hp = 1; @@ -1023,7 +1023,7 @@ static void es8326_init(struct snd_soc_component *component) struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); regmap_write(es8326->regmap, ES8326_RESET, 0x1f); - regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x3E); regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0); usleep_range(10000, 15000); regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9); -- 2.17.1

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[GIT PULL] ASoC updates for v6.14
by Mark Brown 20 Jan '25

20 Jan '25

The following changes since commit 9d89551994a430b50c4fffcb1e617a057fa76e20: Linux 6.13-rc6 (2025-01-05 14:13:40 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.14 for you to fetch changes up to fee89ddd76e45841a2b01d87b481bc02483f4572: ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled() (2025-01-16 15:20:41 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.14 This was quite a quiet release for what I imagine are holiday related reasons, the diffstat is dominated by some Cirrus Logic Kunit tests. There's the usual mix of small improvements and fixes, plus a few new drivers and features. The diffstat includes some DRM changes due to work on HDMI audio. - Allow clocking on each DAI in an audio graph card to be configured separately. - Improved power management for Renesas RZ-SSI. - KUnit testing for the Cirrus DSP framework. - Memory to meory operation support for Freescale/NXP platforms. - Support for pause operations in SOF. - Support for Allwinner suinv F1C100s, Awinc AW88083, Realtek ALC5682I-VE ---------------------------------------------------------------- Alexey Klimov (1): ASoC: qcom: sdm845: add handling of secondary MI2S clock Amadeusz Sławiński (4): ASoC: Intel: avs: Fix init-config parsing ASoC: Intel: avs: Print IPC error messages in lower layer ASoC: Intel: avs: Add MODULE_FIRMWARE to inform about FW ASoC: Intel: avs: Update ASRC definition Andrew Davis (21): ASoC: ad193x: Remove use of i2c_match_id() ASoC: adau1761: Remove use of i2c_match_id() ASoC: adau1781: Remove use of i2c_match_id() ASoC: adau1977: Remove use of i2c_match_id() ASoC: alc5623: Remove use of i2c_match_id() ASoC: alc5632: Remove use of i2c_match_id() ASoC: max98088: Remove use of i2c_match_id() ASoC: max98090: Remove use of i2c_match_id() ASoC: max98095: Remove use of i2c_match_id() ASoC: pcm186x: Remove use of i2c_match_id() ASoc: pcm6240: Remove use of i2c_match_id() ASoC: ssm2602: Remove use of i2c_match_id() ASoC: tas2562: Remove use of i2c_match_id() ASoC: tas2781: Remove use of i2c_match_id() ASoC: tas5720: Remove use of i2c_match_id() ASoC: tlv320adc3xxx: Remove use of i2c_match_id() ASoC: tlv320aic31xx: Remove use of i2c_match_id() ASoC: tlv320aic3x: Remove use of i2c_match_id() ASoC: tpa6130a2: Remove use of i2c_match_id() ASoC: wm8904: Remove use of i2c_match_id() ASoC: wm8985: Remove use of i2c_match_id() Andy Shevchenko (1): ASoc: mediatek: mt8365: Don't use "proxy" headers Arnd Bergmann (1): firmware: cs_dsp: avoid large local variables Bard Liao (7): ASoC: SOF: Intel: hda-dai: Ensure DAI widget is valid during params ASoC: Intel: sof_sdw: correct mach_params->dmic_num ASoC: Intel: sof_sdw: reduce log level for not using internal dmic ASoC: Intel: sof_sdw: improve the log of DAI link numbers ASoC: Intel: soc-acpi-intel-ptl-match: add rt712_vb + rt1320 support ASoC: Intel: soc-acpi-intel-lnl-match: add rt713_vb_l2_rt1320_l13 support ASoC: Intel: soc-acpi-intel-ptl-match: add rt713_vb_l2_rt1320_l13 support Bence Csókás (1): ASoC: sun4i-codec: Use new devm clk and reset APIs Cezary Rojewski (9): ASoC: Intel: avs: Do not readq() u32 registers ASoC: Intel: avs: Fix the minimum firmware version numbers ASoC: Intel: avs: Fix theoretical infinite loop ASoC: Intel: avs: Update hda component teardown sequences ASoC: Intel: avs: Clearly state assumptions of hw_params() ASoC: Intel: avs: Improve logging of firmware loading ASoC: Intel: avs: Adjust DSP status register names ASoC: Intel: avs: Adjust IPC traces ASoC: Intel: avs: Add missing includes Chancel Liu (4): ASoC: fsl_utils: Add function to constrain rates ASoC: fsl_micfil: Switch to common sample rate constraint function ASoC: fsl_xcvr: Add sample rate constraint ASoC: fsl_sai: Add sample rate constraint Charles Keepax (5): ASoC: SDCA: Add missing header includes ASoC: SDCA: Clean up error messages ASoC: SDCA: Add bounds check for function address ASoC: SDCA: Add missing function type names ASoC: SDCA: Split function type patching and function naming Christophe JAILLET (1): ASoC: cs42l51: Constify struct i2c_device_id Claudiu Beznea (16): ASoC: renesas: rz-ssi: Terminate all the DMA transactions ASoC: renesas: rz-ssi: Use only the proper amount of dividers ASoC: renesas: rz-ssi: Fix typo on SSI_RATES macro comment ASoC: renesas: rz-ssi: Remove pdev member of struct rz_ssi_priv ASoC: renesas: rz-ssi: Remove the rz_ssi_get_dai() function ASoC: renesas: rz-ssi: Remove the first argument of rz_ssi_stream_is_play() ASoC: renesas: rz-ssi: Use readl_poll_timeout_atomic() ASoC: renesas: rz-ssi: Use temporary variable for struct device ASoC: renesas: rz-ssi: Use goto label names that specify their actions ASoC: renesas: rz-ssi: Rely on the ASoC subsystem to runtime resume/suspend the SSI ASoC: renesas: rz-ssi: Enable runtime PM autosuspend support ASoC: renesas: rz-ssi: Add runtime PM support ASoC: renesas: rz-ssi: Issue software reset in hw_params API ASoC: renesas: rz-ssi: Add suspend to RAM support ASoC: dt-bindings: renesas,rz-ssi: Remove DMA description ASoC: dt-bindings: renesas,rz-ssi: Document the Renesas RZ/G3S SoC Colin Ian King (1): ASoC: soc-core: remove redundant assignment to variable ret Csókás Bence (1): ASoC: dt-bindings: Add Allwinner suniv F1C100s Audio Codec Dan Carpenter (1): ASoC: renesas: rz-ssi: Add a check for negative sample_space Derek Fang (1): ASoC: rt5682: Support the ALC5682I-VE codec Dheeraj Reddy Jonnalagadda (1): ASoC: rockchip: i2s-tdm: Fix a useless call issue Dr. David Alan Gilbert (1): ASoC: rt715: Remove unused hda_to_sdw George Lander (1): ASoC: sun4i-spdif: Add clock multiplier settings Krzysztof Kozlowski (3): ASoC: codecs: wcd9335: Add define for number of DAIs ASoC: dt-bindings: qcom,wcd9335: Drop number of DAIs from the header ASoC: dt-bindings: Correct indentation and style in DTS example Kuninori Morimoto (18): ASoC: simple-card-utils: use __free(device_node) for device node ASoC: simple-card-utils: check port reg first on graph_get_dai_id() ASoC: simple-card-utils: use for_each_of_graph_port() on graph_get_dai_id() ASoC: soc-core: tidyup ret handling for card->disable_route_checks ASoC: soc-pcm: don't use soc_pcm_ret() on .prepare callback ASoC: simple-card-utils: Unify clock direction by clk_direction ASoC: dt-bindings: renesas,rsnd: remove post-init-providers property ASoC: soc-card: remove card check ASoC: intel: remove disable_route_checks ASoC: remove disable_route_checks ASoC: audio-graph-card2: use __free(device_node) for device node ASoC: audio-graph-card: use __free(device_node) for device node ASoC: simple-card: use __free(device_node) for device node ASoC: soc-core: return 0 if np was NULL on snd_soc_daifmt_parse_clock_provider_raw() ASoC: audio-graph-card2: use of_graph_get_port_by_id() at graph_get_next_multi_ep() ASoC: soc-core: Enable to use extra format on each DAI ASoC: audio-graph-card2: Use extra format on each DAI ASoC: soc-dapm: remove !card check from snd_soc_dapm_set_bias_level() Marcus Cooper (2): ASoC: sun4i-spdif: Always set the valid data to be the MSB ASoC: sun4i-spdif: Add working 24bit audio support Mark Brown (23): ASoC: fsl_xcvr: Add suspend and resume support ASoC: Correct *-objs usages Add support for codec of F1C100s ASoC: Splitting cs35l56 SoundWire DAI into separate ASoC: Merge up origin to resolve interaction with manline symbol changes ASoC: Merge up v6.12-rc2 ASoC: Intel: boards: updates for 6.14 ASoC: sun4i-spdif: Add 24bit support Add function to constrain rates ASoC: simple-card-utils: tidyup for Multi connection Add audio support for the Renesas RZ/G3S SoC ASoC: SOF: core/Intel: Handle pause supported token firmware: cirrus: Add KUnit tests for cs_dsp ASoC: fsl: add memory to memory function for ASRC ASoC: Intel: Add matches for Cirrus Logic CDB35L56 Minor debugging improvements ASoC: codecs: Add aw88083 amplifier driver ASoC: Merge up v6.13-rc6 ASoC: Intel: avs: Fixes and cleanups ASoC: remove disable_route_checks ASoC: fsl: Support MQS on i.MX943 ASoC: extra format on each DAI ASoC: fsl: Support micfil on i.MX943 Martin Blumenstingl (2): ASoC: soc-dai: add snd_soc_dai_prepare() and use it internally ASoC: dapm: add support for preparing streams Mesih Kilinc (2): ASoC: sun4i-codec: Add DMA Max Burst field ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s Michal Simek (1): ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled() Neil Armstrong (1): ASoC: dt-bindings: convert rt5682.txt to dt-schema Nícolas F. R. A. Prado (2): ASoC: dt-bindings: mediatek,mt8188-mt6359: Add compatible for mt8390 evk ASoC: dt-bindings: mediatek,mt8188-mt6359: Allow DL_SRC/UL_SRC dai-links Peter Ujfalusi (4): ASoC: SOF: Add support for pause supported tokens from topology ASoC: SOF: Intel: hda-pcm: Follow the pause_supported flag to drop PAUSE support ASoC: SOF: ipc4-topology: Use macro to set the EXT_PARAM_SIZE in widget setup ASoC: SOF: sof-priv: Remove unused SOF_DAI_STREAM() and SOF_FORMATS Richard Fitzgerald (21): ASoC: cs35l56: Split SoundWire DAI into separate playback and capture ASoC: sdw_utils: cs_amp: Assign non-overlapping TDM masks for each codec on a bus firmware: cs_dsp: Add mock regmap for KUnit testing firmware: cs_dsp: Add mock DSP memory map for KUnit testing firmware: cs_dsp: Add mock wmfw file generator for KUnit testing firmware: cs_dsp: Add mock bin file generator for KUnit testing firmware: cs_dsp: Add KUnit testing of bin file download firmware: cs_dsp: Add KUnit testing of wmfw download firmware: cs_dsp: Add KUnit testing of control parsing firmware: cs_dsp: Add KUnit testing of control cache firmware: cs_dsp: Add KUnit testing of control read/write firmware: cs_dsp: Add KUnit testing of bin error cases firmware: cs_dsp: Add KUnit testing of wmfw error cases firmware: cs_dsp: Add KUnit testing of client callbacks ASoC: Intel: tgl-match: Add entries for CS35L56 on CDB35L56-EIGHT-C ASoC: Intel: mtl-match: Add CDB35L56-EIGHT-C with aggregated speakers ASoC: Intel: mtl-match: Add CDB35L56-EIGHT-C 8x CS35L56 without CS42L43 firmware: cs_dsp: Fix kerneldoc typos in cs_dsp_mock_bin.c firmware: cs_dsp: Avoid using a u32 as a __be32 in cs_dsp_mock_mem_maps.c firmware: cs_dsp: Fix endianness conversion in cs_dsp_mock_wmfw.c firmware: cs_dsp: Delete redundant assignments in cs_dsp_test_bin.c Rob Herring (Arm) (2): ASoC: Use of_property_present() for non-boolean properties ASoC: simple-card: Drop unnecessary "dai-tdm-slot-width-map" property presence check Shengjiu Wang (13): ASoC: fsl_xcvr: Use regmap for PHY and PLL registers ASoC: fsl_xcvr: Add suspend and resume support ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device ASoC: fsl_asrc_m2m: force cast for snd_pcm_format_t type ASoC: fsl_mqs: Add i.MX943 platform support ASoC: dt-bindings: fsl,mqs: Add compatible string for i.MX943 platform ASoC: fsl_micfil: Add i.MX943 platform support ASoC: dt-bindings: fsl,micfil: Add compatible string for i.MX943 platform Simon Trimmer (4): ASoC: Intel: sof_sdw: Correct quirk for Lenovo Yoga Slim 7 ASoC: Intel: sof_sdw: Add a dev_dbg message for the SOC_SDW_CODEC_MIC quirk ASoC: Intel: soc-acpi: arl: Correct naming of a cs35l56 address struct ASoC: Intel: soc-acpi: arl: Add match entries for new cs42l43 laptops Stephen Gordon (2): ASoC: simple_card: Improve debugging messages ASoC: simple_card: Show if link is unidirectional Takashi Iwai (5): ASoC: cs40l50: Use *-y for Makefile ASoC: mediatek: mt8365: Use *-y for Makefile ASoC: SDCA: Use *-y for Makefile ASoC: cs42l84: Use *-y for Makefile ASoC: wcd937x: Use *-y for Makefile Tang Bin (2): ASoC: mediatek: mt8192-afe-pcm: Simplify probe() with local dev variable ASoC: tas2781: Fix redundant logical jump Thorsten Blum (4): ASoC: SOF: Intel: Use str_yes_no() helper in atom_dump() ASoC: madera: Use str_enabled_disabled() helper function ASoC: SOF: Intel: Use str_yes_no() to improve bdw_dump() ASoC: codecs: Use ARRAY_SIZE() to calculate PEB2466_TLV_SIZE Uwe Kleine-König (1): ASoC: Drop explicit initialization of struct i2c_device_id::driver_data to 0 Vijendar Mukunda (2): ASoC: amd: ps: update mach params subsystem_rev variable ASoC: amd: ps: add ZSC control register programming sequence Weidong Wang (2): ASoC: dt-bindings: Add schema for "awinic,aw88083" ASoC: codecs: Add aw88083 amplifier driver Wolfram Sang (1): ASoC: cs42l43: don't include '<linux/find.h>' directly Yu-Chun Lin (1): ASoC: amd: Add ACPI dependency to fix build error .../devicetree/bindings/sound/adi,ssm2518.yaml | 20 +- .../bindings/sound/allwinner,sun4i-a10-codec.yaml | 31 + .../devicetree/bindings/sound/awinic,aw88395.yaml | 2 + .../devicetree/bindings/sound/everest,es71x4.yaml | 10 +- .../devicetree/bindings/sound/everest,es7241.yaml | 19 +- .../devicetree/bindings/sound/fsl,easrc.yaml | 32 +- .../devicetree/bindings/sound/fsl,micfil.yaml | 1 + .../devicetree/bindings/sound/fsl,mqs.yaml | 2 + .../devicetree/bindings/sound/fsl,xcvr.yaml | 34 +- .../bindings/sound/intel,keembay-i2s.yaml | 32 +- .../bindings/sound/mediatek,mt8188-mt6359.yaml | 16 +- .../bindings/sound/neofidelity,ntp8918.yaml | 26 +- .../devicetree/bindings/sound/realtek,rt5682.yaml | 156 + .../devicetree/bindings/sound/renesas,rsnd.yaml | 6 - .../devicetree/bindings/sound/renesas,rz-ssi.yaml | 19 +- Documentation/devicetree/bindings/sound/rt5682.txt | 98 - .../devicetree/bindings/sound/ti,pcm6240.yaml | 32 +- .../devicetree/bindings/sound/ti,tas2562.yaml | 30 +- .../devicetree/bindings/sound/ti,tas2770.yaml | 34 +- .../devicetree/bindings/sound/ti,tas2781.yaml | 38 +- .../devicetree/bindings/sound/ti,tas27xx.yaml | 34 +- .../devicetree/bindings/sound/ti,tas57xx.yaml | 36 +- MAINTAINERS | 4 +- drivers/firmware/cirrus/Kconfig | 20 + drivers/firmware/cirrus/Makefile | 2 + drivers/firmware/cirrus/test/Makefile | 23 + drivers/firmware/cirrus/test/cs_dsp_mock_bin.c | 199 ++ .../firmware/cirrus/test/cs_dsp_mock_mem_maps.c | 752 +++++ drivers/firmware/cirrus/test/cs_dsp_mock_regmap.c | 367 +++ drivers/firmware/cirrus/test/cs_dsp_mock_utils.c | 13 + drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c | 473 +++ drivers/firmware/cirrus/test/cs_dsp_test_bin.c | 2556 +++++++++++++++ .../firmware/cirrus/test/cs_dsp_test_bin_error.c | 600 ++++ .../firmware/cirrus/test/cs_dsp_test_callbacks.c | 688 ++++ .../cirrus/test/cs_dsp_test_control_cache.c | 3282 ++++++++++++++++++++ .../cirrus/test/cs_dsp_test_control_parse.c | 1851 +++++++++++ .../firmware/cirrus/test/cs_dsp_test_control_rw.c | 2669 ++++++++++++++++ drivers/firmware/cirrus/test/cs_dsp_test_wmfw.c | 2211 +++++++++++++ .../firmware/cirrus/test/cs_dsp_test_wmfw_error.c | 1347 ++++++++ drivers/firmware/cirrus/test/cs_dsp_tests.c | 14 + include/dt-bindings/sound/qcom,wcd9335.h | 1 - include/linux/firmware/cirrus/cs_dsp_test_utils.h | 160 + include/sound/sdca.h | 7 +- include/sound/sdca_function.h | 3 + include/sound/simple_card_utils.h | 8 +- include/sound/soc-dai.h | 3 + include/sound/soc.h | 12 +- include/sound/soc_sdw_utils.h | 2 + include/uapi/sound/compress_params.h | 23 +- include/uapi/sound/sof/tokens.h | 2 + sound/soc/amd/Kconfig | 2 +- sound/soc/amd/ps/pci-ps.c | 16 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/Makefile | 8 +- sound/soc/codecs/ad193x-i2c.c | 3 +- sound/soc/codecs/adau1761-i2c.c | 5 +- sound/soc/codecs/adau1781-i2c.c | 5 +- sound/soc/codecs/adau1977-i2c.c | 5 +- sound/soc/codecs/alc5623.c | 10 +- sound/soc/codecs/alc5632.c | 6 +- sound/soc/codecs/aw88081.c | 333 +- sound/soc/codecs/aw88081.h | 43 + sound/soc/codecs/cs35l56.c | 8 +- sound/soc/codecs/cs42l43.c | 2 +- sound/soc/codecs/cs42l51-i2c.c | 6 +- sound/soc/codecs/cs42l84.c | 2 +- sound/soc/codecs/es8323.c | 2 +- sound/soc/codecs/madera.c | 7 +- sound/soc/codecs/max98088.c | 4 +- sound/soc/codecs/max98090.c | 18 +- sound/soc/codecs/max98095.c | 4 +- sound/soc/codecs/ntp8835.c | 2 +- sound/soc/codecs/ntp8918.c | 2 +- sound/soc/codecs/pcm186x-i2c.c | 3 +- sound/soc/codecs/pcm6240.c | 3 +- sound/soc/codecs/peb2466.c | 3 +- sound/soc/codecs/rt5682-i2c.c | 6 + sound/soc/codecs/rt5682.c | 12 +- sound/soc/codecs/rt5682.h | 2 + sound/soc/codecs/rt715-sdw.c | 41 - sound/soc/codecs/rt715.h | 3 - sound/soc/codecs/sma1307.c | 4 +- sound/soc/codecs/ssm2602-i2c.c | 5 +- sound/soc/codecs/tas2562.c | 4 +- sound/soc/codecs/tas2781-i2c.c | 71 +- sound/soc/codecs/tas5720.c | 10 +- sound/soc/codecs/tlv320adc3xxx.c | 4 +- sound/soc/codecs/tlv320aic31xx.c | 6 +- sound/soc/codecs/tlv320aic3x-i2c.c | 3 +- sound/soc/codecs/tpa6130a2.c | 4 +- sound/soc/codecs/uda1342.c | 2 +- sound/soc/codecs/wcd9335.c | 2 + sound/soc/codecs/wm8904.c | 13 +- sound/soc/codecs/wm8985.c | 4 +- sound/soc/fsl/Kconfig | 2 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/fsl_asrc.c | 179 +- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 + sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++ sound/soc/fsl/fsl_easrc.c | 261 +- sound/soc/fsl/fsl_easrc.h | 4 + sound/soc/fsl/fsl_micfil.c | 131 +- sound/soc/fsl/fsl_micfil.h | 2 +- sound/soc/fsl/fsl_mqs.c | 28 + sound/soc/fsl/fsl_sai.c | 7 +- sound/soc/fsl/fsl_sai.h | 3 + sound/soc/fsl/fsl_utils.c | 45 + sound/soc/fsl/fsl_utils.h | 5 + sound/soc/fsl/fsl_xcvr.c | 404 ++- sound/soc/fsl/fsl_xcvr.h | 13 + sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-card.c | 2 +- sound/soc/fsl/imx-rpmsg.c | 2 +- sound/soc/generic/audio-graph-card.c | 48 +- sound/soc/generic/audio-graph-card2.c | 262 +- sound/soc/generic/simple-card-utils.c | 79 +- sound/soc/generic/simple-card.c | 58 +- sound/soc/intel/avs/apl.c | 2 +- sound/soc/intel/avs/core.c | 24 +- sound/soc/intel/avs/debugfs.c | 1 + sound/soc/intel/avs/ipc.c | 25 +- sound/soc/intel/avs/loader.c | 36 +- sound/soc/intel/avs/messages.c | 22 +- sound/soc/intel/avs/messages.h | 3 +- sound/soc/intel/avs/pcm.c | 5 +- sound/soc/intel/avs/registers.h | 2 +- sound/soc/intel/avs/topology.c | 4 +- sound/soc/intel/avs/trace.h | 38 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 - sound/soc/intel/boards/sof_sdw.c | 33 +- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 45 +- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 70 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 289 +- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 148 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 194 ++ sound/soc/intel/keembay/kmb_platform.c | 2 +- sound/soc/mediatek/common/mtk-soundcard-driver.c | 4 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 19 +- sound/soc/mediatek/mt8365/Makefile | 2 +- sound/soc/mediatek/mt8365/mt8365-mt6357.c | 11 +- sound/soc/qcom/common.c | 6 +- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sdm845.c | 5 +- sound/soc/renesas/rz-ssi.c | 228 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 28 - sound/soc/sdca/Makefile | 2 +- sound/soc/sdca/sdca_device.c | 2 + sound/soc/sdca/sdca_functions.c | 132 +- sound/soc/sdw_utils/soc_sdw_cs_amp.c | 46 + sound/soc/sdw_utils/soc_sdw_utils.c | 13 +- sound/soc/soc-card.c | 4 +- sound/soc/soc-core.c | 58 +- sound/soc/soc-dai.c | 27 +- sound/soc/soc-dapm.c | 14 +- sound/soc/soc-pcm.c | 32 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/atom.c | 16 +- sound/soc/sof/intel/bdw.c | 16 +- sound/soc/sof/intel/hda-dai.c | 12 + sound/soc/sof/intel/hda-pcm.c | 15 + sound/soc/sof/intel/hda.c | 5 + sound/soc/sof/ipc4-topology.c | 2 +- sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/sof-priv.h | 8 - sound/soc/sof/topology.c | 4 + sound/soc/sunxi/sun4i-codec.c | 409 ++- sound/soc/sunxi/sun4i-spdif.c | 24 +- sound/soc/xilinx/xlnx_spdif.c | 38 +- 170 files changed, 21739 insertions(+), 1441 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5682.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt5682.txt create mode 100644 drivers/firmware/cirrus/test/Makefile create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_bin.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_mem_maps.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_regmap.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_utils.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_bin.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_bin_error.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_callbacks.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_control_cache.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_control_parse.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_control_rw.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_wmfw.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_wmfw_error.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_tests.c create mode 100644 include/linux/firmware/cirrus/cs_dsp_test_utils.h create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c

2 1

[PATCH V2 7/7] soundwire: amd: add soundwire host wake interrupt enable/disable sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

For wake event, SoundWire host wake interrupt will be asserted based on below pre-conditions for ACP7.0 & ACP7.1 platforms. - ACP device should be in D0 state. - SoundWire manager instance should be in D3 state. - SoundWire manager device state should be set to D3. - ACP_PME_EN should be set to 1. Implement code changes to enable/disable SoundWire host wake interrupt mask during suspend and resume as per design flow for ACP7.0 & ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 58 +++++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 2 ++ 2 files changed, 60 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 03c9e15dce5b..161d963c80e6 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -166,6 +166,34 @@ static int amd_sdw_set_device_state(struct amd_sdw_manager *amd_manager, u32 tar return 0; } +static int amd_sdw_host_wake_enable(struct amd_sdw_manager *amd_manager, bool enable) +{ + u32 intr_cntl1; + u32 sdw_host_wake_irq_mask; + + if (!amd_manager->wake_en_mask) + return 0; + + switch (amd_manager->instance) { + case ACP_SDW0: + sdw_host_wake_irq_mask = AMD_SDW0_HOST_WAKE_INTR_MASK; + break; + case ACP_SDW1: + sdw_host_wake_irq_mask = AMD_SDW1_HOST_WAKE_INTR_MASK; + break; + default: + return -EINVAL; + } + + intr_cntl1 = readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(ACP_SDW1)); + if (enable) + intr_cntl1 |= sdw_host_wake_irq_mask; + else + intr_cntl1 &= ~sdw_host_wake_irq_mask; + writel(intr_cntl1, amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(ACP_SDW1)); + return 0; +} + static void amd_sdw_ctl_word_prep(u32 *lower_word, u32 *upper_word, struct sdw_msg *msg, int cmd_offset) { @@ -1186,11 +1214,21 @@ static int __maybe_unused amd_suspend(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, false); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, false); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } /* * As per hardware programming sequence on AMD platforms, * clock stop should be invoked first before powering-off @@ -1224,11 +1262,21 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) } if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, true); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, true); + if (ret) + return ret; + } ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, true); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, true); + if (ret) + return ret; + } ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; @@ -1269,8 +1317,18 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) ret = amd_sdw_clock_stop_exit(amd_manager); if (ret) return ret; + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 1d5e94371f81..6cc916b0c820 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -194,6 +194,8 @@ #define AMD_SDW_CLK_RESUME_DONE 3 #define AMD_SDW_WAKE_STAT_MASK BIT(16) #define AMD_SDW_WAKE_INTR_MASK BIT(16) +#define AMD_SDW0_HOST_WAKE_INTR_MASK BIT(22) +#define AMD_SDW1_HOST_WAKE_INTR_MASK BIT(23) #define AMD_SDW_DEVICE_STATE 0x1430 #define AMD_SDW0_DEVICE_STATE_MASK GENMASK(1, 0) #define AMD_SDW1_DEVICE_STATE_MASK GENMASK(3, 2) -- 2.34.1

1 0

[PATCH V2 6/7] soundwire: amd: set ACP_PME_EN during runtime suspend sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Set ACP_PME_EN to 1 during runtime suspend sequence as per design flow for ACP7.0 & ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 9 +++++++++ drivers/soundwire/amd_manager.h | 1 + 2 files changed, 10 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 42d553878e91..03c9e15dce5b 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1215,6 +1215,7 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) struct amd_sdw_manager *amd_manager = dev_get_drvdata(dev); struct sdw_bus *bus = &amd_manager->bus; int ret; + u32 val; if (bus->prop.hw_disabled) { dev_dbg(bus->dev, "SoundWire manager %d is disabled,\n", @@ -1239,6 +1240,14 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D3); if (ret) return ret; + if (amd_manager->wake_en_mask) { + val = readl(amd_manager->acp_mmio + ACP_PME_EN); + if (!val) { + writel(1, amd_manager->acp_mmio + ACP_PME_EN); + val = readl(amd_manager->acp_mmio + ACP_PME_EN); + dev_dbg(amd_manager->dev, "ACP_PME_EN:0x%x\n", val); + } + } } return 0; } diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 8430f279d88e..1d5e94371f81 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -199,6 +199,7 @@ #define AMD_SDW1_DEVICE_STATE_MASK GENMASK(3, 2) #define AMD_SDW_DEVICE_STATE_D0 0 #define AMD_SDW_DEVICE_STATE_D3 3 +#define ACP_PME_EN 0x0001400 static u32 amd_sdw_freq_tbl[AMD_SDW_MAX_FREQ_NUM] = { AMD_SDW_DEFAULT_CLK_FREQ, -- 2.34.1

1 0

[PATCH V2 5/7] soundwire: amd: set device power state during suspend/resume sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Set SoundWire manager device power state during suspend and resume sequence for ACP7.0 & ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 58 ++++++++++++++++++++++++++++++--- drivers/soundwire/amd_manager.h | 5 +++ 2 files changed, 58 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index d611498d5488..42d553878e91 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -143,6 +143,29 @@ static void amd_sdw_wake_enable(struct amd_sdw_manager *amd_manager, bool enable writel(wake_ctrl, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_8TO11); } +static int amd_sdw_set_device_state(struct amd_sdw_manager *amd_manager, u32 target_device_state) +{ + u32 sdw_dev_state; + + sdw_dev_state = readl(amd_manager->acp_mmio + AMD_SDW_DEVICE_STATE); + switch (amd_manager->instance) { + case ACP_SDW0: + u32p_replace_bits(&sdw_dev_state, target_device_state, + AMD_SDW0_DEVICE_STATE_MASK); + break; + case ACP_SDW1: + u32p_replace_bits(&sdw_dev_state, target_device_state, + AMD_SDW1_DEVICE_STATE_MASK); + break; + default: + return -EINVAL; + } + writel(sdw_dev_state, amd_manager->acp_mmio + AMD_SDW_DEVICE_STATE); + sdw_dev_state = readl(amd_manager->acp_mmio + AMD_SDW_DEVICE_STATE); + dev_dbg(amd_manager->dev, "AMD_SDW_DEVICE_STATE:0x%x\n", sdw_dev_state); + return 0; +} + static void amd_sdw_ctl_word_prep(u32 *lower_word, u32 *upper_word, struct sdw_msg *msg, int cmd_offset) { @@ -1163,7 +1186,9 @@ static int __maybe_unused amd_suspend(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, false); - return amd_sdw_clock_stop(amd_manager); + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, false); /* @@ -1173,7 +1198,14 @@ static int __maybe_unused amd_suspend(struct device *dev) ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; - return amd_deinit_sdw_manager(amd_manager); + ret = amd_deinit_sdw_manager(amd_manager); + if (ret) + return ret; + } + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D3); + if (ret) + return ret; } return 0; } @@ -1191,13 +1223,22 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) } if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, true); - return amd_sdw_clock_stop(amd_manager); + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, true); ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; - return amd_deinit_sdw_manager(amd_manager); + ret = amd_deinit_sdw_manager(amd_manager); + if (ret) + return ret; + } + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D3); + if (ret) + return ret; } return 0; } @@ -1216,7 +1257,9 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) } if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { - return amd_sdw_clock_stop_exit(amd_manager); + ret = amd_sdw_clock_stop_exit(amd_manager); + if (ret) + return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); @@ -1239,6 +1282,11 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) return ret; amd_sdw_set_frameshape(amd_manager); } + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D0); + if (ret) + return ret; + } return 0; } diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 30244a31c21c..8430f279d88e 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -194,6 +194,11 @@ #define AMD_SDW_CLK_RESUME_DONE 3 #define AMD_SDW_WAKE_STAT_MASK BIT(16) #define AMD_SDW_WAKE_INTR_MASK BIT(16) +#define AMD_SDW_DEVICE_STATE 0x1430 +#define AMD_SDW0_DEVICE_STATE_MASK GENMASK(1, 0) +#define AMD_SDW1_DEVICE_STATE_MASK GENMASK(3, 2) +#define AMD_SDW_DEVICE_STATE_D0 0 +#define AMD_SDW_DEVICE_STATE_D3 3 static u32 amd_sdw_freq_tbl[AMD_SDW_MAX_FREQ_NUM] = { AMD_SDW_DEFAULT_CLK_FREQ, -- 2.34.1

1 0

[PATCH V2 4/7] soundwire: amd: add support for ACP7.0 & ACP7.1 platforms
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Add SoundWire support for ACP7.0 and ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 22 ++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 18 ++++++++++++++++++ include/linux/soundwire/sdw_amd.h | 2 ++ 3 files changed, 42 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 6831b3729db5..d611498d5488 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -446,6 +446,10 @@ static int amd_sdw_port_params(struct sdw_bus *bus, struct sdw_port_params *p_pa return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + frame_fmt_reg = acp70_sdw_dp_reg[p_params->num].frame_fmt_reg; + break; default: return -EINVAL; } @@ -494,6 +498,14 @@ static int amd_sdw_transport_params(struct sdw_bus *bus, return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + frame_fmt_reg = acp70_sdw_dp_reg[params->port_num].frame_fmt_reg; + sample_int_reg = acp70_sdw_dp_reg[params->port_num].sample_int_reg; + hctrl_dp0_reg = acp70_sdw_dp_reg[params->port_num].hctrl_dp0_reg; + offset_reg = acp70_sdw_dp_reg[params->port_num].offset_reg; + lane_ctrl_ch_en_reg = acp70_sdw_dp_reg[params->port_num].lane_ctrl_ch_en_reg; + break; default: return -EINVAL; } @@ -549,6 +561,10 @@ static int amd_sdw_port_enable(struct sdw_bus *bus, return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + lane_ctrl_ch_en_reg = acp70_sdw_dp_reg[enable_ch->port_num].lane_ctrl_ch_en_reg; + break; default: return -EINVAL; } @@ -898,6 +914,7 @@ int amd_sdw_manager_start(struct amd_sdw_manager *amd_manager) ret = amd_enable_sdw_manager(amd_manager); if (ret) return ret; + amd_sdw_set_frameshape(amd_manager); } /* Enable runtime PM */ @@ -969,6 +986,11 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + amd_manager->num_dout_ports = AMD_ACP70_SDW_MAX_TX_PORTS; + amd_manager->num_din_ports = AMD_ACP70_SDW_MAX_RX_PORTS; + break; default: return -EINVAL; } diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index cc2170e4521e..30244a31c21c 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -159,8 +159,11 @@ #define AMD_ACP63_SDW0_MAX_RX_PORTS 3 #define AMD_ACP63_SDW1_MAX_TX_PORTS 1 #define AMD_ACP63_SDW1_MAX_RX_PORTS 1 +#define AMD_ACP70_SDW_MAX_TX_PORTS 3 +#define AMD_ACP70_SDW_MAX_RX_PORTS 3 #define AMD_ACP63_SDW0_MAX_DAI 6 #define AMD_ACP63_SDW1_MAX_DAI 2 +#define AMD_ACP70_SDW_MAX_DAI 6 #define AMD_SDW_SLAVE_0_ATTACHED 5 #define AMD_SDW_SSP_COUNTER_VAL 3 @@ -244,6 +247,21 @@ static struct sdw_manager_dp_reg acp63_sdw1_dp_reg[AMD_ACP63_SDW1_MAX_DAI] = { ACP_SW_AUDIO1_RX_OFFSET, ACP_SW_AUDIO1_RX_CHANNEL_ENABLE_DP0} }; +static struct sdw_manager_dp_reg acp70_sdw_dp_reg[AMD_ACP70_SDW_MAX_DAI] = { + {ACP_SW_AUDIO0_TX_FRAME_FORMAT, ACP_SW_AUDIO0_TX_SAMPLEINTERVAL, ACP_SW_AUDIO0_TX_HCTRL_DP0, + ACP_SW_AUDIO0_TX_OFFSET_DP0, ACP_SW_AUDIO0_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO1_TX_FRAME_FORMAT, ACP_SW_AUDIO1_TX_SAMPLEINTERVAL, ACP_SW_AUDIO1_TX_HCTRL, + ACP_SW_AUDIO1_TX_OFFSET, ACP_SW_AUDIO1_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO2_TX_FRAME_FORMAT, ACP_SW_AUDIO2_TX_SAMPLEINTERVAL, ACP_SW_AUDIO2_TX_HCTRL, + ACP_SW_AUDIO2_TX_OFFSET, ACP_SW_AUDIO2_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO0_RX_FRAME_FORMAT, ACP_SW_AUDIO0_RX_SAMPLEINTERVAL, ACP_SW_AUDIO0_RX_HCTRL_DP0, + ACP_SW_AUDIO0_RX_OFFSET_DP0, ACP_SW_AUDIO0_RX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO1_RX_FRAME_FORMAT, ACP_SW_AUDIO1_RX_SAMPLEINTERVAL, ACP_SW_AUDIO1_RX_HCTRL, + ACP_SW_AUDIO1_RX_OFFSET, ACP_SW_AUDIO1_RX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO2_RX_FRAME_FORMAT, ACP_SW_AUDIO2_RX_SAMPLEINTERVAL, ACP_SW_AUDIO2_RX_HCTRL, + ACP_SW_AUDIO2_RX_OFFSET, ACP_SW_AUDIO2_RX_CHANNEL_ENABLE_DP0}, +}; + static u32 sdw_manager_reg_mask_array[AMD_SDW_MAX_MANAGER_COUNT] = { AMD_SDW0_EXT_INTR_MASK, AMD_SDW1_EXT_INTR_MASK diff --git a/include/linux/soundwire/sdw_amd.h b/include/linux/soundwire/sdw_amd.h index 799f8578137b..6b839987f14c 100644 --- a/include/linux/soundwire/sdw_amd.h +++ b/include/linux/soundwire/sdw_amd.h @@ -28,6 +28,8 @@ #define ACP_SDW1 1 #define AMD_SDW_MAX_MANAGER_COUNT 2 #define ACP63_PCI_REV_ID 0x63 +#define ACP70_PCI_REV_ID 0x70 +#define ACP71_PCI_REV_ID 0x71 struct acp_sdw_pdata { u16 instance; -- 2.34.1

1 0

[PATCH V2 2/7] soundwire: amd: add debug log for soundwire wake event
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Add debug log in amd_sdw_process_wake_event() function. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 187006e68ad6..60be5805715e 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -849,6 +849,7 @@ static void amd_sdw_update_slave_status(u32 status_change_0to7, u32 status_chang static void amd_sdw_process_wake_event(struct amd_sdw_manager *amd_manager) { + dev_dbg(amd_manager->dev, "SoundWire Wake event reported\n"); pm_request_resume(amd_manager->dev); writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_8TO11); -- 2.34.1

1 0

[PATCH V2 1/7] soundwire: amd: change the soundwire wake enable/disable sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

During runtime suspend scenario, SoundWire wake should be enabled and during system level suspend scenario SoundWire wake should be disabled. Implement the SoundWire wake enable/disable sequence as per design flow for SoundWire poweroff mode. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 71b3cf2df89d..187006e68ad6 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1139,6 +1139,7 @@ static int __maybe_unused amd_suspend(struct device *dev) amd_sdw_wake_enable(amd_manager, false); return amd_sdw_clock_stop(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + amd_sdw_wake_enable(amd_manager, false); /* * As per hardware programming sequence on AMD platforms, * clock stop should be invoked first before powering-off @@ -1166,6 +1167,7 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) amd_sdw_wake_enable(amd_manager, true); return amd_sdw_clock_stop(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + amd_sdw_wake_enable(amd_manager, true); ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; -- 2.34.1

1 0

[PATCH v1 1/3] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 20 Jan '25

20 Jan '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - update the year from 2024 to 2025. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. --- include/sound/tas2781-dsp.h | 26 +++++++++++++++++++++++--- include/sound/tas2781.h | 13 ++++++++++++- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..5cfbb9e0a626 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); -- 2.34.1

1 2

Is function delay of the pulse plugin correct?
by Pavel Hofman 19 Jan '25

19 Jan '25

Hi, Please correct me if I am wrong: "avail" means (buffer - buffer_fill), while "delay" means (buffer_fill + device/downstream delay/latency). Some drivers/devices do not track the device delay, and then delay = buffer_fill. While troubleshooting some playback code, I hit that snd_pcm_delay on the plugin pulse (device pulse) was always returning 0 frames, while snd_pcm_avail was returning dynamic (correct) values. IIUC while pulse_pcm.c uses generic code for avail, it overrides the generic snd_pcm_ioplug_delay https://github.com/alsa-project/alsa-lib/blob/35d2efefa95d28e0cd963a06e051f… with specific pulse_delay https://github.com/alsa-project/alsa-plugins/blob/52574cb5ccbb8b546df2759e4… . IIUC this method returns only delay obtained from Pulseaudio, i.e. no fill of the pulse plugin buffer is included. And IIUC in this specific case pulseaudio keeps returning zero for the pa_stream_get_latency call, making snd_pcm_delay return zero too. Therefore the equation (buffer size = avail + delay - driver/device delay) does not seem to hold for the pulse plugin. The new pipewire plugin does not seem to override the generic snd_pcm_ioplug_delay method https://github.com/PipeWire/pipewire-alsa/blob/master/alsa-plugins/pcm_pipe… . Please, is that behavior correct? With regards, Pavel.

1 0

[tiwai-sound:for-next 50/52] sound/usb/fcp.c:889:15: error: incompatible function pointer types assigning to 'long (*)(struct snd_hwdep *, char *, long, loff_t *)' (aka 'long (*)(struct snd_hwdep *, char *, long, long long *)') from 'ssize_t (struct snd_hwdep *, char *, ssize_t, loff...
by kernel test robot 19 Jan '25

19 Jan '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: dd85bbb7fc482b5cdb78f8b5e8f9ef7a3919d6ef commit: 46757a3e7d50dac923888e7fbe68377736f13c70 [50/52] ALSA: FCP: Add Focusrite Control Protocol driver config: hexagon-allmodconfig (https://download.01.org/0day-ci/archive/20250119/202501192122.ngosycnT-lkp@…) compiler: clang version 20.0.0git (https://github.com/llvm/llvm-project c23f2417dc5f6dc371afb07af5627ec2a9d373a0) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250119/202501192122.ngosycnT-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202501192122.ngosycnT-lkp@intel.com/ All errors (new ones prefixed by >>): >> sound/usb/fcp.c:889:15: error: incompatible function pointer types assigning to 'long (*)(struct snd_hwdep *, char *, long, loff_t *)' (aka 'long (*)(struct snd_hwdep *, char *, long, long long *)') from 'ssize_t (struct snd_hwdep *, char *, ssize_t, loff_t *)' (aka 'int (struct snd_hwdep *, char *, int, long long *)') [-Wincompatible-function-pointer-types] 889 | hw->ops.read = fcp_hwdep_read; | ^ ~~~~~~~~~~~~~~ 1 error generated. vim +889 sound/usb/fcp.c 874 875 static int fcp_hwdep_init(struct usb_mixer_interface *mixer) 876 { 877 struct snd_hwdep *hw; 878 int err; 879 880 err = snd_hwdep_new(mixer->chip->card, "Focusrite Control", 0, &hw); 881 if (err < 0) 882 return err; 883 884 hw->private_data = mixer; 885 hw->exclusive = 1; 886 hw->ops.open = fcp_hwdep_open; 887 hw->ops.ioctl = fcp_hwdep_ioctl; 888 hw->ops.ioctl_compat = fcp_hwdep_ioctl; > 889 hw->ops.read = fcp_hwdep_read; 890 hw->ops.poll = fcp_hwdep_poll; 891 hw->ops.release = fcp_hwdep_release; 892 893 return 0; 894 } 895 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 49/52] sound/pci/hda/tas2781_hda_spi.c:110:6: warning: variable 'ret' is used uninitialized whenever 'if' condition is false
by kernel test robot 19 Jan '25

19 Jan '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: dd85bbb7fc482b5cdb78f8b5e8f9ef7a3919d6ef commit: bb5f86ea50ffb292f42eb1ebdb99991d5c5ac3ba [49/52] ALSA: hda/tas2781: Add tas2781 hda SPI driver config: arm64-allmodconfig (https://download.01.org/0day-ci/archive/20250119/202501192006.Hm9GmKiV-lkp@…) compiler: clang version 18.1.8 (https://github.com/llvm/llvm-project 3b5b5c1ec4a3095ab096dd780e84d7ab81f3d7ff) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250119/202501192006.Hm9GmKiV-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202501192006.Hm9GmKiV-lkp@intel.com/ All warnings (new ones prefixed by >>): >> sound/pci/hda/tas2781_hda_spi.c:110:6: warning: variable 'ret' is used uninitialized whenever 'if' condition is false [-Wsometimes-uninitialized] 110 | if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/tas2781_hda_spi.c:119:9: note: uninitialized use occurs here 119 | return ret; | ^~~ sound/pci/hda/tas2781_hda_spi.c:110:2: note: remove the 'if' if its condition is always true 110 | if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/tas2781_hda_spi.c:108:9: note: initialize the variable 'ret' to silence this warning 108 | int ret; | ^ | = 0 1 warning generated. vim +110 sound/pci/hda/tas2781_hda_spi.c 104 105 static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) 106 { 107 struct regmap *map = tas_priv->regmap; 108 int ret; 109 > 110 if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { 111 ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, 112 TASDEVICE_BOOK_ID(reg)); 113 if (ret < 0) { 114 dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); 115 return ret; 116 } 117 tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); 118 } 119 return ret; 120 } 121 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 50/51] sound/usb/fcp.c:889:22: error: assignment to 'long int (*)(struct snd_hwdep *, char *, long int, loff_t *)' {aka 'long int (*)(struct snd_hwdep *, char *, long int, long long int *)'} from incompatible pointer type 'ssize_t (*)(struct snd_hwdep *, char ...
by kernel test robot 18 Jan '25

18 Jan '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 0ce204d3af3beca1825018e9ca128635ccc8aa85 commit: 46757a3e7d50dac923888e7fbe68377736f13c70 [50/51] ALSA: FCP: Add Focusrite Control Protocol driver config: i386-buildonly-randconfig-004-20250119 (https://download.01.org/0day-ci/archive/20250119/202501190536.A2CiOP0a-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250119/202501190536.A2CiOP0a-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202501190536.A2CiOP0a-lkp@intel.com/ All errors (new ones prefixed by >>): sound/usb/fcp.c: In function 'fcp_hwdep_init': >> sound/usb/fcp.c:889:22: error: assignment to 'long int (*)(struct snd_hwdep *, char *, long int, loff_t *)' {aka 'long int (*)(struct snd_hwdep *, char *, long int, long long int *)'} from incompatible pointer type 'ssize_t (*)(struct snd_hwdep *, char *, ssize_t, loff_t *)' {aka 'int (*)(struct snd_hwdep *, char *, int, long long int *)'} [-Werror=incompatible-pointer-types] 889 | hw->ops.read = fcp_hwdep_read; | ^ cc1: some warnings being treated as errors vim +889 sound/usb/fcp.c 874 875 static int fcp_hwdep_init(struct usb_mixer_interface *mixer) 876 { 877 struct snd_hwdep *hw; 878 int err; 879 880 err = snd_hwdep_new(mixer->chip->card, "Focusrite Control", 0, &hw); 881 if (err < 0) 882 return err; 883 884 hw->private_data = mixer; 885 hw->exclusive = 1; 886 hw->ops.open = fcp_hwdep_open; 887 hw->ops.ioctl = fcp_hwdep_ioctl; 888 hw->ops.ioctl_compat = fcp_hwdep_ioctl; > 889 hw->ops.read = fcp_hwdep_read; 890 hw->ops.poll = fcp_hwdep_poll; 891 hw->ops.release = fcp_hwdep_release; 892 893 return 0; 894 } 895 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

No audio on Acer Swift SF315-52
by GitHub issues - opened 18 Jan '25

18 Jan '25

alsa-project/alsa-ucm-conf issue #497 was opened from pavethem: [Alsa info](http://alsa-project.org/db/?f=8e6dc59510d7aaa42899f81a8f29a740032e9ce… Ever since kernel 6.12 dropped for Fedora, my laptop only shows a dummy output. This is probably due to the kernel removing legacy sound drivers and replacing them with avs, which doesn’t seem work for the kaby lake architecture present in this model. ([phoronix](https://www.phoronix.com/news/Linux-6.12-Sound-Drivers)). I am unsure whether or not I am filing this bug in the right place. Since I am able to get the microphone to work by copying a file: `cp /usr/share/alsa/ucm2/conf.d/avs_dmic/Acer-Lars-1.0.conf /usr/share/alsa/ucm2/conf.d/avs_dmic/avs_dmic.conf` I am guessing this is a case of a missing ucm file for this particular hardware. If this is incorrect, I would appreciate if someone could point me in the right direction and tell me if this is rather a bug with alsa-firmware or the kernel itself. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/497 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.13-rc7
by Mark Brown 18 Jan '25

18 Jan '25

The following changes since commit 139fa599cea0fd9d38e00246ea9f79af6c59acbd: ASoC: rsnd: check rsnd_adg_clk_enable() return value (2025-01-09 13:17:02 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc7 for you to fetch changes up to 75a7f7d6e7e10b9052d287be37f965103e1abae1: ASoC: Intel: sof_sdw: Fix DMI match entries for a (2025-01-15 19:50:01 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A few final driver specific fixes - a couple of x86 ID field changes, plus bug fixes for simple-card-utils and nau8824. ---------------------------------------------------------------- Laurentiu Mihalcea (1): ASoC: simple-card-utils: fix priv->dai_props indexing Mark Brown (1): ASoC: Intel: sof_sdw: Fix DMI match entries for a Maxim Kochetkov (1): ASoC: codecs: nau8824: fix max volume for Speaker output Simon Trimmer (2): ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83LC ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83JX, 83MC and 83NM include/sound/simple_card_utils.h | 7 +++++++ sound/soc/codecs/nau8824.c | 8 ++++---- sound/soc/generic/simple-card-utils.c | 10 +++++----- sound/soc/intel/boards/sof_sdw.c | 23 +++++++++++++++++++---- 4 files changed, 35 insertions(+), 13 deletions(-)

2 1

RE: [REGRESSION] Distorted sound on Acer Aspire A115-31 laptop
by Kailang 17 Jan '25

17 Jan '25

Attached. > -----Original Message----- > From: Takashi Iwai <tiwai(a)suse.de> > Sent: Thursday, January 16, 2025 10:27 PM > To: Evgeny Kapun <abacabadabacaba(a)gmail.com> > Cc: Kailang <kailang(a)realtek.com>; Takashi Iwai <tiwai(a)suse.de>; Linux > Sound Mailing List <linux-sound(a)vger.kernel.org>; Linux Kernel Mailing List > <linux-kernel(a)vger.kernel.org>; Linux Regressions Mailing List > <regressions(a)lists.linux.dev>; Linux Stable Mailing List > <stable(a)vger.kernel.org> > Subject: Re: [REGRESSION] Distorted sound on Acer Aspire A115-31 laptop > > > External mail. > > > > On Sat, 11 Jan 2025 16:00:33 +0100, > Evgeny Kapun wrote: > > > > On 12/24/24 04:54, Kailang wrote: > > > Please test attach patch. > > > > This patch, when applied to kernel version 6.12.8, appears to fix the > > issue. There are no distortions, and the left and the right channel > > can be controlled independently. > > Good to hear. > > Kailang, care to submit a proper patch for merging? > > > thanks, > > Takashi

2 1

Topology: NHLT: Intel: Fix DMA slots config in SSP blob
by GitHub pull_request - opened 16 Jan '25

16 Jan '25

alsa-project/alsa-utils pull request #289 was opened from singalsu: This fixes a mistake in function ssp_calculate_intern_v30(). The rx_dir[0] and tx_dir[0] value was hard-coded to 0x3 for typical stereo slots configuration. It should be instead retrieved from topology parameters for the SSP. Request URL : https://github.com/alsa-project/alsa-utils/pull/289 Patch URL : https://github.com/alsa-project/alsa-utils/pull/289.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

no internal sound with snd_hda_intel, ALC256/ Lenovo ThinkCentre M90a Pro Gen4
by Ronny Sattler 15 Jan '25

15 Jan '25

Hello, I have a problem with the snd-hda-intel. There is no sound. Only when I plug in headphones do I hear the sound on the headphones. The speakers are onboard (All in One - PC) and always remain mute. I suspect it is because the amplifier is not activated or a PIN assignment is incorrect. Please let me know if there is anything else I can do for you on the PC. Attached are previous log files and general information. Thank you very much for your work :-) Best regards Ronny

1 0

[RFC PATCH] soc: audio-graph-card2: use correct endpoint when getting link parameters
by Ivaylo Dimitrov 15 Jan '25

15 Jan '25

We may have multiple links between ports, with each link having different parameters. Currently, no matter the topology, it is always port endpoint 0 that is used when setting parameters. On a complex sound system, like the one found on Motorola droid4, hifi and voice DAIs require differents formats (i2s vs dsp_a) and curently it is impossible to use DT to set that. Implementing the change leads to partially dropping of at least 0dedbde5062d (ASoC: cpcap: Implement set_tdm_slot for voice call support), as core does most of what is needed to configure voice DAI. We (on Maemo Leste ) use the patch (along with few others) to have voice calls working properly on d4 through UCM. The patch is for linux 6.6, I want to know whether the approach would be accepted before sending a proper patch for current master. the original commit message follows: When link parameters are parsed, it is always endpoint@0 that is used and parameters set to other endpoints are ignored. Fix that by using endpoint that is set in DT when parsing link parameters. Signed-off-by: Ivaylo Dimitrov <ivo.g.dimitrov.75(a)gmail.com> --- sound/soc/generic/audio-graph-card2.c | 59 +++++++++++++-------------- 1 file changed, 28 insertions(+), 31 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index b1c675c6b6db..163a20c8ffee 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -508,17 +508,16 @@ static int __graph_parse_node(struct asoc_simple_priv *priv, static int graph_parse_node(struct asoc_simple_priv *priv, enum graph_type gtype, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu) { - struct device_node *ep; int ret = 0; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); - if (graph_lnk_is_multi(port)) { + if (is_multi) { int idx; - of_node_get(port); - for (idx = 0;; idx++) { ep = graph_get_next_multi_ep(&port); if (!ep) @@ -532,9 +531,8 @@ static int graph_parse_node(struct asoc_simple_priv *priv, } } else { /* Single CPU / Codec */ - ep = port_to_endpoint(port); + of_node_put(port); ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, 0); - of_node_put(ep); } return ret; @@ -591,22 +589,20 @@ static void graph_parse_daifmt(struct device_node *node, } static void graph_link_init(struct asoc_simple_priv *priv, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu_node) { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct device_node *ep; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); struct device_node *ports; unsigned int daifmt = 0, daiclk = 0; unsigned int bit_frame = 0; - if (graph_lnk_is_multi(port)) { - of_node_get(port); + if (is_multi) { ep = graph_get_next_multi_ep(&port); port = of_get_parent(ep); - } else { - ep = port_to_endpoint(port); } ports = of_get_parent(port); @@ -642,6 +638,9 @@ static void graph_link_init(struct asoc_simple_priv *priv, dai_link->ops = &graph_ops; if (priv->ops) dai_link->ops = priv->ops; + + of_node_put(port); + of_node_put(ports); } int audio_graph2_link_normal(struct asoc_simple_priv *priv, @@ -650,7 +649,7 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, { struct device_node *cpu_port = lnk; struct device_node *cpu_ep = port_to_endpoint(cpu_port); - struct device_node *codec_port = of_graph_get_remote_port(cpu_ep); + struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; /* @@ -658,20 +657,20 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_NORMAL, codec_port, li, 0); + ret = graph_parse_node(priv, GRAPH_NORMAL, codec_ep, li, 0); if (ret < 0) goto err; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_port, li, 1); + ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_ep, li, 1); if (ret < 0) goto err; - graph_link_init(priv, cpu_port, li, 1); + graph_link_init(priv, cpu_ep, li, 1); err: - of_node_put(codec_port); + of_node_put(codec_ep); of_node_put(cpu_ep); return ret; @@ -684,7 +683,6 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, { struct device_node *ep = port_to_endpoint(lnk); struct device_node *rep = of_graph_get_remote_endpoint(ep); - struct device_node *rport = of_graph_get_remote_port(ep); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); int is_cpu = asoc_graph_is_ports0(lnk); @@ -718,7 +716,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 1); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 1); if (ret) goto err; } else { @@ -751,7 +749,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 0); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 0); if (ret < 0) goto err; } @@ -761,11 +759,10 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, snd_soc_dai_link_set_capabilities(dai_link); - graph_link_init(priv, rport, li, is_cpu); + graph_link_init(priv, rep, li, is_cpu); err: of_node_put(ep); of_node_put(rep); - of_node_put(rport); return ret; } @@ -777,7 +774,7 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct device_node *port0, *port1, *ports; - struct device_node *codec0_port, *codec1_port; + struct device_node *codec0_ep, *codec1_ep; struct device_node *ep0, *ep1; u32 val = 0; int ret = -EINVAL; @@ -834,31 +831,31 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ep0 = port_to_endpoint(port0); ep1 = port_to_endpoint(port1); - codec0_port = of_graph_get_remote_port(ep0); - codec1_port = of_graph_get_remote_port(ep1); + codec0_ep = of_graph_get_remote_endpoint(ep0); + codec1_ep = of_graph_get_remote_endpoint(ep1); /* * call Codec first. * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_C2C, codec1_port, li, 0); + ret = graph_parse_node(priv, GRAPH_C2C, codec1_ep, li, 0); if (ret < 0) goto err2; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_C2C, codec0_port, li, 1); + ret = graph_parse_node(priv, GRAPH_C2C, codec0_ep, li, 1); if (ret < 0) goto err2; - graph_link_init(priv, codec0_port, li, 1); + graph_link_init(priv, codec0_ep, li, 1); err2: of_node_put(ep0); of_node_put(ep1); - of_node_put(codec0_port); - of_node_put(codec1_port); + of_node_put(codec0_ep); + of_node_put(codec1_ep); err1: of_node_put(ports); of_node_put(port0); -- 2.30.2

3 9

[PATCH v2 0/5] Updates for rawmidi / sequencer
by Takashi Iwai 14 Jan '25

14 Jan '25

The kernel 6.14 will receive a few small enhancements for rawmidi and sequencer APIs, and this patch set covers those. v1->v2: sync with the latest rawmidi API change to pass 0 to tied_device as default v1: https://lore.kernel.org/20250114085515.11880-1-tiwai@suse.de Takashi === Takashi Iwai (5): Sync UAPI asound.h with 6.14 kernel Sync UAPI asequencer.h with 6.14 kernel rawmidi: Extensions for tied device and substream inactive flag rawmidi: Make rawmidi flag bits doxygen-style comments seq: Define new events for UMP EP/FB change notifications include/local.h | 1 + include/rawmidi.h | 4 +++- include/seq_event.h | 12 ++++++++++++ include/sound/uapi/asequencer.h | 12 +++++++++++- include/sound/uapi/asound.h | 8 ++++++-- src/rawmidi/rawmidi.c | 16 ++++++++++++++++ 6 files changed, 49 insertions(+), 4 deletions(-) -- 2.43.0

1 5

[PATCH alsa-lib 0/5] Updates for rawmidi / sequencer
by Takashi Iwai 14 Jan '25

14 Jan '25

The kernel 6.14 will receive a few small enhancements for rawmidi and sequencer APIs, and this patch set covers those. Takashi === Takashi Iwai (5): Sync UAPI asound.h with 6.14 kernel Sync UAPI asequencer.h with 6.14 kernel rawmidi: Extensions for tied device and substream inactive flag rawmidi: Make rawmidi flag bits doxygen-style comments seq: Define new events for UMP EP/FB change notifications include/local.h | 1 + include/rawmidi.h | 4 +++- include/seq_event.h | 12 ++++++++++++ include/sound/uapi/asequencer.h | 12 +++++++++++- include/sound/uapi/asound.h | 8 ++++++-- src/rawmidi/rawmidi.c | 22 +++++++++++++++++++++- 6 files changed, 54 insertions(+), 5 deletions(-) -- 2.43.0

1 5

rt722: change output volume of headphone to 0dB
by GitHub pull_request - opened 14 Jan '25

14 Jan '25

alsa-project/alsa-ucm-conf pull request #496 was opened from shumingfan: rt722: change output volume of headphone to 0dB Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/496 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/496.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 1/1] ASoC: codecs: nau8824: fix max volume for Speaker output
by Maxim Kochetkov 13 Jan '25

13 Jan '25

There is no audio output if Speaker volume is set above 25. According to datasheet Rev 2.5 maximum allowed value for the Speaker output is 0b11001 (25) 0x6D CLASSD_GAIN_1/ 0x6E CLASSD_GAIN_2: Left/Right Channel Class-D Driver Gain For DAC Left/Right Input (Step size is 1dB.) 00000 = Mute (DEFAULT) 00001 = 0dB 00002 = 1dB ... 11000 = 23dB 11001 = 24dB So adjust this value in accordance with the datasheet. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/nau8824.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 12540397fd4d..5aaf8c496300 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -368,13 +368,13 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), SOC_SINGLE_TLV("Speaker Right DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Right DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Headphone Right DACR Volume", NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), -- 2.45.2

2 1

speaker-test.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 11 Jan '25

11 Jan '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' <file>" to find trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:183: misuse, warning: .BR is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.6-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-4 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

[GIT PULL] ASoC fixes for v6.13-rc6
by Mark Brown 10 Jan '25

10 Jan '25

The following changes since commit 32c9c06adb5b157ef259233775a063a43746d699: ASoC: mediatek: disable buffer pre-allocation (2024-12-19 11:15:09 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc6 for you to fetch changes up to 139fa599cea0fd9d38e00246ea9f79af6c59acbd: ASoC: rsnd: check rsnd_adg_clk_enable() return value (2025-01-09 13:17:02 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A collection of device specific fixes that came in over the holidays, plus a MAINTAINERS update and some documentation to help users debug problems with some of the Cirrus CODECs found in modern laptops. ---------------------------------------------------------------- Charles Keepax (3): ASoC: wm8994: Add depends on MFD core ASoC: samsung: Add missing selects for MFD_WM8994 ASoC: samsung: Add missing depends on I2C Kuninori Morimoto (2): MAINTAINERS: add missing maintainers for Simple Audio Card ASoC: rsnd: check rsnd_adg_clk_enable() return value Maciej Strozek (1): ASoC: cs42l43: Add codec force suspend/resume ops Marian Postevca (1): ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK Richard Fitzgerald (2): ALSA: doc: cs35l56: Add information about Cirrus Logic CS35L54/56/57 ALSA: doc: Add codecs/index.rst to top-level index Shenghao Ding (1): ASoC: tas2781: Fix occasional calibration failture Zhang Yi (1): ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise Documentation/sound/codecs/cs35l56.rst | 292 +++++++++++++++++++++++++++++++++ Documentation/sound/codecs/index.rst | 9 + Documentation/sound/index.rst | 1 + MAINTAINERS | 2 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs42l43.c | 1 + sound/soc/codecs/es8316.c | 10 +- sound/soc/codecs/es8326.c | 23 ++- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/renesas/rcar/adg.c | 28 +++- sound/soc/renesas/rcar/core.c | 4 +- sound/soc/renesas/rcar/rsnd.h | 2 +- sound/soc/samsung/Kconfig | 6 +- 13 files changed, 364 insertions(+), 17 deletions(-) create mode 100644 Documentation/sound/codecs/cs35l56.rst create mode 100644 Documentation/sound/codecs/index.rst

2 1

alsactl info fails with exit code 25
by GitHub issues - opened 09 Jan '25

09 Jan '25

alsa-project/alsa-utils issue #287 was opened from z-s-e: Here is the output of running alsactl info on my machine: ``` $ alsactl info 0 # # Sound card # - card: 0 id: Generic name: HD-Audio Generic longname: HD-Audio Generic at 0xb88c8000 irq 87 driver_name: HDA-Intel mixer_name: ATI R6xx HDMI components: HDA:1002aa01,00aa0100,00100800 controls_count: 15 pcm: - stream: PLAYBACK devices: - device: 3 id: HDMI 0 name: G246HYL subdevices: - subdevice: 0 name: subdevice #0 - device: 7 id: HDMI 1 name: HDMI 1 subdevices: - subdevice: 0 name: subdevice #0 alsactl: rawmidi_device_list:105: snd_ctl_rawmidi_next_device $ echo $? 25 ``` Apparently it fails when trying to enumerate raw midi devices. I don't think this is supposed to happen, right? alsa-info for reference: [alsa-info.txt](https://github.com/user-attachments/files/18362369/alsa-info… Issue URL : https://github.com/alsa-project/alsa-utils/issues/287 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 09 Jan '25

09 Jan '25

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 30cbc3d44cd5..4c806ad2fb16 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23172,25 +23172,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23200,9 +23217,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

Topology: NHLT: Intel: Fix mono DMIC configure for MTL platform
by GitHub pull_request - opened 08 Jan '25

08 Jan '25

alsa-project/alsa-utils pull request #286 was opened from singalsu: This change fixes the blob generator for mono microphone configuration. As difference to previous platforms the FIFO packer mono/stereo mode set up in OUTCONTROLx IPM_SOURCE_MODE bit-field. Request URL : https://github.com/alsa-project/alsa-utils/pull/286 Patch URL : https://github.com/alsa-project/alsa-utils/pull/286.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aseqdump version 1.2.11, There are small bugs
by GitHub issues - opened 08 Jan '25

08 Jan '25

alsa-project/alsa-utils issue #285 was opened from masahirokakishita: Hello, I found small bugs in aseqdump.c static void dump_ump_midi2_event(const unsigned int *ump) ... case SND_UMP_MSG_PITCHBEND: printf("Channel pressure %2d, value %s", channel, midi2_pitchbend(m->channel_pressure.data)); break; It is not "Channel pressure", but "Pitch Bend". case SND_UMP_MSG_NOTE_ON: printf("Note on %2d, note %d, velocity %s, attr type = %d, data = 0x%x", channel, m->note_off.note, midi2_velocity(m->note_off.velocity), m->note_off.attr_type, m->note_off.attr_data); break; "note->on" is correct. This bug does not affect the display. Issue URL : https://github.com/alsa-project/alsa-utils/issues/285 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: rt5682: Support the ALC5682I-VE codec
by derek.fang＠realtek.com 07 Jan '25

07 Jan '25

From: Derek Fang <derek.fang(a)realtek.com> The ALC5682I-VD and ALC5682I-VE use the same I2C codec driver with different calibration settings. This patch aims to handle their differences. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index ff9e14fad0cd..a8820435d1e0 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -186,6 +186,12 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return -ENODEV; } + regmap_read(rt5682->regmap, RT5682_INT_DEVICE_ID, &val); + if (val == 0x6956) { + dev_dbg(&i2c->dev, "ALC5682I-VE device\n"); + rt5682->ve_ic = true; + } + mutex_init(&rt5682->calibrate_mutex); rt5682_calibrate(rt5682); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index aa163ec40862..b4d72fc4a44d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -395,6 +395,7 @@ bool rt5682_volatile_register(struct device *dev, unsigned int reg) case RT5682_4BTN_IL_CMD_1: case RT5682_AJD1_CTRL: case RT5682_HP_CALIB_CTRL_1: + case RT5682_INT_DEVICE_ID: case RT5682_DEVICE_ID: case RT5682_I2C_MODE: case RT5682_HP_CALIB_CTRL_10: @@ -419,6 +420,7 @@ bool rt5682_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5682_RESET: + case RT5682_INT_DEVICE_ID: case RT5682_VERSION_ID: case RT5682_VENDOR_ID: case RT5682_DEVICE_ID: @@ -3139,7 +3141,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x7005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); @@ -3168,7 +3173,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x2005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0c0c); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index b2d9e87af259..de43a5d99403 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -22,6 +22,7 @@ /* Info */ #define RT5682_RESET 0x0000 +#define RT5682_INT_DEVICE_ID 0x00f9 #define RT5682_VERSION_ID 0x00fd #define RT5682_VENDOR_ID 0x00fe #define RT5682_DEVICE_ID 0x00ff @@ -1446,6 +1447,7 @@ struct rt5682_priv { bool hw_init; bool first_hw_init; bool is_sdw; + bool ve_ic; #ifdef CONFIG_COMMON_CLK struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; -- 2.34.1

2 1

[PATCH v1] ASoC: tas2781: Fix occasional calibration failture
by Shenghao Ding 07 Jan '25

07 Jan '25

The root cause is that TAS2781 internal register is not unlocked before writing data into TAS2781_PRM_TEST_57_REG. Fixes: 49e2e353fb0d ("ASoC: tas2781: Add Calibration Kcontrols for Chromebook") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index fb8cd2284fe8..728bf78ae71f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -78,7 +78,7 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false), X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false), X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false), - X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, true), X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true), X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false), X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false), -- 2.34.1

2 1

[PATCH RESEND] ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK
by Marian Postevca 07 Jan '25

07 Jan '25

For 48Mhz MCLK systems the calculation of the HW rate is broken, and will not produce even one sane rate. Since es83xx supports the option to halve MCLK, calculate also rates with MCLK/2. Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/codecs/es8316.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 61729e5b50a8..f508df01145b 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -39,7 +39,9 @@ struct es8316_priv { struct snd_soc_jack *jack; int irq; unsigned int sysclk; - unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios)]; + /* ES83xx supports halving the MCLK so it supports twice as many rates + */ + unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios) * 2]; struct snd_pcm_hw_constraint_list sysclk_constraints; bool jd_inverted; }; @@ -386,6 +388,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq % ratio == 0) es8316->allowed_rates[count++] = freq / ratio; + + /* We also check if the halved MCLK produces a valid rate + * since the codec supports halving the MCLK. + */ + if ((freq / ratio) % 2 == 0) + es8316->allowed_rates[count++] = freq / ratio / 2; } if (count) { -- 2.45.2

2 1

[PATCH] ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise
by Zhang Yi 07 Jan '25

07 Jan '25

We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a5603b617688..b06eead7e0f6 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); } } } else { @@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00); if (direction == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); @@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { msleep(300); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, @@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); + } break; case SND_SOC_BIAS_PREPARE: break; @@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80); break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); @@ -773,7 +785,10 @@ static void es8326_jack_button_handler(struct work_struct *work) case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + if ((iface == 0x6f) && (es8326->version > ES8326_VERSION_B)) + cur_button = SND_JACK_BTN_0; + else + cur_button = SND_JACK_BTN_1; break; case 0x27: /* button volume down */ @@ -1082,7 +1097,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

2 1

headers: make more headers self-contained
by GitHub pull_request - opened 06 Jan '25

06 Jan '25

alsa-project/alsa-lib pull request #435 was opened from timblechmann: Some IDEs parse headers in isolation to populate their code model. This exposes some missing headers, mainly for alsa internal struct definitions, ssize_t (requiring sys/types.h), size_t (requiring string.h), va_args (requiring stdargs.h) or FILE (requiring stdio.h) This patch makes the main headers parsable by qtcreator Request URL : https://github.com/alsa-project/alsa-lib/pull/435 Patch URL : https://github.com/alsa-project/alsa-lib/pull/435.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

bug in the asla driver asla-sink-viritio-snd
by GitHub issues - opened 06 Jan '25

06 Jan '25

alsa-project/alsa-lib issue #434 was opened from LoreFiore: <img width="1440" alt="Screenshot 2025-01-05 alle 23 31 08" src="https://github.com/user-attachments/assets/34b279a0-b256-4e7b-82ec-a9df46f9…" /> Issue URL : https://github.com/alsa-project/alsa-lib/issues/434 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ld.lld default version symbol snd_dlopen@@ALSA_1.1.6 must be defined
by GitHub issues - edited 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #433 was edited from trcrsired: ``` clang: warning: argument unused during compilation: '--unwindlib=libunwind' [-Wunused-command-line-argument] CCLD libasound.la ld.lld: error: seq/.libs/libseq.a(seq.o at 50494) <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:1:1: default version symbol snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_format_first, snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:2:1: default version symbol snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_channels_near, snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:3:1: default version symbol snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_rate_near, snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:4:1: default version symbol snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_time_near, snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:5:1: default version symbol snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_size_near, snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:6:1: default version symbol snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_periods_near, snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:7:1: default version symbol snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_time_near, snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:8:1: default version symbol snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_size_near, snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:9:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: .libs/conf.o <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_hw.o at 720480) <inline asm>:1:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:1:1: default version symbol snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_access, snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:2:1: default version symbol snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_format, snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:3:1: default version symbol snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_subformat, snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:4:1: default version symbol snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_channels, snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:5:1: default version symbol snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_rate, snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:6:1: default version symbol snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:7:1: default version symbol snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_size, snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:8:1: default version symbol snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_buffer_size, snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 ^ ld.lld: error: too many errors emitted, stopping now (use --error-limit=0 to see all errors) clang: error: linker command failed with exit code 1 (use -v to see invocation) make[3]: *** [Makefile:506: libasound.la] Error 1 make[3]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[2]: *** [Makefile:567: all-recursive] Error 1 make[2]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[1]: *** [Makefile:436: all] Error 2 make[1]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make: *** [Makefile:414: all-recursive] Error 1 alsa-lib build failure ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/433 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

default version symbol snd_dlopen@@ALSA_1.1.6 must be defined
by GitHub issues - opened 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #433 was opened from trcrsired: ``` clang: warning: argument unused during compilation: '--unwindlib=libunwind' [-Wunused-command-line-argument] CCLD libasound.la ld.lld: error: seq/.libs/libseq.a(seq.o at 50494) <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:1:1: default version symbol snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_format_first, snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:2:1: default version symbol snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_channels_near, snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:3:1: default version symbol snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_rate_near, snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:4:1: default version symbol snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_time_near, snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:5:1: default version symbol snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_size_near, snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:6:1: default version symbol snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_periods_near, snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:7:1: default version symbol snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_time_near, snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:8:1: default version symbol snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_size_near, snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:9:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: .libs/conf.o <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_hw.o at 720480) <inline asm>:1:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:1:1: default version symbol snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_access, snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:2:1: default version symbol snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_format, snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:3:1: default version symbol snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_subformat, snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:4:1: default version symbol snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_channels, snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:5:1: default version symbol snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_rate, snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:6:1: default version symbol snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:7:1: default version symbol snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_size, snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:8:1: default version symbol snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_buffer_size, snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 ^ ld.lld: error: too many errors emitted, stopping now (use --error-limit=0 to see all errors) clang: error: linker command failed with exit code 1 (use -v to see invocation) make[3]: *** [Makefile:506: libasound.la] Error 1 make[3]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[2]: *** [Makefile:567: all-recursive] Error 1 make[2]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[1]: *** [Makefile:436: all] Error 2 make[1]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make: *** [Makefile:414: all-recursive] Error 1 alsa-lib build failure ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/433 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm2: USB-Audio: add Presonus Revelator IO 44 (USB194f:0424)
by GitHub pull_request - edited 04 Jan '25

04 Jan '25

alsa-project/alsa-ucm-conf pull request #495 was edited from markulovi: [alsa-info.txt](https://github.com/user-attachments/files/18306150/alsa-info… This adds support for [Presonus Revelator io44](https://eu.presonus.com/products/revelator-io44). This device has 3 virtual stereo channels out and 4 virtual stereo channels in as follows: ![out](https://github.com/user-attachments/assets/f4cacf14-e75e-4d33-ad64-cca4ca7ee18e) ![in](https://github.com/user-attachments/assets/60515091-51a5-4299-97c7-a12d3c091b8a) Microphone inputs have been separated into two mono channels for convenience. As far as I can tell, the [Universal Control](https://www.presonus.com/pages/support-documents-downloads?product… software is required to configure levels and effects of these channels, but the settings can be saved to the device and operation without the software is possible outside Windows. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: USB-Audio: add Presonus Revelator IO 44 (USB194f:0424)
by GitHub pull_request - opened 04 Jan '25

04 Jan '25

alsa-project/alsa-ucm-conf pull request #495 was opened from markulovi: [alsa-info.txt](https://github.com/user-attachments/files/18306150/alsa-info… This adds support for [Presonus Revelator io44](https://eu.presonus.com/products/revelator-io44). This device has 3 virtual stereo channels out and 4 virtual stereo channels in as follows: ![out](https://github.com/user-attachments/assets/f4cacf14-e75e-4d33-ad64-cca4ca7ee18e) ![in](https://github.com/user-attachments/assets/60515091-51a5-4299-97c7-a12d3c091b8a) Microphone inputs have been separated to two mono channels for convenience. As far as I can tell, the [Universal Control](https://www.presonus.com/pages/support-documents-downloads?product… software is required to configure levels and effects of these channels, but the settings can be saved to the device and operation without the software is possible outside Windows. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/5] soundwire: amd: code improvements and new platform
by Vijendar Mukunda 02 Jan '25

02 Jan '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Vijendar Mukunda (5): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: add soundwire host wake interrupt enable/disable sequence soundwire: amd: clear soundwire pme wake status drivers/soundwire/amd_manager.c | 102 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 27 ++++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 131 insertions(+) -- 2.34.1

2 7

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when (at most week after Baldur's Gate 3 release ;). What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` I had this monitor connected before and alsacap reported "flipped" results - not like now that both cards are limited to 48000. What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when (at most week after Baldur's Gate 3 release ;). What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when. What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

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Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - opened 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was opened from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when. What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

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