October 2024 - 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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[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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[RESEND PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 30 Dec '24

30 Dec '24

Hi Mark, resending this as it failed to apply in my last submission. Added Neil Armstrong to Cc: as hopefully he will be able to better review this. Thanks, Geraldo Nascimento --- Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10), for example, if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,28 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending on the HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

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[PATCH v4 0/9] ASoC: samsung: midas-audio: Add GPIO-based headset jack detection
by Artur Weber 29 Dec '24

29 Dec '24

Many of Samsung's Exynos 4 devices share the same midas-audio driver to handle the codec setup. While most of these devices, including the Midas itself, use the jack detection provided by the WM8994 driver, other devices such as the Samsung Galaxy Tab 3 8.0 (lt01) use two GPIOs and an ADC channel to determine jack insertion, the jack's type, and button presses (for headsets with volume up/down/play buttons). In the downstream kernel, this behavior is implemented in the sec-jack driver[1], and the per-device settings are configured in *-jack.c files in the mach folder (see e.g. the Tab 3's implementation[2]). This patchset implements this mechanism in the midas_wm1811.c driver, and adds new DTS options to allow for its configuration. It also enables jack detection for the Samsung Galaxy Tab 3 8.0. A very similar mechanism was already present in the aries_wm8994.c driver[3]; this implementation heavily borrows from it, though there are a few extra cleanups as well. Signed-off-by: Artur Weber <aweber.kernel(a)gmail.com> [1] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [2] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [3] https://github.com/torvalds/linux/blob/master/sound/soc/samsung/aries_wm899… --- Changes in v4: - Replaced manual regulator toggles with SND_SOC_DAPM_REGULATOR_SUPPLY, both for existing mic bias regulators and for the new headset mic bias regulator - Link to v3: https://lore.kernel.org/r/20240519-midas-wm1811-gpio-jack-v3-0-0c1736144c0e… Changes in v3: - Re-added pipe (|) character to samsung,headset-button-threshold-microvolt description to prevent it from being parsed as a mapping (fixes syntax error) - Split out "ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection" into two patches - Link to v2: https://lore.kernel.org/r/20240508-midas-wm1811-gpio-jack-v2-0-b4d36cd02c6e… Changes in v2: - Added vendor prefix to threshold properties - Added separate headset mic bias regulator - Changed some cases of dev_err + return with return dev_err_probe - Added an extra patch to replace some previous dev_err + return cases with dev_err_probe - Moved tab3 DTS wm1811 codec config changes to separate commit --- Artur Weber (9): ASoC: dt-bindings: samsung,midas-audio: Add headset mic bias supply ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use SND_SOC_DAPM_REGULATOR_SUPPLY for bias regulators ASoC: samsung: midas_wm1811: Add headset mic bias supply support ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use dev_err_probe where appropriate ARM: dts: samsung: exynos4212-tab3: Fix headset mic, add jack detection ARM: dts: samsung: exynos4212-tab3: Add MCLK2 clock to WM1811 codec config ARM: dts: samsung: exynos4212-tab3: Drop interrupt from WM1811 codec .../bindings/sound/samsung,midas-audio.yaml | 33 ++ arch/arm/boot/dts/samsung/exynos4212-tab3.dtsi | 31 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/midas_wm1811.c | 348 +++++++++++++++++---- 4 files changed, 343 insertions(+), 71 deletions(-) --- base-commit: a38297e3fb012ddfa7ce0321a7e5a8daeb1872b6 change-id: 20240502-midas-wm1811-gpio-jack-b10226b17ecc Best regards, -- Artur Weber <aweber.kernel(a)gmail.com>

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[PATCH 00/35] Reorganize kerneldoc parameter names
by Julia Lawall 20 Dec '24

20 Dec '24

Reorganize kerneldoc parameter names to match the parameter order in the function header. The misordered cases were identified using the following Coccinelle semantic patch: // <smpl> @initialize:ocaml@ @@ let parse_doc l = let pieces = List.map String.trim (String.split_on_char '*' l) in let l = String.concat " " pieces in match String.split_on_char ':' l with x::xs -> x | _ -> "" let params ps = List.rev (List.fold_left (fun prev (pm,_) -> let ty = String.trim(Pretty_print_c.string_of_fullType pm.Ast_c.p_type) in if ty = "void" && pm.Ast_c.p_namei = None then prev else let name = match pm.Ast_c.p_namei with Some name -> name | None -> failwith "function parameter has no name" in (String.trim (Pretty_print_c.string_of_name name),ty)::prev) [] ps) @r@ comments c; identifier fn; position p; parameter list ps; type T; @@ T@c fn@p(ps) { ... } @script:ocaml@ p << r.p; c << r.c; (_,ps) << r.ps; @@ let isdoc c ps = List.length ps > 1 && (let c = String.trim c in String.length c > 3 && String.sub c 0 3 = "/**" && String.get c 3 != '*') in let subset l1 l2 = List.for_all (fun x -> List.mem x l2) l1 in let (cb,cm,ca) = List.hd c in match List.rev cb with c::_ when isdoc c ps -> let pieces = String.split_on_char '@' c in (match pieces with _::tl -> let d_names = List.map parse_doc tl in (* check parameters *) let p_names = List.map fst (params ps) in if d_names <> [] && not(d_names = p_names) then begin if List.sort compare d_names = List.sort compare p_names then Coccilib.print_main "out of order" p else if subset d_names p_names then Coccilib.print_main "doc is missing a parameter" p else if subset d_names p_names then Coccilib.print_main "doc has an extra parameter" p end | _ -> ()) | _ -> () // </smpl> --- arch/arm/mach-omap2/prm2xxx_3xxx.c | 1 - arch/powerpc/platforms/ps3/interrupt.c | 2 +- arch/powerpc/platforms/ps3/repository.c | 2 +- drivers/base/firmware_loader/main.c | 2 +- drivers/comedi/drivers/comedi_8254.c | 2 +- drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c | 2 +- drivers/gpu/drm/amd/display/dc/core/dc.c | 2 +- drivers/gpu/drm/drm_gem_framebuffer_helper.c | 3 +-- drivers/gpu/drm/drm_gpuvm.c | 2 +- drivers/gpu/drm/radeon/radeon_ib.c | 2 +- drivers/iommu/iommu.c | 2 +- drivers/leds/leds-gpio-register.c | 2 +- drivers/mfd/atmel-smc.c | 4 ++-- drivers/misc/mei/bus.c | 2 +- drivers/mtd/ubi/eba.c | 2 +- drivers/net/ethernet/broadcom/bnxt/bnxt_hwrm.c | 2 +- drivers/net/ethernet/intel/e1000/e1000_hw.c | 2 +- drivers/net/ethernet/intel/i40e/i40e_common.c | 7 +++---- drivers/net/ethernet/intel/ice/ice_common.c | 2 +- drivers/net/ethernet/intel/ixgbe/ixgbe_common.c | 2 +- drivers/nvdimm/dimm_devs.c | 2 +- drivers/pci/hotplug/pci_hotplug_core.c | 2 +- drivers/pinctrl/pinmux.c | 2 +- drivers/slimbus/messaging.c | 2 +- drivers/soc/qcom/qmi_interface.c | 2 +- drivers/soundwire/stream.c | 2 +- drivers/usb/gadget/config.c | 4 ++-- fs/char_dev.c | 2 +- fs/dcache.c | 4 ++-- fs/seq_file.c | 2 +- kernel/audit.c | 2 +- kernel/resource.c | 2 +- kernel/sysctl.c | 1 - kernel/trace/ring_buffer.c | 2 +- lib/lru_cache.c | 2 +- lib/maple_tree.c | 2 +- mm/mmu_notifier.c | 2 +- net/dccp/feat.c | 2 +- net/mac80211/mesh_hwmp.c | 6 +++--- net/mac80211/mesh_pathtbl.c | 10 +++++----- net/socket.c | 2 +- net/sunrpc/xprt.c | 2 +- net/tipc/link.c | 14 +++++++------- net/tipc/msg.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- 45 files changed, 60 insertions(+), 64 deletions(-)

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hda: how to implement component master_unbind?
by Kai Vehmanen 20 Dec '24

20 Dec '24

Hi Takashi et al, I've been having multiple discussions with our i915 team w.r.t. audio driver behaviour when the aggregate component is unbound, triggered by i915 unbind. This came up again in the recent regression with devres allocations and I now dug into the topic again. In short, I'm puzzled how to really implement this. ALSA (or ASoC) don't really have support for individual components of a card disappearing in a hotplug fashion. At card level, we do have such support (USB, firewire and recent work for PCI hotplug). But not individual components of a card getting unplugged. In current code we have this: static void hdac_component_master_unbind(struct device *dev) { » struct drm_audio_component *acomp = hdac_get_acomp(dev); » if (acomp->audio_ops && acomp->audio_ops->master_unbind) » » acomp->audio_ops->master_unbind(dev, acomp); » module_put(acomp->ops->owner); » component_unbind_all(dev, acomp); » WARN_ON(acomp->ops || acomp->dev); } ... when e.g. i915 driver is unbound (but could be any driver using the component framework, not jus Intel hw), i915 calls component_del() and HDA gets call to the above. After this, acomp ops are null are audio no longer has ability to talk to i915 driver (which makes sense given it's unbound). If audio was runtime suspended, this kind of works (but relies on some good luck). Upon HDA controller resume, we note acomp ops are NULL and HDMI/DP operations (like snd_hdac_display_power()) silently become no-ops. PCM streaming will go to /dev/null, but this is ok'ish since no monitor can be connected anyways. If audio was active, we start to get warnings or worse. Audio expects hw display codec to be powered and programmed, but suddenly it mey lose state. If the audio controller is actually part of the display hardware (e.g. discrete GPUs), then controller registers can stop functioning (they should be still available, but given the main diplay driver is unbound, odds of suprising behaviour are high). So this is less than ideal. I've been looking at options: 1) prevent/block the unbind if audio device is busy The component framework does not allow individual components to return -EBUSY, so there's no way to let others know that unbind is not possible now. This would force anyone testing DRM driver unbind to first unbind the audio driver and stop any active audio use-cases if needed. 2) unbind the ALSA card I've experimented doing a device_unregister() from the above callback, but this has not really worked (onion peeling exercise of new probelms). The code is shared by multiple controllers, so getting a handle to an snd_card object is not straighforward (drvdata is differnet for SOF and snd-hda-intel for instance). But with some new work, maybe we could call snd_card_disconnect() to detach the card and inform user-space. 3) continue as-is and try to fix bugs If audio is active, maybe we could force a acomp->put_power() to ensure a clean unregister of the display part. But this leaves a big chunk of issues especially with HDA controllers that require the display to be powered on, to function. .. Any ideas, and/or has there been prior work? It seems Takashi it's been mostly you who has been active working on the acomp integration recently. I also included Chris, Daniel and Jani who've had patches to hdac_component.c. Br, Kai

3 5

[PATCH] dt-bindings: sound:atmel-at91sam9g20ek: convert bindings to json-schema
by Balakrishnan Sambath 10 Dec '24

10 Dec '24

Convert atmel-at91sam9g20ek-wm8731-audio DT binding to yaml based json-schema.Change file name to match json-scheme naming. Signed-off-by: Balakrishnan Sambath <balakrishnan.s(a)microchip.com> --- .../bindings/sound/atmel,at91sam9g20ek-wm8731.yaml | 60 ++++++++++++++++++++++ .../sound/atmel-at91sam9g20ek-wm8731-audio.txt | 26 ---------- 2 files changed, 60 insertions(+), 26 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml new file mode 100644 index 000000000000..f6330707fe1b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,at91sam9g20ek-wm8731.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel at91sam9g20ek wm8731 audio complex + +maintainers: + - Balakrishnan Sambath <balakrishnan.s(a)microchip.com> + +properties: + compatible: + const: atmel,at91sam9g20ek-wm8731-audio + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + atmel,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: A list of the connections between audio components. + minItems: 2 + items: + enum: + # Board Connectors + - "Ext Spk" + - "Int MIC" + + # CODEC Pins + - LHPOUT + - MICIN + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of WM8731 audio codec + +required: + - compatible + - atmel,model + - atmel,audio-routing + - atmel,ssc-controller + - atmel,audio-codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,at91sam9g20ek-wm8731-audio"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + atmel,model = "wm8731 @ AT91SAMG20EK"; + atmel,audio-routing = + "Ext Spk", "LHPOUT", + "Int MIC", "MICIN"; + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt deleted file mode 100644 index 9c5a9947b64d..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt +++ /dev/null @@ -1,26 +0,0 @@ -* Atmel at91sam9g20ek wm8731 audio complex - -Required properties: - - compatible: "atmel,at91sam9g20ek-wm8731-audio" - - atmel,model: The user-visible name of this sound complex. - - atmel,audio-routing: A list of the connections between audio components. - - atmel,ssc-controller: The phandle of the SSC controller - - atmel,audio-codec: The phandle of the WM8731 audio codec -Optional properties: - - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt - -Example: -sound { - compatible = "atmel,at91sam9g20ek-wm8731-audio"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_pck0_as_mck>; - - atmel,model = "wm8731 @ AT91SAMG20EK"; - - atmel,audio-routing = - "Ext Spk", "LHPOUT", - "Int MIC", "MICIN"; - - atmel,ssc-controller = <&ssc0>; - atmel,audio-codec = <&wm8731>; -}; --- base-commit: 0dd3ee31125508cd67f7e7172247f05b7fd1753a change-id: 20240214-at91sam9g20ek-wm8731-yaml-404ddddbb91a Best regards, -- Balakrishnan Sambath <balakrishnan.s(a)microchip.com>

5 5

[PATCH treewide 00/11] ASoC: Clean up {hp,mic}-det-gpio handling
by Geert Uytterhoeven 09 Dec '24

09 Dec '24

Hi all, The "gpio" suffix for GPIO consumers was deprecated a while ago, in favor of the "gpios" suffix. However, there are still several users of the "hp-det-gpio" and "mic-det-gpio" properties, in DT bindings, drivers, and DT source files. Drivers that use gpiolib handle both the deprecated and new properties transparently, but that is not the case when using of_property_*() calls directly. - The first patch fixes a bug, as the Freescale Generic ASoC Sound driver handles only the deprecated properties, while there is already a user of the new variants, - The second patch updates the Audio Graph and Simple Audio Card DT bindings, - Patches 3-9 converts various DTS files to use the new properties, - The last 2 patches convert Freescale sound device nodes to use the new properties. All patches can be applied independently, except for the last two, which depend on the first patch to avoid regressions. This series was compile-tested only. Thanks for your comments! Geert Uytterhoeven (11): ASoC: fsl-asoc-card: Add missing handling of {hp,mic}-dt-gpios ASoC: dt-bindings: Deprecate {hp,mic}-det-gpio ARM: dts: marvell: mmp2-olpc-xo-1-75: Switch to {hp,mic}-det-gpios arm64: dts: freescale: imx: Switch to simple-audio-card,hp-det-gpios arm64: dts: renesas: beacon-renesom: Switch to mic-det-gpios arm64: dts: rockchip: Switch to simple-audio-card,hp-det-gpios arm64: dts: rockchip: Switch to hp-det-gpios arm64: dts: uniphier: Switch to hp-det-gpios mips: dts: ingenic: Switch to simple-audio-card,hp-det-gpios ARM: dts: nxp: imx: Switch to {hp,mic}-det-gpios arm64: dts: freescale: imx: Switch to hp-det-gpios .../devicetree/bindings/sound/audio-graph.yaml | 6 ++++++ .../devicetree/bindings/sound/simple-card.yaml | 12 ++++++++++++ arch/arm/boot/dts/marvell/mmp2-olpc-xo-1-75.dts | 4 ++-- arch/arm/boot/dts/nxp/imx/imx6qdl-sabresd.dtsi | 4 ++-- arch/arm/boot/dts/nxp/imx/imx6sl-evk.dts | 2 +- arch/arm/boot/dts/nxp/imx/imx6sll-evk.dts | 2 +- arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi | 2 +- arch/arm/boot/dts/nxp/imx/imx6ul-14x14-evk.dtsi | 2 +- arch/arm/boot/dts/nxp/imx/imx7d-sdb.dts | 2 +- .../boot/dts/freescale/imx8mq-librem5-devkit.dts | 2 +- arch/arm64/boot/dts/freescale/imx8mq-librem5.dtsi | 2 +- arch/arm64/boot/dts/freescale/imx8qm-mek.dts | 2 +- arch/arm64/boot/dts/freescale/imx8qxp-mek.dts | 2 +- arch/arm64/boot/dts/freescale/imx95-19x19-evk.dts | 2 +- .../boot/dts/renesas/beacon-renesom-baseboard.dtsi | 2 +- .../boot/dts/rockchip/rk3326-anbernic-rg351m.dtsi | 2 +- .../arm64/boot/dts/rockchip/rk3326-gameforce-chi.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3326-odroid-go.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3368-lba3368.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-eaidk-610.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-pinebook-pro.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-roc-pc-plus.dts | 2 +- .../boot/dts/rockchip/rk3399-rock-pi-4a-plus.dts | 2 +- .../boot/dts/rockchip/rk3399-rock-pi-4b-plus.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-rock-pi-4c.dts | 2 +- .../boot/dts/rockchip/rk3399-sapphire-excavator.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg-arc.dtsi | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353p.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353ps.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353v.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353vs.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg503.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3566-pinetab2.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3566-powkiddy-x55.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3568-odroid-m1.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-armsom-sige7.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-evb1-v10.dts | 2 +- .../dts/rockchip/rk3588-friendlyelec-cm3588-nas.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-nanopc-t6.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3588-ok3588-c.dts | 2 +- .../boot/dts/rockchip/rk3588-orangepi-5-plus.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-quartzpro64.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-rock-5-itx.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-rock-5b.dts | 2 +- .../boot/dts/rockchip/rk3588s-gameforce-ace.dts | 2 +- .../boot/dts/socionext/uniphier-ld11-global.dts | 2 +- .../boot/dts/socionext/uniphier-ld20-global.dts | 2 +- arch/mips/boot/dts/ingenic/gcw0.dts | 2 +- arch/mips/boot/dts/ingenic/rs90.dts | 2 +- sound/soc/fsl/fsl-asoc-card.c | 8 +++++--- 50 files changed, 72 insertions(+), 52 deletions(-) -- 2.34.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

8 30

[PATCH v2 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 06 Dec '24

06 Dec '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Change since v1: Support audio function for it6505 bridge to hdmi-codec Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 37 ++++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 116 insertions(+), 12 deletions(-) -- 2.18.0

3 16

[PATCH] ASoC: tas2781: Fix redundant logical jump
by Tang Bin 02 Dec '24

02 Dec '24

In these functions, some logical jump of "goto" and variable are redundant, thus remove them. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/tas2781-i2c.c | 68 +++++++++++++--------------------- 1 file changed, 25 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 68887799e..27f2f84a4 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -489,14 +489,11 @@ static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); const int sum = ARRAY_SIZE(tas2563_cali_start_reg); - int rc = 1; int i, j; guard(mutex)(&tas_priv->codec_lock); - if (tas_priv->chip_id != TAS2563) { - rc = -1; - goto out; - } + if (tas_priv->chip_id != TAS2563) + return -1; for (i = 0; i < tas_priv->ndev; i++) { struct tasdevice *tasdev = tas_priv->tasdevice; @@ -523,8 +520,8 @@ static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, q[j].val, 4); } } -out: - return rc; + + return 1; } static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) @@ -576,7 +573,7 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, struct cali_reg *p = &cali_data->cali_reg_array; unsigned char *src = ucontrol->value.bytes.data; unsigned char *dst = cali_data->data; - int rc = 1, i = 0; + int i = 0; int j; guard(mutex)(&priv->codec_lock); @@ -605,7 +602,7 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, i += 3; memcpy(dst, &src[i], cali_data->total_sz); - return rc; + return 1; } static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, @@ -1115,25 +1112,21 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) char *conf_name, *prog_name; int nr_controls = 4; int mix_index = 0; - int ret; /* Alloc kcontrol via devm_kzalloc, which don't manually * free the kcontrol */ dsp_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, sizeof(dsp_ctrls[0]), GFP_KERNEL); - if (!dsp_ctrls) { - ret = -ENOMEM; - goto out; - } + if (!dsp_ctrls) + return -ENOMEM; /* Create mixer items for selecting the active Program and Config */ prog_name = devm_kstrdup(tas_priv->dev, "Speaker Program Id", GFP_KERNEL); - if (!prog_name) { - ret = -ENOMEM; - goto out; - } + if (!prog_name) + return -ENOMEM; + dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -1143,10 +1136,9 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) conf_name = devm_kstrdup(tas_priv->dev, "Speaker Config Id", GFP_KERNEL); - if (!conf_name) { - ret = -ENOMEM; - goto out; - } + if (!conf_name) + return -ENOMEM; + dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; @@ -1156,10 +1148,9 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) active_dev_num = devm_kstrdup(tas_priv->dev, "Activate Tasdevice Num", GFP_KERNEL); - if (!active_dev_num) { - ret = -ENOMEM; - goto out; - } + if (!active_dev_num) + return -ENOMEM; + dsp_ctrls[mix_index].name = active_dev_num; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_active_num; @@ -1168,21 +1159,17 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) mix_index++; chip_id = devm_kstrdup(tas_priv->dev, "Tasdevice Chip Id", GFP_KERNEL); - if (!chip_id) { - ret = -ENOMEM; - goto out; - } + if (!chip_id) + return -ENOMEM; + dsp_ctrls[mix_index].name = chip_id; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_chip_id; dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; - ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); - -out: - return ret; } static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) @@ -1469,7 +1456,6 @@ static int tasdevice_hw_params(struct snd_pcm_substream *substream, unsigned int slot_width; unsigned int fsrate; int bclk_rate; - int rc = 0; fsrate = params_rate(params); switch (fsrate) { @@ -1479,8 +1465,7 @@ static int tasdevice_hw_params(struct snd_pcm_substream *substream, default: dev_err(tas_priv->dev, "%s: incorrect sample rate = %u\n", __func__, fsrate); - rc = -EINVAL; - goto out; + return -EINVAL; } slot_width = params_width(params); @@ -1493,20 +1478,17 @@ static int tasdevice_hw_params(struct snd_pcm_substream *substream, default: dev_err(tas_priv->dev, "%s: incorrect slot width = %u\n", __func__, slot_width); - rc = -EINVAL; - goto out; + return -EINVAL; } bclk_rate = snd_soc_params_to_bclk(params); if (bclk_rate < 0) { dev_err(tas_priv->dev, "%s: incorrect bclk rate = %d\n", __func__, bclk_rate); - rc = bclk_rate; - goto out; + return bclk_rate; } -out: - return rc; + return 0; } static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, -- 2.33.0

3 2

[RFC PATCH 0/2] Add support for stacked and parallel memories
by Amit Kumar Mahapatra 25 Nov '24

25 Nov '24

Hello Everyone, Following an email discussion with Miquel regarding the binding changes and overall architecture for implementing support for stacked and parallel memories, I’m sharing this RFC to initiate a discussion on the proposed updates to current bindings and to finalize the implementation architecture. Before diving into the main topic, here is some background on stacked and parallel memories. The AMD QSPI controller supports two advanced connection modes(Stacked and Parallel) which allow the controller to treat two different flashes as one storage. Stacked: Flashes share the same SPI bus, but different CS line, controller driver asserts the CS of the flash to which it needs to communicate. Stacked mode is a software abstraction rather than a controller feature or capability. At any given time, the controller communicates with one of the two connected flash devices, as determined by the requested address and data length. If an operation starts on one flash and ends on the other, the core needs to split it into two separate operations and adjust the data length accordingly. Parallel(Multi-CS): Both the flashes have their separate SPI bus, CS of both the flashes will be asserted/de-asserted at the same time. In this mode data will be split across both the flashes by enabling the STRIPE setting in the controller. Parallel mode is a controller feature where if the STRIPE bit is set then the controller internally handles the data split during data write to the flashes and while reading data from the flash the controller internally merges data from both the flashes before writing to the controller FIFO. If STRIPE is not enabled, then same data will be sent to both the devices. In parallel mode both the flashes should be identical. For more information on the modes please feel free to go through the controller flash interface below [1]. Mirochip QSPI controller[2] also supports "Dual Parallel 8-bit IO mode", but they call it "Twin Quad Mode". Initially in [3] [4] [5] Miquel had tried to extend MTD-CONCAT driver to support Stacked mode, but the bindings were not accepted. So, the MTD-CONCAT approach was dropped and the DT bindings that got accepted [6] [7] [8] that describes the two flash devices as being one. SPI core changes to support the above bindings were added [9]. While adding the support in SPI-NOR Tudor provided additional feedback, leading to a discussion on updating the current stacked and parallel DT bindings. Proposed Solution: The solution has two parts: 1. Update MTD-CONCAT Update MTD-CONCAT to create virtual concatinated mtd devices as defined in the device tree. 2. Add a New Layer Add a new layer between the SPI-NOR and MTD layers to support stacked and parallel configurations. This new layer will be part of spi-nor, located in mtd/spi-nor/, can be included/excluded via Kconfig, will be maintained by AMD and will: - During probing, store information from all connected flashes in stacked or parallel mode and present them as a single device to the MTD layer. - Register callbacks and manage MTD device registration within the new layer instead of spi-nor/core.c. - Make minimal changes in spi-nor/core.c, as stacked and parallel handling will be managed by the new layer on top of SPI-NOR. - Handle odd byte count requests from the MTD layer during flash operations in parallel mode. [1] https://docs.amd.com/r/en-US/am011-versal-acap-trm/QSPI-Flash-Device-Interf… [2] https://ww1.microchip.com/downloads/aemDocuments/documents/MPU32/ProductDoc… [3] https://lore.kernel.org/all/20191113171505.26128-4-miquel.raynal@bootlin.co… [4] https://lore.kernel.org/all/20191127105522.31445-5-miquel.raynal@bootlin.co… [5]https://lore.kernel.org/all/20211112152411.818321-1-miquel.raynal@bootlin… [6] https://github.com/torvalds/linux/commit/f89504300e94524d5d5846ff8b728592ac… [7] https://github.com/torvalds/linux/commit/eba5368503b4291db7819512600fa014ea… [8] https://github.com/torvalds/linux/commit/e2edd1b64f1c79e8abda365149ed62a2a9… [9]https://github.com/torvalds/linux/commit/4d8ff6b0991d5e86b17b235fc46ec62e… Thanks, Amit Amit Kumar Mahapatra (2): dt-bindings: mtd: Add bindings for describing concatinated MTD devices dt-bindings: spi: Update stacked and parallel bindings .../mtd/partitions/fixed-partitions.yaml | 18 +++++++++++++++ .../bindings/mtd/partitions/partitions.yaml | 6 +++++ .../bindings/spi/spi-controller.yaml | 23 +++++++++++++++++-- .../bindings/spi/spi-peripheral-props.yaml | 9 +++----- 4 files changed, 48 insertions(+), 8 deletions(-) -- 2.34.1

6 15

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

3 10

ASoC Kernel 6.1: Query regarding dynamic dai link/dapm registration during bootup
by Shalini Manjunatha (Consultant) (QUIC) 08 Nov '24

08 Nov '24

Hi All, On kernel6.1 based Qualcomm target for automotive platform based on Android U, we have a feature early services to launch certain early apps at early init of android bootup cycle. Hence in order to launch early chime sound of audio at early init of android boot up cycle, we need to load all required audio kernel modules at early init stage in order to get sound card registered. We have certain KPI goals for every early apps to get completed within lesser duration so that we can achieve final KPI goal of android boot complete as per the target specification. In this journey, while optimizing the sound card registration through machine driver probe to achieve lower KPI for audio early chime sound, we find that for early chime launch we required only selected pcm front end dai links only. Hence if we keep FE dai link structure in our machine driver with only required items for early chime, overall sound card registrations gets completed within 200msec from the time it starts, which is a very good KPI number to achieve. So we were trying out code change to split overall FE dai link structure into two, one for early chime and another for all other required dai link's, then some how we need to register early chime related dai link during early init so that sound card registration process gets completed quickly, later stage we want to dynamically load all other FE dai links. Could you please help us to suggest on how can we achieve this?. Is there any possible way out to avoid static FE dai link registration for all required ones from machine driver during machine driver probe, instead register only limited list of dai links. Later can we dynamically register the dai links in order to get PCM devices entries created based on use case. Currently we have this API devm_snd_soc_register_card() being called with snd_soc_card parameter which has all possible FE dai links populated at once. Queries: 1) can we register sound card twice during bootup, once with limited dai links and next with all required ones? 2) can we register dai links/dapm dynamically based on use case requriement? 3) If it is possible, could you share any reference from any other vendor who is able to achieve the same?. Regards, Shalini M

3 3

[PATCH 0/8] ASoC: cleasnup rtd and its ID
by Kuninori Morimoto 05 Nov '24

05 Nov '24

Hi Mark rtd has own ID, but it is naming "num" for it. The naming is confusable. This patch rename it to "id". And many functions request both "rtd" and its "id" as function parameter, but rtd itself has rtd->id. This patch cleanup it. And, Qcom driver want to use irregular rtd ID because of its topology, and thus, soc-core need irregular calculation. I'm not sure why only Qcom needs such calculation, but this patch also cleanup it. But I guess we want to cleanup is not soc-core but Qcom side (?) Kuninori Morimoto (8): ASoC: rename rtd->num to rtd->id ASoC: fsl: switch to use rtd->id from rtd->num ASoC: meson: switch to use rtd->id from rtd->num ASoC: sh: switch to use rtd->id from rtd->num ASoC: generic: switch to use rtd->id from rtd->num ASoC: remove rtd->num ASoC: soc-core: do rtd->id trick at snd_soc_add_pcm_runtime() ASoC: cleanup function parameter for rtd and its id include/sound/soc-dai.h | 5 ++- include/sound/soc.h | 8 ++--- sound/soc/fsl/imx-card.c | 6 ++-- sound/soc/generic/simple-card-utils.c | 10 +++--- sound/soc/meson/axg-card.c | 6 ++-- sound/soc/meson/gx-card.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 9 +++-- sound/soc/soc-core.c | 50 +++++++++++++++------------ sound/soc/soc-dai.c | 4 +-- sound/soc/soc-pcm.c | 16 ++++----- 11 files changed, 60 insertions(+), 58 deletions(-) -- 2.43.0

3 11

[PATCH] ASoC: qcom: x1e80100: Support boards with two speakers
by Krzysztof Kozlowski 05 Nov '24

05 Nov '24

Some Qualcomm X1E laptops have only two speakers. Regardless whether this sound card driver is suitable for them (we could re-use one for some older SoC), we should set reasonable channel map depending on the number of channels, not always 4-speaker setup. This change is necessary for bringing audio support on Lenovo Thinkpad T14s with Qualcomm X1E78100 and only two speakers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/x1e80100.c | 40 ++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index 898b5c26bf1e..8eb57fc12f0d 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -95,23 +95,53 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); } +static int x1e80100_snd_hw_map_channels(unsigned int *ch_map, int num) +{ + switch (num) { + case 1: + ch_map[0] = PCM_CHANNEL_FC; + break; + case 2: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + break; + case 3: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_FC; + break; + case 4: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_LB; + ch_map[2] = PCM_CHANNEL_FR; + ch_map[3] = PCM_CHANNEL_RB; + break; + default: + return -EINVAL; + } + + return 0; +} + static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; - const unsigned int rx_slot[4] = { PCM_CHANNEL_FL, - PCM_CHANNEL_LB, - PCM_CHANNEL_FR, - PCM_CHANNEL_RB }; + unsigned int channels = substream->runtime->channels; + unsigned int rx_slot[4]; int ret; switch (cpu_dai->id) { case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: + ret = x1e80100_snd_hw_map_channels(rx_slot, channels); + if (ret) + return ret; + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, - ARRAY_SIZE(rx_slot), rx_slot); + channels, rx_slot); if (ret) return ret; break; -- 2.43.0

2 1

[PATCH] ASoC: codecs: ES8326: Reduce pop noise
by Zhang Yi 05 Nov '24

05 Nov '24

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 | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index aa3e364827c8..a5603b617688 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, 0x00); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); } } } else { @@ -1082,7 +1082,7 @@ 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, 0x03); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

2 1

Add support for ASRock X870E Taichi
by GitHub pull_request - opened 01 Nov '24

01 Nov '24

alsa-project/alsa-ucm-conf pull request #457 was opened from julmajustus: Included id 26ce:0a0b to realtek-alc4080 list. USB Audio S/PDIF Output, USB Audio Rear input and USB Audio Front Headphones seems to work now without issues with HiFi 2.0 channels. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/457 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/457.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Direct MIDI event delivery to ALSA sequencer's hardware port without any connections/subscriptions fails with -ENODEV.
by giedrius＠blokas.io 31 Oct '24

31 Oct '24

Hello, I had written an `aseqsend` utility to send MIDI events from hex strings to ALSA sequencer ports, similar to the -S arg of `amidi`, but I just noticed that while sending events directly to application ALSA Sequencer ports works just fine, trying to send to a hardware MIDI port fails with -19 (-ENODEV). Here's the aseqsend utility code, the highlighted line is the one returning -ENODEV: https://github.com/BlokasLabs/aseqsend/blob/main/aseqsend.c#L173 Here's some sample commands showcasing the problem (I've added a printf with drain result just after the indicated line above locally): https://paste.debian.net/hidden/9690d660/ It shows that in case there's no active subscriptions to the port, the event delivery fails, with no data ever reaching the hardware device. As long as there's at least one subscription to the port, it works ok. I've reproduced this on 5.15.36-rt41-v7+ and 6.6.58-v8-16k (running on Raspberry Pi, based on https://github.com/raspberrypi/linux) I've traced the -ENODEV to be returned at this location: https://elixir.bootlin.com/linux/v6.6.58/source/sound/core/seq/seq_midi.c#L… Whenever there's a subscription, the 'substream' is not NULL. But what I would naturally expect is for the events to get delivered to the hardware port when using direct addressing, regardless of its subscription status, the same way how it's possible to send events to application aseq ports: -- Terminal 1 -- patch@patchbox:~ $ aseqdump Waiting for data at port 128:0. Press Ctrl+C to end. Source Event Ch Data 129:0 Note on 0, note 64, velocity 48 -- Terminal 2 -- patch@patchbox:~/work/aseqsend $ ./aseqsend 128:0 90 40 30 drain: 0 Is this a bug in ALSA Sequencer, or is my expectation of being able to write directly to a hardware ALSA sequencer port incorrect?

3 3

[PATCH v29 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 31 Oct '24

31 Oct '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v29: - Fixed some phrases/wording within the SOC USB documentation, and also added an output with aplay -l for the example output. - Fixed allocated string buffer for creating the USB SND offload mixer, and added a PCM index check to ensure that the pcm index is less than the expected number. - Added a complement enable jack call if USB backend DAI link drivers need access to it. Changes in v28: - Updated comments and commit log in the stop endpoint sync patch. Clarified that the default stop endpoint completion routine won't fully run as expected since it has a completion associated w/ the command. - Added a null check for sb->xhci within xhci_sideband_create_interrupter(). This is to just ensure that caller has registered sideband before calling create interrupter. Changes in v27: - Added some comments and notes about the offload design. Enforcing the q6routing to only allow one USB mixer (PCM device) to be enabled at a time. - Modified SND_JACK_USB notifications for all USB audio offloadable devices plugged in - Rebased on latest XHCI secondary interrupter IMOD changes upstream. Modified the change in this series to allow for XHCI sideband to set the IMOD for sideband clients. - Updated documentation on how USB SND kcontrols are involved in the overall design. - Remove mutex locking from suspend/resume platform ops, as USB core ensures that the interface and device are in the RPM_ACTIVE state while disconnect is handled. Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (3): xhci: support setting interrupt moderation IMOD for secondary interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (30): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: xhci-sideband: Set IMOD for xHCI sideband clients usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 491 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 37 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 51 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 70 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 147 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 32 +- sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 369 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 106 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1974 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 +++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5957 insertions(+), 57 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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[GIT PULL] ASoC fixes for v6.12-rc5
by Mark Brown 30 Oct '24

30 Oct '24

The following changes since commit db7e59e6a39a4d3d54ca8197c796557e6d480b0d: ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc (2024-10-22 21:00:38 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc5 for you to fetch changes up to 2db63e92186d7201ee1cb2f5af11757c5e5a1020: wcd937x codec fixes (2024-10-29 19:18:48 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.12 The biggest set of changes here is Hans' fixes and quirks for various Baytrail based platforms with RT5640 CODECs, and there's one core fix for a missed length assignment for __counted_by() checking. Otherwise it's small device specific fixes, several of them in the DT bindings. ---------------------------------------------------------------- Aleksei Vetrov (1): ASoC: dapm: fix bounds checker error in dapm_widget_list_create Alexey Klimov (2): ASoC: codecs: wcd937x: add missing LO Switch control ASoC: codecs: wcd937x: relax the AUX PDM watchdog Christophe JAILLET (1): ASoC: cs42l51: Fix some error handling paths in cs42l51_probe() Dmitry Yashin (1): ASoC: dt-bindings: rockchip,rk3308-codec: add port property Hans de Goede (5): ASoC: codecs: rt5640: Always disable IRQs from rt5640_cancel_work() ASoC: Intel: bytcr_rt5640: Add support for non ACPI instantiated codec ASoC: Intel: bytcr_rt5640: Add DMI quirk for Vexia Edu Atla 10 tablet ASoC: Intel: sst: Support LPE0F28 ACPI HID ASoC: Intel: sst: Fix used of uninitialized ctx to log an error Mark Brown (1): wcd937x codec fixes .../bindings/sound/rockchip,rk3308-codec.yaml | 4 ++ sound/hda/intel-dsp-config.c | 4 ++ sound/soc/codecs/cs42l51.c | 7 ++- sound/soc/codecs/rt5640.c | 27 +++++---- sound/soc/codecs/wcd937x.c | 12 +++- sound/soc/codecs/wcd937x.h | 4 ++ sound/soc/intel/atom/sst/sst_acpi.c | 64 +++++++++++++++++++--- sound/soc/intel/boards/bytcr_rt5640.c | 48 +++++++++++++++- sound/soc/soc-dapm.c | 2 + 9 files changed, 144 insertions(+), 28 deletions(-)

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Why Choose a Dedicated Server for Your Business?
by David Menk 29 Oct '24

29 Oct '24

Hello everyone, I wanted to share some insights on why dedicated server hosting can be a game-changer for businesses, especially those with high-traffic websites or specific performance needs. What is a Dedicated Server? A dedicated server is a physical server that's exclusively used by one client. Unlike shared hosting, where resources are split between multiple users, a dedicated server gives you complete control over CPU, RAM, storage, and bandwidth, making it ideal for websites or applications that need consistent performance, security, and reliability. Benefits of Using a Dedicated Server: Performance and Speed: With dedicated resources, your website or application can handle high volumes of traffic without slowing down. This is especially important for e-commerce sites or any business where speed impacts the user experience. Enhanced Security: Dedicated servers offer stronger security, as you're not sharing the server with other users. You can implement custom security measures tailored to your business, helping to protect against potential breaches and data loss. Full Control and Customization: Since the server is dedicated to you, you have complete control over the operating system, software, and applications you want to run. This level of customization can be essential for companies with specific software requirements. Scalability: As your business grows, dedicated servers can be upgraded with additional resources to meet your increasing demands, providing flexibility without the need to migrate to another server. Better Reliability: Dedicated servers are less prone to outages or downtime compared to shared hosting, providing a stable environment that’s crucial for businesses that rely on 24/7 availability. Is a Dedicated Server Right for You? If your website or application has high traffic, complex applications, or sensitive data, a dedicated server could be the right choice. It's a bit more of an investment, but the control, performance, and reliability often pay off in the long run. If anyone has experience with dedicated servers or is considering making the switch, I’d love to hear your thoughts and any advice you might have! Looking forward to the discussion!

1 1

[PATCH] ALSA: usb-audio: Add Pioneer DJ/AlphaTheta DJM-A9 Mixer
by Sarah Grant 29 Oct '24

29 Oct '24

This also includes a change to the capture level as they differ with this device, as well as additional values for prefader channels. Acked-by: Olivia Mackintosh <livvy(a)base.nu> Signed-off-by: Sarah Grant <s(a)srd.tw> --- sound/usb/mixer_quirks.c | 68 +++++++++++++++++++++++++++++++++++++--- sound/usb/quirks-table.h | 57 +++++++++++++++++++++++++++++++++ 2 files changed, 121 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2a9594f34dac..15c300eddf53 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3498,7 +3498,7 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) } /* - * Pioneer DJ DJM Mixers + * Pioneer DJ / AlphaTheta DJM Mixers * * These devices generally have options for soft-switching the playback and * capture sources in addition to the recording level. Although different @@ -3515,17 +3515,23 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_CAP_CDLINE 0x01 #define SND_DJM_CAP_DIGITAL 0x02 #define SND_DJM_CAP_PHONO 0x03 +#define SND_DJM_CAP_PREFADER 0x05 #define SND_DJM_CAP_PFADER 0x06 #define SND_DJM_CAP_XFADERA 0x07 #define SND_DJM_CAP_XFADERB 0x08 #define SND_DJM_CAP_MIC 0x09 #define SND_DJM_CAP_AUX 0x0d #define SND_DJM_CAP_RECOUT 0x0a +#define SND_DJM_CAP_RECOUT_NOMIC 0x0e #define SND_DJM_CAP_NONE 0x0f #define SND_DJM_CAP_CH1PFADER 0x11 #define SND_DJM_CAP_CH2PFADER 0x12 #define SND_DJM_CAP_CH3PFADER 0x13 #define SND_DJM_CAP_CH4PFADER 0x14 +#define SND_DJM_CAP_CH1PREFADER 0x31 +#define SND_DJM_CAP_CH2PREFADER 0x32 +#define SND_DJM_CAP_CH3PREFADER 0x33 +#define SND_DJM_CAP_CH4PREFADER 0x34 // Playback types #define SND_DJM_PB_CH1 0x00 @@ -3551,6 +3557,7 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_900NXS2_IDX 0x3 #define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_450_IDX 0x5 +#define SND_DJM_A9_IDX 0x6 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -3579,7 +3586,7 @@ struct snd_djm_ctl { u16 wIndex; }; -static const char *snd_djm_get_label_caplevel(u16 wvalue) +static const char *snd_djm_get_label_caplevel_common(u16 wvalue) { switch (wvalue) { case 0x0000: return "-19dB"; @@ -3590,6 +3597,20 @@ static const char *snd_djm_get_label_caplevel(u16 wvalue) } }; +// The DJM-A9 has different capture levels than other, older models +static const char *snd_djm_get_label_caplevel_a9(u16 wvalue) +{ + switch (wvalue) { + case 0x0000: return "+15dB"; + case 0x0100: return "+12dB"; + case 0x0200: return "+9dB"; + case 0x0300: return "+6dB"; + case 0x0400: return "+3dB"; + case 0x0500: return "0dB"; + default: return NULL; + } +}; + static const char *snd_djm_get_label_cap_common(u16 wvalue) { switch (wvalue & 0x00ff) { @@ -3602,8 +3623,13 @@ static const char *snd_djm_get_label_cap_common(u16 wvalue) case SND_DJM_CAP_XFADERB: return "Cross Fader B"; case SND_DJM_CAP_MIC: return "Mic"; case SND_DJM_CAP_RECOUT: return "Rec Out"; + case SND_DJM_CAP_RECOUT_NOMIC: return "Rec Out without Mic"; case SND_DJM_CAP_AUX: return "Aux"; case SND_DJM_CAP_NONE: return "None"; + case SND_DJM_CAP_CH1PREFADER: return "Pre Fader Ch1"; + case SND_DJM_CAP_CH2PREFADER: return "Pre Fader Ch2"; + case SND_DJM_CAP_CH3PREFADER: return "Pre Fader Ch3"; + case SND_DJM_CAP_CH4PREFADER: return "Pre Fader Ch4"; case SND_DJM_CAP_CH1PFADER: return "Post Fader Ch1"; case SND_DJM_CAP_CH2PFADER: return "Post Fader Ch2"; case SND_DJM_CAP_CH3PFADER: return "Post Fader Ch3"; @@ -3623,6 +3649,14 @@ static const char *snd_djm_get_label_cap_850(u16 wvalue) } }; +static const char *snd_djm_get_label_caplevel(u8 device_idx, u16 wvalue) +{ + switch (device_idx) { + case SND_DJM_A9_IDX: return snd_djm_get_label_caplevel_a9(wvalue); + default: return snd_djm_get_label_caplevel_common(wvalue); + } +}; + static const char *snd_djm_get_label_cap(u8 device_idx, u16 wvalue) { switch (device_idx) { @@ -3644,7 +3678,7 @@ static const char *snd_djm_get_label_pb(u16 wvalue) static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex) { switch (windex) { - case SND_DJM_WINDEX_CAPLVL: return snd_djm_get_label_caplevel(wvalue); + case SND_DJM_WINDEX_CAPLVL: return snd_djm_get_label_caplevel(device_idx, wvalue); case SND_DJM_WINDEX_CAP: return snd_djm_get_label_cap(device_idx, wvalue); case SND_DJM_WINDEX_PB: return snd_djm_get_label_pb(wvalue); default: return NULL; @@ -3653,7 +3687,7 @@ static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex) // common DJM capture level option values static const u16 snd_djm_opts_cap_level[] = { - 0x0000, 0x0100, 0x0200, 0x0300 }; + 0x0000, 0x0100, 0x0200, 0x0300, 0x400, 0x500 }; // DJM-250MK2 @@ -3795,6 +3829,28 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { }; +// DJM-A9 +static const u16 snd_djm_opts_a9_cap1[] = { + 0x0107, 0x0108, 0x0109, 0x010a, 0x010e, + 0x111, 0x112, 0x113, 0x114, 0x0131, 0x132, 0x133, 0x134 }; +static const u16 snd_djm_opts_a9_cap2[] = { + 0x0201, 0x0202, 0x0203, 0x0205, 0x0206, 0x0207, 0x0208, 0x0209, 0x020a, 0x020e }; +static const u16 snd_djm_opts_a9_cap3[] = { + 0x0301, 0x0302, 0x0303, 0x0305, 0x0306, 0x0307, 0x0308, 0x0309, 0x030a, 0x030e }; +static const u16 snd_djm_opts_a9_cap4[] = { + 0x0401, 0x0402, 0x0403, 0x0405, 0x0406, 0x0407, 0x0408, 0x0409, 0x040a, 0x040e }; +static const u16 snd_djm_opts_a9_cap5[] = { + 0x0501, 0x0502, 0x0503, 0x0505, 0x0506, 0x0507, 0x0508, 0x0509, 0x050a, 0x050e }; + +static const struct snd_djm_ctl snd_djm_ctls_a9[] = { + SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Master Input", a9_cap1, 3, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch1 Input", a9_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch2 Input", a9_cap3, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch3 Input", a9_cap4, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch4 Input", a9_cap5, 2, SND_DJM_WINDEX_CAP) +}; + static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_250MK2_IDX] = SND_DJM_DEVICE(250mk2), [SND_DJM_750_IDX] = SND_DJM_DEVICE(750), @@ -3802,6 +3858,7 @@ static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_900NXS2_IDX] = SND_DJM_DEVICE(900nxs2), [SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2), [SND_DJM_450_IDX] = SND_DJM_DEVICE(450), + [SND_DJM_A9_IDX] = SND_DJM_DEVICE(a9), }; @@ -4076,6 +4133,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ err = snd_djm_controls_create(mixer, SND_DJM_900NXS2_IDX); break; + case USB_ID(0x2b73, 0x003c): /* Pioneer DJ / AlphaTheta DJM-A9 */ + err = snd_djm_controls_create(mixer, SND_DJM_A9_IDX); + break; } return err; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 24c981c9b240..b758655b948c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3119,6 +3119,63 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* + * Pioneer DJ / AlphaTheta DJM-A9 + * 10 channels playback & 12 channels capture @ 44.1/48/96kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x003c), + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 44100, 48000, 96000 + } + } + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 44100, 48000, 96000 + } + } + }, + QUIRK_COMPOSITE_END + } + } +}, + /* * MacroSilicon MS2100/MS2106 based AV capture cards * -- 2.40.1

2 1

[PATCH] ALSA: hda/realtek: Add subwoofer quirk for Infinix ZERO BOOK 13
by Piyush Raj Chouhan 29 Oct '24

29 Oct '24

Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Also The mic in this laptop suffers too high gain resulting in mostly fan noise being recorded, This patch Also limit mic boost. HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b Test: All 4 speaker works, Mic has low noise. Signed-off-by: Piyush Raj Chouhan <piyushchouhan1598(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3567b14b52b7..075935707f8a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7551,6 +7551,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7996,6 +7997,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 }, /* use as internal speaker (back) */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10989,6 +11000,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), -- 2.46.2

2 1

[PATCH v5 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 29 Oct '24

29 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20241002093904.180979… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): 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 include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 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 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

1 6

[PATCH] ASoC: codecs: ES8326: Modify the configuration of and micbias
by Zhang Yi 28 Oct '24

28 Oct '24

Because we designed a new version of ES8326, the configuration of micbias needed to be modified.We tested the new driver, on both the new version and the old one. It works well. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index d5362b3be484..aa3e364827c8 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -614,6 +614,10 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, 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, 0x00); + } } } else { if (!es8326->calibrated) { @@ -640,6 +644,10 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) ES8326_MUTE_MASK, ~(ES8326_MUTE)); } 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_VMIDSEL, 0x40, 0x00); + } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, 0x0F, 0x00); } @@ -821,7 +829,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) iface = snd_soc_component_read(comp, ES8326_HPDET_STA); dev_dbg(comp->dev, "gpio flag %#04x", iface); - if ((es8326->jack_remove_retry == 1) && (es8326->version != ES8326_VERSION_B)) { + if ((es8326->jack_remove_retry == 1) && (es8326->version < ES8326_VERSION_B)) { if (iface & ES8326_HPINSERT_FLAG) es8326->jack_remove_retry = 2; else @@ -859,7 +867,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) /* * Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event */ - if ((es8326->jack_remove_retry == 0) && (es8326->version != ES8326_VERSION_B)) { + if ((es8326->jack_remove_retry == 0) && (es8326->version < ES8326_VERSION_B)) { es8326->jack_remove_retry = 1; dev_dbg(comp->dev, "remove event check, invert HPJACK_POL, cnt = %d\n", es8326->jack_remove_retry); @@ -954,7 +962,7 @@ static int es8326_calibrate(struct snd_soc_component *component) regmap_read(es8326->regmap, ES8326_CHIP_VERSION, &reg); es8326->version = reg; - if ((es8326->version == ES8326_VERSION_B) && (es8326->calibrated == false)) { + if ((es8326->version >= ES8326_VERSION_B) && (es8326->calibrated == false)) { dev_dbg(component->dev, "ES8326_VERSION_B, calibrating\n"); regmap_write(es8326->regmap, ES8326_CLK_INV, 0xc0); regmap_write(es8326->regmap, ES8326_CLK_DIV1, 0x03); @@ -1047,7 +1055,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_DAC_VPPSCALE, 0x15); regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x80 | - ((es8326->version == ES8326_VERSION_B) ? + ((es8326->version >= ES8326_VERSION_B) ? (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol) : (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol | 0x04))); usleep_range(5000, 10000); @@ -1073,6 +1081,10 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); 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, 0x03); + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + } msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

2 1

[PATCH] ALSA: hda/realtek: Add subwoofer quirk for Infinix ZERO BOOK 13
by Piyush Raj Chouhan 28 Oct '24

28 Oct '24

From: pc1598 <piyushchouhan1598(a)gmail.com> Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Also The mic in this laptop suffers too high gain resulting in mostly fan noise being recorded, This patch Also limit mic boost. HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b Test: All 4 speaker works, Mic has low noise. Signed-off-by: Piyush Raj Chouhan <piyushchouhan1598(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3567b14b52b7..075935707f8a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7551,6 +7551,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7996,6 +7997,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 }, /* use as internal speaker (back) */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10989,6 +11000,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), -- 2.46.2

2 1

[tiwai-sound:topic/compress-accel 1/1] sound/core/compress_offload.c:409:35: warning: variable 'runtime' set but not used
by kernel test robot 25 Oct '24

25 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/compress-accel head: 04177158cf98a79744937893b100020d77e6f9ac commit: 04177158cf98a79744937893b100020d77e6f9ac [1/1] ALSA: compress_offload: introduce accel operation mode config: arm-randconfig-001-20241026 (https://download.01.org/0day-ci/archive/20241026/202410260257.PNGCF9yq-lkp@…) compiler: arm-linux-gnueabi-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241026/202410260257.PNGCF9yq-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/202410260257.PNGCF9yq-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/core/compress_offload.c: In function 'snd_compr_poll': >> sound/core/compress_offload.c:409:35: warning: variable 'runtime' set but not used [-Wunused-but-set-variable] 409 | struct snd_compr_runtime *runtime; | ^~~~~~~ vim +/runtime +409 sound/core/compress_offload.c 404 405 static __poll_t snd_compr_poll(struct file *f, poll_table *wait) 406 { 407 struct snd_compr_file *data = f->private_data; 408 struct snd_compr_stream *stream; > 409 struct snd_compr_runtime *runtime; 410 size_t avail; 411 __poll_t retval = 0; 412 413 if (snd_BUG_ON(!data)) 414 return EPOLLERR; 415 416 stream = &data->stream; 417 runtime = stream->runtime; 418 419 guard(mutex)(&stream->device->lock); 420 421 switch (stream->runtime->state) { 422 case SNDRV_PCM_STATE_OPEN: 423 case SNDRV_PCM_STATE_XRUN: 424 return snd_compr_get_poll(stream) | EPOLLERR; 425 default: 426 break; 427 } 428 429 poll_wait(f, &stream->runtime->sleep, wait); 430 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Help to solve pop when sound loops
by Charles Mulder 25 Oct '24

25 Oct '24

Hi. I hope this is the correct place to ask for help relating to ALSA's C lib? I'm working on a wavetable project for college. Please see attached. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on buffer > period > frames, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C -- Sent with Tuta; enjoy secure & ad-free emails: https://tuta.com

1 0

Help to solve pop when sound loops
by GitHub issues - edited 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #417 was edited from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ``` #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

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Help to address pop at end of buffer
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #417 was opened from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ``` #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALC225 the microphone does not work in jack3.5
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #416 was opened from PavelGu: I have notebook clevo NL55AU with Ubuntu24.04 installed. The microphone does not work on a headset connected to a combined audio output jack 3.5. Logs: https://alsa-project.org/db/?f=89d2b631cb1635dd48357610d54e062bddef81b4 Can you help me? Issue URL : https://github.com/alsa-project/alsa-lib/issues/416 Repository URL: https://github.com/alsa-project/alsa-lib

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[Discussion] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #415 was opened from xeroxslayer: I raised this discussion a few months ago on the ALSA mailing list, but there was no response regarding this. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). Now, this profile will be autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software, I wouldn't know, I haven't actually tried adding a mono profile manually), but it seemed odd that one has to go through such trouble just to listen to mono sound... something that should be available out of the box IMO. Issue URL : https://github.com/alsa-project/alsa-lib/issues/415 Repository URL: https://github.com/alsa-project/alsa-lib

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Limit internal Mic boost on Dell platform
by Kailang 25 Oct '24

25 Oct '24

Hi Takashi, Attach patch will limit internal Mic boost on all Dell platform. BR, Kailang

2 1

Update ucm conf for MediaTek Genio platforms
by GitHub pull_request - opened 24 Oct '24

24 Oct '24

alsa-project/alsa-ucm-conf pull request #456 was opened from zoran055: 1. Add dynamic configuration to detect HDMI/DP devices for mt8370-evk, mt8390-evk and mt8395-evk. 2. Add headset jack detection support for mt8370-evk, mt8390-evk and mt8395-evk. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/456 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/456.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.12-rc4
by Mark Brown 24 Oct '24

24 Oct '24

The following changes since commit 8cf0b93919e13d1e8d4466eb4080a4c4d9d66d7b: Linux 6.12-rc2 (2024-10-06 15:32:27 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc4 for you to fetch changes up to db7e59e6a39a4d3d54ca8197c796557e6d480b0d: ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc (2024-10-22 21:00:38 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.12 An uncomfortably large set of fixes due to me not getting round to sending them for longer than I should due to travel and illness. This is mostly smaller driver specific changes, but there are a couple of generic changes: - Bumping the minimal topology ABI we check for during validation, the code had support for v4 removed previously but the update of the define used for initial validation was missed. - Fix the assumption that DAPM structs will be embedded in a component which isn't true for card widgets when doing name comparisons, though fortunately this is rarely triggered. We've pulled in one Soundwire fix which was part of a larger series fixing cleanup issues in on Intel Soundwire systems. ---------------------------------------------------------------- Alexey Klimov (2): ASoC: qcom: sdm845: add missing soundwire runtime stream alloc ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs Amadeusz Sławiński (2): ASoC: Intel: avs: Update stream status in a separate thread ASoC: topology: Bump minimal topology ABI version Andrei Simion (1): MAINTAINERS: Update maintainer list for MICROCHIP ASOC, SSC and MCP16502 drivers Benjamin Bara (1): ASoC: dapm: avoid container_of() to get component Binbin Zhou (1): ASoC: loongson: Fix component check failed on FDT systems Chancel Liu (1): ASoC: fsl_micfil: Add a flag to distinguish with different volume control types Christian Heusel (1): ASoC: amd: yc: Add quirk for ASUS Vivobook S15 M3502RA Colin Ian King (1): ASoC: max98388: Fix missing increment of variable slot_found Derek Fang (1): ASoC: Intel: soc-acpi: lnl: Add match entry for TM2 laptops Ilya Dudikov (1): ASoC: amd: yc: Fix non-functional mic on ASUS E1404FA Jack Yu (1): ASoC: rt722-sdca: increase clk_stop_timeout to fix clock stop issue Kai Vehmanen (1): ASoC: SOF: Intel: hda-loader: do not wait for HDaudio IOC Kirill Marinushkin (1): ASoC: Change my e-mail to gmail Krzysztof Kozlowski (2): ASoC: qcom: Select missing common Soundwire module code on SDM845 ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc Lad Prabhakar (1): ASoC: rsnd: Fix probe failure on HiHope boards due to endpoint parsing Mark Brown (1): ASoC/SoundWire: clean up link DMA during stop for IPC4 Miquel Raynal (1): ASoC: dt-bindings: davinci-mcasp: Fix interrupt properties Ranjani Sridharan (4): ASoC: SOF: ipc4-topology: Do not set ALH node_id for aggregated DAIs ASoC: SOF: Intel: hda: Handle prepare without close for non-HDA DAI's soundwire: intel_ace2x: Send PDI stream number during prepare ASoC: SOF: Intel: hda: Always clean up link DMA during stop Shengjiu Wang (2): ASoC: fsl_esai: change dev_warn to dev_dbg in irq handler ASoC: fsl_micfil: Add sample rate constraint Venkata Prasad Potturu (2): ASoC: SOF: amd: Add error log for DSP firmware validation failure ASoC: SOF: amd: Fix for ACP SRAM addr for acp7.0 platform Zhu Jun (1): ASoC: codecs: Fix error handling in aw_dev_get_dsp_status function Zichen Xie (1): ASoC: qcom: Fix NULL Dereference in asoc_qcom_lpass_cpu_platform_probe() .../bindings/sound/davinci-mcasp-audio.yaml | 18 ++--- MAINTAINERS | 5 +- drivers/soundwire/intel_ace2x.c | 19 ++--- include/uapi/sound/asoc.h | 2 +- sound/soc/amd/yc/acp6x-mach.c | 14 ++++ sound/soc/codecs/aw88399.c | 2 +- sound/soc/codecs/lpass-rx-macro.c | 15 ++-- sound/soc/codecs/max98388.c | 1 + sound/soc/codecs/pcm3060-i2c.c | 4 +- sound/soc/codecs/pcm3060-spi.c | 4 +- sound/soc/codecs/pcm3060.c | 4 +- sound/soc/codecs/pcm3060.h | 2 +- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/fsl/fsl_esai.c | 4 +- sound/soc/fsl/fsl_micfil.c | 81 +++++++++++++++++++++- sound/soc/intel/avs/core.c | 3 +- sound/soc/intel/avs/pcm.c | 19 +++++ sound/soc/intel/avs/pcm.h | 16 +++++ sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 38 ++++++++++ sound/soc/loongson/loongson_card.c | 1 + sound/soc/qcom/Kconfig | 2 + sound/soc/qcom/lpass-cpu.c | 2 + sound/soc/qcom/sc7280.c | 10 ++- sound/soc/qcom/sdm845.c | 7 +- sound/soc/sh/rcar/core.c | 7 +- sound/soc/soc-dapm.c | 4 +- sound/soc/sof/amd/acp-loader.c | 5 +- sound/soc/sof/amd/acp.c | 4 +- sound/soc/sof/intel/hda-dai-ops.c | 23 +++--- sound/soc/sof/intel/hda-dai.c | 37 ++++++++-- sound/soc/sof/intel/hda-loader.c | 17 ----- sound/soc/sof/ipc4-topology.c | 15 +++- 32 files changed, 298 insertions(+), 89 deletions(-) create mode 100644 sound/soc/intel/avs/pcm.h

2 1

[PATCH v4 0/7] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 24 Oct '24

24 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Add it to this patch set. Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Jaroslav Kysela (1): ALSA: compress_offload: introduce accel operation mode Shengjiu Wang (6): 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 .../sound/designs/compress-accel.rst | 136 ++++ include/sound/compress_driver.h | 46 ++ include/uapi/sound/compress_offload.h | 63 +- include/uapi/sound/compress_params.h | 23 +- sound/core/Kconfig | 3 + sound/core/compress_offload.c | 351 ++++++++- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 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 + 14 files changed, 1851 insertions(+), 17 deletions(-) create mode 100644 Documentation/sound/designs/compress-accel.rst create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

4 10

Demo application for accel operation.
by GitHub pull_request - opened 24 Oct '24

24 Oct '24

alsa-project/tinycompress pull request #26 was opened from TE-N-ShengjiuWang: This 'caccel' applicaton is a demo to show how to call the new introduced accel interface. The kernel patch is: ALSA: compress_offload: introduce accel operation mode As there is conflict when using the API of alsa-lib, add alsa_wrap file to separate the header files: alsa/asoundlib.h sound/asound.h command for conversion to 48kHz: caccel -i in.wav -o out.wav -r 48000 Request URL : https://github.com/alsa-project/tinycompress/pull/26 Patch URL : https://github.com/alsa-project/tinycompress/pull/26.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH v5 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 23 Oct '24

23 Oct '24

Hi Mark I got kernel-test report that v4 had compile warning. This is v5 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v4 -> v5 - fixup sumsung odroid compile warning - rebased on latest asoc/for-6.13 v3 -> v4 - adjust latest amd - rebased on asoc/for-6.13 v2 -> v3 - tidyup Intel files v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r091lwhg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87y138vbv7.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87o73mqj5b.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 11 +-- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 584 deletions(-) -- 2.43.0

2 14

[PATCH v4 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 23 Oct '24

23 Oct '24

Hi Mark 2 weeks past. This is v4 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v3 -> v4 - adjust latest amd - rebased on asoc/for-6.13 v2 -> v3 - tidyup Intel files v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r091lwhg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87y138vbv7.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 583 deletions(-) -- 2.43.0

2 14

Update default depop procedure
by Kailang 23 Oct '24

23 Oct '24

Hi Takashi, Attach patch will solve issue which Headphone has a chance no sound. Many Thanks. BR, Kailang

2 1

ALC4080 onboard audio support on the MSI X870 Tomahawk motherboard
by GitHub issues - opened 23 Oct '24

23 Oct '24

alsa-project/alsa-ucm-conf issue #455 was opened from nfp0: @perexg As mentioned [here](https://github.com/alsa-project/alsa-ucm-conf/issues/447#issuecomment…, it seems the ALC4080 audio on the new MSI X870 Tomahawk motherboard is not supported. USB ID is `0db0:cd0e` I have tested adding `cd0e` on the `USB-Audio.conf` file like so: ``` Regex "USB((0414:a0(0e|1[0124]))|(0b05:(19(84|9[69])|1a(16|2[07]|5[23c]|97|f1)))|(0db0:(cd0e|005a|124b|151f|1feb|3130|36e7|4(19c|22d|240|88c)|62a4|6c[0c]9|70d3|7696|82c7|8af7|961e|a(073|228|47c|74b)|b202|d1d7|d6e7))|(26ce:0a0[68]))" ``` With this, the microphone now seems to be correctly mapped and works. But the problem is that it's volume is way too low. It is set to 100% in both KDE and alsamixer (it's the Line one). ![image](https://github.com/user-attachments/assets/034e8f45-9e17-4338-9c1b-fda0b56dae29) How can I help solve this problem? Should I open a separate issue? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/455 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [RFC PATCH 00/10] Add support for DMA and audio codec of F1C100s
by Mesih Kilinc 23 Oct '24

23 Oct '24

This is RFC patchset for Allwinner suniv F1C100s to support DMA and audio codec. Allwinner F1C100s has a audio codec that has necessary digital and analog parts. It has r-l headphone output and microphone, line, r-l FM inputs. ADC can capture any inputs and also output channels via mux. Any input channels or DAC samples can feed output channels. Add support for this audio codec. F1C100s utilizes DMA channels to send and receive ADC-DAC samples. So DMA support needed. Patch 1~5 adds support for DMA. Suniv F1C100s has very similar DMA to sun4i. But there is some dissimilarities also. Suniv features a DMA reset bit in clock control unit. It has smaller number of DMA channels. Several registers has different addresses. It's max burst size is 4 instead of 8. Also DMA endpoint numbers are different. Patch 6 adds DMA max burst option to sun4i-codec. Patch 7~8 Add support for suniv F1C100s audio codec. Patch 9 adds audio codec to suniv-f1c100s.dtsi Patch 10 adds audio codec support to Lichee Pi Nano board. Thanks! Mesih Kilinc (10): dma-engine: sun4i: Add a quirk to support different chips dma-engine: sun4i: Add has_reset option to quirk dt-bindings: dmaengine: Add Allwinner suniv F1C100s DMA dma-engine: sun4i: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for DMA ASoC: sun4i-codec: Add DMA Max Burst field dt-bindigs: sound: Add Allwinner suniv F1C100s Audio Codec ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for Audio Codec ARM: dts: suniv: f1c100s: Activate Audio Codec for Lichee Pi Nano .../devicetree/bindings/dma/sun4i-dma.txt | 4 +- .../devicetree/bindings/sound/sun4i-codec.txt | 5 + arch/arm/boot/dts/suniv-f1c100s-licheepi-nano.dts | 8 + arch/arm/boot/dts/suniv-f1c100s.dtsi | 25 ++ drivers/dma/Kconfig | 4 +- drivers/dma/sun4i-dma.c | 221 ++++++++++-- sound/soc/sunxi/sun4i-codec.c | 371 ++++++++++++++++++++- 7 files changed, 601 insertions(+), 37 deletions(-) -- 2.7.4

5 18

[PATCH 0/2] ASoC: fsl_xcvr: enable some interrupts
by Shengjiu Wang 23 Oct '24

23 Oct '24

Enable interrupt of cmdc status update and the interrupts for wrong preamble received. Shengjiu Wang (2): ASoC: fsl_xcvr: enable interrupt of cmdc status update ASoC: fsl_xcvr: reset RX dpath after wrong preamble sound/soc/fsl/fsl_xcvr.c | 94 ++++++++++++++++++++++++++++++++++------ sound/soc/fsl/fsl_xcvr.h | 5 +++ 2 files changed, 86 insertions(+), 13 deletions(-) -- 2.34.1

2 3

ALSA problem , audio doesn't work
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA problem , audio on/off across multiple ubuntu releases
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

AlSA problem , audio on/off across multiple ubuntu releases
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

AlSA problem , audio on/off across multiple ubuntu releases
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

AlSA problem , audio on/off across multiple ubutnu
by GitHub issues - opened 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was opened from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: rt722-sdca: increase clk_stop_timeout to fix clock stop issue
by Jack Yu 22 Oct '24

22 Oct '24

clk_stop_timeout should be increased to 900ms to fix clock stop issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index 87354bb1564e..d5c985ff5ac5 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -253,7 +253,7 @@ static int rt722_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 200; + prop->clk_stop_timeout = 900; /* wake-up event */ prop->wake_capable = 1; -- 2.34.1

2 1

ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument.
by GitHub issues - opened 21 Oct '24

21 Oct '24

alsa-project/alsa-lib issue #413 was opened from linuxandos2: Playing a flac with a samplerate higer than 96000 with audacious gives: > _ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument._ Output device for audacious is hw:1.0 - USB Audio the device is a soudbaster blasterx G6. output of _cat /proc/asound/card1/stream0_ > Creative Technology Ltd Sound BlasterX G6 at usb-0000:00:14.0-4.1, high speed : > USB Audio > > Playback: > Status: Stop > Interface 1 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 1 > Implicit Feedback Mode: No > Interface 1 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 2 > Implicit Feedback Mode: No > > Capture: > Status: Stop > Interface 2 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Interface 2 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > linux: gentoo kernel: 6.9.9-gentoo-x86_64 alsa-lib-1.2.12 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.12 alsa-plugins-1.2.12 alsa-utils-1.2.12 selecting > pulse - PulseAudio Sound Server in audacious results in no sound and audacious hangs. Pipewire and wireplumber are used. On an old gentoo- system on another partition all samplerates play successfully. kernel: linux-6.8.1-gentoo the versions of alsa are: alsa-lib-1.2.10-r2 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.10-r1 alsa-plugins-1.2.7.1-r1 alsa-utils-1.2.10-r1 On the new system the 'old' kernel linux-6.8.1-gentoo is also available. Rolled back alsa-lib from alsa-lib-1.2.12 to alsa-lib-1.2.10 and booted kernel kernel linux-6.8.1-gentoo Still same problem. Issue URL : https://github.com/alsa-project/alsa-lib/issues/413 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: firewire-lib: Avoid division by zero in apply_constraint_to_size()
by Andrey Shumilin 21 Oct '24

21 Oct '24

The step variable is initialized to zero. It is changed in the loop, but if it's not changed it will remain zero. Add a variable check before the division. The observed behavior was introduced by commit 826b5de90c0b ("ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size"), and it is difficult to show that any of the interval parameters will satisfy the snd_interval_test() condition with data from the amdtp_rate_table[] table. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 826b5de90c0b ("ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size") Signed-off-by: Andrey Shumilin <shum.sdl(a)nppct.ru> --- sound/firewire/amdtp-stream.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4e2f2bb7879f..6c45ee3545f9 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -163,6 +163,9 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params, step = max(step, amdtp_syt_intervals[i]); } + if (step == 0) + return -EINVAL; + t.min = roundup(s->min, step); t.max = rounddown(s->max, step); t.integer = 1; -- 2.30.2

2 1

snd_pcm_hw_open open '/dev/snd/pcmC0D0p' failed (-16)
by GitHub issues - edited 21 Oct '24

21 Oct '24

alsa-project/alsa-lib issue #412 was edited from SVENsuzhou: Hi, I am using pjsip to make a sip call function. But I got the following error in alsa-lib. ``` Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 pjsua_call.c !Answering call 0: code=200 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 inv0x7f989c003fc8 ..SDP negotiation done: Success Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c ...Call 0: updating media.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c .....Media stream call00:0 is destroyed Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 ....UDP media transport started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_aud.c ....Audio channel update.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 .....UDP media transport attached Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Encoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Decoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_media.c ....Audio updated, stream #0: PCMA (sendrecv) Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ...Conf connect: 1 --> 0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ....Set sound device: capture=-1, playback=-2, mode=0, use_default_settings=0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@44100/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: start media transmit failed: pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@48000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@32000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@8000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c .....Unable to open sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c ....Error opening sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 media.cpp ...pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.321 pjsua_core.c ....TX 966 bytes Response msg 200/INVITE/cseq=30946 (tdta0x5559ce2e7378) to UDP 10.57.0.200:5060: Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: SIP/2.0 200 OK Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Via: SIP/2.0/UDP 10.57.0.200:5060;rport=5060;received=10.57.0.200;branch=z9hG4bKPje5a12777-db72-4db0-a5a5-a72743589a27 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Call-ID: 0bc91889-099c-42de-beb2-4fedefac6e7f Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: answer incoming call, remote uri: <sip:passenger_pei@10.57.0.200>, state: CONNECTING Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: From: <sip:passenger_pei@10.57.0.200>;tag=9efc50b5-c57a-4aa7-823e-2c3df064a27c ``` ``` root@tos-hmi-02:~# dmesg | grep snd [ 16.251073] snd_hda_intel 0000:00:0e.0: DSP detected with PCI class/subclass/prog-if info 0x040380 [ 16.261081] snd_hda_intel 0000:00:0e.0: bound 0000:00:02.0 (ops 0xffffffff8df312a0) [ 16.351558] snd_hda_codec_idt hdaudioC0D0: autoconfig for 92HD89D3: line_outs=3 (0xd/0xf/0x10/0x0/0x0) type:line [ 16.362460] snd_hda_codec_idt hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 16.362470] snd_hda_codec_idt hdaudioC0D0: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) [ 16.362476] snd_hda_codec_idt hdaudioC0D0: mono: mono_out=0x0 [ 16.383430] snd_hda_codec_idt hdaudioC0D0: dig-out=0x22/0x23 [ 16.383447] snd_hda_codec_idt hdaudioC0D0: inputs: [ 16.383452] snd_hda_codec_idt hdaudioC0D0: Internal Mic=0x14 [ 16.383455] snd_hda_codec_idt hdaudioC0D0: Rear Mic=0xe [ 16.383459] snd_hda_codec_idt hdaudioC0D0: Front Mic=0xb [ 16.383461] snd_hda_codec_idt hdaudioC0D0: Internal Mic=0x13 [ 16.383464] snd_hda_codec_idt hdaudioC0D0: Line=0xc [ 16.383466] snd_hda_codec_idt hdaudioC0D0: CD=0x12 [ 16.383469] snd_hda_codec_idt hdaudioC0D0: dig-in=0x24 ``` I try to use `lsof /dev/snd/*` to check which process hold the file handle. And I got nothing. Can you give me some leads on troubleshooting? Issue URL : https://github.com/alsa-project/alsa-lib/issues/412 Repository URL: https://github.com/alsa-project/alsa-lib

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snd_pcm_hw_open open '/dev/snd/pcmC0D0p' failed (-16)
by GitHub issues - opened 21 Oct '24

21 Oct '24

alsa-project/alsa-lib issue #412 was opened from SVENsuzhou: Hi, I am using pjsip to make a sip call function. But I got the following error in alsa-lib. ``` Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 pjsua_call.c !Answering call 0: code=200 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 inv0x7f989c003fc8 ..SDP negotiation done: Success Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c ...Call 0: updating media.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c .....Media stream call00:0 is destroyed Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 ....UDP media transport started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_aud.c ....Audio channel update.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 .....UDP media transport attached Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Encoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Decoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_media.c ....Audio updated, stream #0: PCMA (sendrecv) Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ...Conf connect: 1 --> 0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ....Set sound device: capture=-1, playback=-2, mode=0, use_default_settings=0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@44100/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: start media transmit failed: pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@48000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@32000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@8000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c .....Unable to open sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c ....Error opening sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 media.cpp ...pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.321 pjsua_core.c ....TX 966 bytes Response msg 200/INVITE/cseq=30946 (tdta0x5559ce2e7378) to UDP 10.57.0.200:5060: Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: SIP/2.0 200 OK Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Via: SIP/2.0/UDP 10.57.0.200:5060;rport=5060;received=10.57.0.200;branch=z9hG4bKPje5a12777-db72-4db0-a5a5-a72743589a27 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Call-ID: 0bc91889-099c-42de-beb2-4fedefac6e7f Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: answer incoming call, remote uri: <sip:passenger_pei@10.57.0.200>, state: CONNECTING Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: From: <sip:passenger_pei@10.57.0.200>;tag=9efc50b5-c57a-4aa7-823e-2c3df064a27c ``` I try to use `lsof /dev/snd/*` to check which process hold the file handle. And I got nothing. Can you give me some leads on troubleshooting? Issue URL : https://github.com/alsa-project/alsa-lib/issues/412 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v6] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 20 Oct '24

20 Oct '24

Due to internal re-org, Kevin has no longer maintained audio driver. Due to 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> --- v6: - fixed spelling mistakes v5: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202410021557.FByBO9Dp-lkp@intel.com/ v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 87198c5e589e..d1484b5658e3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22910,25 +22910,42 @@ 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*.* @@ -22938,9 +22955,8 @@ 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

[PATCH v2] ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs
by Alexey Klimov 19 Oct '24

19 Oct '24

Turns out some registers of pre-2.5 version of rxmacro codecs are not located at the expected offsets but 0xc further away in memory. So far the detected registers are CDC_RX_RX2_RX_PATH_SEC7 and CDC_RX_RX2_RX_PATH_DSM_CTL. CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macro incorrectly generates the address 0x540 for RX2 but it should be 0x54C and it also overwrites CDC_RX_RX2_RX_PATH_SEC7 which is located at 0x540. The same goes for CDC_RX_RXn_RX_PATH_SEC7(rx, n). Fix this by introducing additional rxn_reg_stride2 offset. For 2.5 version and above this offset will be equal to 0. With such change the corresponding RXn() macros will generate the same values for 2.5 codec version for all RX paths and the same old values for pre-2.5 version for RX0 and RX1. However for the latter case with RX2 path it will also add rxn_reg_stride2 on top. While at this, also remove specific if-check for INTERP_AUX from rx_macro_digital_mute() and rx_macro_enable_interp_clk(). These if-check was used to handle such special offset for AUX interpolator but since CDC_RX_RXn_RX_PATH_SEC7(rx, n) and CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macros will generate the correst addresses of dsm register, they are no longer needed. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Signed-off-by: Alexey Klimov <alexey.klimov(a)linaro.org> --- Changes in v2: - updated macros as suggested by Mark and Dmitry; - removed INTERP_AUX if-check in two functions; Link to previous patch: https://lore.kernel.org/lkml/20240925043823.520218-1-alexey.klimov@linaro.o… sound/soc/codecs/lpass-rx-macro.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ef7a70fa6966..febbbe073962 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -202,12 +202,14 @@ #define CDC_RX_RXn_RX_PATH_SEC3(rx, n) (0x042c + rx->rxn_reg_stride * n) #define CDC_RX_RX0_RX_PATH_SEC4 (0x0430) #define CDC_RX_RX0_RX_PATH_SEC7 (0x0434) -#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) (0x0434 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) \ + (0x0434 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_DSM_OUT_DELAY_SEL_MASK GENMASK(2, 0) #define CDC_RX_DSM_OUT_DELAY_TWO_SAMPLE 0x2 #define CDC_RX_RX0_RX_PATH_MIX_SEC0 (0x0438) #define CDC_RX_RX0_RX_PATH_MIX_SEC1 (0x043C) -#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) (0x0440 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) \ + (0x0440 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_RXn_DSM_CLK_EN_MASK BIT(0) #define CDC_RX_RX0_RX_PATH_DSM_CTL (0x0440) #define CDC_RX_RX0_RX_PATH_DSM_DATA1 (0x0444) @@ -645,6 +647,7 @@ struct rx_macro { int rx_mclk_cnt; enum lpass_codec_version codec_version; int rxn_reg_stride; + int rxn_reg_stride2; bool is_ear_mode_on; bool hph_pwr_mode; bool hph_hd2_mode; @@ -1929,9 +1932,6 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) CDC_RX_PATH_PGA_MUTE_MASK, 0x0); } - if (j == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - int_mux_cfg0 = CDC_RX_INP_MUX_RX_INT0_CFG0 + j * 8; int_mux_cfg1 = int_mux_cfg0 + 4; int_mux_cfg0_val = snd_soc_component_read(component, int_mux_cfg0); @@ -2702,9 +2702,6 @@ static int rx_macro_enable_interp_clk(struct snd_soc_component *component, main_reg = CDC_RX_RXn_RX_PATH_CTL(rx, interp_idx); dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, interp_idx); - if (interp_idx == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - rx_cfg2_reg = CDC_RX_RXn_RX_PATH_CFG2(rx, interp_idx); if (SND_SOC_DAPM_EVENT_ON(event)) { @@ -3821,6 +3818,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_0: case LPASS_CODEC_VERSION_2_1: rx->rxn_reg_stride = 0x80; + rx->rxn_reg_stride2 = 0xc; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_pre_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) @@ -3834,6 +3832,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_7: case LPASS_CODEC_VERSION_2_8: rx->rxn_reg_stride = 0xc0; + rx->rxn_reg_stride2 = 0x0; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) -- 2.45.2

3 2

[PATCH] ASoC: fsl_micfil: Add a flag to distinguish with different volume control types
by Chancel Liu 19 Oct '24

19 Oct '24

On i.MX8MM the register of volume control has positive and negative values. It is different from other platforms like i.MX8MP and i.MX93 which only have positive values. Add a volume_sx flag to use SX_TLV volume control for this kind of platform. Use common TLV volume control for other platforms. Fixes: cdfa92eb90f5 ("ASoC: fsl_micfil: Correct the number of steps on SX controls") Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 43 +++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 4161b674745b..0679d762f08a 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -78,6 +78,7 @@ struct fsl_micfil_soc_data { bool imx; bool use_edma; bool use_verid; + bool volume_sx; u64 formats; }; @@ -87,6 +88,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mm = { .fifo_depth = 8, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .volume_sx = true, }; static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { @@ -95,6 +97,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { .fifo_depth = 32, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S32_LE, + .volume_sx = false, }; static struct fsl_micfil_soc_data fsl_micfil_imx93 = { @@ -105,6 +108,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx93 = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .use_edma = true, .use_verid = true, + .volume_sx = false, }; static const struct of_device_id fsl_micfil_dt_ids[] = { @@ -328,7 +332,26 @@ static int hwvad_detected(struct snd_kcontrol *kcontrol, return 0; } -static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { +static const struct snd_kcontrol_new fsl_micfil_volume_controls[] = { + SOC_SINGLE_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_volume_sx_controls[] = { SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, @@ -345,6 +368,9 @@ static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { SOC_ENUM_EXT("MICFIL Quality Select", fsl_micfil_quality_enum, micfil_quality_get, micfil_quality_set), @@ -844,6 +870,20 @@ static int fsl_micfil_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } +static int fsl_micfil_component_probe(struct snd_soc_component *component) +{ + struct fsl_micfil *micfil = snd_soc_component_get_drvdata(component); + + if (micfil->soc->volume_sx) + snd_soc_add_component_controls(component, fsl_micfil_volume_sx_controls, + ARRAY_SIZE(fsl_micfil_volume_sx_controls)); + else + snd_soc_add_component_controls(component, fsl_micfil_volume_controls, + ARRAY_SIZE(fsl_micfil_volume_controls)); + + return 0; +} + static const struct snd_soc_dai_ops fsl_micfil_dai_ops = { .probe = fsl_micfil_dai_probe, .startup = fsl_micfil_startup, @@ -865,6 +905,7 @@ static struct snd_soc_dai_driver fsl_micfil_dai = { static const struct snd_soc_component_driver fsl_micfil_component = { .name = "fsl-micfil-dai", + .probe = fsl_micfil_component_probe, .controls = fsl_micfil_snd_controls, .num_controls = ARRAY_SIZE(fsl_micfil_snd_controls), .legacy_dai_naming = 1, -- 2.46.2

3 2

[PATCH] Use card->dev in replace of the &pdev->dev argument in the dev_err function
by Liu Jing 18 Oct '24

18 Oct '24

Because card->dev = &pdev->dev is already defined in the rx51_soc_probe function, and then &pdev->dev is still used. Signed-off-by: Liu Jing <liujing(a)cmss.chinamobile.com> --- sound/soc/ti/rx51.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c index 77296237575a..d9900c69e536 100644 --- a/sound/soc/ti/rx51.c +++ b/sound/soc/ti/rx51.c @@ -371,7 +371,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,cpu-dai", 0); if (!dai_node) { - dev_err(&pdev->dev, "McBSP node is not provided\n"); + dev_err(card->dev, "McBSP node is not provided\n"); return -EINVAL; } rx51_dai[0].cpus->dai_name = NULL; @@ -381,7 +381,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,audio-codec", 0); if (!dai_node) { - dev_err(&pdev->dev, "Codec node is not provided\n"); + dev_err(card->dev, "Codec node is not provided\n"); return -EINVAL; } rx51_dai[0].codecs->name = NULL; @@ -389,7 +389,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,audio-codec", 1); if (!dai_node) { - dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n"); + dev_err(card->dev, "Auxiliary Codec node is not provided\n"); return -EINVAL; } rx51_aux_dev[0].dlc.name = NULL; @@ -399,7 +399,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0); if (!dai_node) { - dev_err(&pdev->dev, "Headphone amplifier node is not provided\n"); + dev_err(card->dev, "Headphone amplifier node is not provided\n"); return -EINVAL; } rx51_aux_dev[1].dlc.name = NULL; @@ -408,7 +408,7 @@ static int rx51_soc_probe(struct platform_device *pdev) rx51_codec_conf[1].dlc.of_node = dai_node; } - pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + pdata = devm_kzalloc(card->dev, sizeof(*pdata), GFP_KERNEL); if (pdata == NULL) return -ENOMEM; @@ -439,7 +439,7 @@ static int rx51_soc_probe(struct platform_device *pdev) err = devm_snd_soc_register_card(card->dev, card); if (err) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err); + dev_err(card->dev, "snd_soc_register_card failed (%d)\n", err); return err; } -- 2.27.0

2 2

[PATCH 5.10 0/1] soundwire: cadence: Fix error check in cdns_xfer_msg()
by Daniil Dulov 18 Oct '24

18 Oct '24

Svacer reports redundant comparison in cdns_xfer_msg(). The problem is present in 5.10 stable release and can be fixed by the following upstream patch that can be cleanly applied to 5.10 stable branch.

1 1

[PATCH alsa-ucm-conf v5 1/2] sof-soundwire: Change map control names to make them unique and user friendly
by Maciej Strozek 18 Oct '24

18 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v5: Change PlaybackMixerElem to just "Speaker" v4: Add "CaptureMixer" to headset config v3: Make Headset Microphone more user-friendly and contain the word "Capture" v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 10 ++++++++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- ucm2/sof-soundwire/cs42l43.conf | 3 ++- 4 files changed, 14 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..b2796e2 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,13 +2,19 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } + "name='cs42l43 Headset Microphone Capture Switch'" { + "name='cs42l43 Decimator 1 Switch'".vindex.0 0 + } + "name='cs42l43 Headset Microphone Capture Volume'" { + "name='cs42l43 Decimator 1 Volume'".vindex.0 0 + } } } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..176f880 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 1702cf0..ddca67a 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -40,7 +40,8 @@ SectionDevice."Headset" { Value { CapturePriority 200 CapturePCM "hw:${CardId},4" - CaptureMixerElem "cs42l43 Decimator 1" + CaptureMixer "default:${CardId}" + CaptureMixerElem "cs42l43 Headset Microphone" JackControl "Headset Mic Jack" } } -- 2.39.5

2 3

Qualcomm: fix codec initialisation
by GitHub pull_request - opened 18 Oct '24

18 Oct '24

alsa-project/alsa-ucm-conf pull request #454 was opened from jhovold: Fix the Qualcomm configurations that are using the same identifier for lazy includes of codec initialisation files which means that only the last one is actually included. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/454 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/454.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Revert "ucm2: Qualcomm: x1e80100: add USB DisplayPort playback"
by GitHub pull_request - opened 18 Oct '24

18 Oct '24

alsa-project/alsa-ucm-conf pull request #453 was opened from jhovold: This reverts commit 1dcb58e4fac989bafe0705e49a7f029727207fb3. The recently added DisplayPort support depends on kernel changes that never made it upstream and has some usability issues that remain to be resolved. Revert for now to avoid breaking existing setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/453 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/453.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument.
by GitHub issues - opened 17 Oct '24

17 Oct '24

alsa-project/alsa-plugins issue #62 was opened from linuxandos2: Playing a flac with a samplerate higer than 96000 with audacious gives: > _ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument._ Output device for audacious is hw:1.0 - USB Audio the device is a soudbaster blasterx G6. output of _cat /proc/asound/card1/stream0_ > Creative Technology Ltd Sound BlasterX G6 at usb-0000:00:14.0-4.1, high speed : > USB Audio > > Playback: > Status: Stop > Interface 1 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 1 > Implicit Feedback Mode: No > Interface 1 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 2 > Implicit Feedback Mode: No > > Capture: > Status: Stop > Interface 2 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Interface 2 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > linux: gentoo kernel: 6.9.9-gentoo-x86_64 alsa-lib-1.2.12 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.12 alsa-plugins-1.2.12 alsa-utils-1.2.12 selecting > pulse - PulseAudio Sound Server in audacious results in no sound and audacious hangs. Pipewire and wireplumber are used. On an old gentoo- system on another partition all samplerates play successfully. Using the same kernel: linux-6.8.1-gentoo as in the old system on the new one does not solve the problem. the versions of alsa are: alsa-lib-1.2.10-r2 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.10-r1 alsa-plugins-1.2.7.1-r1 alsa-utils-1.2.10-r1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/62 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

acppdmmach: add support for ACP 7.0
by GitHub pull_request - opened 17 Oct '24

17 Oct '24

alsa-project/alsa-ucm-conf pull request #452 was opened from superm1: Starting with ACP 7.0 there is a generic 'acppdmmach' driver that will be utilized. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/452 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/452.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH alsa-ucm-conf v4 1/2] sof-soundwire: Change map control names to make them unique and user friendly
by Maciej Strozek 17 Oct '24

17 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v4: Add "CaptureMixer" to headset config v3: Make Headset Microphone more user-friendly and contain the word "Capture" v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 10 ++++++++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- ucm2/sof-soundwire/cs42l43.conf | 3 ++- 4 files changed, 14 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..b2796e2 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,13 +2,19 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } + "name='cs42l43 Headset Microphone Capture Switch'" { + "name='cs42l43 Decimator 1 Switch'".vindex.0 0 + } + "name='cs42l43 Headset Microphone Capture Volume'" { + "name='cs42l43 Decimator 1 Volume'".vindex.0 0 + } } } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..a3cf6ad 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker Playback" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 1702cf0..ddca67a 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -40,7 +40,8 @@ SectionDevice."Headset" { Value { CapturePriority 200 CapturePCM "hw:${CardId},4" - CaptureMixerElem "cs42l43 Decimator 1" + CaptureMixer "default:${CardId}" + CaptureMixerElem "cs42l43 Headset Microphone" JackControl "Headset Mic Jack" } } -- 2.39.5

2 2

MUSL static build - conf.c still insists on using dynamic link libraries
by GitHub issues - opened 16 Oct '24

16 Oct '24

alsa-project/alsa-lib issue #411 was opened from pietervandermeer: alsa-lib v 1.2.12 configured for musl static build like so: ```CC=arm-linux-musleabihf-gcc ./configure --enable-shared=no --enable-static=yes --with-pic --host=armv7-unknown-linux-musleabihf --prefix=/home/vagrant/alsa-lib-1.2.12/build --libdir=/usr/lib/arm-linux-musleabihf --includedir=/usr/include/arm-linux-musleabihf/``` This builds and links fine against my executable. But when I run it, there's an issue: ``` INFO opening alsa audio playback device [hw:CARD=sndrpihifiberry,DEV=0] ALSA lib conf.c:3558:(snd_config_hooks_call) Cannot open shared library (null) ((null): Dynamic loading not supported) ALSA lib conf.c:4026:(snd_config_update_r) hooks failed, removing configuration WARN Not found playback device, error ALSA function 'snd_pcm_open' failed with error 'No such file or directory (2)'. Trying next.. ``` Note the "cannot open shared library". Well, yes, because you should be configured as 100% static, as per configure arguments ```--enable-shared=no --enable-static=yes``` Upon inspecting the code in ```conf.c```, there are various functions like ```snd_config_hooks_call()``` that just call ```snd_dlopen()``` and ```snd_dlsym()```. Right now I'm just replacing these with ```strcmp``` on the function_name followed by a static function call to the respective function. But I don't know how much work I'll need, and of course it is really hacky. It would be great if a 100% static build was supported. Issue URL : https://github.com/alsa-project/alsa-lib/issues/411 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Working on audio for qcom apq8098 / msm8998
by Marc Gonzalez 15 Oct '24

15 Oct '24

Hello everyone, I've been trying to get audio working on a qcom apq8098 (cousin of msm8998) board. Code is mostly based on work from here: https://gitlab.com/msm8998-mainline/linux This is the diff I have on top of v6.12-rc1 https://paste.debian.net/1331666/ Kernel log after init starts: [ 13.311201] random: crng init done [ 14.724797] atl1c 0000:01:00.0 enp1s0: renamed from eth0 (while UP) [ 14.895978] Detected VIPT I-cache on CPU1 [ 14.896041] GICv3: CPU1: found redistributor 1 region 0:0x0000000017b20000 [ 14.896113] CPU1: Booted secondary processor 0x0000000001 [0x51af8014] [ 14.949128] Detected VIPT I-cache on CPU2 [ 14.949198] GICv3: CPU2: found redistributor 2 region 0:0x0000000017b40000 [ 14.949272] CPU2: Booted secondary processor 0x0000000002 [0x51af8014] [ 15.025271] Detected VIPT I-cache on CPU3 [ 15.025349] GICv3: CPU3: found redistributor 3 region 0:0x0000000017b60000 [ 15.025423] CPU3: Booted secondary processor 0x0000000003 [0x51af8014] [ 15.106412] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:3 [ 15.109935] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:4 [ 15.119041] qcom-q6afe aprsvc:apr-service:4:4: YO: q6afe_probe:1759 [ 15.134672] q6afe-dai 17300000.remoteproc:glink-edge:apr:apr-service@4:dais: YOA: q6afe_dai_dev_probe [ 15.137948] q6afe-dai 17300000.remoteproc:glink-edge:apr:apr-service@4:dais: YOA: of_q6afe_parse_dai_data [ 15.146461] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:7 [ 15.153563] remoteproc remoteproc0: powering up 4080000.remoteproc [ 15.166371] remoteproc remoteproc0: Booting fw image mba.mbn, size 234152 [ 15.169747] q6asm-dai 17300000.remoteproc:glink-edge:apr:apr-service@7:dais: Adding to iommu group 5 [ 15.170559] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:8 [ 15.179732] q6afe-dai 17300000.remoteproc:glink-edge:apr:apr-service@4:dais: YOA: msm_dai_q6_dai_probe [ 15.199345] qcom-q6afe aprsvc:apr-service:4:4: YO: q6afe_find_port:860 [ 15.208834] msm-snd-msm8998 sound: ASoC: registered pcm #0 MultiMedia1 (*) [ 15.216194] msm-snd-msm8998 sound: snd-soc-dummy-dai <-> MultiMedia1 mapping ok [ 15.221994] msm-snd-msm8998 sound: ASoC: registered pcm #1 MultiMedia2 (*) [ 15.229582] msm-snd-msm8998 sound: snd-soc-dummy-dai <-> MultiMedia2 mapping ok [ 15.236045] msm-snd-msm8998 sound: ASoC: registered pcm #2 (HDMI) [ 15.243211] msm-snd-msm8998 sound: i2s-hifi <-> HDMI mapping ok [ 15.285770] input: batfish Headset Jack as /devices/platform/sound/sound/card0/input2 [ 15.298010] qcom-q6v5-mss 4080000.remoteproc: MBA booted without debug policy, loading mpss [ 15.542937] remoteproc remoteproc0: remote processor 4080000.remoteproc is now up [ 17.727724] ath10k_snoc 18800000.wifi: qmi chip_id 0x30214 chip_family 0x4001 board_id 0xff soc_id 0x40010002 [ 17.727923] ath10k_snoc 18800000.wifi: qmi fw_version 0x100204b2 fw_build_timestamp 2019-09-04 03:01 fw_build_id QC_IMAGE_VERSION_STRING=WLAN.HL.1.0-01202-QCAHLSWMTPLZ-1.221523.2 [ 17.736928] ath10k_snoc 18800000.wifi: qmi not waiting for msa_ready indicator [ 17.903361] platform a800000.usb: deferred probe pending: dwc3: failed to initialize core [ 21.512409] arm-smmu 16c0000.iommu: Unhandled context fault: fsr=0x402, iova=0x00000000, fsynr=0x1, cbfrsynra=0x1900, cb=0 [ 21.512619] arm-smmu 16c0000.iommu: FSR = 00000402 [Format=2 TF], SID=0x1900 [ 21.522450] arm-smmu 16c0000.iommu: FSYNR0 = 00000001 [S1CBNDX=0 PLVL=1] [ 21.599761] ath10k_snoc 18800000.wifi: wcn3990 hw1.0 target 0x00000008 chip_id 0x00000000 sub 0000:0000 [ 21.599935] ath10k_snoc 18800000.wifi: kconfig debug 1 debugfs 1 tracing 0 dfs 0 testmode 0 [ 21.608221] ath10k_snoc 18800000.wifi: firmware ver api 5 features wowlan,mgmt-tx-by-reference,non-bmi crc32 3ec1ac4b [ 21.657531] ath10k_snoc 18800000.wifi: htt-ver 3.44 wmi-op 4 htt-op 3 cal file max-sta 32 raw 0 hwcrypto 1 [ 21.713363] arm-smmu 16c0000.iommu: Unhandled context fault: fsr=0x402, iova=0x00000000, fsynr=0x1, cbfrsynra=0x1900, cb=0 [ 21.713577] arm-smmu 16c0000.iommu: FSR = 00000402 [Format=2 TF], SID=0x1900 [ 21.723404] arm-smmu 16c0000.iommu: FSYNR0 = 00000001 [S1CBNDX=0 PLVL=1] [ 21.732374] ath10k_snoc 18800000.wifi: invalid MAC address; choosing random [ 21.737662] ath: EEPROM regdomain: 0x0 [ 21.744356] ath: EEPROM indicates default country code should be used [ 21.748148] ath: doing EEPROM country->regdmn map search [ 21.754648] ath: country maps to regdmn code: 0x3a [ 21.759986] ath: Country alpha2 being used: US [ 21.764605] ath: Regpair used: 0x3a Does anyone see something obvious I might have missed? Regards

3 6

ucm-validator verify fails when including codecs/hda/hdmi.conf
by GitHub issues - opened 15 Oct '24

15 Oct '24

alsa-project/alsa-ucm-conf issue #451 was opened from ChrisLane: After increasing the verbosity of the validator in the GitHub workflow, we can see that the "Validate Syntax" step is failing on the `codecs/hda/hdmi.conf` file. Here is the relevant output: https://github.com/alsa-project/alsa-ucm-conf/actions/runs/11346809053/job/… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/451 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v28 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 15 Oct '24

15 Oct '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v28: - Fixed some phrases/wording within the SOC USB documentation, and also added an output with aplay -l for the example output. - Fixed allocated string buffer for creating the USB SND offload mixer, and added a PCM index check to ensure that the pcm index is less than the expected number. - Added a complement enable jack call if USB backend DAI link drivers need access to it. Changes in v27: - Added some comments and notes about the offload design. Enforcing the q6routing to only allow one USB mixer (PCM device) to be enabled at a time. - Modified SND_JACK_USB notifications for all USB audio offloadable devices plugged in - Rebased on latest XHCI secondary interrupter IMOD changes upstream. Modified the change in this series to allow for XHCI sideband to set the IMOD for sideband clients. - Updated documentation on how USB SND kcontrols are involved in the overall design. - Remove mutex locking from suspend/resume platform ops, as USB core ensures that the interface and device are in the RPM_ACTIVE state while disconnect is handled. Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (3): xhci: support setting interrupt moderation IMOD for secondary interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (30): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: xhci-sideband: Set IMOD for xHCI sideband clients usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 491 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 37 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 51 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 70 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 147 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 32 +- sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 369 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 106 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1974 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 +++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5957 insertions(+), 57 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 68

ucm2: wsa884x: fix typo in mixer names
by GitHub pull_request - opened 14 Oct '24

14 Oct '24

alsa-project/alsa-ucm-conf pull request #450 was opened from joske: DTS fix for x1e80100-CRD will be submitted upstream separately Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/450 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/450.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: rt-sdw-common: Enhance switch case to prevent uninitialized variable
by Jack Yu 14 Oct '24

14 Oct '24

If det_mode is not 0, 3 or 5 then function will return jack_type with an uninitialzed value. Enhance switch case to prevent uninitialized variable issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt-sdw-common.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt-sdw-common.c b/sound/soc/codecs/rt-sdw-common.c index a422da6cf702..9f51fec383f9 100644 --- a/sound/soc/codecs/rt-sdw-common.c +++ b/sound/soc/codecs/rt-sdw-common.c @@ -150,15 +150,15 @@ int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id) goto io_error; switch (det_mode) { - case 0x00: - jack_type = 0; - break; case 0x03: jack_type = SND_JACK_HEADPHONE; break; case 0x05: jack_type = SND_JACK_HEADSET; break; + default: + jack_type = 0; + break; } /* write selected_mode */ -- 2.34.1

2 1

Debian 12 : no audio
by GitHub issues - opened 14 Oct '24

14 Oct '24

alsa-project/alsa-ucm-conf issue #449 was opened from ezawadzki: After fresh debian 12 installation : - Speakers audio : ok - Headphones : no audio After installed `alsa-ucm-conf` and followed : ``` systemctl stop alsa-state rm /var/lib/alsa/asound.state systemctl start alsa-state ``` Now : - Speakers audio : no audio (speakers not recognized) - Headphones : no audio ``` cat /etc/os-release PRETTY_NAME="Debian GNU/Linux 12 (bookworm)" ``` ``` uname -a Linux debian 6.10.11+bpo-amd64 #1 SMP PREEMPT_DYNAMIC Debian 6.10.11-1~bpo12+1 (2024-10-03) x86_64 GNU/Linux ``` ``` alsamixer Card and Chipset : PipeWire ``` Any help ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/449 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[tiwai-sound:for-next 10/14] sound/pci/hda/patch_realtek.c:10711:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk'
by kernel test robot 14 Oct '24

14 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 commit: 504f052aa3435ab2f15af8b20bc4f4de8ff259c7 [10/14] ALSA: hda/realtek: Use codec SSID matching for Lenovo devices config: i386-buildonly-randconfig-003-20241014 (https://download.01.org/0day-ci/archive/20241014/202410141704.uIhTGCED-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/20241014/202410141704.uIhTGCED-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/202410141704.uIhTGCED-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): >> sound/pci/hda/patch_realtek.c:10711:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 10711 | HDA_CODEC_QUIRK(0x17aa, 0x3802, "DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ sound/pci/hda/patch_realtek.c:10716:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 10716 | HDA_CODEC_QUIRK(0x17aa, 0x3820, "IdeaPad 330-17IKB 81DM", ALC269_FIXUP_ASPIRE_HEADSET_MIC), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ sound/pci/hda/patch_realtek.c:10731:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 10731 | HDA_CODEC_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ sound/pci/hda/patch_realtek.c:10733:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 10733 | HDA_CODEC_QUIRK(0x17aa, 0x386f, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ 4 errors generated. vim +10711 sound/pci/hda/patch_realtek.c 9995 9996 static const struct hda_quirk alc269_fixup_tbl[] = { 9997 SND_PCI_QUIRK(0x1025, 0x0283, "Acer TravelMate 8371", ALC269_FIXUP_INV_DMIC), 9998 SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), 9999 SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), 10000 SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), 10001 SND_PCI_QUIRK(0x1025, 0x072d, "Acer Aspire V5-571G", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10002 SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), 10003 SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), 10004 SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), 10005 SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), 10006 SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), 10007 SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10008 SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF), 10009 SND_PCI_QUIRK(0x1025, 0x100c, "Acer Aspire E5-574G", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST), 10010 SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK), 10011 SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), 10012 SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), 10013 SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), 10014 SND_PCI_QUIRK(0x1025, 0x1094, "Acer Aspire E5-575T", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST), 10015 SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10016 SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10017 SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK), 10018 SND_PCI_QUIRK(0x1025, 0x1167, "Acer Veriton N6640G", ALC269_FIXUP_LIFEBOOK), 10019 SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), 10020 SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), 10021 SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE), 10022 SND_PCI_QUIRK(0x1025, 0x1269, "Acer SWIFT SF314-54", ALC256_FIXUP_ACER_HEADSET_MIC), 10023 SND_PCI_QUIRK(0x1025, 0x126a, "Acer Swift SF114-32", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), 10024 SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 10025 SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 10026 SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 10027 SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), 10028 SND_PCI_QUIRK(0x1025, 0x129d, "Acer SWIFT SF313-51", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), 10029 SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), 10030 SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 10031 SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), 10032 SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), 10033 SND_PCI_QUIRK(0x1025, 0x141f, "Acer Spin SP513-54N", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10034 SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10035 SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), 10036 SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), 10037 SND_PCI_QUIRK(0x1025, 0x1534, "Acer Predator PH315-54", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10038 SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), 10039 SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), 10040 SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X), 10041 SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), 10042 SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), 10043 SND_PCI_QUIRK(0x1028, 0x05be, "Dell Latitude E6540", ALC292_FIXUP_DELL_E7X), 10044 SND_PCI_QUIRK(0x1028, 0x05ca, "Dell Latitude E7240", ALC292_FIXUP_DELL_E7X), 10045 SND_PCI_QUIRK(0x1028, 0x05cb, "Dell Latitude E7440", ALC292_FIXUP_DELL_E7X), 10046 SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER), 10047 SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), 10048 SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), 10049 SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), 10050 SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), 10051 SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), 10052 SND_PCI_QUIRK(0x1028, 0x062c, "Dell Latitude E5550", ALC292_FIXUP_DELL_E7X), 10053 SND_PCI_QUIRK(0x1028, 0x062e, "Dell Latitude E7450", ALC292_FIXUP_DELL_E7X), 10054 SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), 10055 SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10056 SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10057 SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), 10058 SND_PCI_QUIRK(0x1028, 0x0669, "Dell Optiplex 9020m", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), 10059 SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", ALC290_FIXUP_SUBWOOFER_HSJACK), 10060 SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), 10061 SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10062 SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10063 SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10064 SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10065 SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10066 SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10067 SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10068 SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), 10069 SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), 10070 SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), 10071 SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), 10072 SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), 10073 SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), 10074 SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), 10075 SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), 10076 SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), 10077 SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), 10078 SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), 10079 SND_PCI_QUIRK(0x1028, 0x0872, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), 10080 SND_PCI_QUIRK(0x1028, 0x0873, "Dell Precision 3930", ALC255_FIXUP_DUMMY_LINEOUT_VERB), 10081 SND_PCI_QUIRK(0x1028, 0x08ad, "Dell WYSE AIO", ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE), 10082 SND_PCI_QUIRK(0x1028, 0x08ae, "Dell WYSE NB", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE), 10083 SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), 10084 SND_PCI_QUIRK(0x1028, 0x097d, "Dell Precision", ALC289_FIXUP_DUAL_SPK), 10085 SND_PCI_QUIRK(0x1028, 0x097e, "Dell Precision", ALC289_FIXUP_DUAL_SPK), 10086 SND_PCI_QUIRK(0x1028, 0x098d, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), 10087 SND_PCI_QUIRK(0x1028, 0x09bf, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), 10088 SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), 10089 SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), 10090 SND_PCI_QUIRK(0x1028, 0x0a38, "Dell Latitude 7520", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET), 10091 SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), 10092 SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), 10093 SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), 10094 SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), 10095 SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), 10096 SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), 10097 SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), 10098 SND_PCI_QUIRK(0x1028, 0x0b27, "Dell", ALC245_FIXUP_CS35L41_SPI_2), 10099 SND_PCI_QUIRK(0x1028, 0x0b28, "Dell", ALC245_FIXUP_CS35L41_SPI_2), 10100 SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), 10101 SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), 10102 SND_PCI_QUIRK(0x1028, 0x0beb, "Dell XPS 15 9530 (2023)", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10103 SND_PCI_QUIRK(0x1028, 0x0c03, "Dell Precision 5340", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), 10104 SND_PCI_QUIRK(0x1028, 0x0c0b, "Dell Oasis 14 RPL-P", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10105 SND_PCI_QUIRK(0x1028, 0x0c0d, "Dell Oasis", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10106 SND_PCI_QUIRK(0x1028, 0x0c0e, "Dell Oasis 16", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10107 SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), 10108 SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), 10109 SND_PCI_QUIRK(0x1028, 0x0c1b, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), 10110 SND_PCI_QUIRK(0x1028, 0x0c1c, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), 10111 SND_PCI_QUIRK(0x1028, 0x0c1d, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), 10112 SND_PCI_QUIRK(0x1028, 0x0c1e, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), 10113 SND_PCI_QUIRK(0x1028, 0x0c28, "Dell Inspiron 16 Plus 7630", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), 10114 SND_PCI_QUIRK(0x1028, 0x0c4d, "Dell", ALC287_FIXUP_CS35L41_I2C_4), 10115 SND_PCI_QUIRK(0x1028, 0x0cbd, "Dell Oasis 13 CS MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10116 SND_PCI_QUIRK(0x1028, 0x0cbe, "Dell Oasis 13 2-IN-1 MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10117 SND_PCI_QUIRK(0x1028, 0x0cbf, "Dell Oasis 13 Low Weight MTU-L", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10118 SND_PCI_QUIRK(0x1028, 0x0cc0, "Dell Oasis 13", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10119 SND_PCI_QUIRK(0x1028, 0x0cc1, "Dell Oasis 14 MTL-H/U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10120 SND_PCI_QUIRK(0x1028, 0x0cc2, "Dell Oasis 14 2-in-1 MTL-H/U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10121 SND_PCI_QUIRK(0x1028, 0x0cc3, "Dell Oasis 14 Low Weight MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10122 SND_PCI_QUIRK(0x1028, 0x0cc4, "Dell Oasis 16 MTL-H/U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10123 SND_PCI_QUIRK(0x1028, 0x0cc5, "Dell Oasis 14", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10124 SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10125 SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10126 SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), 10127 SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), 10128 SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), 10129 SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10130 SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10131 SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10132 SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10133 SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), 10134 SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10135 SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10136 SND_PCI_QUIRK(0x103c, 0x2236, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10137 SND_PCI_QUIRK(0x103c, 0x2237, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10138 SND_PCI_QUIRK(0x103c, 0x2238, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10139 SND_PCI_QUIRK(0x103c, 0x2239, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10140 SND_PCI_QUIRK(0x103c, 0x224b, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10141 SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10142 SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10143 SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10144 SND_PCI_QUIRK(0x103c, 0x2256, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10145 SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10146 SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10147 SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED), 10148 SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY), 10149 SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10150 SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10151 SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10152 SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10153 SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10154 SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10155 SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10156 SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10157 SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED), 10158 SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10159 SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS), 10160 SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10161 SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC280_FIXUP_HP_DOCK_PINS), 10162 SND_PCI_QUIRK(0x103c, 0x2278, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10163 SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10164 SND_PCI_QUIRK(0x103c, 0x2282, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10165 SND_PCI_QUIRK(0x103c, 0x228b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10166 SND_PCI_QUIRK(0x103c, 0x228e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10167 SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10168 SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10169 SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10170 SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10171 SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10172 SND_PCI_QUIRK(0x103c, 0x22c5, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10173 SND_PCI_QUIRK(0x103c, 0x22c7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10174 SND_PCI_QUIRK(0x103c, 0x22c8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10175 SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10176 SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC280_FIXUP_HP_9480M), 10177 SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10178 SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10179 SND_PCI_QUIRK(0x103c, 0x2334, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10180 SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10181 SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10182 SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10183 SND_PCI_QUIRK(0x103c, 0x2b5e, "HP 288 Pro G2 MT", ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE), 10184 SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), 10185 SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), 10186 SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC), 10187 SND_PCI_QUIRK(0x103c, 0x8158, "HP", ALC256_FIXUP_HP_HEADSET_MIC), 10188 SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC295_FIXUP_HP_X360), 10189 SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), 10190 SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), 10191 SND_PCI_QUIRK(0x103c, 0x827f, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10192 SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), 10193 SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), 10194 SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10195 SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10196 SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10197 SND_PCI_QUIRK(0x103c, 0x84a6, "HP 250 G7 Notebook PC", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10198 SND_PCI_QUIRK(0x103c, 0x84ae, "HP 15-db0403ng", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10199 SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), 10200 SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10201 SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), 10202 SND_PCI_QUIRK(0x103c, 0x8537, "HP ProBook 440 G6", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10203 SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360), 10204 SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10205 SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10206 SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), 10207 SND_PCI_QUIRK(0x103c, 0x86c1, "HP Laptop 15-da3001TU", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10208 SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), 10209 SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), 10210 SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), 10211 SND_PCI_QUIRK(0x103c, 0x86f9, "HP Spectre x360 13-aw0xxx", ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED), 10212 SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10213 SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10214 SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), 10215 SND_PCI_QUIRK(0x103c, 0x8728, "HP EliteBook 840 G7", ALC285_FIXUP_HP_GPIO_LED), 10216 SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), 10217 SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10218 SND_PCI_QUIRK(0x103c, 0x8735, "HP ProBook 435 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10219 SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10220 SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), 10221 SND_PCI_QUIRK(0x103c, 0x876e, "HP ENVY x360 Convertible 13-ay0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), 10222 SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), 10223 SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), 10224 SND_PCI_QUIRK(0x103c, 0x8780, "HP ZBook Fury 17 G7 Mobile Workstation", 10225 ALC285_FIXUP_HP_GPIO_AMP_INIT), 10226 SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", 10227 ALC285_FIXUP_HP_GPIO_AMP_INIT), 10228 SND_PCI_QUIRK(0x103c, 0x8786, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), 10229 SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), 10230 SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), 10231 SND_PCI_QUIRK(0x103c, 0x87b7, "HP Laptop 14-fq0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10232 SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), 10233 SND_PCI_QUIRK(0x103c, 0x87d3, "HP Laptop 15-gw0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10234 SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10235 SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10236 SND_PCI_QUIRK(0x103c, 0x87f1, "HP ProBook 630 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10237 SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10238 SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), 10239 SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), 10240 SND_PCI_QUIRK(0x103c, 0x87f6, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), 10241 SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), 10242 SND_PCI_QUIRK(0x103c, 0x87fd, "HP Laptop 14-dq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10243 SND_PCI_QUIRK(0x103c, 0x87fe, "HP Laptop 15s-fq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10244 SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10245 SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10246 SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), 10247 SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), 10248 SND_PCI_QUIRK(0x103c, 0x881d, "HP 250 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10249 SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10250 SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10251 SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10252 SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10253 SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), 10254 SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), 10255 SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10256 SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10257 SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10258 SND_PCI_QUIRK(0x103c, 0x887a, "HP Laptop 15s-eq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10259 SND_PCI_QUIRK(0x103c, 0x888a, "HP ENVY x360 Convertible 15-eu0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), 10260 SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED), 10261 SND_PCI_QUIRK(0x103c, 0x8895, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED), 10262 SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), 10263 SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), 10264 SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), 10265 SND_PCI_QUIRK(0x103c, 0x88dd, "HP Pavilion 15z-ec200", ALC285_FIXUP_HP_MUTE_LED), 10266 SND_PCI_QUIRK(0x103c, 0x8902, "HP OMEN 16", ALC285_FIXUP_HP_MUTE_LED), 10267 SND_PCI_QUIRK(0x103c, 0x890e, "HP 255 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10268 SND_PCI_QUIRK(0x103c, 0x8919, "HP Pavilion Aero Laptop 13-be0xxx", ALC287_FIXUP_HP_GPIO_LED), 10269 SND_PCI_QUIRK(0x103c, 0x896d, "HP ZBook Firefly 16 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10270 SND_PCI_QUIRK(0x103c, 0x896e, "HP EliteBook x360 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10271 SND_PCI_QUIRK(0x103c, 0x8971, "HP EliteBook 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10272 SND_PCI_QUIRK(0x103c, 0x8972, "HP EliteBook 840 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10273 SND_PCI_QUIRK(0x103c, 0x8973, "HP EliteBook 860 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10274 SND_PCI_QUIRK(0x103c, 0x8974, "HP EliteBook 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10275 SND_PCI_QUIRK(0x103c, 0x8975, "HP EliteBook x360 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10276 SND_PCI_QUIRK(0x103c, 0x897d, "HP mt440 Mobile Thin Client U74", ALC236_FIXUP_HP_GPIO_LED), 10277 SND_PCI_QUIRK(0x103c, 0x8981, "HP Elite Dragonfly G3", ALC245_FIXUP_CS35L41_SPI_4), 10278 SND_PCI_QUIRK(0x103c, 0x898e, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), 10279 SND_PCI_QUIRK(0x103c, 0x898f, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), 10280 SND_PCI_QUIRK(0x103c, 0x8991, "HP EliteBook 845 G9", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10281 SND_PCI_QUIRK(0x103c, 0x8992, "HP EliteBook 845 G9", ALC287_FIXUP_CS35L41_I2C_2), 10282 SND_PCI_QUIRK(0x103c, 0x8994, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10283 SND_PCI_QUIRK(0x103c, 0x8995, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2), 10284 SND_PCI_QUIRK(0x103c, 0x89a4, "HP ProBook 440 G9", ALC236_FIXUP_HP_GPIO_LED), 10285 SND_PCI_QUIRK(0x103c, 0x89a6, "HP ProBook 450 G9", ALC236_FIXUP_HP_GPIO_LED), 10286 SND_PCI_QUIRK(0x103c, 0x89aa, "HP EliteBook 630 G9", ALC236_FIXUP_HP_GPIO_LED), 10287 SND_PCI_QUIRK(0x103c, 0x89ac, "HP EliteBook 640 G9", ALC236_FIXUP_HP_GPIO_LED), 10288 SND_PCI_QUIRK(0x103c, 0x89ae, "HP EliteBook 650 G9", ALC236_FIXUP_HP_GPIO_LED), 10289 SND_PCI_QUIRK(0x103c, 0x89c0, "HP ZBook Power 15.6 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10290 SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), 10291 SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10292 SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10293 SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10294 SND_PCI_QUIRK(0x103c, 0x89e7, "HP Elite x2 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10295 SND_PCI_QUIRK(0x103c, 0x8a0f, "HP Pavilion 14-ec1xxx", ALC287_FIXUP_HP_GPIO_LED), 10296 SND_PCI_QUIRK(0x103c, 0x8a20, "HP Laptop 15s-fq5xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10297 SND_PCI_QUIRK(0x103c, 0x8a25, "HP Victus 16-d1xxx (MB 8A25)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), 10298 SND_PCI_QUIRK(0x103c, 0x8a28, "HP Envy 13", ALC287_FIXUP_CS35L41_I2C_2), 10299 SND_PCI_QUIRK(0x103c, 0x8a29, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10300 SND_PCI_QUIRK(0x103c, 0x8a2a, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10301 SND_PCI_QUIRK(0x103c, 0x8a2b, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10302 SND_PCI_QUIRK(0x103c, 0x8a2c, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10303 SND_PCI_QUIRK(0x103c, 0x8a2d, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10304 SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10305 SND_PCI_QUIRK(0x103c, 0x8a30, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10306 SND_PCI_QUIRK(0x103c, 0x8a31, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10307 SND_PCI_QUIRK(0x103c, 0x8a6e, "HP EDNA 360", ALC287_FIXUP_CS35L41_I2C_4), 10308 SND_PCI_QUIRK(0x103c, 0x8a74, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10309 SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), 10310 SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), 10311 SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), 10312 SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), 10313 SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), 10314 SND_PCI_QUIRK(0x103c, 0x8ab9, "HP EliteBook 840 G8 (MB 8AB8)", ALC285_FIXUP_HP_GPIO_LED), 10315 SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10316 SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10317 SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10318 SND_PCI_QUIRK(0x103c, 0x8ad8, "HP 800 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10319 SND_PCI_QUIRK(0x103c, 0x8b0f, "HP Elite mt645 G7 Mobile Thin Client U81", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10320 SND_PCI_QUIRK(0x103c, 0x8b2f, "HP 255 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10321 SND_PCI_QUIRK(0x103c, 0x8b3a, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10322 SND_PCI_QUIRK(0x103c, 0x8b3f, "HP mt440 Mobile Thin Client U91", ALC236_FIXUP_HP_GPIO_LED), 10323 SND_PCI_QUIRK(0x103c, 0x8b42, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10324 SND_PCI_QUIRK(0x103c, 0x8b43, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10325 SND_PCI_QUIRK(0x103c, 0x8b44, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10326 SND_PCI_QUIRK(0x103c, 0x8b45, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10327 SND_PCI_QUIRK(0x103c, 0x8b46, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10328 SND_PCI_QUIRK(0x103c, 0x8b47, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10329 SND_PCI_QUIRK(0x103c, 0x8b59, "HP Elite mt645 G7 Mobile Thin Client U89", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10330 SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10331 SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10332 SND_PCI_QUIRK(0x103c, 0x8b63, "HP Elite Dragonfly 13.5 inch G4", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), 10333 SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10334 SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10335 SND_PCI_QUIRK(0x103c, 0x8b70, "HP EliteBook 835 G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10336 SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10337 SND_PCI_QUIRK(0x103c, 0x8b74, "HP EliteBook 845W G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10338 SND_PCI_QUIRK(0x103c, 0x8b77, "HP ElieBook 865 G10", ALC287_FIXUP_CS35L41_I2C_2), 10339 SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), 10340 SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED), 10341 SND_PCI_QUIRK(0x103c, 0x8b87, "HP", ALC236_FIXUP_HP_GPIO_LED), 10342 SND_PCI_QUIRK(0x103c, 0x8b8a, "HP", ALC236_FIXUP_HP_GPIO_LED), 10343 SND_PCI_QUIRK(0x103c, 0x8b8b, "HP", ALC236_FIXUP_HP_GPIO_LED), 10344 SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), 10345 SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), 10346 SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10347 SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10348 SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10349 SND_PCI_QUIRK(0x103c, 0x8bb3, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2), 10350 SND_PCI_QUIRK(0x103c, 0x8bb4, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2), 10351 SND_PCI_QUIRK(0x103c, 0x8bdd, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10352 SND_PCI_QUIRK(0x103c, 0x8bde, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10353 SND_PCI_QUIRK(0x103c, 0x8bdf, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10354 SND_PCI_QUIRK(0x103c, 0x8be0, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10355 SND_PCI_QUIRK(0x103c, 0x8be1, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10356 SND_PCI_QUIRK(0x103c, 0x8be2, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10357 SND_PCI_QUIRK(0x103c, 0x8be3, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10358 SND_PCI_QUIRK(0x103c, 0x8be5, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10359 SND_PCI_QUIRK(0x103c, 0x8be6, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10360 SND_PCI_QUIRK(0x103c, 0x8be7, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10361 SND_PCI_QUIRK(0x103c, 0x8be8, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10362 SND_PCI_QUIRK(0x103c, 0x8be9, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10363 SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), 10364 SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre x360 2-in-1 Laptop 14-eu0xxx", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), 10365 SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), 10366 SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), 10367 SND_PCI_QUIRK(0x103c, 0x8c21, "HP Pavilion Plus Laptop 14-ey0XXX", ALC245_FIXUP_HP_X360_MUTE_LEDS), 10368 SND_PCI_QUIRK(0x103c, 0x8c30, "HP Victus 15-fb1xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), 10369 SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10370 SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10371 SND_PCI_QUIRK(0x103c, 0x8c48, "HP EliteBook 860 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10372 SND_PCI_QUIRK(0x103c, 0x8c49, "HP Elite x360 830 2-in-1 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10373 SND_PCI_QUIRK(0x103c, 0x8c4d, "HP Omen", ALC287_FIXUP_CS35L41_I2C_2), 10374 SND_PCI_QUIRK(0x103c, 0x8c4e, "HP Omen", ALC287_FIXUP_CS35L41_I2C_2), 10375 SND_PCI_QUIRK(0x103c, 0x8c4f, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10376 SND_PCI_QUIRK(0x103c, 0x8c50, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10377 SND_PCI_QUIRK(0x103c, 0x8c51, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10378 SND_PCI_QUIRK(0x103c, 0x8c52, "HP EliteBook 1040 G11", ALC285_FIXUP_HP_GPIO_LED), 10379 SND_PCI_QUIRK(0x103c, 0x8c53, "HP Elite x360 1040 2-in-1 G11", ALC285_FIXUP_HP_GPIO_LED), 10380 SND_PCI_QUIRK(0x103c, 0x8c66, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10381 SND_PCI_QUIRK(0x103c, 0x8c67, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10382 SND_PCI_QUIRK(0x103c, 0x8c68, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10383 SND_PCI_QUIRK(0x103c, 0x8c6a, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10384 SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10385 SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10386 SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10387 SND_PCI_QUIRK(0x103c, 0x8c7b, "HP ProBook 445 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10388 SND_PCI_QUIRK(0x103c, 0x8c7c, "HP ProBook 445 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10389 SND_PCI_QUIRK(0x103c, 0x8c7d, "HP ProBook 465 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10390 SND_PCI_QUIRK(0x103c, 0x8c7e, "HP ProBook 465 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10391 SND_PCI_QUIRK(0x103c, 0x8c7f, "HP EliteBook 645 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10392 SND_PCI_QUIRK(0x103c, 0x8c80, "HP EliteBook 645 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10393 SND_PCI_QUIRK(0x103c, 0x8c81, "HP EliteBook 665 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10394 SND_PCI_QUIRK(0x103c, 0x8c89, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED), 10395 SND_PCI_QUIRK(0x103c, 0x8c8a, "HP EliteBook 630", ALC236_FIXUP_HP_GPIO_LED), 10396 SND_PCI_QUIRK(0x103c, 0x8c8c, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED), 10397 SND_PCI_QUIRK(0x103c, 0x8c8d, "HP ProBook 440 G11", ALC236_FIXUP_HP_GPIO_LED), 10398 SND_PCI_QUIRK(0x103c, 0x8c8e, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED), 10399 SND_PCI_QUIRK(0x103c, 0x8c90, "HP EliteBook 640", ALC236_FIXUP_HP_GPIO_LED), 10400 SND_PCI_QUIRK(0x103c, 0x8c91, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED), 10401 SND_PCI_QUIRK(0x103c, 0x8c96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10402 SND_PCI_QUIRK(0x103c, 0x8c97, "HP ZBook", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10403 SND_PCI_QUIRK(0x103c, 0x8ca1, "HP ZBook Power", ALC236_FIXUP_HP_GPIO_LED), 10404 SND_PCI_QUIRK(0x103c, 0x8ca2, "HP ZBook Power", ALC236_FIXUP_HP_GPIO_LED), 10405 SND_PCI_QUIRK(0x103c, 0x8ca4, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10406 SND_PCI_QUIRK(0x103c, 0x8ca7, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10407 SND_PCI_QUIRK(0x103c, 0x8caf, "HP Elite mt645 G8 Mobile Thin Client", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10408 SND_PCI_QUIRK(0x103c, 0x8cbd, "HP Pavilion Aero Laptop 13-bg0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), 10409 SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), 10410 SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), 10411 SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10412 SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10413 SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), 10414 SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), 10415 SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), 10416 SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10417 SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), 10418 SND_PCI_QUIRK(0x1043, 0x10c0, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), 10419 SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), 10420 SND_PCI_QUIRK(0x1043, 0x10d3, "ASUS K6500ZC", ALC294_FIXUP_ASUS_SPK), 10421 SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10422 SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), 10423 SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), 10424 SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), 10425 SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), 10426 SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), 10427 SND_PCI_QUIRK(0x1043, 0x12a3, "Asus N7691ZM", ALC269_FIXUP_ASUS_N7601ZM), 10428 SND_PCI_QUIRK(0x1043, 0x12af, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), 10429 SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), 10430 SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), 10431 SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), 10432 SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), 10433 SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), 10434 SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650PY/PZ/PV/PU/PYV/PZV/PIV/PVV", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), 10435 SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X/GA402N", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), 10436 SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604VI/VC/VE/VG/VJ/VQ/VU/VV/VY/VZ", ALC285_FIXUP_ASUS_HEADSET_MIC), 10437 SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603VQ/VU/VV/VJ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC), 10438 SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601VV/VU/VJ/VQ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC), 10439 SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G614JY/JZ/JG", ALC245_FIXUP_CS35L41_SPI_2), 10440 SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS G513PI/PU/PV", ALC287_FIXUP_CS35L41_I2C_2), 10441 SND_PCI_QUIRK(0x1043, 0x1503, "ASUS G733PY/PZ/PZV/PYV", ALC287_FIXUP_CS35L41_I2C_2), 10442 SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), 10443 SND_PCI_QUIRK(0x1043, 0x1533, "ASUS GV302XA/XJ/XQ/XU/XV/XI", ALC287_FIXUP_CS35L41_I2C_2), 10444 SND_PCI_QUIRK(0x1043, 0x1573, "ASUS GZ301VV/VQ/VU/VJ/VA/VC/VE/VVC/VQC/VUC/VJC/VEC/VCC", ALC285_FIXUP_ASUS_HEADSET_MIC), 10445 SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), 10446 SND_PCI_QUIRK(0x1043, 0x1663, "ASUS GU603ZI/ZJ/ZQ/ZU/ZV", ALC285_FIXUP_ASUS_HEADSET_MIC), 10447 SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), 10448 SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS UX3402VA", ALC245_FIXUP_CS35L41_SPI_2), 10449 SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), 10450 SND_PCI_QUIRK(0x1043, 0x16d3, "ASUS UX5304VA", ALC245_FIXUP_CS35L41_SPI_2), 10451 SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), 10452 SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS UX7602VI/BZ", ALC245_FIXUP_CS35L41_SPI_2), 10453 SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), 10454 SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), 10455 SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY), 10456 SND_PCI_QUIRK(0x1043, 0x1eb3, "ROG Ally X RC72LA", ALC294_FIXUP_ASUS_ALLY_X), 10457 SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404VI/VV", ALC245_FIXUP_CS35L41_SPI_2), 10458 SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), 10459 SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), 10460 SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS UM3504DA", ALC294_FIXUP_CS35L41_I2C_2), 10461 SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), 10462 SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), 10463 SND_PCI_QUIRK(0x1043, 0x1970, "ASUS UX550VE", ALC289_FIXUP_ASUS_GA401), 10464 SND_PCI_QUIRK(0x1043, 0x1982, "ASUS B1400CEPE", ALC256_FIXUP_ASUS_HPE), 10465 SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), 10466 SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), 10467 SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), 10468 SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), 10469 SND_PCI_QUIRK(0x1043, 0x1a63, "ASUS UX3405MA", ALC245_FIXUP_CS35L41_SPI_2), 10470 SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2), 10471 SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), 10472 SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), 10473 SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC), 10474 SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), 10475 SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), 10476 SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2), 10477 SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10478 SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2), 10479 SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), 10480 SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), 10481 SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1), 10482 SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), 10483 SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC), 10484 SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10485 SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), 10486 SND_PCI_QUIRK(0x1043, 0x1ccf, "ASUS G814JU/JV/JI", ALC245_FIXUP_CS35L41_SPI_2), 10487 SND_PCI_QUIRK(0x1043, 0x1cdf, "ASUS G814JY/JZ/JG", ALC245_FIXUP_CS35L41_SPI_2), 10488 SND_PCI_QUIRK(0x1043, 0x1cef, "ASUS G834JY/JZ/JI/JG", ALC285_FIXUP_ASUS_HEADSET_MIC), 10489 SND_PCI_QUIRK(0x1043, 0x1d1f, "ASUS G713PI/PU/PV/PVN", ALC287_FIXUP_CS35L41_I2C_2), 10490 SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), 10491 SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), 10492 SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), 10493 SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), 10494 SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), 10495 SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), 10496 SND_PCI_QUIRK(0x1043, 0x1e1f, "ASUS Vivobook 15 X1504VAP", ALC2XX_FIXUP_HEADSET_MIC), 10497 SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), 10498 SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), 10499 SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), 10500 SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), 10501 SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), 10502 SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), 10503 SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), 10504 SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2), 10505 SND_PCI_QUIRK(0x1043, 0x1f1f, "ASUS H7604JI/JV/J3D", ALC245_FIXUP_CS35L41_SPI_2), 10506 SND_PCI_QUIRK(0x1043, 0x1f62, "ASUS UX7602ZM", ALC245_FIXUP_CS35L41_SPI_2), 10507 SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), 10508 SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), 10509 SND_PCI_QUIRK(0x1043, 0x3a20, "ASUS G614JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10510 SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10511 SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10512 SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10513 SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10514 SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), 10515 SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), 10516 SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), 10517 SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), 10518 SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), 10519 SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), 10520 SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), 10521 SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), 10522 SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), 10523 SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), 10524 SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), 10525 SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), 10526 SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT), 10527 SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN), 10528 SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), 10529 SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), 10530 SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), 10531 SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), 10532 SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), 10533 SND_PCI_QUIRK(0x10ec, 0x119e, "Positivo SU C1400", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10534 SND_PCI_QUIRK(0x10ec, 0x11bc, "VAIO VJFE-IL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10535 SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10536 SND_PCI_QUIRK(0x10ec, 0x124c, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10537 SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10538 SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10539 SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10540 SND_PCI_QUIRK(0x10ec, 0x12f6, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10541 SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10542 SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), 10543 SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), 10544 SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP), 10545 SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP), 10546 SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_AMP), 10547 SND_PCI_QUIRK(0x144d, 0xc1a3, "Samsung Galaxy Book Pro (NP935XDB-KC1SE)", ALC298_FIXUP_SAMSUNG_AMP), 10548 SND_PCI_QUIRK(0x144d, 0xc1a4, "Samsung Galaxy Book Pro 360 (NT935QBD)", ALC298_FIXUP_SAMSUNG_AMP), 10549 SND_PCI_QUIRK(0x144d, 0xc1a6, "Samsung Galaxy Book Pro 360 (NP930QBD)", ALC298_FIXUP_SAMSUNG_AMP), 10550 SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), 10551 SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_AMP), 10552 SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), 10553 SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), 10554 SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), 10555 SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), 10556 SND_PCI_QUIRK(0x144d, 0xc870, "Samsung Galaxy Book2 Pro (NP950XED)", ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS), 10557 SND_PCI_QUIRK(0x144d, 0xc872, "Samsung Galaxy Book2 Pro (NP950XEE)", ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS), 10558 SND_PCI_QUIRK(0x144d, 0xc886, "Samsung Galaxy Book3 Pro (NP964XFG)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10559 SND_PCI_QUIRK(0x144d, 0xc1ca, "Samsung Galaxy Book3 Pro 360 (NP960QFG)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10560 SND_PCI_QUIRK(0x144d, 0xc1cc, "Samsung Galaxy Book3 Ultra (NT960XFH)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10561 SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), 10562 SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), 10563 SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), 10564 SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), 10565 SND_PCI_QUIRK(0x152d, 0x1262, "Huawei NBLB-WAX9N", ALC2XX_FIXUP_HEADSET_MIC), 10566 SND_PCI_QUIRK(0x1558, 0x0353, "Clevo V35[05]SN[CDE]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10567 SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10568 SND_PCI_QUIRK(0x1558, 0x1325, "Clevo N15[01][CW]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10569 SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10570 SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10571 SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10572 SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10573 SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10574 SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10575 SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10576 SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10577 SND_PCI_QUIRK(0x1558, 0x4041, "Clevo NV4[15]PZ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10578 SND_PCI_QUIRK(0x1558, 0x40a1, "Clevo NL40GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10579 SND_PCI_QUIRK(0x1558, 0x40c1, "Clevo NL40[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10580 SND_PCI_QUIRK(0x1558, 0x40d1, "Clevo NL41DU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10581 SND_PCI_QUIRK(0x1558, 0x5015, "Clevo NH5[58]H[HJK]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10582 SND_PCI_QUIRK(0x1558, 0x5017, "Clevo NH7[79]H[HJK]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10583 SND_PCI_QUIRK(0x1558, 0x50a3, "Clevo NJ51GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10584 SND_PCI_QUIRK(0x1558, 0x50b3, "Clevo NK50S[BEZ]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10585 SND_PCI_QUIRK(0x1558, 0x50b6, "Clevo NK50S5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10586 SND_PCI_QUIRK(0x1558, 0x50b8, "Clevo NK50SZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10587 SND_PCI_QUIRK(0x1558, 0x50d5, "Clevo NP50D5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10588 SND_PCI_QUIRK(0x1558, 0x50e1, "Clevo NH5[58]HPQ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10589 SND_PCI_QUIRK(0x1558, 0x50e2, "Clevo NH7[79]HPQ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10590 SND_PCI_QUIRK(0x1558, 0x50f0, "Clevo NH50A[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10591 SND_PCI_QUIRK(0x1558, 0x50f2, "Clevo NH50E[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10592 SND_PCI_QUIRK(0x1558, 0x50f3, "Clevo NH58DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10593 SND_PCI_QUIRK(0x1558, 0x50f5, "Clevo NH55EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10594 SND_PCI_QUIRK(0x1558, 0x50f6, "Clevo NH55DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10595 SND_PCI_QUIRK(0x1558, 0x5101, "Clevo S510WU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10596 SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10597 SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10598 SND_PCI_QUIRK(0x1558, 0x51b1, "Clevo NS50AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10599 SND_PCI_QUIRK(0x1558, 0x51b3, "Clevo NS70AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10600 SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10601 SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10602 SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10603 SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10604 SND_PCI_QUIRK(0x1558, 0x70f3, "Clevo NH77DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10605 SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10606 SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10607 SND_PCI_QUIRK(0x1558, 0x7716, "Clevo NS50PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10608 SND_PCI_QUIRK(0x1558, 0x7717, "Clevo NS70PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10609 SND_PCI_QUIRK(0x1558, 0x7718, "Clevo L140PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10610 SND_PCI_QUIRK(0x1558, 0x7724, "Clevo L140AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10611 SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10612 SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10613 SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10614 SND_PCI_QUIRK(0x1558, 0x8535, "Clevo NH50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10615 SND_PCI_QUIRK(0x1558, 0x8536, "Clevo NH79D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10616 SND_PCI_QUIRK(0x1558, 0x8550, "Clevo NH[57][0-9][ER][ACDH]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10617 SND_PCI_QUIRK(0x1558, 0x8551, "Clevo NH[57][0-9][ER][ACDH]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10618 SND_PCI_QUIRK(0x1558, 0x8560, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), 10619 SND_PCI_QUIRK(0x1558, 0x8561, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), 10620 SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), 10621 SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10622 SND_PCI_QUIRK(0x1558, 0x866d, "Clevo NP5[05]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10623 SND_PCI_QUIRK(0x1558, 0x867c, "Clevo NP7[01]PNP", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10624 SND_PCI_QUIRK(0x1558, 0x867d, "Clevo NP7[01]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10625 SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10626 SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME), 10627 SND_PCI_QUIRK(0x1558, 0x8a20, "Clevo NH55DCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10628 SND_PCI_QUIRK(0x1558, 0x8a51, "Clevo NH70RCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10629 SND_PCI_QUIRK(0x1558, 0x8d50, "Clevo NH55RCQ-M", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10630 SND_PCI_QUIRK(0x1558, 0x951d, "Clevo N950T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10631 SND_PCI_QUIRK(0x1558, 0x9600, "Clevo N960K[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10632 SND_PCI_QUIRK(0x1558, 0x961d, "Clevo N960S[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10633 SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10634 SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10635 SND_PCI_QUIRK(0x1558, 0xa554, "VAIO VJFH52", ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE), 10636 SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10637 SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10638 SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10639 SND_PCI_QUIRK(0x1558, 0xa741, "Clevo V54x_6x_TNE", ALC245_FIXUP_CLEVO_NOISY_MIC), 10640 SND_PCI_QUIRK(0x1558, 0xa763, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), 10641 SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10642 SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10643 SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10644 SND_PCI_QUIRK(0x1558, 0xc018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10645 SND_PCI_QUIRK(0x1558, 0xc019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10646 SND_PCI_QUIRK(0x1558, 0xc022, "Clevo NH77[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10647 SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), 10648 SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), 10649 SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), 10650 SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), 10651 SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), 10652 SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), 10653 SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), 10654 SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), 10655 SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST), 10656 SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), 10657 SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), 10658 SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), 10659 SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), 10660 SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440), 10661 SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), 10662 SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), 10663 SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK), 10664 SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), 10665 SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), 10666 SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10667 SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), 10668 SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), 10669 SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), 10670 SND_PCI_QUIRK(0x17aa, 0x222d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10671 SND_PCI_QUIRK(0x17aa, 0x222e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10672 SND_PCI_QUIRK(0x17aa, 0x2231, "Thinkpad T560", ALC292_FIXUP_TPT460), 10673 SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), 10674 SND_PCI_QUIRK(0x17aa, 0x2234, "Thinkpad ICE-1", ALC287_FIXUP_TAS2781_I2C), 10675 SND_PCI_QUIRK(0x17aa, 0x2245, "Thinkpad T470", ALC298_FIXUP_TPT470_DOCK), 10676 SND_PCI_QUIRK(0x17aa, 0x2246, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10677 SND_PCI_QUIRK(0x17aa, 0x2247, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10678 SND_PCI_QUIRK(0x17aa, 0x2249, "Thinkpad", ALC292_FIXUP_TPT460), 10679 SND_PCI_QUIRK(0x17aa, 0x224b, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10680 SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10681 SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10682 SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10683 SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), 10684 SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), 10685 SND_PCI_QUIRK(0x17aa, 0x22c1, "Thinkpad P1 Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), 10686 SND_PCI_QUIRK(0x17aa, 0x22c2, "Thinkpad X1 Extreme Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), 10687 SND_PCI_QUIRK(0x17aa, 0x22f1, "Thinkpad", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10688 SND_PCI_QUIRK(0x17aa, 0x22f2, "Thinkpad", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10689 SND_PCI_QUIRK(0x17aa, 0x22f3, "Thinkpad", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10690 SND_PCI_QUIRK(0x17aa, 0x2316, "Thinkpad P1 Gen 6", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10691 SND_PCI_QUIRK(0x17aa, 0x2317, "Thinkpad P1 Gen 6", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10692 SND_PCI_QUIRK(0x17aa, 0x2318, "Thinkpad Z13 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10693 SND_PCI_QUIRK(0x17aa, 0x2319, "Thinkpad Z16 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10694 SND_PCI_QUIRK(0x17aa, 0x231a, "Thinkpad Z16 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10695 SND_PCI_QUIRK(0x17aa, 0x231e, "Thinkpad", ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318), 10696 SND_PCI_QUIRK(0x17aa, 0x231f, "Thinkpad", ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318), 10697 SND_PCI_QUIRK(0x17aa, 0x2326, "Hera2", ALC287_FIXUP_TAS2781_I2C), 10698 SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), 10699 SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), 10700 SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10701 SND_PCI_QUIRK(0x17aa, 0x3111, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10702 SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10703 SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10704 SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10705 SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), 10706 SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), 10707 SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), 10708 SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), 10709 SND_PCI_QUIRK(0x17aa, 0x334b, "Lenovo ThinkCentre M70 Gen5", ALC283_FIXUP_HEADSET_MIC), 10710 SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), 10711 HDA_CODEC_QUIRK(0x17aa, 0x3802, "DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10712 SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga Pro 9 14IRP8", ALC287_FIXUP_TAS2781_I2C), 10713 SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), 10714 SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), 10715 SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), 10716 HDA_CODEC_QUIRK(0x17aa, 0x3820, "IdeaPad 330-17IKB 81DM", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10717 SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10718 SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), 10719 SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), 10720 SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10721 SND_PCI_QUIRK(0x17aa, 0x383d, "Legion Y9000X 2019", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), 10722 SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), 10723 SND_PCI_QUIRK(0x17aa, 0x3847, "Legion 7 16ACHG6", ALC287_FIXUP_LEGION_16ACHG6), 10724 SND_PCI_QUIRK(0x17aa, 0x384a, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10725 SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10726 SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10727 SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), 10728 SND_PCI_QUIRK(0x17aa, 0x3865, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), 10729 SND_PCI_QUIRK(0x17aa, 0x3866, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), 10730 SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), 10731 HDA_CODEC_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2), 10732 SND_PCI_QUIRK(0x17aa, 0x386e, "Yoga Pro 7 14ARP8", ALC285_FIXUP_SPEAKER2_TO_DAC1), 10733 HDA_CODEC_QUIRK(0x17aa, 0x386f, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), 10734 SND_PCI_QUIRK(0x17aa, 0x386f, "Legion Pro 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), 10735 SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), 10736 SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), 10737 SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), 10738 SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), 10739 SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), 10740 SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C), 10741 SND_PCI_QUIRK(0x17aa, 0x3882, "Lenovo Yoga Pro 7 14APH8", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), 10742 SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), 10743 SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), 10744 SND_PCI_QUIRK(0x17aa, 0x3891, "Lenovo Yoga Pro 7 14AHP9", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), 10745 SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), 10746 SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), 10747 SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10748 SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10749 SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), 10750 SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), 10751 SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), 10752 SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), 10753 SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), 10754 SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), 10755 SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), 10756 SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", ALC287_FIXUP_TAS2781_I2C), 10757 SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), 10758 SND_PCI_QUIRK(0x17aa, 0x38c7, "Thinkbook 13x Gen 4", ALC287_FIXUP_CS35L41_I2C_4), 10759 SND_PCI_QUIRK(0x17aa, 0x38c8, "Thinkbook 13x Gen 4", ALC287_FIXUP_CS35L41_I2C_4), 10760 SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), 10761 SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), 10762 SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), 10763 SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), 10764 SND_PCI_QUIRK(0x17aa, 0x38df, "Y990 YG DUAL", ALC287_FIXUP_TAS2781_I2C), 10765 SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), 10766 SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), 10767 SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), 10768 SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC), 10769 SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), 10770 SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), 10771 SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), 10772 SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10773 SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), 10774 SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), 10775 SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10776 SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), 10777 SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), 10778 SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), 10779 SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK), 10780 SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), 10781 SND_PCI_QUIRK(0x17aa, 0x5050, "Thinkpad T560p", ALC292_FIXUP_TPT460), 10782 SND_PCI_QUIRK(0x17aa, 0x5051, "Thinkpad L460", ALC292_FIXUP_TPT460), 10783 SND_PCI_QUIRK(0x17aa, 0x5053, "Thinkpad T460", ALC292_FIXUP_TPT460), 10784 SND_PCI_QUIRK(0x17aa, 0x505d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10785 SND_PCI_QUIRK(0x17aa, 0x505f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10786 SND_PCI_QUIRK(0x17aa, 0x5062, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10787 SND_PCI_QUIRK(0x17aa, 0x508b, "Thinkpad X12 Gen 1", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), 10788 SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10789 SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10790 SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10791 SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), 10792 SND_PCI_QUIRK(0x17aa, 0x9e56, "Lenovo ZhaoYang CF4620Z", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), 10793 SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), 10794 SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), 10795 SND_PCI_QUIRK(0x1854, 0x0440, "LG CQ6", ALC256_FIXUP_HEADPHONE_AMP_VOL), 10796 SND_PCI_QUIRK(0x1854, 0x0441, "LG CQ6 AIO", ALC256_FIXUP_HEADPHONE_AMP_VOL), 10797 SND_PCI_QUIRK(0x1854, 0x0488, "LG gram 16 (16Z90R)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10798 SND_PCI_QUIRK(0x1854, 0x048a, "LG gram 17 (17ZD90R)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10799 SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), 10800 SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), 10801 SND_PCI_QUIRK(0x19e5, 0x3212, "Huawei KLV-WX9 ", ALC256_FIXUP_ACER_HEADSET_MIC), 10802 SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), 10803 SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), 10804 SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), 10805 SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ 10806 SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), 10807 SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), 10808 SND_PCI_QUIRK(0x1c6c, 0x122a, "Positivo N14AP7", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10809 SND_PCI_QUIRK(0x1c6c, 0x1251, "Positivo N14KP6-TG", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE), 10810 SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS), 10811 SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMRxx", ALC269_FIXUP_NO_SHUTUP), 10812 SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNRxx", ALC269_FIXUP_NO_SHUTUP), 10813 SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), 10814 SND_PCI_QUIRK(0x1d05, 0x1119, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), 10815 SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), 10816 SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), 10817 SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), 10818 SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), 10819 SND_PCI_QUIRK(0x1d05, 0x1387, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), 10820 SND_PCI_QUIRK(0x1d17, 0x3288, "Haier Boyue G42", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), 10821 SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), 10822 SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), 10823 SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), 10824 SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), 10825 SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), 10826 SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10827 SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), 10828 SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), 10829 SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), 10830 SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), 10831 SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), 10832 SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), 10833 SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), 10834 SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), 10835 SND_PCI_QUIRK(0xf111, 0x0009, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), 10836 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[PATCH v1] ALSA: hda/tas2781: Add new quirk for Lenovo, ASUS, Dell projects
by Baojun Xu 14 Oct '24

14 Oct '24

Add new vendor_id and subsystem_id in quirk for Lenovo, ASUS, and Dell projects. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/pci/hda/patch_realtek.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e2e927656cd..28d8e58e7fee 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10198,6 +10198,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0c1e, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c28, "Dell Inspiron 16 Plus 7630", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0c4d, "Dell", ALC287_FIXUP_CS35L41_I2C_4), + SND_PCI_QUIRK(0x1028, 0x0c94, "Dell Polaris 3 metal", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1028, 0x0c96, "Dell Polaris 2in1", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1028, 0x0cbd, "Dell Oasis 13 CS MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbe, "Dell Oasis 13 2-IN-1 MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbf, "Dell Oasis 13 Low Weight MTU-L", ALC289_FIXUP_DELL_CS35L41_SPI_2), @@ -10501,11 +10503,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), + SND_PCI_QUIRK(0x1043, 0x10a4, "ASUS TP3407SA", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x10c0, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x10d3, "ASUS K6500ZC", ALC294_FIXUP_ASUS_SPK), + SND_PCI_QUIRK(0x1043, 0x1154, "ASUS TP3607SH", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x1204, "ASUS Strix G615JHR_JMR_JPR", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x1214, "ASUS Strix G615LH_LM_LP", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), @@ -10583,6 +10589,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1eb3, "ASUS Ally RCLA72", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), @@ -10597,6 +10604,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), + SND_PCI_QUIRK(0x1043, 0x3e30, "ASUS TP3607SA", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3ee0, "ASUS Strix G815_JHR_JMR_JPR", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3ef0, "ASUS Strix G635LR_LW_LX", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3f00, "ASUS Strix G815LH_LM_LP", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3f10, "ASUS Strix G835LR_LW_LX", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3f20, "ASUS Strix G615LR_LW", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3f30, "ASUS Strix G815LR_LW", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), @@ -10819,11 +10833,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387f, "Yoga S780-16 pro dual LX", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3880, "Yoga S780-16 pro dual YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3882, "Lenovo Yoga Pro 7 14APH8", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3891, "Lenovo Yoga Pro 7 14AHP9", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38a5, "Y580P AMD dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), @@ -10832,6 +10849,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38b8, "Yoga S780-14.5 proX AMD YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38b9, "Yoga S780-14.5 proX AMD LX Dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), @@ -10842,12 +10861,22 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38d3, "Yoga S990-16 Pro IMH YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38d4, "Yoga S990-16 Pro IMH VECO Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38d5, "Yoga S990-16 Pro IMH YC Quad", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38d6, "Yoga S990-16 Pro IMH VECO Quad", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38df, "Yoga Y990 Intel YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38e0, "Yoga Y990 Intel VECO Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38f8, "Yoga Book 9i", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38df, "Y990 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38fd, "ThinkBook plus Gen5 Hybrid", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC), + SND_PCI_QUIRK(0x17aa, 0x391f, "Yoga S990-16 pro Quad YC Quad", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3920, "Yoga S990-16 pro Quad VECO Quad", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), -- 2.43.0

2 1

[PATCH] Support for Line 6 TonePort KB37
by Seth Poulsen 14 Oct '24

14 Oct '24

The Line 6 TonePort KB37 is a device with the same ports as the TonePort UX2, but with a MIDI keyboard onboard. It can use the same driver as the UX2, it just needed to be enabled. -Seth Poulsen

2 1

[PATCH alsa-ucm-conf v3 1/2] sof-soundwire: Change map control names to make them unique and user friendly
by Maciej Strozek 14 Oct '24

14 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v3: Make Headset Microphone more user-friendly and contain the word "Capture" v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 10 ++++++++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- ucm2/sof-soundwire/cs42l43.conf | 2 +- 4 files changed, 13 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..b2796e2 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,13 +2,19 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } + "name='cs42l43 Headset Microphone Capture Switch'" { + "name='cs42l43 Decimator 1 Switch'".vindex.0 0 + } + "name='cs42l43 Headset Microphone Capture Volume'" { + "name='cs42l43 Decimator 1 Volume'".vindex.0 0 + } } } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..a3cf6ad 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker Playback" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 1702cf0..7e74fb8 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -40,7 +40,7 @@ SectionDevice."Headset" { Value { CapturePriority 200 CapturePCM "hw:${CardId},4" - CaptureMixerElem "cs42l43 Decimator 1" + CaptureMixerElem "cs42l43 Headset Microphone" JackControl "Headset Mic Jack" } } -- 2.39.5

2 2

[PATCH] ASoC: rt-sdw-common: Enhance switch case to prevent uninitialized variable
by Jack Yu 14 Oct '24

14 Oct '24

If det_mode is not 0, 3 or 5 then function will return jack_type with an uninitialzed value. Enhance switch case to prevent uninitialized variable issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt-sdw-common.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt-sdw-common.c b/sound/soc/codecs/rt-sdw-common.c index a422da6cf702..3ec576c576be 100644 --- a/sound/soc/codecs/rt-sdw-common.c +++ b/sound/soc/codecs/rt-sdw-common.c @@ -150,15 +150,14 @@ int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id) goto io_error; switch (det_mode) { - case 0x00: - jack_type = 0; - break; case 0x03: jack_type = SND_JACK_HEADPHONE; break; case 0x05: jack_type = SND_JACK_HEADSET; break; + default: + jack_type = 0; } /* write selected_mode */ -- 2.34.1

2 1

[tiwai-sound:for-next 9/14] sound/pci/hda/hda_local.h:312:58: error: 'const struct hda_quirk' has no member named 'name'
by kernel test robot 14 Oct '24

14 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 commit: 1f55e3699fc9ced72400cdca39fe248bf2b288a2 [9/14] ALSA: hda/conexant: Use the new codec SSID matching config: i386-buildonly-randconfig-006-20241014 (https://download.01.org/0day-ci/archive/20241014/202410141700.bF6I8r36-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/20241014/202410141700.bF6I8r36-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/202410141700.bF6I8r36-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 builds fine. It only hurts bisectability. All error/warnings (new ones prefixed by >>): In file included from sound/pci/hda/patch_conexant.c:18: >> sound/pci/hda/hda_local.h:312:58: error: 'const struct hda_quirk' has no member named 'name' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^~~~ sound/pci/hda/patch_conexant.c:1081:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1081 | HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ >> sound/pci/hda/hda_local.h:312:65: warning: excess elements in struct initializer 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^ sound/pci/hda/patch_conexant.c:1081:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1081 | HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:65: note: (near initialization for 'cxt5066_fixups[47]') 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^ sound/pci/hda/patch_conexant.c:1081:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1081 | HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ >> sound/pci/hda/hda_local.h:312:58: error: 'const struct hda_quirk' has no member named 'name' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^~~~ sound/pci/hda/patch_conexant.c:1082:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1082 | HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ >> sound/pci/hda/hda_local.h:312:65: warning: excess elements in struct initializer 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^ sound/pci/hda/patch_conexant.c:1082:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1082 | HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:65: note: (near initialization for 'cxt5066_fixups[48]') 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^ sound/pci/hda/patch_conexant.c:1082:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1082 | HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ vim +312 sound/pci/hda/hda_local.h 5b1913a79c3e05 Takashi Iwai 2024-10-08 310 5b1913a79c3e05 Takashi Iwai 2024-10-08 311 #define HDA_CODEC_QUIRK(vend, dev, xname, val) \ 5b1913a79c3e05 Takashi Iwai 2024-10-08 @312 { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ 5b1913a79c3e05 Takashi Iwai 2024-10-08 313 .match_codec_ssid = true } 5b1913a79c3e05 Takashi Iwai 2024-10-08 314 :::::: The code at line 312 was first introduced by commit :::::: 5b1913a79c3e0518d9c5db343fa9fc4edcea041f ALSA: hda: Use own quirk lookup helper :::::: TO: Takashi Iwai <tiwai(a)suse.de> :::::: CC: Takashi Iwai <tiwai(a)suse.de> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 9/14] sound/pci/hda/patch_conexant.c:1081:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk'
by kernel test robot 14 Oct '24

14 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 commit: 1f55e3699fc9ced72400cdca39fe248bf2b288a2 [9/14] ALSA: hda/conexant: Use the new codec SSID matching config: i386-buildonly-randconfig-003-20241014 (https://download.01.org/0day-ci/archive/20241014/202410141648.JOPFB2du-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/20241014/202410141648.JOPFB2du-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/202410141648.JOPFB2du-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): >> sound/pci/hda/patch_conexant.c:1081:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 1081 | HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ sound/pci/hda/patch_conexant.c:1082:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 1082 | HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ 2 errors generated. vim +1081 sound/pci/hda/patch_conexant.c 1029 1030 static const struct hda_quirk cxt5066_fixups[] = { 1031 SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), 1032 SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), 1033 SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), 1034 SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), 1035 SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), 1036 SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), 1037 SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), 1038 SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), 1039 SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), 1040 SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), 1041 SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), 1042 SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1043 SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1044 SND_PCI_QUIRK(0x103c, 0x82b4, "HP ProDesk 600 G3", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1045 SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), 1046 SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), 1047 SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), 1048 SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), 1049 SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), 1050 SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO), 1051 SND_PCI_QUIRK(0x103c, 0x8427, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED), 1052 SND_PCI_QUIRK(0x103c, 0x844f, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED), 1053 SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1054 SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1055 SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1056 SND_PCI_QUIRK(0x103c, 0x8458, "HP Z2 G4 mini premium", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1057 SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), 1058 SND_PCI_QUIRK(0x14f1, 0x0265, "SWS JS201D", CXT_PINCFG_SWS_JS201D), 1059 SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), 1060 SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), 1061 SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), 1062 SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), 1063 SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), 1064 SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), 1065 SND_PCI_QUIRK(0x17aa, 0x21d2, "Lenovo T420s", CXT_PINCFG_LENOVO_TP410), 1066 SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), 1067 SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), 1068 SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), 1069 SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), 1070 SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), 1071 SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), 1072 /* NOTE: we'd need to extend the quirk for 17aa:3977 as the same 1073 * PCI SSID is used on multiple Lenovo models 1074 */ 1075 SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), 1076 SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), 1077 SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), 1078 SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), 1079 SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), 1080 SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), > 1081 HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), 1082 HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), 1083 {} 1084 }; 1085 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH 6.1] ASoC: rt5682s: Return devm_of_clk_add_hw_provider to transfer the error
by Vitaliy Shevtsov 14 Oct '24

14 Oct '24

From: Ma Ke <make24(a)iscas.ac.cn> commit 3ff810b9bebe5578a245cfa97c252ab602e703f1 upstream. Return devm_of_clk_add_hw_provider() in order to transfer the error, if it fails due to resource allocation failure or device tree clock provider registration failure. Fixes: bdd229ab26be ("ASoC: rt5682s: Add driver for ALC5682I-VS codec") Signed-off-by: Ma Ke <make24(a)iscas.ac.cn> Link: https://patch.msgid.link/20240717115436.3449492-1-make24@iscas.ac.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Vitaliy Shevtsov <v.shevtsov(a)maxima.ru> --- sound/soc/codecs/rt5682s.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 80c673aa14db..07d514b4ce70 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2828,7 +2828,9 @@ static int rt5682s_register_dai_clks(struct snd_soc_component *component) } if (dev->of_node) { - devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); + ret = devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); + if (ret) + return ret; } else { ret = devm_clk_hw_register_clkdev(dev, dai_clk_hw, init.name, dev_name(dev)); -- 2.46.2

1 0

[PATCH v2] ASoC: rt-sdw-common: Common functions for Realtek soundwire driver
by Jack Yu 13 Oct '24

13 Oct '24

This is the first version of common functions for Realtek soundwire codec driver. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt-sdw-common.c | 238 +++++++++++++++++++++++++++++++ sound/soc/codecs/rt-sdw-common.h | 66 +++++++++ 4 files changed, 311 insertions(+) create mode 100644 sound/soc/codecs/rt-sdw-common.c create mode 100644 sound/soc/codecs/rt-sdw-common.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7092842480ef..b095eba9d634 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1545,6 +1545,11 @@ config SND_SOC_RL6231 default m if SND_SOC_RT1305=m default m if SND_SOC_RT1308=m +config SND_SOC_RT_SDW_COMMON + tristate + default y if SND_SOC_RT721_SDCA_SDW=y + default m if SND_SOC_RT721_SDCA_SDW=m + config SND_SOC_RL6347A tristate default y if SND_SOC_RT274=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 54cbc3feae32..4497b2e30cd3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -216,6 +216,7 @@ snd-soc-rk3308-y := rk3308_codec.o snd-soc-rk3328-y := rk3328_codec.o snd-soc-rk817-y := rk817_codec.o snd-soc-rl6231-y := rl6231.o +snd-soc-rt-sdw-common-y := rt-sdw-common.o snd-soc-rl6347a-y := rl6347a.o snd-soc-rt1011-y := rt1011.o snd-soc-rt1015-y := rt1015.o @@ -618,6 +619,7 @@ obj-$(CONFIG_SND_SOC_RK3308) += snd-soc-rk3308.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RT_SDW_COMMON) += snd-soc-rt-sdw-common.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o obj-$(CONFIG_SND_SOC_RT1015) += snd-soc-rt1015.o diff --git a/sound/soc/codecs/rt-sdw-common.c b/sound/soc/codecs/rt-sdw-common.c new file mode 100644 index 000000000000..9ed0e9855699 --- /dev/null +++ b/sound/soc/codecs/rt-sdw-common.c @@ -0,0 +1,238 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt-sdw-common.c +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// + +/* + * This file defines common functions used with Realtek soundwire codecs. + */ + +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/bitops.h> +#include <linux/soundwire/sdw_registers.h> +#include <sound/jack.h> + +#include "rt-sdw-common.h" + +/** + * rt_sdca_index_write - Write a value to Realtek defined register. + * + * @map: map for setting. + * @nid: Realtek-defined ID. + * @reg: register. + * @value: value. + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int rt_sdca_index_write(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int value) +{ + unsigned int addr = (nid << 20) | reg; + int ret; + + ret = regmap_write(map, addr, value); + if (ret < 0) + pr_err("Failed to set value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_index_write); + +/** + * rt_sdca_index_read - Read value from Realtek defined register. + * + * @map: map for setting. + * @nid: Realtek-defined ID. + * @reg: register. + * @value: value. + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int rt_sdca_index_read(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int *value) +{ + unsigned int addr = (nid << 20) | reg; + int ret; + + ret = regmap_read(map, addr, value); + if (ret < 0) + pr_err("Failed to get value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_index_read); + +/** + * rt_sdca_index_update_bits - Update value on Realtek defined register. + * + * @map: map for setting. + * @nid: Realtek-defined ID. + * @reg: register. + * @mask: Bitmask to change + * @value: New value for bitmask + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ + +int rt_sdca_index_update_bits(struct regmap *map, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp; + int ret; + + ret = rt_sdca_index_read(map, nid, reg, &tmp); + if (ret < 0) + return ret; + + set_mask_bits(&tmp, mask, val); + return rt_sdca_index_write(map, nid, reg, tmp); +} +EXPORT_SYMBOL_GPL(rt_sdca_index_update_bits); + +/** + * rt_sdca_btn_type - Decision of button type. + * + * @buffer: UMP message buffer. + * + * A button type will be returned regarding to buffer, + * it returns zero if buffer cannot be recognized. + */ +int rt_sdca_btn_type(unsigned char *buffer) +{ + u8 btn_type = 0; + int ret; + + btn_type |= buffer[0] & 0xf; + btn_type |= (buffer[0] >> 4) & 0xf; + btn_type |= buffer[1] & 0xf; + btn_type |= (buffer[1] >> 4) & 0xf; + + if (btn_type & BIT(0)) + ret |= SND_JACK_BTN_2; + if (btn_type & BIT(1)) + ret |= SND_JACK_BTN_3; + if (btn_type & BIT(2)) + ret |= SND_JACK_BTN_0; + if (btn_type & BIT(3)) + ret |= SND_JACK_BTN_1; + + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_btn_type); + +/** + * rt_sdca_headset_detect - Headset jack type detection. + * + * @map: map for setting. + * @entity_id: SDCA entity ID. + * + * A headset jack type will be returned, a negative errno will + * be returned in error cases. + */ +int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id) +{ + unsigned int det_mode, jack_type; + int ret; + + /* get detected_mode */ + ret = regmap_read(map, SDW_SDCA_CTL(SDCA_NUM_JACK_CODEC, entity_id, + RT_SDCA_CTL_DETECTED_MODE, 0), &det_mode); + + if (ret < 0) + goto io_error; + + switch (det_mode) { + case 0x00: + jack_type = 0; + break; + case 0x03: + jack_type = SND_JACK_HEADPHONE; + break; + case 0x05: + jack_type = SND_JACK_HEADSET; + break; + } + + /* write selected_mode */ + if (det_mode) { + ret = regmap_write(map, SDW_SDCA_CTL(SDCA_NUM_JACK_CODEC, entity_id, + RT_SDCA_CTL_SELECTED_MODE, 0), det_mode); + if (ret < 0) + goto io_error; + } + + return jack_type; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_headset_detect); + +/** + * rt_sdca_button_detect - Read UMP message and decide button type. + * + * @map: map for setting. + * @entity_id: SDCA entity ID. + * @hid_buf_addr: HID buffer address. + * @hid_id: Report ID for HID. + * + * A button type will be returned regarding to buffer, + * it returns zero if buffer cannot be recognized. + */ +int rt_sdca_button_detect(struct regmap *map, unsigned int entity_id, + unsigned int hid_buf_addr, unsigned int hid_id) +{ + unsigned int btn_type = 0, offset, idx, val, owner; + unsigned char buf[3]; + int ret; + + /* get current UMP message owner */ + ret = regmap_read(map, SDW_SDCA_CTL(SDCA_NUM_HID, entity_id, + RT_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), &owner); + if (ret < 0) + return 0; + + /* if owner is device then there is no button event from device */ + if (owner == 1) + return 0; + + /* read UMP message offset */ + ret = regmap_read(map, SDW_SDCA_CTL(SDCA_NUM_HID, entity_id, + RT_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), &offset); + if (ret < 0) + goto _end_btn_det_; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(map, hid_buf_addr + offset + idx, &val); + if (ret < 0) + goto _end_btn_det_; + buf[idx] = val & 0xff; + } + /* Report ID for HID */ + if (buf[0] == hid_id) + btn_type = rt_sdca_btn_type(&buf[1]); + +_end_btn_det_: + /* Host is owner, so set back to device */ + if (owner == 0) + /* set owner to device */ + regmap_write(map, + SDW_SDCA_CTL(SDCA_NUM_HID, entity_id, + RT_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), 0x01); + + return btn_type; +} +EXPORT_SYMBOL_GPL(rt_sdca_button_detect); + +MODULE_DESCRIPTION("Realtek soundwire common functions"); +MODULE_AUTHOR("jack yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt-sdw-common.h b/sound/soc/codecs/rt-sdw-common.h new file mode 100644 index 000000000000..4759516feb38 --- /dev/null +++ b/sound/soc/codecs/rt-sdw-common.h @@ -0,0 +1,66 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +// +// rt-sdw-common.h +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// + +/* + * This file defines common functions used with Realtek soundwire codecs. + */ + +#ifndef __RT_SDW_COMMON_H__ +#define __RT_SDW_COMMON_H__ + +#define SDCA_NUM_JACK_CODEC 0x01 +#define SDCA_NUM_MIC_ARRAY 0x02 +#define SDCA_NUM_HID 0x03 +#define SDCA_NUM_AMP 0x04 +#define RT_SDCA_CTL_SELECTED_MODE 0x01 +#define RT_SDCA_CTL_DETECTED_MODE 0x02 +#define RT_SDCA_CTL_HIDTX_CURRENT_OWNER 0x10 +#define RT_SDCA_CTL_HIDTX_MESSAGE_OFFSET 0x12 + +struct rt_sdca_dmic_kctrl_priv { + unsigned int reg_base; + unsigned int count; + unsigned int max; + unsigned int invert; +}; + +#define RT_SDCA_PR_VALUE(xreg_base, xcount, xmax, xinvert) \ + ((unsigned long)&(struct rt_sdca_dmic_kctrl_priv) \ + {.reg_base = xreg_base, .count = xcount, .max = xmax, \ + .invert = xinvert}) + +#define RT_SDCA_FU_CTRL(xname, reg_base, xmax, xinvert, xcount, \ + xinfo, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = xinfo, \ + .get = xget, \ + .put = xput, \ + .private_value = RT_SDCA_PR_VALUE(reg_base, xcount, xmax, xinvert)} + +#define RT_SDCA_EXT_TLV(xname, reg_base, xhandler_get,\ + xhandler_put, xcount, xmax, tlv_array, xinfo) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = xinfo, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = RT_SDCA_PR_VALUE(reg_base, xcount, xmax, 0) } + + +int rt_sdca_index_write(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int value); +int rt_sdca_index_read(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int *value); +int rt_sdca_index_update_bits(struct regmap *map, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val); +int rt_sdca_btn_type(unsigned char *buffer); +int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id); +int rt_sdca_button_detect(struct regmap *map, unsigned int entity_id, + unsigned int hid_buf_addr, unsigned int hid_id); + +#endif /* __RT_SDW_COMMON_H__ */ -- 2.34.1

3 5

[PATCH 0/2] ASoC: imx-card: add cs42888 codec support
by Shengjiu Wang 12 Oct '24

12 Oct '24

add cs42888 codec support Chancel Liu (2): ASoC: imx-card: Set mclk for codec ASoC: imx-card: Add CS42888 support sound/soc/fsl/imx-card.c | 59 +++++++++++++++++++++++++++++++++++----- 1 file changed, 52 insertions(+), 7 deletions(-) -- 2.34.1

3 4

[PATCH AUTOSEL 6.11 12/16] ASoC: qcom: sm8250: add qrb4210-rb2-sndcard compatible string
by Sasha Levin 12 Oct '24

12 Oct '24

From: Alexey Klimov <alexey.klimov(a)linaro.org> [ Upstream commit b97bc0656a66f89f78098d4d72dc04fa9518ab11 ] Add "qcom,qrb4210-rb2-sndcard" to the list of recognizable devices. Signed-off-by: Alexey Klimov <alexey.klimov(a)linaro.org> Link: https://patch.msgid.link/20241002022015.867031-3-alexey.klimov@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/sm8250.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index a15dafb99b337..50e175fd521ce 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -166,6 +166,7 @@ static int sm8250_platform_probe(struct platform_device *pdev) static const struct of_device_id snd_sm8250_dt_match[] = { {.compatible = "qcom,sm8250-sndcard"}, + {.compatible = "qcom,qrb4210-rb2-sndcard"}, {.compatible = "qcom,qrb5165-rb5-sndcard"}, {} }; -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 04/16] ASoC: fsl_sai: Enable 'FIFO continue on error' FCONT bit
by Sasha Levin 12 Oct '24

12 Oct '24

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 72455e33173c1a00c0ce93d2b0198eb45d5f4195 ] FCONT=1 means On FIFO error, the SAI will continue from the same word that caused the FIFO error to set after the FIFO warning flag has been cleared. Set FCONT bit in control register to avoid the channel swap issue after SAI xrun. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://patch.msgid.link/1727676508-22830-1-git-send-email-shengjiu.wang@nx… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_sai.c | 5 ++++- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d03b0172b8ad2..a1f03c97b7bb8 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -613,6 +613,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* Set to avoid channel swap */ + val_cr4 |= FSL_SAI_CR4_FCONT; + /* Set to output mode to avoid tri-stated data pins */ if (tx) val_cr4 |= FSL_SAI_CR4_CHMOD; @@ -699,7 +702,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | - FSL_SAI_CR4_CHMOD_MASK, + FSL_SAI_CR4_CHMOD_MASK | FSL_SAI_CR4_FCONT_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index dadbd16ee3945..9c4d19fe22c65 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -137,6 +137,7 @@ /* SAI Transmit and Receive Configuration 4 Register */ +#define FSL_SAI_CR4_FCONT_MASK BIT(28) #define FSL_SAI_CR4_FCONT BIT(28) #define FSL_SAI_CR4_FCOMB_SHIFT BIT(26) #define FSL_SAI_CR4_FCOMB_SOFT BIT(27) -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 03/16] ASoC: codecs: lpass-rx-macro: add missing CDC_RX_BCL_VBAT_RF_PROC2 to default regs values
by Sasha Levin 12 Oct '24

12 Oct '24

From: Alexey Klimov <alexey.klimov(a)linaro.org> [ Upstream commit e249786b2188107a7c50e7174d35f955a60988a1 ] CDC_RX_BCL_VBAT_RF_PROC1 is listed twice and its default value is 0x2a which is overwriten by its next occurence in rx_defaults[]. The second one should be missing CDC_RX_BCL_VBAT_RF_PROC2 instead and its default value is expected 0x0. Signed-off-by: Alexey Klimov <alexey.klimov(a)linaro.org> Link: https://patch.msgid.link/20240925043823.520218-2-alexey.klimov@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ce42749660c87..ac759f4a880d0 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -958,7 +958,7 @@ static const struct reg_default rx_defaults[] = { { CDC_RX_BCL_VBAT_PK_EST2, 0x01 }, { CDC_RX_BCL_VBAT_PK_EST3, 0x40 }, { CDC_RX_BCL_VBAT_RF_PROC1, 0x2A }, - { CDC_RX_BCL_VBAT_RF_PROC1, 0x00 }, + { CDC_RX_BCL_VBAT_RF_PROC2, 0x00 }, { CDC_RX_BCL_VBAT_TAC1, 0x00 }, { CDC_RX_BCL_VBAT_TAC2, 0x18 }, { CDC_RX_BCL_VBAT_TAC3, 0x18 }, -- 2.43.0

1 0

[PATCH] ASoC: fsl_esai: change dev_warn to dev_dbg in irq handler
by Shengjiu Wang 11 Oct '24

11 Oct '24

Irq handler need to be executed as fast as possible, so the log in irq handler is better to use dev_dbg which needs to be enabled when debugging. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index a65f5b9935a2..0b247f16a163 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -119,10 +119,10 @@ static irqreturn_t esai_isr(int irq, void *devid) dev_dbg(&pdev->dev, "isr: Transmission Initialized\n"); if (esr & ESAI_ESR_RFF_MASK) - dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + dev_dbg(&pdev->dev, "isr: Receiving overrun\n"); if (esr & ESAI_ESR_TFE_MASK) - dev_warn(&pdev->dev, "isr: Transmission underrun\n"); + dev_dbg(&pdev->dev, "isr: Transmission underrun\n"); if (esr & ESAI_ESR_TLS_MASK) dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); -- 2.34.1

3 2

[PATCH v1] ALSA: hda/tas2781: Add speaker id check for ASUS projects
by Baojun Xu 11 Oct '24

11 Oct '24

Add speaker id check by gpio in ACPI for ASUS projects. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781.h | 3 +++ sound/pci/hda/tas2781_hda_i2c.c | 39 ++++++++++++++++++++++++++++++--- 2 files changed, 39 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 8cd6da0480b7..f8a3dc13cbbc 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -107,6 +107,8 @@ #define TASDEVICE_CMD_DELAY 0x3 #define TASDEVICE_CMD_FIELD_W 0x4 +#define TAS2781_ASUS_ID 0x10430000 + enum audio_device { TAS2563, TAS2781, @@ -156,6 +158,7 @@ struct tasdevice_priv { struct tasdevice_rca rcabin; struct calidata cali_data; struct tasdevice_fw *fmw; + struct gpio_desc *speaker_id; struct gpio_desc *reset; struct mutex codec_lock; struct regmap *regmap; diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 370d847517f9..1f71927825b2 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -615,7 +615,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) struct tasdevice_priv *tas_priv = context; struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); struct hda_codec *codec = tas_priv->codec; - int i, ret; + int i, ret, spk_id; pm_runtime_get_sync(tas_priv->dev); mutex_lock(&tas_priv->codec_lock); @@ -648,8 +648,23 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) tasdevice_dsp_remove(tas_priv); tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; - scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", - codec->core.subsystem_id & 0xffff); + if ((codec->core.subsystem_id & 0xffff0000) == TAS2781_ASUS_ID) { + // Speaker id need to be checked. + if (tas_priv->speaker_id) + spk_id = gpiod_get_value(tas_priv->speaker_id); + else + spk_id = 0; + if (spk_id < 0 || spk_id > 1) { + // Speaker id is not valid, use default. + dev_dbg(tas_priv->dev, "Wrong spk_id = %d\n", spk_id); + spk_id = 0; + } + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X%01d.bin", + codec->core.subsystem_id & 0xffff, spk_id); + } else { + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", + codec->core.subsystem_id & 0xffff); + } ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", @@ -787,6 +802,15 @@ static void tas2781_hda_remove(struct device *dev) tasdevice_remove(tas_hda->priv); } +static const struct acpi_gpio_params speakerid_gpios = { 0, 0, false }; +static const struct acpi_gpio_params interrupt_gpios = { 1, 0, false }; + +static const struct acpi_gpio_mapping tas2781_speaker_id_gpios[] = { + { "speakerid-gpios", &speakerid_gpios, 1 }, + { "interrupt-gpios", &interrupt_gpios, 1 }, + { } +}; + static int tas2781_hda_i2c_probe(struct i2c_client *clt) { struct tas2781_hda *tas_hda; @@ -823,6 +847,15 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) if (ret) return dev_err_probe(tas_hda->dev, ret, "Platform not supported\n"); + ret = devm_acpi_dev_add_driver_gpios(tas_hda->dev, + tas2781_speaker_id_gpios); + if (ret) + dev_info(tas_hda->dev, "Unable to add GPIO mapping table\n"); + + tas_hda->priv->speaker_id = devm_gpiod_get(tas_hda->dev, "speakerid", + GPIOD_IN); + if (IS_ERR(tas_hda->priv->speaker_id)) + dev_info(tas_hda->dev, "Failed to get Speaker id gpio.\n"); ret = tasdevice_init(tas_hda->priv); if (ret) -- 2.43.0

3 3

[PATCH] ALSA: hda/realtek: tas2781: Fix ROG ALLY X audio
by Jonathan LoBue 11 Oct '24

11 Oct '24

This patch enables the TI TAS2781 amplifier SoC for the ASUS ROG ALLY X. This is a design change from the original ASUS ROG ALLY, creating the need for this patch. All other Realtek Codec settings seem to be re-used from the original ROG ALLY design (on the ROG ALLY X). This patch maintains the previous settings for the Realtek codec portion, but enables the I2C binding for the TI TAS2781 amplifier (instead of the Cirrus CS35L41 amp used on the original ASUS ROG ALLY). One other requirement must be met for audio to work on the ASUS ROG ALLY X. A proper firmware file in the correct location with a proper symlink. We had reached out to TI engineers and confirmed that the firmware found in the Windows' driver package has a GPL license. Bazzite Github is hosting this firmware file for now until proper linux-firmware upstreaming can occur. https://github.com/ublue-os/bazzite This firmware file should be placed in /usr/lib/firmware/ti/tas2781/TAS2XXX1EB3.bin with a symlink to it from /usr/lib/firmware/TAS2XXX1EB3.bin Co-developed by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Signed-off-by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Co-developed by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Antheas Kapenekakis <lkml(a)antheas.dev> Tested-by: Richard Alvarez <alvarez.richard(a)gmail.com> Tested-by: Miles Montierth <cyber_dopamine(a)intheblackmedia.com> Signed-off-by: Jonathan LoBue <jlobue10(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 480e82df7a4c..9bc39c2ee6b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7410,6 +7410,7 @@ enum { ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_ALLY, + ALC294_FIXUP_ASUS_ALLY_X, ALC294_FIXUP_ASUS_ALLY_PINS, ALC294_FIXUP_ASUS_ALLY_VERBS, ALC294_FIXUP_ASUS_ALLY_SPEAKER, @@ -8877,6 +8878,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS }, + [ALC294_FIXUP_ASUS_ALLY_X] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS + }, [ALC294_FIXUP_ASUS_ALLY_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10313,6 +10320,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY), + SND_PCI_QUIRK(0x1043, 0x1eb3, "ROG Ally X RC72LA", ALC294_FIXUP_ASUS_ALLY_X), SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404VI/VV", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.46.0

4 5

[PATCH 6.1] ASoC: Intel: sof_realtek_common: set ret = 0 as initial value
by Vitaliy Shevtsov 10 Oct '24

10 Oct '24

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> commit 47d2b66fec133cb27da3a551334686e465d19469 upstream. 'ret' will not be initialized if dai_fmt is not DSP_A or DSP_B. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20221206212507.359993-4-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Vitaliy Shevtsov <v.shevtsov(a)maxima.ru> --- sound/soc/intel/boards/sof_realtek_common.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index ff2851fc8930..6c12ca92f371 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -267,7 +267,8 @@ static int rt1015_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *codec_dai; - int i, clk_freq, ret; + int i, clk_freq; + int ret = 0; clk_freq = sof_dai_get_bclk(rtd); -- 2.46.2

1 0

Add Toradex Hardware Configuration Files
by GitHub pull_request - opened 10 Oct '24

10 Oct '24

alsa-project/alsa-ucm-conf pull request #448 was opened from hiagofranco: Add necessary configuration files related to Toradex Hardware. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/448 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/448.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add stacked and parallel memories support in spi-nor
by Mahapatra, Amit Kumar 10 Oct '24

10 Oct '24

Hello Everyone, As https://lore.kernel.org/all/20231125092137.2948-8-amit.kumar-mahapatra@amd.… thread discussing the addition of stacked and parallel mode support in SPI-NOR has become too long and difficult to follow. Therefore, I'm starting a new email thread to focus on the DT binding changes and the overall implementation architecture for adding stacked and parallel (multi-CS) support in the Kernel. Please share your input on this approach. Background: The AMD QSPI controller supports two advanced connection modes(Stacked and Dual Parallel) which allow the controller to treat two different flashes as one storage. Stacked: Flashes share the same SPI bus, but different CS line, controller driver asserts the CS of the flash to which it needs to communicate. Stacked mode is a software abstraction rather than a controller feature or capability. At any given time, the controller communicates with one of the two connected flash devices, as determined by the requested address and data length. If an operation starts on one flash and ends on the other, the core needs to split it into two separate operations and adjust the data length accordingly. Dual Parallel: Both the flashes have their separate SPI bus, CS of both the flashes will be asserted/de-asserted at the same time. In this mode data will be split across both the flashes by enabling the STRIPE setting in the controller. Parallel mode is a controller feature where if the STRIPE bit is set then the controller internally handles the data split during data write to the flashes and while reading data from the flash the controller internally merges data from both the flashes before writing to the controller FIFO. If STRIPE is not enabled, then same data will be sent to both the devices. In parallel mode both the flashes should be identical. For more information on the modes please feel free to go through the controller flash interface below https://docs.amd.com/r/en-US/am011-versal-acap-trm/QSPI-Flash-Device-Interf… Mirochip QSPI controller also supports "Dual Parallel 8-bit IO mode", but they call it "Twin Quad Mode". https://ww1.microchip.com/downloads/aemDocuments/documents/MPU32/ProductDoc… Initially in [1] [2] [3] Miquel had tried to extend MTD-CONCAT driver to support Stacked mode, but the bindings were not accepted. [1] https://lore.kernel.org/all/20191113171505.26128-4-miquel.raynal@bootlin.co… [2] https://lore.kernel.org/all/20191127105522.31445-5-miquel.raynal@bootlin.co… [3]https://lore.kernel.org/all/20211112152411.818321-1-miquel.raynal@bootlin… So, the MTD-CONCAT approach was dropped and the DT bindings that got accepted [4] [5] [6] describe the two flash devices as being one. [4] https://github.com/torvalds/linux/commit/f89504300e94524d5d5846ff8b728592ac… [5] https://github.com/torvalds/linux/commit/eba5368503b4291db7819512600fa014ea… [6] https://github.com/torvalds/linux/commit/e2edd1b64f1c79e8abda365149ed62a2a9… SPI core changes to support the above bindings were added through the following commit: https://github.com/torvalds/linux/commit/4d8ff6b0991d5e86b17b235fc46ec62e91… While adding the support in SPI-NOR we received some inputs/suggestions from Tudor, based on that I am planning to add a new layer between the SPI-NOR and MTD layers to support stacked and parallel configurations. This new layer will be part of the spi-nor and located in mtd/spi-nor/. This layer would perform the following tasks: - During probing, store information from all the connected flashes, whether in stacked or parallel mode, and present it as a single device to the MTD layer. - Register callbacks and manage MTD device registration within the new layer instead of in spi-nor/core.c - There will very minimal change in spinor/core.c as the stacked & parallel handling will be managed by the new layer on top of SPI-NOR. - In stacked mode, the new layer will select the appropriate spi-nor flash based on the address provided by the MTD layer during flash operations. - Manage flash crossover operations in stacked mode. - Handle odd byte count requests from the MTD layer during flash operations in parallel mode. For implementing this the current DT binding[1] [2] [3] need to be updated as follows. stacked-memories DT changes: - Flash size information can be retrieved directly from the flash, so it has been removed from the DT binding. - Each stacked flash will have its own flash node. This approach allows flashes of different makes and sizes to be stacked together, as each flash will be probed individually. - Each of the flash node will have its own "reg" property that will contain its physical CS. - The stacked-memories DT bindings will contain the phandles of the flash nodes connected in stacked mode. - The first flash node will contain the mtd partition that would have the cross over memory staring at a memory location in the first flash and ending at some memory location of the 2nd flash - The new layer will update the mtd->size and other mtd_info parameters after both the flashes are probed and will call mtd_device_register with the combined information. spi@0 { flash@0 { compatible = "jedec,spi-nor" reg = <0x00>; stacked-memories = <&flash@0 &flash@1>; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-0"; reg = <0x0 0xf00000>; }; } flash@1 { compatible = "jedec,spi-nor" reg = <0x01>; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-1"; reg = <0x0 0x800000>; }; } } parallel-memories DT changes: - Flash size information can be retrieved directly from the flash, so it has been removed from the DT binding. - Each flash connected in parallel mode should be identical and will have one flash node for both the flash devices. - The "reg" prop will contain the physical CS number for both the connected flashes. - The new layer will double the mtd-> size and register it with the mtd layer. spi@0 { flash@0 { compatible = "jedec,spi-nor" reg = <0x00 0x01>; parallel-memories; spi-max-frequency = <50000000>; ... partition@0 { label = "qspi-0"; reg = <0x0 0xf00000>; } }; } Regards, Amit

2 6

[PATCH 1/2] ASoC: Intel: Add rt721-sdca support for PTL platform
by Jack Yu 10 Oct '24

10 Oct '24

Add rt721-sdca support for PTL platform. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/intel/boards/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index cc10ae58b0c7..9b80b19bb8d0 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -519,6 +519,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_RT712_SDCA_DMIC_SDW select SND_SOC_RT715_SDW select SND_SOC_RT715_SDCA_SDW + select SND_SOC_RT721_SDCA_SDW select SND_SOC_RT722_SDCA_SDW select SND_SOC_RT1308_SDW select SND_SOC_RT1308 -- 2.34.1

3 2

[PATCH 6.1/6.6] ASoC: SOF: Fix runtime pm usage counter balance after fw exception
by Vitaliy Shevtsov 10 Oct '24

10 Oct '24

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> commit b30289e7fa927f921bfb4d0d04727461706ae822 upstream. If the retain context is enabled we will unconditionally increment the device's pm use count on each exception and when the drivers are unloaded we do not correct this (as we don't know how many times we 'prevented d3 entry'). Introduce a flag to make sure that we do not increment the use count more than once and on module unload decrement the use count if needed to balance it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240213114729.7055-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Vitaliy Shevtsov <v.shevtsov(a)maxima.ru> --- sound/soc/sof/core.c | 10 ++++++++++ sound/soc/sof/debug.c | 8 +++++--- sound/soc/sof/sof-priv.h | 1 + 3 files changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 0938b259f703..cba0af5b86aa 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -477,6 +477,16 @@ int snd_sof_device_remove(struct device *dev) */ snd_sof_machine_unregister(sdev, pdata); + /* + * Balance the runtime pm usage count in case we are faced with an + * exception and we forcably prevented D3 power state to preserve + * context + */ + if (sdev->d3_prevented) { + sdev->d3_prevented = false; + pm_runtime_put_noidle(sdev->dev); + } + if (sdev->fw_state > SOF_FW_BOOT_NOT_STARTED) { sof_fw_trace_free(sdev); ret = snd_sof_dsp_power_down_notify(sdev); diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index d547318e0d32..7c8aafca8fde 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -433,13 +433,15 @@ static void snd_sof_ipc_dump(struct snd_sof_dev *sdev) void snd_sof_handle_fw_exception(struct snd_sof_dev *sdev, const char *msg) { - if (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_RETAIN_DSP_CONTEXT) || - sof_debug_check_flag(SOF_DBG_RETAIN_CTX)) { + if ((IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_RETAIN_DSP_CONTEXT) || + sof_debug_check_flag(SOF_DBG_RETAIN_CTX)) && !sdev->d3_prevented) { /* should we prevent DSP entering D3 ? */ if (!sdev->ipc_dump_printed) dev_info(sdev->dev, "Attempting to prevent DSP from entering D3 state to preserve context\n"); - pm_runtime_get_if_in_use(sdev->dev); + + if (pm_runtime_get_if_in_use(sdev->dev) == 1) + sdev->d3_prevented = true; } /* dump vital information to the logs */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index d4f6702e93dc..b27a5f9e1bc8 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -593,6 +593,7 @@ struct snd_sof_dev { struct list_head dfsentry_list; bool dbg_dump_printed; bool ipc_dump_printed; + bool d3_prevented; /* runtime pm use count incremented to prevent context lost */ /* firmware loader */ struct sof_ipc_fw_ready fw_ready; -- 2.46.2

1 0

[PATCH 2/2] ASoC: rt721-sdca: Fix issue of warning message
by Jack Yu 10 Oct '24

10 Oct '24

Fix issue of warning messages caused by some variables. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt-sdw-common.c | 2 +- sound/soc/codecs/rt721-sdca.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt-sdw-common.c b/sound/soc/codecs/rt-sdw-common.c index 9ed0e9855699..a422da6cf702 100644 --- a/sound/soc/codecs/rt-sdw-common.c +++ b/sound/soc/codecs/rt-sdw-common.c @@ -108,7 +108,7 @@ EXPORT_SYMBOL_GPL(rt_sdca_index_update_bits); int rt_sdca_btn_type(unsigned char *buffer) { u8 btn_type = 0; - int ret; + int ret = 0; btn_type |= buffer[0] & 0xf; btn_type |= (buffer[0] >> 4) & 0xf; diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c index 36056cb7a3ca..201cb667c8c1 100644 --- a/sound/soc/codecs/rt721-sdca.c +++ b/sound/soc/codecs/rt721-sdca.c @@ -30,7 +30,7 @@ static void rt721_sdca_jack_detect_handler(struct work_struct *work) { struct rt721_sdca_priv *rt721 = container_of(work, struct rt721_sdca_priv, jack_detect_work.work); - int btn_type = 0, ret; + int btn_type = 0; if (!rt721->hs_jack) return; @@ -42,7 +42,7 @@ static void rt721_sdca_jack_detect_handler(struct work_struct *work) if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { rt721->jack_type = rt_sdca_headset_detect(rt721->regmap, RT721_SDCA_ENT_GE49); - if (ret < 0) + if (rt721->jack_type < 0) return; } -- 2.34.1

2 1

[PATCH 5.15.y 0/1] Fix of CVE-2023-52904 for stable-5.15
by Harshvardhan Jha 09 Oct '24

09 Oct '24

Folowing is a backport of commit a474d4ad59cd ("ALSA: usb-audio: Fix possible NULL pointer dereference in snd_usb_pcm_has_fixed_rate()") to stable-5.15 branch which fixes broken commit bfd36b1d1869 ("ALSA: usb-audio: Always initialize fixed_rate in snd_usb_find_implicit_fb_sync_format()"). This was a clean pick and shouldn't be break anything. Basic smoke tests were performed on this patch. Jaroslav Kysela (1): ALSA: usb-audio: Fix possible NULL pointer dereference in snd_usb_pcm_has_fixed_rate() sound/usb/pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) -- 2.46.0

1 1

[PATCH v3] ASoC: rt721-sdca: Add RT721 SDCA driver
by Jack Yu 08 Oct '24

08 Oct '24

This is the initial codec driver for rt721-sdca. It's a three functions (jack,mic,amp) soundwire driver. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> v2: Fix typo in mbq default registers. v3: Include soundwire common functions for Realtek. --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt721-sdca-sdw.c | 551 ++++++++++ sound/soc/codecs/rt721-sdca-sdw.h | 150 +++ sound/soc/codecs/rt721-sdca.c | 1547 +++++++++++++++++++++++++++++ sound/soc/codecs/rt721-sdca.h | 268 +++++ 6 files changed, 2525 insertions(+) create mode 100644 sound/soc/codecs/rt721-sdca-sdw.c create mode 100644 sound/soc/codecs/rt721-sdca-sdw.h create mode 100644 sound/soc/codecs/rt721-sdca.c create mode 100644 sound/soc/codecs/rt721-sdca.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7092842480ef..74c14da4ae09 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -222,6 +222,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT712_SDCA_DMIC_SDW imply SND_SOC_RT715_SDW imply SND_SOC_RT715_SDCA_SDW + imply SND_SOC_RT721_SDCA_SDW imply SND_SOC_RT722_SDCA_SDW imply SND_SOC_RT1308_SDW imply SND_SOC_RT1316_SDW @@ -1743,6 +1744,12 @@ config SND_SOC_RT712_SDCA_DMIC_SDW select REGMAP_SOUNDWIRE select REGMAP_SOUNDWIRE_MBQ +config SND_SOC_RT721_SDCA_SDW + tristate "Realtek RT721 SDCA Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + select REGMAP_SOUNDWIRE_MBQ + config SND_SOC_RT722_SDCA_SDW tristate "Realtek RT722 SDCA Codec - SDW" depends on SOUNDWIRE diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 54cbc3feae32..3477c6352acc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -259,6 +259,7 @@ snd-soc-rt712-sdca-y := rt712-sdca.o rt712-sdca-sdw.o snd-soc-rt712-sdca-dmic-y := rt712-sdca-dmic.o snd-soc-rt715-y := rt715.o rt715-sdw.o snd-soc-rt715-sdca-y := rt715-sdca.o rt715-sdca-sdw.o +snd-soc-rt721-sdca-y := rt721-sdca.o rt721-sdca-sdw.o snd-soc-rt722-sdca-y := rt722-sdca.o rt722-sdca-sdw.o snd-soc-rt9120-y := rt9120.o snd-soc-rtq9128-y := rtq9128.o @@ -662,6 +663,7 @@ obj-$(CONFIG_SND_SOC_RT712_SDCA_SDW) += snd-soc-rt712-sdca.o obj-$(CONFIG_SND_SOC_RT712_SDCA_DMIC_SDW) += snd-soc-rt712-sdca-dmic.o obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o obj-$(CONFIG_SND_SOC_RT715_SDCA_SDW) += snd-soc-rt715-sdca.o +obj-$(CONFIG_SND_SOC_RT721_SDCA_SDW) += snd-soc-rt721-sdca.o obj-$(CONFIG_SND_SOC_RT722_SDCA_SDW) += snd-soc-rt722-sdca.o obj-$(CONFIG_SND_SOC_RT9120) += snd-soc-rt9120.o obj-$(CONFIG_SND_SOC_RTQ9128) += snd-soc-rtq9128.o diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c new file mode 100644 index 000000000000..c0f8cccae3b2 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca-sdw.c @@ -0,0 +1,551 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt721-sdca-sdw.c -- rt721 SDCA ALSA SoC audio driver +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/pm_runtime.h> +#include <linux/soundwire/sdw_registers.h> + +#include "rt721-sdca.h" +#include "rt721-sdca-sdw.h" +#include "rt-sdw-common.h" + +static bool rt721_sdca_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f01 ... 0x2f0a: + case 0x2f35: + case 0x2f50: + case 0x2f51: + case 0x2f58 ... 0x2f5d: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XUV, + RT721_SDCA_CTL_XUV, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_SELECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ... SDW_SDCA_CTL(FUNC_NUM_HID, + RT721_SDCA_ENT_HID01, RT721_SDCA_CTL_HIDTX_MESSAGE_LENGTH, 0): + case RT721_BUF_ADDR_HID1 ... RT721_BUF_ADDR_HID2: + return true; + default: + return false; + } +} + +static bool rt721_sdca_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f01: + case 0x2f51: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XUV, + RT721_SDCA_CTL_XUV, 0): + case SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ... SDW_SDCA_CTL(FUNC_NUM_HID, + RT721_SDCA_ENT_HID01, RT721_SDCA_CTL_HIDTX_MESSAGE_LENGTH, 0): + case RT721_BUF_ADDR_HID1 ... RT721_BUF_ADDR_HID2: + return true; + default: + return false; + } +} + +static bool rt721_sdca_mbq_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0900007: + case 0x0a00005: + case 0x0c00005: + case 0x0d00014: + case 0x0310100: + case 0x2000001: + case 0x2000002: + case 0x2000003: + case 0x2000013: + case 0x200003c: + case 0x2000046: + case 0x5810000: + case 0x5810036: + case 0x5810037: + case 0x5810038: + case 0x5810039: + case 0x5b10018: + case 0x5b10019: + case 0x5f00045: + case 0x5f00048: + case 0x6100000: + case 0x6100005: + case 0x6100006: + case 0x610000d: + case 0x6100010: + case 0x6100011: + case 0x6100013: + case 0x6100015: + case 0x6100017: + case 0x6100025: + case 0x6100029: + case 0x610002c ... 0x610002f: + case 0x6100053 ... 0x6100055: + case 0x6100057: + case 0x610005a: + case 0x610005b: + case 0x610006a: + case 0x610006d: + case 0x6100092: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_L): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_R): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_L): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_R): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_01): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_02): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_03): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_04): + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_R): + return true; + default: + return false; + } +} + +static bool rt721_sdca_mbq_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0310100: + case 0x0a00005: + case 0x0c00005: + case 0x0d00014: + case 0x2000000: + case 0x200000d: + case 0x2000019: + case 0x2000020: + case 0x2000030: + case 0x2000046: + case 0x2000067: + case 0x2000084: + case 0x2000086: + case 0x5810000: + case 0x5810036: + case 0x5810037: + case 0x5810038: + case 0x5810039: + case 0x5b10018: + case 0x5b10019: + return true; + default: + return false; + } +} + +static const struct regmap_config rt721_sdca_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt721_sdca_readable_register, + .volatile_reg = rt721_sdca_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = rt721_sdca_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt721_sdca_reg_defaults), + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct regmap_config rt721_sdca_mbq_regmap = { + .name = "sdw-mbq", + .reg_bits = 32, + .val_bits = 16, + .readable_reg = rt721_sdca_mbq_readable_register, + .volatile_reg = rt721_sdca_mbq_volatile_register, + .max_register = 0x41000312, + .reg_defaults = rt721_sdca_mbq_defaults, + .num_reg_defaults = ARRAY_SIZE(rt721_sdca_mbq_defaults), + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt721_sdca_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_UNATTACHED) + rt721->hw_init = false; + + if (status == SDW_SLAVE_ATTACHED) { + if (rt721->hs_jack) { + /* + * Due to the SCP_SDCA_INTMASK will be cleared by any reset, and then + * if the device attached again, we will need to set the setting back. + * It could avoid losing the jack detection interrupt. + * This also could sync with the cache value as the rt721_sdca_jack_init set. + */ + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + } + } + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt721->hw_init || status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt721_sdca_io_init(&slave->dev, slave); +} + +static int rt721_sdca_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + sdw_slave_read_prop(slave); + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* + * port = 1 for headphone playback + * port = 2 for headset-mic capture + * port = 3 for speaker playback + * port = 6 for digital-mic capture + */ + prop->source_ports = BIT(6) | BIT(2); /* BITMAP: 01000100 */ + prop->sink_ports = BIT(3) | BIT(1); /* BITMAP: 00001010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 900; + + /* wake-up event */ + prop->wake_capable = 1; + + /* Three data lanes are supported by rt721-sdca codec */ + prop->lane_control_support = true; + + return 0; +} + +static int rt721_sdca_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(&slave->dev); + int ret, stat; + int count = 0, retry = 3; + unsigned int sdca_cascade, scp_sdca_stat1, scp_sdca_stat2 = 0; + + if (cancel_delayed_work_sync(&rt721->jack_detect_work)) { + dev_warn(&slave->dev, "%s the pending delayed_work was cancelled", __func__); + /* avoid the HID owner doesn't change to device */ + if (rt721->scp_sdca_stat2) + scp_sdca_stat2 = rt721->scp_sdca_stat2; + } + + /* + * The critical section below intentionally protects a rather large piece of code. + * We don't want to allow the system suspend to disable an interrupt while we are + * processing it, which could be problematic given the quirky SoundWire interrupt + * scheme. We do want however to prevent new workqueues from being scheduled if + * the disable_irq flag was set during system suspend. + */ + mutex_lock(&rt721->disable_irq_lock); + + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + + rt721->scp_sdca_stat1 = ret; + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + + rt721->scp_sdca_stat2 = ret; + if (scp_sdca_stat2) + rt721->scp_sdca_stat2 |= scp_sdca_stat2; + do { + /* clear flag */ + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + if (ret & SDW_SCP_SDCA_INTMASK_SDCA_0) { + ret = sdw_update_no_pm(rt721->slave, SDW_SCP_SDCA_INT1, + SDW_SCP_SDCA_INT_SDCA_0, SDW_SCP_SDCA_INT_SDCA_0); + if (ret < 0) + goto io_error; + } else if (ret & SDW_SCP_SDCA_INTMASK_SDCA_6) { + ret = sdw_update_no_pm(rt721->slave, SDW_SCP_SDCA_INT1, + SDW_SCP_SDCA_INT_SDCA_6, SDW_SCP_SDCA_INT_SDCA_6); + if (ret < 0) + goto io_error; + } + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + if (ret & SDW_SCP_SDCA_INTMASK_SDCA_8) { + ret = sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INT2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + if (ret < 0) + goto io_error; + } + + /* check if flag clear or not */ + ret = sdw_read_no_pm(rt721->slave, SDW_DP0_INT); + if (ret < 0) + goto io_error; + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + scp_sdca_stat1 = ret & SDW_SCP_SDCA_INTMASK_SDCA_0; + + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + scp_sdca_stat2 = ret & SDW_SCP_SDCA_INTMASK_SDCA_8; + + stat = scp_sdca_stat1 || scp_sdca_stat2 || sdca_cascade; + + count++; + } while (stat != 0 && count < retry); + + if (stat) + dev_warn(&slave->dev, + "%s scp_sdca_stat1=0x%x, scp_sdca_stat2=0x%x\n", __func__, + rt721->scp_sdca_stat1, rt721->scp_sdca_stat2); + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + + if (status->sdca_cascade && !rt721->disable_irq) + mod_delayed_work(system_power_efficient_wq, + &rt721->jack_detect_work, msecs_to_jiffies(280)); + + mutex_unlock(&rt721->disable_irq_lock); + + return 0; + +io_error: + mutex_unlock(&rt721->disable_irq_lock); + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} + +static const struct sdw_slave_ops rt721_sdca_slave_ops = { + .read_prop = rt721_sdca_read_prop, + .interrupt_callback = rt721_sdca_interrupt_callback, + .update_status = rt721_sdca_update_status, +}; + +static int rt721_sdca_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap, *mbq_regmap; + + /* Regmap Initialization */ + mbq_regmap = devm_regmap_init_sdw_mbq(slave, &rt721_sdca_mbq_regmap); + if (IS_ERR(mbq_regmap)) + return PTR_ERR(mbq_regmap); + + regmap = devm_regmap_init_sdw(slave, &rt721_sdca_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return rt721_sdca_init(&slave->dev, regmap, mbq_regmap, slave); +} + +static int rt721_sdca_sdw_remove(struct sdw_slave *slave) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(&slave->dev); + + if (rt721->hw_init) { + cancel_delayed_work_sync(&rt721->jack_detect_work); + cancel_delayed_work_sync(&rt721->jack_btn_check_work); + } + + if (rt721->first_hw_init) + pm_runtime_disable(&slave->dev); + + mutex_destroy(&rt721->calibrate_mutex); + mutex_destroy(&rt721->disable_irq_lock); + + return 0; +} + +static const struct sdw_device_id rt721_sdca_id[] = { + SDW_SLAVE_ENTRY_EXT(0x025d, 0x721, 0x3, 0x1, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt721_sdca_id); + +static int __maybe_unused rt721_sdca_dev_suspend(struct device *dev) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); + + if (!rt721->hw_init) + return 0; + + cancel_delayed_work_sync(&rt721->jack_detect_work); + cancel_delayed_work_sync(&rt721->jack_btn_check_work); + + regcache_cache_only(rt721->regmap, true); + regcache_cache_only(rt721->mbq_regmap, true); + + return 0; +} + +static int __maybe_unused rt721_sdca_dev_system_suspend(struct device *dev) +{ + struct rt721_sdca_priv *rt721_sdca = dev_get_drvdata(dev); + struct sdw_slave *slave = dev_to_sdw_dev(dev); + int ret1, ret2; + + if (!rt721_sdca->hw_init) + return 0; + + /* + * prevent new interrupts from being handled after the + * deferred work completes and before the parent disables + * interrupts on the link + */ + mutex_lock(&rt721_sdca->disable_irq_lock); + rt721_sdca->disable_irq = true; + ret1 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6, 0); + ret2 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8, 0); + mutex_unlock(&rt721_sdca->disable_irq_lock); + + if (ret1 < 0 || ret2 < 0) { + /* log but don't prevent suspend from happening */ + dev_dbg(&slave->dev, "%s: could not disable SDCA interrupts\n:", __func__); + } + + return rt721_sdca_dev_suspend(dev); +} + +#define RT721_PROBE_TIMEOUT 5000 + +static int __maybe_unused rt721_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt721->first_hw_init) + return 0; + + if (!slave->unattach_request) { + mutex_lock(&rt721->disable_irq_lock); + if (rt721->disable_irq == true) { + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt721->disable_irq = false; + } + mutex_unlock(&rt721->disable_irq_lock); + goto regmap_sync; + } + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT721_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt721->regmap, false); + regcache_sync(rt721->regmap); + regcache_cache_only(rt721->mbq_regmap, false); + regcache_sync(rt721->mbq_regmap); + return 0; +} + +static const struct dev_pm_ops rt721_sdca_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt721_sdca_dev_system_suspend, rt721_sdca_dev_resume) + SET_RUNTIME_PM_OPS(rt721_sdca_dev_suspend, rt721_sdca_dev_resume, NULL) +}; + +static struct sdw_driver rt721_sdca_sdw_driver = { + .driver = { + .name = "rt721-sdca", + .owner = THIS_MODULE, + .pm = &rt721_sdca_pm, + }, + .probe = rt721_sdca_sdw_probe, + .remove = rt721_sdca_sdw_remove, + .ops = &rt721_sdca_slave_ops, + .id_table = rt721_sdca_id, +}; +module_sdw_driver(rt721_sdca_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT721 SDCA SDW driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt721-sdca-sdw.h b/sound/soc/codecs/rt721-sdca-sdw.h new file mode 100644 index 000000000000..214b31b82583 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca-sdw.h @@ -0,0 +1,150 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt721-sdca-sdw.h -- RT721 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2024 Realtek Semiconductor Corp. + */ + +#ifndef __RT721_SDW_H__ +#define __RT721_SDW_H__ + +#include <linux/regmap.h> +#include <linux/soundwire/sdw_registers.h> + +static const struct reg_default rt721_sdca_reg_defaults[] = { + { 0x202d, 0x00 }, + { 0x2f01, 0x00 }, + { 0x2f02, 0x09 }, + { 0x2f03, 0x08 }, + { 0x2f04, 0x00 }, + { 0x2f05, 0x0e }, + { 0x2f06, 0x01 }, + { 0x2f09, 0x00 }, + { 0x2f0a, 0x00 }, + { 0x2f35, 0x00 }, + { 0x2f50, 0xf0 }, + { 0x2f58, 0x07 }, + { 0x2f59, 0x07 }, + { 0x2f5a, 0x00 }, + { 0x2f5b, 0x07 }, + { 0x2f5c, 0x27 }, + { 0x2f5d, 0x07 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS01, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS11, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE12, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE40, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_03), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_04), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_CS1F, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_IT26, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_PDE2A, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_CS31, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_OT23, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, +}; + +static const struct reg_default rt721_sdca_mbq_defaults[] = { + { 0x0900007, 0xc004 }, + { 0x2000001, 0x0000 }, + { 0x2000002, 0x0000 }, + { 0x2000003, 0x0000 }, + { 0x2000013, 0x8001 }, + { 0x200003c, 0x0000 }, + { 0x2000046, 0x3400 }, + { 0x5f00044, 0x6040 }, + { 0x5f00045, 0x3333 }, + { 0x5f00048, 0x0000 }, + { 0x6100005, 0x0005 }, + { 0x6100006, 0x0000 }, + { 0x610000d, 0x0051 }, + { 0x6100010, 0x0180 }, + { 0x6100011, 0x0000 }, + { 0x6100013, 0x0000 }, + { 0x6100015, 0x0000 }, + { 0x6100017, 0x8049 }, + { 0x6100025, 0x1000 }, + { 0x6100029, 0x0809 }, + { 0x610002c, 0x2828 }, + { 0x610002d, 0x2929 }, + { 0x610002e, 0x3529 }, + { 0x610002f, 0x2901 }, + { 0x6100053, 0x2630 }, + { 0x6100054, 0x2a2a }, + { 0x6100055, 0x152f }, + { 0x6100057, 0x2200 }, + { 0x610005a, 0x2a4b }, + { 0x610005b, 0x2a00 }, + { 0x610006a, 0x0102 }, + { 0x610006d, 0x0102 }, + { 0x6100092, 0x4f61 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_L), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_R), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_L), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_R), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, RT721_SDCA_CTL_FU_CH_GAIN, + CH_L), 0xfe00 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, RT721_SDCA_CTL_FU_CH_GAIN, + CH_R), 0xfe00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_01), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_02), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_03), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_04), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_01), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_02), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_03), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_04), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_L), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_R), + 0x0000 }, +}; + +#endif /* __RT721_SDW_H__ */ diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c new file mode 100644 index 000000000000..36056cb7a3ca --- /dev/null +++ b/sound/soc/codecs/rt721-sdca.c @@ -0,0 +1,1547 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt721-sdca.c -- rt721 SDCA ALSA SoC audio driver +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// +// + +#include <linux/bitops.h> +#include <sound/core.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <sound/initval.h> +#include <sound/jack.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/pcm.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/slab.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "rt721-sdca.h" +#include "rt-sdw-common.h" + +static void rt721_sdca_jack_detect_handler(struct work_struct *work) +{ + struct rt721_sdca_priv *rt721 = + container_of(work, struct rt721_sdca_priv, jack_detect_work.work); + int btn_type = 0, ret; + + if (!rt721->hs_jack) + return; + + if (!rt721->component->card || !rt721->component->card->instantiated) + return; + + /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ + if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { + rt721->jack_type = rt_sdca_headset_detect(rt721->regmap, + RT721_SDCA_ENT_GE49); + if (ret < 0) + return; + } + + /* SDW_SCP_SDCA_INT_SDCA_8 is used for button detection */ + if (rt721->scp_sdca_stat2 & SDW_SCP_SDCA_INT_SDCA_8) + btn_type = rt_sdca_button_detect(rt721->regmap, + RT721_SDCA_ENT_HID01, RT721_BUF_ADDR_HID1, + RT721_SDCA_HID_ID); + + if (rt721->jack_type == 0) + btn_type = 0; + + dev_dbg(&rt721->slave->dev, + "in %s, jack_type=%d\n", __func__, rt721->jack_type); + dev_dbg(&rt721->slave->dev, + "in %s, btn_type=0x%x\n", __func__, btn_type); + dev_dbg(&rt721->slave->dev, + "in %s, scp_sdca_stat1=0x%x, scp_sdca_stat2=0x%x\n", __func__, + rt721->scp_sdca_stat1, rt721->scp_sdca_stat2); + + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt721->jack_btn_check_work, msecs_to_jiffies(200)); + } +} + +static void rt721_sdca_btn_check_handler(struct work_struct *work) +{ + struct rt721_sdca_priv *rt721 = + container_of(work, struct rt721_sdca_priv, jack_btn_check_work.work); + int btn_type = 0, ret, idx; + unsigned int det_mode, offset, val; + unsigned char buf[3]; + + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0), &det_mode); + if (ret < 0) + goto io_error; + + /* pin attached */ + if (det_mode) { + /* read UMP message offset */ + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), &offset); + if (ret < 0) + goto io_error; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(rt721->regmap, + RT721_BUF_ADDR_HID1 + offset + idx, &val); + if (ret < 0) + goto io_error; + buf[idx] = val & 0xff; + } + /* Report ID for HID1 */ + if (buf[0] == 0x11) + btn_type = rt_sdca_btn_type(&buf[1]); + } else + rt721->jack_type = 0; + + dev_dbg(&rt721->slave->dev, "%s, btn_type=0x%x\n", __func__, btn_type); + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt721->jack_btn_check_work, msecs_to_jiffies(200)); + } + + return; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); +} + +static void rt721_sdca_dmic_preset(struct rt721_sdca_priv *rt721) +{ + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8000); + rt_sdca_index_write(rt721->mbq_regmap, RT721_ANA_POW_PART, + RT721_VREF1_HV_CTRL1, 0xe000); + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8007); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL9, 0x2a2a); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL10, 0x2a00); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL6, 0x2a2a); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL5, 0x2626); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL8, 0x1e00); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL7, 0x1515); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_CH_FLOAT_CTL3, 0x0304); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_CH_FLOAT_CTL4, 0x0304); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_CTL1, 0x0000); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_IT26, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x01); + regmap_write(rt721->mbq_regmap, 0x5910009, 0x2e01); + rt_sdca_index_write(rt721->mbq_regmap, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b00); + rt_sdca_index_write(rt721->mbq_regmap, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b40); + regmap_write(rt721->regmap, 0x2f5c, 0x25); +} + +static void rt721_sdca_amp_preset(struct rt721_sdca_priv *rt721) +{ + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8000); + rt_sdca_index_write(rt721->mbq_regmap, RT721_ANA_POW_PART, + RT721_VREF1_HV_CTRL1, 0xe000); + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8007); + regmap_write(rt721->mbq_regmap, 0x5810000, 0x6420); + regmap_write(rt721->mbq_regmap, 0x5810000, 0x6421); + regmap_write(rt721->mbq_regmap, 0x5810000, 0xe421); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_CH_FLOAT_CTL6, 0x5561); + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_REG, + RT721_GPIO_PAD_CTRL5, 0x8003); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_OT23, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x04); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x00); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x00); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x00); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x00); +} + +static void rt721_sdca_jack_preset(struct rt721_sdca_priv *rt721) +{ + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8000); + rt_sdca_index_write(rt721->mbq_regmap, RT721_ANA_POW_PART, + RT721_VREF1_HV_CTRL1, 0xe000); + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8007); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_GE_REL_CTRL1, 0x8011); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL3, 0xcf00); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL4, 0x000f); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL1, 0x1100); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL5, 0x0c12); + rt_sdca_index_write(rt721->mbq_regmap, RT721_JD_CTRL, + RT721_JD_1PIN_GAT_CTRL2, 0xc002); + rt_sdca_index_write(rt721->mbq_regmap, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b00); + rt_sdca_index_write(rt721->mbq_regmap, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b40); + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON14, 0x3333); + regmap_write(rt721->mbq_regmap, 0x5810035, 0x0036); + regmap_write(rt721->mbq_regmap, 0x5810030, 0xee00); + rt_sdca_index_write(rt721->mbq_regmap, RT721_CAP_PORT_CTRL, + RT721_HP_AMP_2CH_CAL1, 0x0140); + regmap_write(rt721->mbq_regmap, 0x5810000, 0x0021); + regmap_write(rt721->mbq_regmap, 0x5810000, 0x8021); + rt_sdca_index_write(rt721->mbq_regmap, RT721_CAP_PORT_CTRL, + RT721_HP_AMP_2CH_CAL18, 0x5522); + regmap_write(rt721->mbq_regmap, 0x5b10007, 0x2000); + regmap_write(rt721->mbq_regmap, 0x5B10017, 0x1b0f); + rt_sdca_index_write(rt721->mbq_regmap, RT721_CBJ_CTRL, + RT721_CBJ_A0_GAT_CTRL1, 0x2a02); + rt_sdca_index_write(rt721->mbq_regmap, RT721_CAP_PORT_CTRL, + RT721_HP_AMP_2CH_CAL4, 0xa105); + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON14, 0x3b33); + regmap_write(rt721->mbq_regmap, 0x310400, 0x3023); + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON14, 0x3f33); + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON13, 0x6048); + regmap_write(rt721->mbq_regmap, 0x310401, 0x3000); + regmap_write(rt721->mbq_regmap, 0x310402, 0x1b00); + regmap_write(rt721->mbq_regmap, 0x310300, 0x000f); + regmap_write(rt721->mbq_regmap, 0x310301, 0x3000); + regmap_write(rt721->mbq_regmap, 0x310302, 0x1b00); + rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON17, 0x0008); + rt_sdca_index_write(rt721->mbq_regmap, RT721_DAC_CTRL, + RT721_DAC_2CH_CTRL3, 0x55ff); + rt_sdca_index_write(rt721->mbq_regmap, RT721_DAC_CTRL, + RT721_DAC_2CH_CTRL4, 0xcc00); + rt_sdca_index_write(rt721->mbq_regmap, RT721_ANA_POW_PART, + RT721_MBIAS_LV_CTRL2, 0x6677); + rt_sdca_index_write(rt721->mbq_regmap, RT721_ANA_POW_PART, + RT721_VREF2_LV_CTRL1, 0x7600); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL2, 0x1234); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL3, 0x3512); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL1, 0x4040); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL4, 0x1201); + regmap_write(rt721->regmap, 0x2f58, 0x07); +} + +static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) +{ + mutex_lock(&rt721->calibrate_mutex); + if (rt721->hs_jack) { + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + dev_dbg(&rt721->slave->dev, "in %s enable\n", __func__); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_UAJ_CTL, 0x036E); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XU03, + RT721_SDCA_CTL_SELECTED_MODE, 0), 0); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XU0D, + RT721_SDCA_CTL_SELECTED_MODE, 0), 0); + rt_sdca_index_update_bits(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_GE_REL_CTRL1, 0x4000, 0x4000); + } + mutex_unlock(&rt721->calibrate_mutex); +} + +static int rt721_sdca_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + int ret; + + rt721->hs_jack = hs_jack; + + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); + return 0; + } + + rt721_sdca_jack_init(rt721); + + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + + return 0; +} + +/* For SDCA control DAC/ADC Gain */ +static int rt721_sdca_set_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned int read_l, read_r, gain_l_val, gain_r_val; + unsigned int adc_vol_flag = 0, changed = 0; + unsigned int lvalue, rvalue; + const unsigned int interval_offset = 0xc0; + const unsigned int tendA = 0x200; + const unsigned int tendB = 0xa00; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume") || + strstr(ucontrol->id.name, "FU0F Capture Volume")) + adc_vol_flag = 1; + + regmap_read(rt721->mbq_regmap, mc->reg, &lvalue); + regmap_read(rt721->mbq_regmap, mc->rreg, &rvalue); + + /* L Channel */ + gain_l_val = ucontrol->value.integer.value[0]; + if (gain_l_val > mc->max) + gain_l_val = mc->max; + + if (mc->shift == 8) { + /* boost gain */ + gain_l_val = gain_l_val * tendB; + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (gain_l_val == 0) + gain_l_val = 0x8000; + else + gain_l_val = (gain_l_val - 1) * tendA; + } else { + /* ADC/DAC gain */ + if (adc_vol_flag) + gain_l_val = 0x1e00 - ((mc->max - gain_l_val) * interval_offset); + else + gain_l_val = 0 - ((mc->max - gain_l_val) * interval_offset); + gain_l_val &= 0xffff; + } + + /* R Channel */ + gain_r_val = ucontrol->value.integer.value[1]; + if (gain_r_val > mc->max) + gain_r_val = mc->max; + + if (mc->shift == 8) { + /* boost gain */ + gain_r_val = gain_r_val * tendB; + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (gain_r_val == 0) + gain_r_val = 0x8000; + else + gain_r_val = (gain_r_val - 1) * tendA; + } else { + /* ADC/DAC gain */ + if (adc_vol_flag) + gain_r_val = 0x1e00 - ((mc->max - gain_r_val) * interval_offset); + else + gain_r_val = 0 - ((mc->max - gain_r_val) * interval_offset); + gain_r_val &= 0xffff; + } + + if (lvalue != gain_l_val || rvalue != gain_r_val) + changed = 1; + else + return 0; + + /* Lch*/ + regmap_write(rt721->mbq_regmap, mc->reg, gain_l_val); + + /* Rch */ + regmap_write(rt721->mbq_regmap, mc->rreg, gain_r_val); + + regmap_read(rt721->mbq_regmap, mc->reg, &read_l); + regmap_read(rt721->mbq_regmap, mc->rreg, &read_r); + if (read_r == gain_r_val && read_l == gain_l_val) + return changed; + + return -EIO; +} + +static int rt721_sdca_set_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int read_l, read_r, ctl_l = 0, ctl_r = 0; + unsigned int adc_vol_flag = 0; + const unsigned int interval_offset = 0xc0; + const unsigned int tendB = 0xa00; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume") || + strstr(ucontrol->id.name, "FU0F Capture Volume")) + adc_vol_flag = 1; + + regmap_read(rt721->mbq_regmap, mc->reg, &read_l); + regmap_read(rt721->mbq_regmap, mc->rreg, &read_r); + + if (mc->shift == 8) /* boost gain */ + ctl_l = read_l / tendB; + else { + if (adc_vol_flag) + ctl_l = mc->max - (((0x1e00 - read_l) & 0xffff) / interval_offset); + else + ctl_l = mc->max - (((0 - read_l) & 0xffff) / interval_offset); + } + + if (read_l != read_r) { + if (mc->shift == 8) /* boost gain */ + ctl_r = read_r / tendB; + else { /* ADC/DAC gain */ + if (adc_vol_flag) + ctl_r = mc->max - (((0x1e00 - read_r) & 0xffff) / interval_offset); + else + ctl_r = mc->max - (((0 - read_r) & 0xffff) / interval_offset); + } + } else { + ctl_r = ctl_l; + } + + ucontrol->value.integer.value[0] = ctl_l; + ucontrol->value.integer.value[1] = ctl_r; + + return 0; +} + +static int rt721_sdca_set_fu1e_capture_ctl(struct rt721_sdca_priv *rt721) +{ + int err, i; + unsigned int ch_mute; + + for (i = 0; i < ARRAY_SIZE(rt721->fu1e_mixer_mute); i++) { + ch_mute = rt721->fu1e_dapm_mute || rt721->fu1e_mixer_mute[i]; + err = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_01) + i, ch_mute); + if (err < 0) + return err; + } + + return 0; +} + +static int rt721_sdca_fu1e_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + unsigned int i; + + for (i = 0; i < p->count; i++) + ucontrol->value.integer.value[i] = !rt721->fu1e_mixer_mute[i]; + + return 0; +} + +static int rt721_sdca_fu1e_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + int err, changed = 0, i; + + for (i = 0; i < p->count; i++) { + if (rt721->fu1e_mixer_mute[i] != !ucontrol->value.integer.value[i]) + changed = 1; + rt721->fu1e_mixer_mute[i] = !ucontrol->value.integer.value[i]; + } + + err = rt721_sdca_set_fu1e_capture_ctl(rt721); + if (err < 0) + return err; + + return changed; +} + +static int rt721_sdca_set_fu0f_capture_ctl(struct rt721_sdca_priv *rt721) +{ + int err; + unsigned int ch_l, ch_r; + + ch_l = (rt721->fu0f_dapm_mute || rt721->fu0f_mixer_l_mute) ? 0x01 : 0x00; + ch_r = (rt721->fu0f_dapm_mute || rt721->fu0f_mixer_r_mute) ? 0x01 : 0x00; + + err = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_L), ch_l); + if (err < 0) + return err; + + err = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_R), ch_r); + if (err < 0) + return err; + + return 0; +} + +static int rt721_sdca_fu0f_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !rt721->fu0f_mixer_l_mute; + ucontrol->value.integer.value[1] = !rt721->fu0f_mixer_r_mute; + return 0; +} + +static int rt721_sdca_fu0f_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + int err, changed = 0; + + if (rt721->fu0f_mixer_l_mute != !ucontrol->value.integer.value[0] || + rt721->fu0f_mixer_r_mute != !ucontrol->value.integer.value[1]) + changed = 1; + + rt721->fu0f_mixer_l_mute = !ucontrol->value.integer.value[0]; + rt721->fu0f_mixer_r_mute = !ucontrol->value.integer.value[1]; + err = rt721_sdca_set_fu0f_capture_ctl(rt721); + if (err < 0) + return err; + + return changed; +} + +static int rt721_sdca_fu_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + + if (p->max == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = p->count; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = p->max; + return 0; +} + +static int rt721_sdca_dmic_set_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + unsigned int boost_step = 0x0a00; + unsigned int vol_max = 0x1e00; + unsigned int regvalue, ctl, i; + unsigned int adc_vol_flag = 0; + const unsigned int interval_offset = 0xc0; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume")) + adc_vol_flag = 1; + + /* check all channels */ + for (i = 0; i < p->count; i++) { + regmap_read(rt721->mbq_regmap, p->reg_base + i, &regvalue); + + if (!adc_vol_flag) /* boost gain */ + ctl = regvalue / boost_step; + else { /* ADC gain */ + if (adc_vol_flag) + ctl = p->max - (((vol_max - regvalue) & 0xffff) / interval_offset); + else + ctl = p->max - (((0 - regvalue) & 0xffff) / interval_offset); + } + + ucontrol->value.integer.value[i] = ctl; + } + + return 0; +} + +static int rt721_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned int boost_step = 0x0a00; + unsigned int vol_max = 0x1e00; + unsigned int gain_val[4]; + unsigned int i, adc_vol_flag = 0, changed = 0; + unsigned int regvalue[4]; + const unsigned int interval_offset = 0xc0; + int err; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume")) + adc_vol_flag = 1; + + /* check all channels */ + for (i = 0; i < p->count; i++) { + regmap_read(rt721->mbq_regmap, p->reg_base + i, &regvalue[i]); + + gain_val[i] = ucontrol->value.integer.value[i]; + if (gain_val[i] > p->max) + gain_val[i] = p->max; + + if (!adc_vol_flag) /* boost gain */ + gain_val[i] = gain_val[i] * boost_step; + else { /* ADC gain */ + gain_val[i] = vol_max - ((p->max - gain_val[i]) * interval_offset); + gain_val[i] &= 0xffff; + } + + if (regvalue[i] != gain_val[i]) + changed = 1; + } + + if (!changed) + return 0; + + for (i = 0; i < p->count; i++) { + err = regmap_write(rt721->mbq_regmap, p->reg_base + i, gain_val[i]); + if (err < 0) + dev_err(&rt721->slave->dev, "%#08x can't be set\n", p->reg_base + i); + } + + return changed; +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol_tlv, 0, 1000, 0); +static const DECLARE_TLV_DB_SCALE(mic2_boost_vol_tlv, -200, 200, 0); + +static const struct snd_kcontrol_new rt721_sdca_controls[] = { + /* Headphone playback settings */ + SOC_DOUBLE_R_EXT_TLV("FU05 Playback Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_VOLUME, CH_R), 0, 0x57, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, out_vol_tlv), + /* Headset mic capture settings */ + SOC_DOUBLE_EXT("FU0F Capture Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt721_sdca_fu0f_capture_get, rt721_sdca_fu0f_capture_put), + SOC_DOUBLE_R_EXT_TLV("FU0F Capture Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_VOLUME, CH_R), 0, 0x3f, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("FU33 Boost Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_R), 1, 0x15, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, mic2_boost_vol_tlv), + /* AMP playback settings */ + SOC_DOUBLE_R_EXT_TLV("FU06 Playback Volume", + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_VOLUME, CH_R), 0, 0x57, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, out_vol_tlv), + /* DMIC capture settings */ + RT_SDCA_FU_CTRL("FU1E Capture Switch", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_01), 1, 1, 4, rt721_sdca_fu_info, + rt721_sdca_fu1e_capture_get, rt721_sdca_fu1e_capture_put), + RT_SDCA_EXT_TLV("FU1E Capture Volume", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_VOLUME, CH_01), + rt721_sdca_dmic_set_gain_get, rt721_sdca_dmic_set_gain_put, + 4, 0x3f, mic_vol_tlv, rt721_sdca_fu_info), + RT_SDCA_EXT_TLV("FU15 Boost Volume", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, + RT721_SDCA_CTL_FU_CH_GAIN, CH_01), + rt721_sdca_dmic_set_gain_get, rt721_sdca_dmic_set_gain_put, + 4, 3, boost_vol_tlv, rt721_sdca_fu_info), +}; + +static int rt721_sdca_adc_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned int val = 0, mask_sft, mask; + + if (strstr(ucontrol->id.name, "ADC 09 Mux")) { + mask_sft = 12; + mask = 0x7; + } else if (strstr(ucontrol->id.name, "ADC 08 R Mux")) { + mask_sft = 10; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 08 L Mux")) { + mask_sft = 8; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 R Mux")) { + mask_sft = 6; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 L Mux")) { + mask_sft = 4; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 R Mux")) { + mask_sft = 2; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 L Mux")) { + mask_sft = 0; + mask = 0x3; + } else + return -EINVAL; + + rt_sdca_index_read(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, &val); + + ucontrol->value.enumerated.item[0] = (val >> mask_sft) & mask; + + return 0; +} + +static int rt721_sdca_adc_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val, val2 = 0, change, mask_sft, mask; + unsigned int check; + + if (item[0] >= e->items) + return -EINVAL; + + if (strstr(ucontrol->id.name, "ADC 09 Mux")) { + mask_sft = 12; + mask = 0x7; + } else if (strstr(ucontrol->id.name, "ADC 08 R Mux")) { + mask_sft = 10; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 08 L Mux")) { + mask_sft = 8; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 R Mux")) { + mask_sft = 6; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 L Mux")) { + mask_sft = 4; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 R Mux")) { + mask_sft = 2; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 L Mux")) { + mask_sft = 0; + mask = 0x3; + } else + return -EINVAL; + + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + rt_sdca_index_read(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, &val2); + + if (strstr(ucontrol->id.name, "ADC 09 Mux")) + val2 = (val2 >> mask_sft) & 0x7; + else + val2 = (val2 >> mask_sft) & 0x3; + + if (val == val2) + change = 0; + else + change = 1; + + if (change) { + rt_sdca_index_read(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, &check); + rt_sdca_index_update_bits(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, mask << mask_sft, + val << mask_sft); + } + + snd_soc_dapm_mux_update_power(dapm, kcontrol, + item[0], e, NULL); + + return change; +} + +static const char * const adc09_mux_text[] = { + "MIC2", + "LINE1", + "LINE2", +}; +static const char * const adc07_10_mux_text[] = { + "DMIC1 RE", + "DMIC1 FE", + "DMIC2 RE", + "DMIC2 FE", +}; + +static SOC_ENUM_SINGLE_DECL( + rt721_adc09_enum, SND_SOC_NOPM, 0, adc09_mux_text); +static SOC_ENUM_SINGLE_DECL( + rt721_dmic_enum, SND_SOC_NOPM, 0, adc07_10_mux_text); + +static const struct snd_kcontrol_new rt721_sdca_adc09_mux = + SOC_DAPM_ENUM_EXT("ADC 09 Mux", rt721_adc09_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc08_r_mux = + SOC_DAPM_ENUM_EXT("ADC 08 R Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc08_l_mux = + SOC_DAPM_ENUM_EXT("ADC 08 L Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc10_r_mux = + SOC_DAPM_ENUM_EXT("ADC 10 R Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc10_l_mux = + SOC_DAPM_ENUM_EXT("ADC 10 L Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc07_r_mux = + SOC_DAPM_ENUM_EXT("ADC 07 R Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc07_l_mux = + SOC_DAPM_ENUM_EXT("ADC 07 L Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); + + +static int rt721_sdca_fu42_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(100); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_L), unmute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_R), unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_L), mute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_R), mute); + break; + } + return 0; +} + +static int rt721_sdca_fu21_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_L), unmute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_R), unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_L), mute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_R), mute); + break; + } + return 0; +} + +static int rt721_sdca_fu23_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_L), unmute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_R), unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_L), mute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_R), mute); + break; + } + return 0; +} + +static int rt721_sdca_fu113_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt721->fu1e_dapm_mute = false; + rt721_sdca_set_fu1e_capture_ctl(rt721); + break; + case SND_SOC_DAPM_PRE_PMD: + rt721->fu1e_dapm_mute = true; + rt721_sdca_set_fu1e_capture_ctl(rt721); + break; + } + return 0; +} + +static int rt721_sdca_fu36_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt721->fu0f_dapm_mute = false; + rt721_sdca_set_fu0f_capture_ctl(rt721); + break; + case SND_SOC_DAPM_PRE_PMD: + rt721->fu0f_dapm_mute = true; + rt721_sdca_set_fu0f_capture_ctl(rt721); + break; + } + return 0; +} + +static int rt721_sdca_pde47_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE40, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE40, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static int rt721_sdca_pde41_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE41, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE41, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static int rt721_sdca_pde11_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_PDE2A, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_PDE2A, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static int rt721_sdca_pde34_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE12, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE12, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget rt721_sdca_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("HP"), + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + SND_SOC_DAPM_INPUT("DMIC1_2"), + SND_SOC_DAPM_INPUT("DMIC3_4"), + + SND_SOC_DAPM_SUPPLY("PDE 41", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde41_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 47", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde47_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 11", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde11_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 34", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde34_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_DAC_E("FU 21", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu21_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("FU 23", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu23_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("FU 42", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu42_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("FU 36", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu36_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("FU 113", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu113_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MUX("ADC 09 Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc09_mux), + SND_SOC_DAPM_MUX("ADC 08 R Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc08_r_mux), + SND_SOC_DAPM_MUX("ADC 08 L Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc08_l_mux), + SND_SOC_DAPM_MUX("ADC 10 R Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc10_r_mux), + SND_SOC_DAPM_MUX("ADC 10 L Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc10_l_mux), + SND_SOC_DAPM_MUX("ADC 07 R Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc07_r_mux), + SND_SOC_DAPM_MUX("ADC 07 L Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc07_l_mux), + + SND_SOC_DAPM_AIF_IN("DP1RX", "DP1 Headphone Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Headset Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DP3RX", "DP3 Speaker Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP6TX", "DP6 DMic Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route rt721_sdca_audio_map[] = { + {"FU 42", NULL, "DP1RX"}, + {"FU 21", NULL, "DP3RX"}, + {"FU 23", NULL, "DP3RX"}, + + {"ADC 09 Mux", "MIC2", "MIC2"}, + {"ADC 09 Mux", "LINE1", "LINE1"}, + {"ADC 09 Mux", "LINE2", "LINE2"}, + {"ADC 07 R Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 07 R Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 07 R Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 07 R Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 07 L Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 07 L Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 07 L Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 07 L Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 08 R Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 08 R Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 08 R Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 08 R Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 08 L Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 08 L Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 08 L Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 08 L Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 10 R Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 10 R Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 10 R Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 10 R Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 10 L Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 10 L Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 10 L Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 10 L Mux", "DMIC2 FE", "DMIC3_4"}, + {"FU 36", NULL, "PDE 34"}, + {"FU 36", NULL, "ADC 09 Mux"}, + {"FU 113", NULL, "PDE 11"}, + {"FU 113", NULL, "ADC 07 R Mux"}, + {"FU 113", NULL, "ADC 07 L Mux"}, + {"FU 113", NULL, "ADC 10 R Mux"}, + {"FU 113", NULL, "ADC 10 L Mux"}, + {"DP2TX", NULL, "FU 36"}, + {"DP6TX", NULL, "FU 113"}, + + {"HP", NULL, "PDE 47"}, + {"HP", NULL, "FU 42"}, + {"SPK", NULL, "PDE 41"}, + {"SPK", NULL, "FU 21"}, + {"SPK", NULL, "FU 23"}, +}; + +static int rt721_sdca_parse_dt(struct rt721_sdca_priv *rt721, struct device *dev) +{ + device_property_read_u32(dev, "realtek,jd-src", &rt721->jd_src); + + return 0; +} + +static int rt721_sdca_probe(struct snd_soc_component *component) +{ + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + int ret; + + rt721_sdca_parse_dt(rt721, &rt721->slave->dev); + rt721->component = component; + + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + + return 0; +} + +static const struct snd_soc_component_driver soc_sdca_dev_rt721 = { + .probe = rt721_sdca_probe, + .controls = rt721_sdca_controls, + .num_controls = ARRAY_SIZE(rt721_sdca_controls), + .dapm_widgets = rt721_sdca_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt721_sdca_dapm_widgets), + .dapm_routes = rt721_sdca_audio_map, + .num_dapm_routes = ARRAY_SIZE(rt721_sdca_audio_map), + .set_jack = rt721_sdca_set_jack_detect, + .endianness = 1, +}; + +static int rt721_sdca_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void rt721_sdca_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int rt721_sdca_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_runtime *sdw_stream; + int retval, port, num_channels; + unsigned int sampling_rate; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!rt721->slave) + return -EINVAL; + + /* + * RT721_AIF1 with port = 1 for headphone playback + * RT721_AIF1 with port = 2 for headset-mic capture + * RT721_AIF2 with port = 3 for speaker playback + * RT721_AIF3 with port = 6 for digital-mic capture + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + if (dai->id == RT721_AIF1) + port = 1; + else if (dai->id == RT721_AIF2) + port = 3; + else + return -EINVAL; + } else { + direction = SDW_DATA_DIR_TX; + if (dai->id == RT721_AIF1) + port = 2; + else if (dai->id == RT721_AIF3) + port = 6; + else + return -EINVAL; + } + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = GENMASK(num_channels - 1, 0); + port_config.num = port; + + retval = sdw_stream_add_slave(rt721->slave, &stream_config, + &port_config, 1, sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + if (params_channels(params) > 16) { + dev_err(component->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = RT721_SDCA_RATE_8000HZ; + break; + case 16000: + sampling_rate = RT721_SDCA_RATE_16000HZ; + break; + case 24000: + sampling_rate = RT721_SDCA_RATE_24000HZ; + break; + case 32000: + sampling_rate = RT721_SDCA_RATE_32000HZ; + break; + case 44100: + sampling_rate = RT721_SDCA_RATE_44100HZ; + break; + case 48000: + sampling_rate = RT721_SDCA_RATE_48000HZ; + break; + case 96000: + sampling_rate = RT721_SDCA_RATE_96000HZ; + break; + case 192000: + sampling_rate = RT721_SDCA_RATE_192000HZ; + break; + case 384000: + sampling_rate = RT721_SDCA_RATE_384000HZ; + break; + case 768000: + sampling_rate = RT721_SDCA_RATE_768000HZ; + break; + default: + dev_err(component->dev, "Rate %d is not supported\n", + params_rate(params)); + return -EINVAL; + } + + /* set sampling frequency */ + if (dai->id == RT721_AIF1) { + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS01, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS11, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + } + + if (dai->id == RT721_AIF2) + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_CS31, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + + if (dai->id == RT721_AIF3) + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_CS1F, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + + return 0; +} + +static int rt721_sdca_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt721->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt721->slave, sdw_stream); + return 0; +} + +#define RT721_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) +#define RT721_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops rt721_sdca_ops = { + .hw_params = rt721_sdca_pcm_hw_params, + .hw_free = rt721_sdca_pcm_hw_free, + .set_stream = rt721_sdca_set_sdw_stream, + .shutdown = rt721_sdca_shutdown, +}; + +static struct snd_soc_dai_driver rt721_sdca_dai[] = { + { + .name = "rt721-sdca-aif1", + .id = RT721_AIF1, + .playback = { + .stream_name = "DP1 Headphone Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .capture = { + .stream_name = "DP2 Headset Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .ops = &rt721_sdca_ops, + }, + { + .name = "rt721-sdca-aif2", + .id = RT721_AIF2, + .playback = { + .stream_name = "DP3 Speaker Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .ops = &rt721_sdca_ops, + }, + { + .name = "rt721-sdca-aif3", + .id = RT721_AIF3, + .capture = { + .stream_name = "DP6 DMic Capture", + .channels_min = 1, + .channels_max = 4, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .ops = &rt721_sdca_ops, + } +}; + +int rt721_sdca_init(struct device *dev, struct regmap *regmap, + struct regmap *mbq_regmap, struct sdw_slave *slave) +{ + struct rt721_sdca_priv *rt721; + + rt721 = devm_kzalloc(dev, sizeof(*rt721), GFP_KERNEL); + if (!rt721) + return -ENOMEM; + + dev_set_drvdata(dev, rt721); + rt721->slave = slave; + rt721->regmap = regmap; + rt721->mbq_regmap = mbq_regmap; + + regcache_cache_only(rt721->regmap, true); + regcache_cache_only(rt721->mbq_regmap, true); + + mutex_init(&rt721->calibrate_mutex); + mutex_init(&rt721->disable_irq_lock); + + INIT_DELAYED_WORK(&rt721->jack_detect_work, rt721_sdca_jack_detect_handler); + INIT_DELAYED_WORK(&rt721->jack_btn_check_work, rt721_sdca_btn_check_handler); + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt721->hw_init = false; + rt721->first_hw_init = false; + rt721->fu1e_dapm_mute = true; + rt721->fu0f_dapm_mute = true; + rt721->fu0f_mixer_l_mute = rt721->fu0f_mixer_r_mute = true; + rt721->fu1e_mixer_mute[0] = rt721->fu1e_mixer_mute[1] = + rt721->fu1e_mixer_mute[2] = rt721->fu1e_mixer_mute[3] = true; + + return devm_snd_soc_register_component(dev, + &soc_sdca_dev_rt721, rt721_sdca_dai, ARRAY_SIZE(rt721_sdca_dai)); +} + +int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); + + rt721->disable_irq = false; + + if (rt721->hw_init) + return 0; + + regcache_cache_only(rt721->regmap, false); + regcache_cache_only(rt721->mbq_regmap, false); + if (rt721->first_hw_init) { + regcache_cache_bypass(rt721->regmap, true); + regcache_cache_bypass(rt721->mbq_regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + rt721_sdca_dmic_preset(rt721); + rt721_sdca_amp_preset(rt721); + rt721_sdca_jack_preset(rt721); + if (rt721->first_hw_init) { + regcache_cache_bypass(rt721->regmap, false); + regcache_mark_dirty(rt721->regmap); + regcache_cache_bypass(rt721->mbq_regmap, false); + regcache_mark_dirty(rt721->mbq_regmap); + } else + rt721->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt721->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +MODULE_DESCRIPTION("ASoC RT721 SDCA SDW driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt721-sdca.h b/sound/soc/codecs/rt721-sdca.h new file mode 100644 index 000000000000..e2f071909da8 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca.h @@ -0,0 +1,268 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt721-sdca.h -- RT721 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2024 Realtek Semiconductor Corp. + */ + +#ifndef __RT721_H__ +#define __RT721_H__ + +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/soc.h> +#include <linux/workqueue.h> + +struct rt721_sdca_priv { + struct regmap *regmap; + struct regmap *mbq_regmap; + struct snd_soc_component *component; + struct sdw_slave *slave; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + struct mutex calibrate_mutex; + struct mutex disable_irq_lock; + bool disable_irq; + /* For Headset jack & Headphone */ + unsigned int scp_sdca_stat1; + unsigned int scp_sdca_stat2; + struct snd_soc_jack *hs_jack; + struct delayed_work jack_detect_work; + struct delayed_work jack_btn_check_work; + int jack_type; + int jd_src; + bool fu0f_dapm_mute; + bool fu0f_mixer_l_mute; + bool fu0f_mixer_r_mute; + /* For DMIC */ + bool fu1e_dapm_mute; + bool fu1e_mixer_mute[4]; +}; + +struct rt721_sdca_dmic_kctrl_priv { + unsigned int reg_base; + unsigned int count; + unsigned int max; + unsigned int invert; +}; + +/* NID */ +#define RT721_ANA_POW_PART 0x01 +#define RT721_DAC_CTRL 0x04 +#define RT721_JD_CTRL 0x09 +#define RT721_CBJ_CTRL 0x0a +#define RT721_CAP_PORT_CTRL 0x0c +#define RT721_CLASD_AMP_CTRL 0x0d +#define RT721_VENDOR_REG 0x20 +#define RT721_RC_CALIB_CTRL 0x40 +#define RT721_VENDOR_EQ_L 0x53 +#define RT721_VENDOR_EQ_R 0x54 +#define RT721_VENDOR_HP_CALI 0x56 +#define RT721_VENDOR_CHARGE_PUMP 0x57 +#define RT721_VENDOR_CLASD_CALI 0x58 +#define RT721_VENDOR_IMS_DRE 0x5b +#define RT721_VENDOR_SPK_EFUSE 0x5c +#define RT721_VENDOR_LEVEL_CTRL 0x5d +#define RT721_VENDOR_ANA_CTL 0x5f +#define RT721_HDA_SDCA_FLOAT 0x61 + +/* Index (NID:01h) */ +#define RT721_MBIAS_LV_CTRL2 0x07 +#define RT721_VREF1_HV_CTRL1 0x0a +#define RT721_VREF2_LV_CTRL1 0x0b + +/* Index (NID:04h) */ +#define RT721_DAC_2CH_CTRL3 0x02 +#define RT721_DAC_2CH_CTRL4 0x03 + +/* Index (NID:09h) */ +#define RT721_JD_1PIN_GAT_CTRL2 0x07 + +/* Index (NID:0ah) */ +#define RT721_CBJ_A0_GAT_CTRL1 0x04 +#define RT721_CBJ_A0_GAT_CTRL2 0x05 + +/* Index (NID:0Ch) */ +#define RT721_HP_AMP_2CH_CAL1 0x05 +#define RT721_HP_AMP_2CH_CAL4 0x08 +#define RT721_HP_AMP_2CH_CAL18 0x1b + +/* Index (NID:0dh) */ +#define RT721_CLASD_AMP_2CH_CAL 0x14 + +/* Index (NID:20h) */ +#define RT721_JD_PRODUCT_NUM 0x00 +#define RT721_ANALOG_BIAS_CTL3 0x04 +#define RT721_JD_CTRL1 0x09 +#define RT721_LDO2_3_CTL1 0x0e +#define RT721_GPIO_PAD_CTRL5 0x13 +#define RT721_LDO1_CTL 0x1a +#define RT721_HP_JD_CTRL 0x24 +#define RT721_VD_HIDDEN_CTRL 0x26 +#define RT721_CLSD_CTRL6 0x3c +#define RT721_COMBO_JACK_AUTO_CTL1 0x45 +#define RT721_COMBO_JACK_AUTO_CTL2 0x46 +#define RT721_COMBO_JACK_AUTO_CTL3 0x47 +#define RT721_DIGITAL_MISC_CTRL4 0x4a +#define RT721_VREFO_GAT 0x63 +#define RT721_FSM_CTL 0x67 +#define RT721_SDCA_INTR_REC 0x82 +#define RT721_SW_CONFIG1 0x8a +#define RT721_SW_CONFIG2 0x8b + +/* Index (NID:40h) */ +#define RT721_RC_CALIB_CTRL0 0x00 + +/* Index (NID:58h) */ +#define RT721_DAC_DC_CALI_CTL1 0x01 +#define RT721_DAC_DC_CALI_CTL2 0x02 +#define RT721_DAC_DC_CALI_CTL3 0x03 + +/* Index (NID:5fh) */ +#define RT721_MISC_POWER_CTL0 0x00 +#define RT721_MISC_POWER_CTL31 0x31 +#define RT721_UAJ_TOP_TCON13 0x44 +#define RT721_UAJ_TOP_TCON14 0x45 +#define RT721_UAJ_TOP_TCON17 0x48 + +/* Index (NID:61h) */ +#define RT721_HDA_LEGACY_MUX_CTL0 0x00 +#define RT721_HDA_LEGACY_UAJ_CTL 0x02 +#define RT721_HDA_LEGACY_CTL1 0x05 +#define RT721_HDA_LEGACY_RESET_CTL 0x06 +#define RT721_GE_REL_CTRL1 0x0d +#define RT721_HDA_LEGACY_GPIO_WAKE_EN_CTL 0x0e +#define RT721_GE_SDCA_RST_CTRL 0x10 +#define RT721_INT_RST_EN_CTRL 0x11 +#define RT721_XU_EVENT_EN 0x13 +#define RT721_INLINE_CTL2 0x17 +#define RT721_UMP_HID_CTRL1 0x18 +#define RT721_UMP_HID_CTRL2 0x19 +#define RT721_UMP_HID_CTRL3 0x1a +#define RT721_UMP_HID_CTRL4 0x1b +#define RT721_UMP_HID_CTRL5 0x1c +#define RT721_FUNC_FLOAT_CTL0 0x22 +#define RT721_FUNC_FLOAT_CTL1 0x23 +#define RT721_FUNC_FLOAT_CTL2 0x24 +#define RT721_FUNC_FLOAT_CTL3 0x25 +#define RT721_ENT_FLOAT_CTL0 0x29 +#define RT721_ENT_FLOAT_CTL1 0x2c +#define RT721_ENT_FLOAT_CTL2 0x2d +#define RT721_ENT_FLOAT_CTL3 0x2e +#define RT721_ENT_FLOAT_CTL4 0x2f +#define RT721_CH_FLOAT_CTL1 0x45 +#define RT721_CH_FLOAT_CTL2 0x46 +#define RT721_ENT_FLOAT_CTL5 0x53 +#define RT721_ENT_FLOAT_CTL6 0x54 +#define RT721_ENT_FLOAT_CTL7 0x55 +#define RT721_ENT_FLOAT_CTL8 0x57 +#define RT721_ENT_FLOAT_CTL9 0x5a +#define RT721_ENT_FLOAT_CTL10 0x5b +#define RT721_CH_FLOAT_CTL3 0x6a +#define RT721_CH_FLOAT_CTL4 0x6d +#define RT721_CH_FLOAT_CTL5 0x70 +#define RT721_CH_FLOAT_CTL6 0x92 + +/* Parameter & Verb control 01 (0x26)(NID:20h) */ +#define RT721_HIDDEN_REG_SW_RESET (0x1 << 14) + +/* Buffer address for HID */ +#define RT721_BUF_ADDR_HID1 0x44030000 +#define RT721_BUF_ADDR_HID2 0x44030020 + +/* RT721 SDCA Control - function number */ +#define FUNC_NUM_JACK_CODEC 0x01 +#define FUNC_NUM_MIC_ARRAY 0x02 +#define FUNC_NUM_HID 0x03 +#define FUNC_NUM_AMP 0x04 + +/* RT721 SDCA entity */ +#define RT721_SDCA_ENT_HID01 0x01 +#define RT721_SDCA_ENT_XUV 0x03 +#define RT721_SDCA_ENT_GE49 0x49 +#define RT721_SDCA_ENT_USER_FU05 0x05 +#define RT721_SDCA_ENT_USER_FU06 0x06 +#define RT721_SDCA_ENT_USER_FU0F 0x0f +#define RT721_SDCA_ENT_USER_FU10 0x19 +#define RT721_SDCA_ENT_USER_FU1E 0x1e +#define RT721_SDCA_ENT_FU15 0x15 +#define RT721_SDCA_ENT_PDE23 0x23 +#define RT721_SDCA_ENT_PDE40 0x40 +#define RT721_SDCA_ENT_PDE41 0x41 +#define RT721_SDCA_ENT_PDE11 0x11 +#define RT721_SDCA_ENT_PDE12 0x12 +#define RT721_SDCA_ENT_PDE2A 0x2a +#define RT721_SDCA_ENT_CS01 0x01 +#define RT721_SDCA_ENT_CS11 0x11 +#define RT721_SDCA_ENT_CS1F 0x1f +#define RT721_SDCA_ENT_CS1C 0x1c +#define RT721_SDCA_ENT_CS31 0x31 +#define RT721_SDCA_ENT_OT23 0x42 +#define RT721_SDCA_ENT_IT26 0x26 +#define RT721_SDCA_ENT_IT09 0x09 +#define RT721_SDCA_ENT_PLATFORM_FU15 0x15 +#define RT721_SDCA_ENT_PLATFORM_FU44 0x44 +#define RT721_SDCA_ENT_XU03 0x03 +#define RT721_SDCA_ENT_XU0D 0x0d +#define RT721_SDCA_ENT_FU55 0x55 + +/* RT721 SDCA control */ +#define RT721_SDCA_CTL_SAMPLE_FREQ_INDEX 0x10 +#define RT721_SDCA_CTL_FU_MUTE 0x01 +#define RT721_SDCA_CTL_FU_VOLUME 0x02 +#define RT721_SDCA_CTL_HIDTX_CURRENT_OWNER 0x10 +#define RT721_SDCA_CTL_HIDTX_SET_OWNER_TO_DEVICE 0x11 +#define RT721_SDCA_CTL_HIDTX_MESSAGE_OFFSET 0x12 +#define RT721_SDCA_CTL_HIDTX_MESSAGE_LENGTH 0x13 +#define RT721_SDCA_CTL_SELECTED_MODE 0x01 +#define RT721_SDCA_CTL_DETECTED_MODE 0x02 +#define RT721_SDCA_CTL_REQ_POWER_STATE 0x01 +#define RT721_SDCA_CTL_VENDOR_DEF 0x30 +#define RT721_SDCA_CTL_XUV 0x34 +#define RT721_SDCA_CTL_FU_CH_GAIN 0x0b + +/* RT721 SDCA channel */ +#define CH_L 0x01 +#define CH_R 0x02 +#define CH_01 0x01 +#define CH_02 0x02 +#define CH_03 0x03 +#define CH_04 0x04 +#define CH_08 0x08 +#define CH_09 0x09 +#define CH_0A 0x0a + +/* sample frequency index */ +#define RT721_SDCA_RATE_8000HZ 0x01 +#define RT721_SDCA_RATE_11025HZ 0x02 +#define RT721_SDCA_RATE_12000HZ 0x03 +#define RT721_SDCA_RATE_16000HZ 0x04 +#define RT721_SDCA_RATE_22050HZ 0x05 +#define RT721_SDCA_RATE_24000HZ 0x06 +#define RT721_SDCA_RATE_32000HZ 0x07 +#define RT721_SDCA_RATE_44100HZ 0x08 +#define RT721_SDCA_RATE_48000HZ 0x09 +#define RT721_SDCA_RATE_88200HZ 0x0a +#define RT721_SDCA_RATE_96000HZ 0x0b +#define RT721_SDCA_RATE_176400HZ 0x0c +#define RT721_SDCA_RATE_192000HZ 0x0d +#define RT721_SDCA_RATE_384000HZ 0x0e +#define RT721_SDCA_RATE_768000HZ 0x0f + +/* RT721 HID ID */ +#define RT721_SDCA_HID_ID 0x11 + +enum { + RT721_AIF1, /* For headset mic and headphone */ + RT721_AIF2, /* For speaker */ + RT721_AIF3, /* For dmic */ + RT721_AIFS, +}; + +int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave); +int rt721_sdca_init(struct device *dev, struct regmap *regmap, + struct regmap *mbq_regmap, struct sdw_slave *slave); +#endif /* __RT721_H__ */ -- 2.34.1

3 3

[PATCH 1/2] ASoC: SOF: amd: Add error log for DSP firmware validation failure
by Venkata Prasad Potturu 08 Oct '24

08 Oct '24

Add dev_err to print ACP_SHA_DSP_FW_QUALIFIER and ACP_SHA_PSP_ACK register values for PSP firmware validation failure case. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index d579c3849392..de3001f5b9bb 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -329,7 +329,9 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, fw_qualifier, fw_qualifier & DSP_FW_RUN_ENABLE, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); if (ret < 0) { - dev_err(sdev->dev, "PSP validation failed\n"); + val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_SHA_PSP_ACK); + dev_err(sdev->dev, "PSP validation failed: fw_qualifier = %#x, ACP_SHA_PSP_ACK = %#x\n", + fw_qualifier, val); return ret; } -- 2.39.2

2 2

[PATCH v5] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 08 Oct '24

08 Oct '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job 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. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v5: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202410021557.FByBO9Dp-lkp@intel.com/ v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index a097afd76ded..ecc987548088 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22901,25 +22901,42 @@ 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*.* @@ -22929,9 +22946,8 @@ 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: 87d6aab2389e5ce0197d8257d5f8ee965a67c4cd -- 2.34.1

1 0

[PATCH alsa-ucm-conf v2 1/2] sof-soundwire: Change mapped control names to make them unique
by Maciej Strozek 08 Oct '24

08 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found as substrings in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Fixes: e87dde51d689 ("sof-soundwire: Add basic support for cs42l43") Fixes: e4277a204cd6 ("sof-soundwire: Add support for cs42l43/cs35l56 bridge configuration") Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 4 ++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..cf61833 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..a3cf6ad 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker Playback" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } -- 2.39.5

1 2

[PATCH v1] ASoC: codecs: lpass-macro: Add support for channel map mixer control
by Mohammad Rafi Shaik 08 Oct '24

08 Oct '24

From: Rohit kumar <quic_rohkumar(a)quicinc.com> Add channel map mixer control for lpass macro codec DAIs. Signed-off-by: Rohit kumar <quic_rohkumar(a)quicinc.com> Co-developed-by: Mohammad Rafi Shaik <quic_mohs(a)quicinc.com> Signed-off-by: Mohammad Rafi Shaik <quic_mohs(a)quicinc.com> --- sound/soc/codecs/lpass-macro-common.c | 55 +++++++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 2 + sound/soc/codecs/lpass-rx-macro.c | 12 ++++++ sound/soc/codecs/lpass-tx-macro.c | 9 +++++ sound/soc/codecs/lpass-va-macro.c | 9 +++++ sound/soc/codecs/lpass-wsa-macro.c | 12 ++++++ 6 files changed, 99 insertions(+) diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c index 6e3b8d0897dd..777af4885245 100644 --- a/sound/soc/codecs/lpass-macro-common.c +++ b/sound/soc/codecs/lpass-macro-common.c @@ -8,12 +8,67 @@ #include <linux/platform_device.h> #include <linux/pm_domain.h> #include <linux/pm_runtime.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> #include "lpass-macro-common.h" static DEFINE_MUTEX(lpass_codec_mutex); static enum lpass_codec_version lpass_codec_version; +static int lpass_macro_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct snd_soc_dai *dai = info->private_data; + u32 *chmap_data = NULL; + u32 rx_ch_cnt = 0; + u32 tx_ch_cnt = 0; + u32 rx_ch, tx_ch; + + chmap_data = kzalloc(sizeof(u32) * 2, GFP_KERNEL); + if (!chmap_data) + return -ENOMEM; + + snd_soc_dai_get_channel_map(dai, &tx_ch_cnt, &tx_ch, &rx_ch_cnt, &rx_ch); + if (rx_ch_cnt) { + chmap_data[0] = rx_ch_cnt; + chmap_data[1] = rx_ch; + } else if (tx_ch_cnt) { + chmap_data[0] = tx_ch_cnt; + chmap_data[1] = tx_ch; + } + memcpy(ucontrol->value.bytes.data, chmap_data, sizeof(u32) * 2); + + kfree(chmap_data); + return 0; +} + +int lpass_macro_add_chmap_ctls(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai, int dir) +{ + struct snd_pcm_chmap *info; + int ret; + + info = kzalloc(sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, dir, NULL, + 2 * sizeof(u32), 0, &info); + if (ret < 0) { + kfree(info); + return ret; + } + + /* override handlers */ + info->private_data = dai; + info->kctl->get = lpass_macro_chmap_ctl_get; + return 0; +} +EXPORT_SYMBOL_GPL(lpass_macro_add_chmap_ctls); + struct lpass_macro *lpass_macro_pds_init(struct device *dev) { struct lpass_macro *l_pds; diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h index fb4b96cb2b23..23ed6836addf 100644 --- a/sound/soc/codecs/lpass-macro-common.h +++ b/sound/soc/codecs/lpass-macro-common.h @@ -40,6 +40,8 @@ struct lpass_macro *lpass_macro_pds_init(struct device *dev); void lpass_macro_pds_exit(struct lpass_macro *pds); void lpass_macro_set_codec_version(enum lpass_codec_version version); enum lpass_codec_version lpass_macro_get_codec_version(void); +int lpass_macro_add_chmap_ctls(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai, int dir); static inline void lpass_macro_pds_exit_action(void *pds) { diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 71e0d3bffd3f..05bd0f8d126b 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -1954,10 +1954,22 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } +static int rx_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_PLAYBACK; + + if (dai->id == RX_MACRO_AIF_ECHO) + dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops rx_macro_dai_ops = { .hw_params = rx_macro_hw_params, .get_channel_map = rx_macro_get_channel_map, .mute_stream = rx_macro_digital_mute, + .pcm_new = rx_macro_pcm_new, }; static struct snd_soc_dai_driver rx_macro_dai[] = { diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index a134584acf90..c7a8e2694e36 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -1211,10 +1211,19 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } +static int tx_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops tx_macro_dai_ops = { .hw_params = tx_macro_hw_params, .get_channel_map = tx_macro_get_channel_map, .mute_stream = tx_macro_digital_mute, + .pcm_new = tx_macro_pcm_new, }; static struct snd_soc_dai_driver tx_macro_dai[] = { diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index c781da476240..1a13f472ed3a 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -939,10 +939,19 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } +static int va_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops va_macro_dai_ops = { .hw_params = va_macro_hw_params, .get_channel_map = va_macro_get_channel_map, .mute_stream = va_macro_digital_mute, + .pcm_new = va_macro_pcm_new, }; static struct snd_soc_dai_driver va_macro_dais[] = { diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index c989d82d1d3c..7c4fd4bf2668 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -1344,9 +1344,21 @@ static int wsa_macro_get_channel_map(const struct snd_soc_dai *dai, return 0; } +static int wsa_macro_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int dir = SNDRV_PCM_STREAM_PLAYBACK; + + if (dai->id == WSA_MACRO_AIF_VI || dai->id == WSA_MACRO_AIF_ECHO) + dir = SNDRV_PCM_STREAM_CAPTURE; + + return lpass_macro_add_chmap_ctls(rtd, dai, dir); +} + static const struct snd_soc_dai_ops wsa_macro_dai_ops = { .hw_params = wsa_macro_hw_params, .get_channel_map = wsa_macro_get_channel_map, + .pcm_new = wsa_macro_pcm_new, }; static struct snd_soc_dai_driver wsa_macro_dai[] = { -- 2.25.1

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