September 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

[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

[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

[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] 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

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 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

2 1

[PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param()
by (학생) 장인규 (컴퓨터공학과) 05 Oct '24

05 Oct '24

From 791716bf359b8540c519810848fd1f8006d7c3c5 Mon Sep 17 00:00:00 2001 From: Ingyu Jang <ingyujang25(a)unist.ac.kr> Date: Thu, 26 Sep 2024 19:40:04 +0900 Subject: [PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param() The aio_src_set_param() function did not previously check the return values of regmap_write() and regmap_update_bits(). If these functions fail, it could lead to silent failures when configuring the sample rate converter (SRC), causing improper behavior in audio processing without any indication of an error. This patch modifies aio_src_set_param to check the return values of regmap_write() and regmap_update_bits(). If either function returns an error, the error code is propagated back to the caller to ensure proper error handling. This change aligns with the existing error-handling behavior in functions like uniphier_aio_prepare(), where a failure in a sub-function should result in an immediate return of the error. Signed-off-by: Ingyu Jang <ingyujang25(a)unist.ac.kr> --- sound/soc/uniphier/aio-core.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 0eba60758134..2c4e8b873253 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -921,16 +921,19 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, { struct regmap *r = sub->aio->chip->regmap; u32 v; + int ret; if (sub->swm->dir != PORT_DIR_OUTPUT) return 0; - regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), + ret = regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), OPORTMXSRC1CTR_THMODE_SRC | OPORTMXSRC1CTR_SRCPATH_CALC | OPORTMXSRC1CTR_SYNC_ASYNC | OPORTMXSRC1CTR_FSIIPSEL_INNER | OPORTMXSRC1CTR_FSISEL_ACLK); + if (ret) + return ret; switch (params_rate(params)) { default: @@ -951,12 +954,18 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, break; } - regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), + + ret = regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), v | OPORTMXRATE_I_ACLKSRC_APLL | OPORTMXRATE_I_LRCKSTP_STOP); - regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), + if (ret) + return ret; + + ret = regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), OPORTMXRATE_I_LRCKSTP_MASK, OPORTMXRATE_I_LRCKSTP_START); + if (ret) + return ret; return 0; } -- 2.34.1

2 2

[PATCH v2] ALSA: hda/realtek: Fix the push button function for the ALC257
by Oder Chiou 04 Oct '24

04 Oct '24

The headset push button cannot work properly in case of the ALC257. This patch reverted the previous commit to correct the side effect. Fixes: ef9718b3d54e ("ALSA: hda/realtek: Fix noise from speakers on Lenovo IdeaPad 3 15IAU7") Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f787ff4182d4..86db1a4ec24c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -587,6 +587,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: case 0x10ec0283: case 0x10ec0285: -- 2.34.1

3 2

[PATCH v1] soundwire: Correct some typos in comments
by Shen Lichuan 03 Oct '24

03 Oct '24

Fixed some confusing spelling errors that were currently identified, the details are as follows: -in the code comments: drivers/soundwire/bus.c:115: enumuration==> enumeration drivers/soundwire/qcom.c:1176: wil ==> will drivers/soundwire/sysfs_slave.c:218: highligh ==> highlight Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- drivers/soundwire/bus.c | 2 +- drivers/soundwire/qcom.c | 2 +- drivers/soundwire/sysfs_slave.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 263ca32f0c5c..d1dc62c34f1c 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -112,7 +112,7 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, /* Set higher order bits */ *bus->assigned = ~GENMASK(SDW_BROADCAST_DEV_NUM, SDW_ENUM_DEV_NUM); - /* Set enumuration device number and broadcast device number */ + /* Set enumeration device number and broadcast device number */ set_bit(SDW_ENUM_DEV_NUM, bus->assigned); set_bit(SDW_BROADCAST_DEV_NUM, bus->assigned); diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aed57002fd0e..2b403b14066c 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1173,7 +1173,7 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, else sconfig.direction = SDW_DATA_DIR_RX; - /* hw parameters wil be ignored as we only support PDM */ + /* hw parameters will be ignored as we only support PDM */ sconfig.ch_count = 1; sconfig.frame_rate = params_rate(params); sconfig.type = stream->type; diff --git a/drivers/soundwire/sysfs_slave.c b/drivers/soundwire/sysfs_slave.c index f4259710dd0f..c5c22d1708ec 100644 --- a/drivers/soundwire/sysfs_slave.c +++ b/drivers/soundwire/sysfs_slave.c @@ -215,7 +215,7 @@ const struct attribute_group *sdw_attr_groups[] = { /* * the status is shown in capital letters for UNATTACHED and RESERVED - * on purpose, to highligh users to the fact that these status values + * on purpose, to highlight users to the fact that these status values * are not expected. */ static const char *const slave_status[] = { -- 2.17.1

2 1

[PATCH 0/4] refactor amd soundwire manager driver for acp 6.3 platform
by Vijendar Mukunda 03 Oct '24

03 Oct '24

This patch series refactors the amd soundwire manager driver to support acp 6.3 platform based on acp pci revision id. This patch series prepared on top of asoc tree for-next branch. This patch series should be applied at one go. @Vinod: Please help to review soundwire patches and provide ack-by for this patch series. Vijendar Mukunda (4): soundwire: amd: pass acp pci revision id as resource data soundwire: amd: refactor existing code for acp 6.3 platform ASoC: SOF: amd: pass acp_rev as soundwire resource data ASoC: amd: ps: pass acp pci revision id as soundwire resource data drivers/soundwire/amd_init.c | 1 + drivers/soundwire/amd_manager.c | 99 ++++++++++++++++++++----------- drivers/soundwire/amd_manager.h | 16 ++--- include/linux/soundwire/sdw_amd.h | 6 ++ sound/soc/amd/ps/acp63.h | 2 + sound/soc/amd/ps/pci-ps.c | 2 + sound/soc/sof/amd/acp.c | 1 + 7 files changed, 84 insertions(+), 43 deletions(-) -- 2.34.1

3 7

[PATCH] ASoC: intel: sof_sdw: Add check devm_kasprintf() returned value
by Charles Han 02 Oct '24

02 Oct '24

devm_kasprintf() can return a NULL pointer on failure but this returned value is not checked. Fixes: b359760d95ee ("ASoC: intel: sof_sdw: Add simple DAI link creation helper") Signed-off-by: Charles Han <hanchunchao(a)inspur.com> --- sound/soc/intel/boards/sof_sdw.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d258728d64cf..26917f6f15cf 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -919,6 +919,9 @@ static int create_ssp_dailinks(struct snd_soc_card *card, char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); char *codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", ssp_info->acpi_id, j++); + if (!name || !cpu_dai_name || !codec_name) + return -ENOMEM; + int playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; int capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; @@ -985,6 +988,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, for (i = 0; i < hdmi_num; i++) { char *name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1); char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1); + if (!name || !cpu_dai_name) + return -ENOMEM; + char *codec_name, *codec_dai_name; if (intel_ctx->hdmi.idisp_codec) { @@ -996,6 +1002,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, codec_dai_name = "snd-soc-dummy-dai"; } + if (!codec_dai_name) + return -ENOMEM; + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, 1, 0, // HDMI only supports playback cpu_dai_name, platform_component->name, @@ -1019,6 +1028,9 @@ static int create_bt_dailinks(struct snd_soc_card *card, SOF_BT_OFFLOAD_SSP_SHIFT; char *name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port); + if (!name || !cpu_dai_name) + return -ENOMEM; + int ret; ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, -- 2.31.1

3 2

[RFC PATCH v3 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 02 Oct '24

02 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… 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 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

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[PATCH] Fix possible NULL Pointer Dereference in 'asoc_qcom_lpass_cpu_platform_probe'
by Gax-c 01 Oct '24

01 Oct '24

A 'devm_kzalloc' in 'asoc_qcom_lpass_cpu_platform_probe' could possibly return NULL pointer. NULL Pointer Dereference may be triggerred in 'asoc_qcom_lpass_cpu_platform_probe' without addtional check. Add a null check for the returned pointer. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zijie Zhao <zzjas98(a)gmail.com> Reported-by: Chenyuan Yang <chenyuan0y(a)gmail.com> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5a47f661e0c6..a8e56f47f237 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1243,6 +1243,9 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); + if (!drvdata->i2sctl) + return -ENOMEM; + /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, drvdata->lpaif_map); -- 2.25.1

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[PATCH 0/3] ASoC: fsl_micfil: fix and improvement
by Shengjiu Wang 01 Oct '24

01 Oct '24

Fix the usage of regmap_write_bits(). Move mclk clock enablement to late stage. Enable the micfil error interrupt. Shengjiu Wang (3): ASoC: fsl_micfil: fix regmap_write_bits usage ASoC: fsl_micfil: Add mclk enable flag ASoC: fsl_micfil: Enable micfil error interrupt sound/soc/fsl/fsl_micfil.c | 49 ++++++++++++++++++++++++++++++-------- 1 file changed, 39 insertions(+), 10 deletions(-) -- 2.34.1

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[PATCH -next 0/5] sound/soc: fix some macro definitions and usages
by Hongbo Li 01 Oct '24

01 Oct '24

Many variable in macro are not used as we used macro_check script to detect and mamually check, let us address these issues. Hongbo Li (5): sound/soc: fix macro definition on TWL4030_OUTPUT_PGA sound/soc remove unused substream in macro soc_component_mark_pop sound/soc remove unused substream in macro soc_dai_mark_pop sound/soc remove unused substream in macro soc_link_mark_pop sound/soc: fix macro definition on STM_SAI_HAS_EXT_SYNC sound/soc/codecs/twl4030.c | 12 ++++++------ sound/soc/soc-component.c | 14 +++++++------- sound/soc/soc-dai.c | 10 +++++----- sound/soc/soc-link.c | 10 +++++----- sound/soc/stm/stm32_sai_sub.c | 2 +- 5 files changed, 24 insertions(+), 24 deletions(-) -- 2.34.1

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[PATCH v2 0/2] {ASoC,media}: constify snd_soc_component_driver when used as read-only
by Javier Carrasco 01 Oct '24

01 Oct '24

Declare `snd_soc_component_driver` as const to move their declarations to read-only sections for the drivers that do not modify the struct after its declaration. Apart from a single case under media/, the affected drivers are members of the ASoC subsystem. To: Tim Harvey <tharvey(a)gateworks.com> To: Mauro Carvalho Chehab <mchehab(a)kernel.org> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> To: Ray Jui <rjui(a)broadcom.com> To: Scott Branden <sbranden(a)broadcom.com> To: Broadcom internal kernel review list <bcm-kernel-feedback-list(a)broadcom.com> To: Shenghao Ding <shenghao-ding(a)ti.com> To: Kevin Lu <kevin-lu(a)ti.com> To: Baojun Xu <baojun.xu(a)ti.com> To: Olivier Moysan <olivier.moysan(a)foss.st.com> To: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> To: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> To: Alexandre Torgue <alexandre.torgue(a)foss.st.com> To: Kunihiko Hayashi <hayashi.kunihiko(a)socionext.com> To: Masami Hiramatsu <mhiramat(a)kernel.org> Cc: linux-media(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Signed-off-by: Javier Carrasco <javier.carrasco.cruz(a)gmail.com> Changes in v2: - drop cs43130 and sti-sas, as they modifiy the struct in the probe function. - Link to v1: https://lore.kernel.org/r/20240725-const_snd_soc_component_driver-v1-0-3d7e… --- Javier Carrasco (2): media: i2c: tda1997x: constify snd_soc_component_driver struct ASoC: constify snd_soc_component_driver struct drivers/media/i2c/tda1997x.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/pcm186x.c | 4 ++-- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/uniphier/evea.c | 2 +- 12 files changed, 13 insertions(+), 13 deletions(-) --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240725-const_snd_soc_component_driver-b9629a95b948 Best regards, -- Javier Carrasco <javier.carrasco.cruz(a)gmail.com>

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[PATCH V2 0/9] ASoC: amd: acp: refactor acp version differentiation logic
by Vijendar Mukunda 01 Oct '24

01 Oct '24

Currently different logics being used in the code for acp version differentiation. This patch series refactors the code to use acp pci revision id for handling acp version specific code. Changes since v1: - Add patch to update mach_params subsystem_rev field in machine select logic. Vijendar Mukunda (9): ASoC: amd: acp: simplify platform conditional checks code ASoC: amd: acp: use acp_rev for platform specific conditional checks ASoC: amd: acp: use acp pci revision id for platform differntiation ASoC: amd: acp: store acp pci rev id in platform driver private structure ASoC: amd: acp: pass acp pci revision id as platform data ASoC: amd: acp: update mach_params subsystem_rev field ASoC: amd: acp: remove unused variable from acp_card_drvdata structure ASoC: amd: acp: replace adata->platform conditional check ASoC: amd: acp: remove unused variable from acp platform driver sound/soc/amd/acp/acp-i2s.c | 38 +++++++++++++-------------- sound/soc/amd/acp/acp-legacy-common.c | 24 ++++++++--------- sound/soc/amd/acp/acp-legacy-mach.c | 8 +++--- sound/soc/amd/acp/acp-mach-common.c | 38 +++++++++++++++++++-------- sound/soc/amd/acp/acp-mach.h | 12 +++------ sound/soc/amd/acp/acp-pci.c | 7 +---- sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 15 ++++++----- sound/soc/amd/acp/acp-rembrandt.c | 4 +-- sound/soc/amd/acp/acp-renoir.c | 4 +-- sound/soc/amd/acp/acp-sof-mach.c | 6 ++--- sound/soc/amd/acp/acp63.c | 4 +-- sound/soc/amd/acp/acp70.c | 14 ++++------ sound/soc/amd/acp/acp_common.h | 19 ++++++++++++++ sound/soc/amd/acp/amd.h | 9 ++----- 15 files changed, 109 insertions(+), 95 deletions(-) create mode 100644 sound/soc/amd/acp/acp_common.h -- 2.34.1

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[PATCH v2 0/2] Updates for Atmel SSC DAI
by Andrei Simion 01 Oct '24

01 Oct '24

This patch set includes two updates for the Atmel SSC DAI driver: - Address the limitation with the S24_LE format. - Add stream names for DPCM and future use-cases. Codrin Ciubotariu (2): ASoC: atmel: atmel_ssc_dai: Add stream names ASoC: atmel: atmel_ssc_dai: Drop S24_LE support due to single channel limitation sound/soc/atmel/atmel_ssc_dai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) base-commit: 7083504315d64199a329de322fce989e1e10f4f7 -- 2.34.1

3 4

[PATCH 0/2] Updates for mchp-spdif(tx/rx)
by Andrei Simion 01 Oct '24

01 Oct '24

This patch set includes two updates for the MCHP SPDIF RX and TX drivers. The patches remove the interface name from the stream_name, allowing the interface name and index to be set in the Device Tree (DT) using the sound-name-prefix string property. Codrin Ciubotariu (2): ASoC: atmel: mchp-spdiftx: Remove interface name from stream_name ASoC: atmel: mchp-spdifrx: Remove interface name from stream_name sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) base-commit: 5acd9952f95fb4b7da6d09a3be39195a80845eb6 -- 2.34.1

2 3

[PATCH 0/8] ASoC: amd: acp: refactor acp version differentiation logic
by Vijendar Mukunda 01 Oct '24

01 Oct '24

Currently different logics being used in the code for acp version differentiation. This patch series refactors the code to use acp pci revision id for handling acp version specific code. Vijendar Mukunda (8): ASoC: amd: acp: simplify platform conditional checks code ASoC: amd: acp: use acp_rev for platform specific conditional checks ASoC: amd: acp: use acp pci revision id for platform differntiation ASoC: amd: acp: store acp pci rev id in platform driver private structure ASoC: amd: acp: pass acp pci revision id as platform data ASoC: amd: acp: remove unused variable from acp_card_drvdata structure ASoC: amd: acp: replace adata->platform conditional check ASoC: amd: acp: remove unused variable from acp platform driver sound/soc/amd/acp/acp-i2s.c | 38 +++++++++++++-------------- sound/soc/amd/acp/acp-legacy-common.c | 24 ++++++++--------- sound/soc/amd/acp/acp-legacy-mach.c | 8 +++--- sound/soc/amd/acp/acp-mach-common.c | 38 +++++++++++++++++++-------- sound/soc/amd/acp/acp-mach.h | 12 +++------ sound/soc/amd/acp/acp-pci.c | 7 +---- sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 14 +++++----- sound/soc/amd/acp/acp-rembrandt.c | 4 +-- sound/soc/amd/acp/acp-renoir.c | 4 +-- sound/soc/amd/acp/acp-sof-mach.c | 6 ++--- sound/soc/amd/acp/acp63.c | 4 +-- sound/soc/amd/acp/acp70.c | 14 ++++------ sound/soc/amd/acp/acp_common.h | 19 ++++++++++++++ sound/soc/amd/acp/amd.h | 9 ++----- 15 files changed, 108 insertions(+), 95 deletions(-) create mode 100644 sound/soc/amd/acp/acp_common.h -- 2.34.1

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[PATCH v28 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 01 Oct '24

01 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: - 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 (2): 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 | 456 ++++ 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 | 31 +- 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 | 47 +- drivers/usb/host/xhci.h | 16 +- .../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 | 139 ++ 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 | 342 +++ 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 | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5870 insertions(+), 55 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

3 41

[PATCH] ASoC: fsl_sai: Enable 'FIFO continue on error' FCONT bit
by Shengjiu Wang 01 Oct '24

01 Oct '24

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> --- 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 ab58a4461073..634168d2bb6e 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 dadbd16ee394..9c4d19fe22c6 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) -- 2.34.1

2 1

[PATCH alsa-ucm-conf 1/2] sof-soundwire: Change mapped control names to make them unique
by Maciej Strozek 01 Oct '24

01 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> --- 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..7c36655 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 Combined Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Combined 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 Combined" } } 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

3 5

[PATCH v2 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 01 Oct '24

01 Oct '24

Hi Mark This is v2 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. v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r0979uhx.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 15

[PATCH] sound: soc: codecs: tlv320aic3x: Fix codec gpio-reset
by Benjamin Hahn 01 Oct '24

01 Oct '24

The TLV320AIC3007 requires a hardware reset after power-up for proper operation. After all power supplies are at their specified values, the RESET pin must be driven low for at least 10 ns. Even though the datasheet sais min 10ns, I needed more than 10ns when testing this out. 15ns worked for me. Signed-off-by: Benjamin Hahn <B.Hahn(a)phytec.de> --- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 56e795a00e22..d002fc0b99b5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1367,8 +1367,11 @@ static int aic3x_regulator_event(struct notifier_block *nb, * Put codec to reset and require cache sync as at least one * of the supplies was disabled */ - if (aic3x->gpio_reset) + if (aic3x->gpio_reset) { gpiod_set_value(aic3x->gpio_reset, 1); + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + } regcache_mark_dirty(aic3x->regmap); } @@ -1813,6 +1816,10 @@ int aic3x_probe(struct device *dev, struct regmap *regmap, kernel_ulong_t driver gpiod_set_consumer_name(aic3x->gpio_reset, "tlv320aic3x reset"); + /* CODEC datasheet says minimum of 10ns */ + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; --- base-commit: ad46e8f95e931e113cb98253daf6d443ac244cde change-id: 20240930-wip-bhahn-tlv320aic3x_fix_reset-cca918a4889d Best regards, -- Benjamin Hahn <B.Hahn(a)phytec.de>

2 1

[PATCH] ASoC: fsl: Use maple tree register cache
by Mark Brown 01 Oct '24

01 Oct '24

Several of the NXP drivers use regmaps with a rbtree register cache. Since the maple tree cache is uisng a generally more modern data structure which makes implementation choices more suitable for modern systems let's convert these drivers to it. This should have no practical impact. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index a6cbaa6364c7..910b4ce7671c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -169,7 +169,7 @@ static const struct regmap_config fsl_aud2htx_regmap_config = { .readable_reg = fsl_aud2htx_readable_reg, .volatile_reg = fsl_aud2htx_volatile_reg, .writeable_reg = fsl_aud2htx_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; static const struct of_device_id fsl_aud2htx_dt_ids[] = { diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 962f30912091..5149e732f763 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1748,7 +1748,7 @@ static const struct regmap_config fsl_easrc_regmap_config = { .rd_table = &fsl_easrc_readable_table, .wr_table = &fsl_easrc_writeable_table, .volatile_table = &fsl_easrc_volatileable_table, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; #ifdef DEBUG diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 22b240a70ad4..c0dc9bebb77b 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -955,7 +955,7 @@ static const struct regmap_config fsl_micfil_regmap_config = { .readable_reg = fsl_micfil_readable_reg, .volatile_reg = fsl_micfil_volatile_reg, .writeable_reg = fsl_micfil_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; /* END OF REGMAP */ --- base-commit: 98f7e32f20d28ec452afb208f9cffc08448a2652 change-id: 20240924-asoc-imx-maple-abb62f013533 Best regards, -- Mark Brown <broonie(a)kernel.org>

3 3

[PATCH] ASoC: meson: axg-iface: set continuous rates
by Jerome Brunet 01 Oct '24

01 Oct '24

The axg TDM HW does not depend on a selected set of rates. The hardware itself, just takes an input clock and work with it, regardless of its rate. In this way, the rates TDM can take are continuous. What might force the use of specific rate are the PLL available as clock and/or the codecs facing the TDM HW. Either way, this constraint does not belong in the TDM interface driver. Allow any rate as far as TDM is concerned by setting SNDRV_PCM_RATE_CONTINUOUS with an interval it has been tested with. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 12 +++++++++--- sound/soc/meson/axg-tdm.h | 2 -- 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 62057c71f742..09103eef2a97 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -442,14 +442,18 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Playback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_PAD, @@ -461,7 +465,9 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Loopback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_LOOPBACK, diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index 1a17f546ce6e..acfcd48f8a00 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -15,8 +15,6 @@ #define AXG_TDM_NUM_LANES 4 #define AXG_TDM_CHANNEL_MAX 128 -#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_768000) #define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ --- base-commit: cd87a98b53518e44cf3c1a7c1c07c869ce33bf83 change-id: 20240920-asoc-axg-iface-continuous-b94cfa234787 Best regards, -- Jerome

2 1

[PATCH] ASoC: tas2781: Fix redundant parameter assignment
by Tang Bin 01 Oct '24

01 Oct '24

In these functions, the variable 'rc' is redundant, thus remove it. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/tas2781-i2c.c | 35 ++++++++-------------------------- 1 file changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a8d97dd7..68887799e 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -650,7 +650,6 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; @@ -659,9 +658,7 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, @@ -673,7 +670,6 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG; @@ -681,9 +677,7 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, reg = TAS2563_RUNTIME_RE_REG; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, @@ -696,7 +690,6 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); @@ -707,9 +700,7 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, else return -1; dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, @@ -721,13 +712,10 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, @@ -739,13 +727,10 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_nop_get( @@ -1115,14 +1100,12 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); int dev_id = ucontrol->value.integer.value[0]; - int max = tas_priv->ndev - 1, rc; + int max = tas_priv->ndev - 1; dev_id = clamp(dev_id, 0, max); guard(mutex)(&tas_priv->codec_lock); - rc = tasdev_chn_switch(tas_priv, dev_id); - - return rc; + return tasdev_chn_switch(tas_priv, dev_id); } static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) @@ -1339,10 +1322,8 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) i++; } - rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, + return snd_soc_add_component_controls(priv->codec, cali_ctrls, nctrls < i ? nctrls : i); - - return rc; } static void tasdevice_fw_ready(const struct firmware *fmw, -- 2.33.0

2 1

[PATCH v2] ALSA: hda/conexant: fix some typos
by Oldherl Oh 30 Sep '24

30 Sep '24

Fix some typos in patch_conexant.c Signed-off-by: Oldherl Oh <me(a)oldherl.one> --- sound/pci/hda/patch_conexant.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e851785ff058..ade42a8209c2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -166,18 +166,18 @@ static void cxt_init_gpio_led(struct hda_codec *codec) static void cx_fixup_headset_recog(struct hda_codec *codec) { - unsigned int mic_persent; + unsigned int mic_present; /* fix some headset type recognize fail issue, such as EDIFIER headset */ - /* set micbiasd output current comparator threshold from 66% to 55%. */ + /* set micbias output current comparator threshold from 66% to 55%. */ snd_hda_codec_write(codec, 0x1c, 0, 0x320, 0x010); - /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias registor + /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias register * value adjustment trim from 2.2K ohms to 2.0K ohms. */ snd_hda_codec_write(codec, 0x1c, 0, 0x3b0, 0xe10); /* fix reboot headset type recognize fail issue */ - mic_persent = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); - if (mic_persent & AC_PINSENSE_PRESENCE) + mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); + if (mic_present & AC_PINSENSE_PRESENCE) /* enable headset mic VREF */ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24); else @@ -249,9 +249,9 @@ static void cx_update_headset_mic_vref(struct hda_codec *codec, struct hda_jack_ { unsigned int mic_present; - /* In cx8070 and sn6140, the node 16 can only be config to headphone or disabled, - * the node 19 can only be config to microphone or disabled. - * Check hp&mic tag to process headset pulgin&plugout. + /* In cx8070 and sn6140, the node 16 can only be configured to headphone or disabled, + * the node 19 can only be configured to microphone or disabled. + * Check hp&mic tag to process headset plugin & plugout. */ mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); if (!(mic_present & AC_PINSENSE_PRESENCE)) /* mic plugout */ -- 2.46.2

2 1

ucm2: USB-Audio: add Steinberg UR22C (USB0499:172f)
by GitHub pull_request - opened 30 Sep '24

30 Sep '24

alsa-project/alsa-ucm-conf pull request #446 was opened from CoelacanthusHex: Similar to UR24C and UR44C, but with 2 inputs 2 outputs. Signed-off-by: Celeste Liu <CoelacanthusHex(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/446 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/446.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: Fix the push button function for the ALC257
by Oder Chiou 30 Sep '24

30 Sep '24

The patch "Fix noise from speakers on Lenovo IdeaPad 3 15IAU7" caused that the headset push button cannot work properly in case of ALC257. This patch reverted it to correct the side effect. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f787ff4182d4..86db1a4ec24c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -587,6 +587,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: case 0x10ec0283: case 0x10ec0285: -- 2.34.1

2 1

Fix some typos in hda/conexant
by Oldherl Oh 30 Sep '24

30 Sep '24

I'm trying to understand Conexant driver code in order to solve my microphone detection problem. As I am new to kernel development, I have made this simple typo-fixing-only patch as an exercise. Hope it is not disturbing.

2 2

[PATCH v2] ALSA: Fix typos in comments across various files
by Yu Jiaoliang 30 Sep '24

30 Sep '24

This patch fixes typos in comments within the ALSA subsystem. These changes improve code readability without affecting functionality. Signed-off-by: Yu Jiaoliang <yujiaoliang(a)vivo.com> --- v2: Update commit message. --- sound/core/compress_offload.c | 2 +- sound/core/oss/rate.c | 2 +- sound/core/pcm_native.c | 2 +- sound/core/sound.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b8c0d6edbdd1..bdf1d78de833 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -288,7 +288,7 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; guard(mutex)(&stream->device->lock); - /* write is allowed when stream is running or has been steup */ + /* write is allowed when stream is running or has been setup */ switch (stream->runtime->state) { case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 98269119347f..b56eeda5e30e 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -294,7 +294,7 @@ static int rate_action(struct snd_pcm_plugin *plugin, default: break; } - return 0; /* silenty ignore other actions */ + return 0; /* silently ignore other actions */ } int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug, diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 99e39b5359cc..b49bc89c0758 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3115,7 +3115,7 @@ struct snd_pcm_sync_ptr32 { } c; } __packed; -/* recalcuate the boundary within 32bit */ +/* recalculate the boundary within 32bit */ static snd_pcm_uframes_t recalculate_boundary(struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t boundary; diff --git a/sound/core/sound.c b/sound/core/sound.c index b9db9aa0bfcb..6531a67f13b3 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -133,7 +133,7 @@ static struct snd_minor *autoload_device(unsigned int minor) /* /dev/aloadSEQ */ snd_request_other(minor); } - mutex_lock(&sound_mutex); /* reacuire lock */ + mutex_lock(&sound_mutex); /* reacquire lock */ return snd_minors[minor]; } #else /* !CONFIG_MODULES */ -- 2.34.1

2 1

[PATCH v1] slimbus: Correct some typos in comments
by Shen Lichuan 29 Sep '24

29 Sep '24

Fixed some confusing typos that were currently identified with codespell, the details are as follows: -in the code comments: drivers/slimbus/messaging.c:226: transfered ==> transferred drivers/slimbus/messaging.c:230: vlaue ==> value drivers/slimbus/qcom-ctrl.c:127: Oder ==> Order drivers/slimbus/slimbus.h:263: Presense ==> Presence drivers/slimbus/slimbus.h:371: resposible ==> responsible Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- drivers/slimbus/messaging.c | 4 ++-- drivers/slimbus/qcom-ctrl.c | 2 +- drivers/slimbus/slimbus.h | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/slimbus/messaging.c b/drivers/slimbus/messaging.c index 242570a5e565..b7dd2badc9d1 100644 --- a/drivers/slimbus/messaging.c +++ b/drivers/slimbus/messaging.c @@ -223,11 +223,11 @@ static u16 slim_slicesize(int code) /** * slim_xfer_msg() - Transfer a value info message on slim device * - * @sbdev: slim device to which this msg has to be transfered + * @sbdev: slim device to which this msg has to be transferred * @msg: value info message pointer * @mc: message code of the message * - * Called by drivers which want to transfer a vlaue or info elements. + * Called by drivers which want to transfer a value or info elements. * * Return: -ETIMEDOUT: If transmission of this message timed out */ diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index e25f9bdd9b23..039384d85e7b 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -124,7 +124,7 @@ static void qcom_slim_queue_tx(struct qcom_slim_ctrl *ctrl, void *buf, __iowrite32_copy(ctrl->base + tx_reg, buf, count); - /* Ensure Oder of subsequent writes */ + /* Ensure Order of subsequent writes */ mb(); } diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 00a7f112574b..2067d47541f2 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -260,7 +260,7 @@ enum slim_ch_aux_bit_fmt { * struct slim_channel - SLIMbus channel, used for state machine * * @id: ID of channel - * @prrate: Presense rate of channel from Table 66 of SLIMbus 2.0 Specs + * @prrate: Presence rate of channel from Table 66 of SLIMbus 2.0 Specs * @seg_dist: segment distribution code from Table 20 of SLIMbus 2.0 Specs * @data_fmt: Data format of channel. * @aux_fmt: Aux format for this channel. @@ -368,7 +368,7 @@ struct slim_stream_runtime { * @get_laddr: It is possible that controller needs to set fixed logical * address table and get_laddr can be used in that case so that controller * can do this assignment. Use case is when the master is on the remote - * processor side, who is resposible for allocating laddr. + * processor side, who is responsible for allocating laddr. * @wakeup: This function pointer implements controller-specific procedure * to wake it up from clock-pause. Framework will call this to bring * the controller out of clock pause. -- 2.17.1

1 0

ucm2: Add Roland Bridge Cast V2
by GitHub pull_request - edited 28 Sep '24

28 Sep '24

alsa-project/alsa-ucm-conf pull request #445 was edited from Fumler: Roland Bridge Cast recently had a firmware update which brings in e.g. virtual surround and is dubbed as "V2" and has a separate usb device id than the existing Bridge Cast in the repo. This PR adds the changes needed for the new firmware to be mapped correctly, as a new USB audio device. Changelog: https://www.roland.com/global/support/by_product/bridge_cast/updates_driver… Thanks to [geoffreybennett](https://old.reddit.com/user/geoffreybennett) on reddit for helping me understand how to debug this stuff. Thread: https://old.reddit.com/r/linuxaudio/comments/1fq06z8/roland_bridge_cast_no_… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/445 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/445.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add Roland Bridge Cast V2
by GitHub pull_request - opened 28 Sep '24

28 Sep '24

alsa-project/alsa-ucm-conf pull request #445 was opened from Fumler: Roland Bridge Cast recently had a firmware update which brings in e.g. virtual surround and is dubbed as "V2" and has a separate usb device id than the existing Bridge Cast in the repo. This PR adds the changes needed for the new firmware to be mapped correctly, as a new USB audio device. Changelog: https://www.roland.com/global/support/by_product/bridge_cast/updates_driver… Thanks to [geoffreybennett](https://old.reddit.com/user/geoffreybennett) on reddit for helping me understand how to debug this stuff. Thread: https://old.reddit.com/r/linuxaudio/comments/1fq06z8/roland_bridge_cast_no_… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/445 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/445.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No sound from XonarDX
by GitHub issues - reopened 28 Sep '24

28 Sep '24

alsa-project/alsa-lib issue #408 was reopened from 3bitbrain: No matter what wiki or advice I try, no sound at all from a known good Xonar DX card. Distro : Gentoo, fresh install. Always un-muted everything in the console alsamixer. alsa-lib version 1.2.11 [alsa.txt](https://github.com/user-attachments/files/16697602/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-lib/issues/408 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param()
by (학생) 장인규 (컴퓨터공학과) 26 Sep '24

26 Sep '24

From 791716bf359b8540c519810848fd1f8006d7c3c5 Mon Sep 17 00:00:00 2001 From: Ingyu Jang <ingyujang25(a)unist.ac.kr> Date: Thu, 26 Sep 2024 19:40:04 +0900 Subject: [PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param() The aio_src_set_param() function did not previously check the return values of regmap_write() and regmap_update_bits(). If these functions fail, it could lead to silent failures when configuring the sample rate converter (SRC), causing improper behavior in audio processing without any indication of an error. This patch modifies aio_src_set_param to check the return values of regmap_write() and regmap_update_bits(). If either function returns an error, the error code is propagated back to the caller to ensure proper error handling. This change aligns with the existing error-handling behavior in functions like uniphier_aio_prepare(), where a failure in a sub-function should result in an immediate return of the error. Signed-off-by: Ingyu Jang <ingyujang25(a)unist.ac.kr> --- sound/soc/uniphier/aio-core.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 0eba60758134..2c4e8b873253 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -921,16 +921,19 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, { struct regmap *r = sub->aio->chip->regmap; u32 v; + int ret; if (sub->swm->dir != PORT_DIR_OUTPUT) return 0; - regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), + ret = regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), OPORTMXSRC1CTR_THMODE_SRC | OPORTMXSRC1CTR_SRCPATH_CALC | OPORTMXSRC1CTR_SYNC_ASYNC | OPORTMXSRC1CTR_FSIIPSEL_INNER | OPORTMXSRC1CTR_FSISEL_ACLK); + if (ret) + return ret; switch (params_rate(params)) { default: @@ -951,12 +954,18 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, break; } - regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), + + ret = regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), v | OPORTMXRATE_I_ACLKSRC_APLL | OPORTMXRATE_I_LRCKSTP_STOP); - regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), + if (ret) + return ret; + + ret = regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), OPORTMXRATE_I_LRCKSTP_MASK, OPORTMXRATE_I_LRCKSTP_START); + if (ret) + return ret; return 0; } -- 2.34.1

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GoXLR - Dynamic Channel Count
by GitHub issues - edited 26 Sep '24

26 Sep '24

alsa-project/alsa-ucm-conf issue #444 was edited from FrostyCoolSlug: The next GoXLR firmware release is going to add two new output channels for the full sized device, labeled a 'Stream Mix 2'. Outside of general usage, there's no way to determine whether these channels exist or not outside of the UCM2 definitions as the device is booting up. Is there currently the facility to configure the devices virtual channels based on the number of channels reported by the hardware (so if there are 23 hardware channels, use the current configuration, if there are 25 hardware channels use an adjusted configuration), and if not, could you suggest the best way to handle this implementation? Regards, Craig McLure Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/444 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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GoXLR - Dynamic Output Channel Count
by GitHub issues - opened 26 Sep '24

26 Sep '24

alsa-project/alsa-ucm-conf issue #444 was opened from FrostyCoolSlug: The next GoXLR firmware release is going to add two new output channels for the full sized device, labeled a 'Stream Mix 2'. Outside of general usage, there's no way to determine whether these channels exist or not outside of the UCM2 definitions as the device is booting up. Is there currently the facility to configure the devices virtual channels based on the number of channels reported by the hardware (so if there are 23 hardware channels, use the current configuration, if there are 25 hardware channels use an adjusted configuration), and if not, could you suggest the best way to handle this implementation? Regards, Craig McLure Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/444 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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