December 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: qcom: x1e80100: Support boards with two speakers
by Krzysztof Kozlowski 17 Jul '25

17 Jul '25

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

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[alsa-devel] ASoC TDM channel re-mapping
by Edward Cragg 17 Jun '25

17 Jun '25

Hi, We have an ASoC driver in which we have added support for TDM over an I2S interface. We would like to have the capability of re-mapping the channel ordering, for which i know there is a callback in place to start this process, set_channel_map. The only backend implementation of this is using the ALSA API and the .copy handler in an ALSA PCM driver to mutate data copied out of ALSA before it's DMA'd to hardware. This is in sound/soc/blackfin/bf5xx-i2s-pcm.c The problem we have is that the driver we're using currently uses a dmaengine based PCM driver, and i'm not sure whether this is compatible with using the .copy callback, without reimplementing the driver entirely as an ALSA PCM driver. Since there aren't very many examples in mainline, I was hoping to ask for advice for the best way of doing this. I wondered: - Would it be possible to add a functional copy callback using a dmaengine based PCM driver? - Otherwise, would there be a good way to implement channel remapping in a TDM frame using the dmaengine based PCM driver? or - Would the best way indeed be to reimplement the PCM driver as a normal ALSA PCM driver? Thanks, Ed Cragg

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[PATCH] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 17 Jun '25

17 Jun '25

Add support for controlling soundwire audio CGCR interface using clock framework to make hclk ungating with software. As per new hardware changes, software has to always ungate hclk if soundwire is operational and keep it running. This requirement is for latest LPASS chipsets for RX, TX and WSA path to work. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- This patch set depends on: -- Clock driver patches with reset control support, yet to be posted. drivers/soundwire/qcom.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index fb183bd..440aad8 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -12,6 +12,7 @@ #include <linux/of_irq.h> #include <linux/of_device.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <linux/soundwire/sdw.h> @@ -129,6 +130,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + struct reset_control *audio_cgcr; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs; #endif @@ -618,6 +620,8 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1269,6 +1273,9 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.ops = &qcom_swrm_ops; ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); + if (IS_ERR(ctrl->audio_cgcr)) + dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); ret = qcom_swrm_get_port_config(ctrl); if (ret) -- 2.7.4

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[PATCH] ASoC: amd: acp: Fix support for a Huawei Matebook laptop
by Marian Postevca 17 Jun '25

17 Jun '25

Previous commit that added support for Huawei MateBook D16 2021 with Ryzen 4600H (HVY-WXX9 M1010) was incomplete. To activate support for this laptop, the DMI table in acp3x-es83xx machine driver must also be updated. Fixes: b5338b1b901e ("ASoC: amd: acp: Add support for a new Huawei Matebook laptop") Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c index f85b85ea4be9..2b0aa270a3e9 100644 --- a/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c +++ b/sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c @@ -354,6 +354,14 @@ static const struct dmi_system_id acp3x_es83xx_dmi_table[] = { }, .driver_data = (void *)(ES83XX_ENABLE_DMIC|ES83XX_48_MHZ_MCLK), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "HVY-WXX9"), + DMI_EXACT_MATCH(DMI_PRODUCT_VERSION, "M1010"), + }, + .driver_data = (void *)(ES83XX_ENABLE_DMIC), + }, { .matches = { DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "HUAWEI"), -- 2.43.0

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[PATCH 0/1] Swapped l/r channels on ThinkBook 13x Gen4
by MadEnvel 28 May '25

28 May '25

Earlier commits introduced audio support for Lenovo ThinBook 13x Gen 4 that needed a patch due to missing _DSD config. Unfortunately, it looks like the left and right channels in the config are swapped. This patch fixes channel assignment. Tested on Lenovo ThinkBook 13x Gen4. MadEnvel (1): Fix swapped left and right channels for Lenovo ThinBook 13x Gen4 sound sound/pci/hda/cs35l41_hda_property.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) -- 2.45.2

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[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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Ask snd_seq_set_client_midi_version
by Correo Alternativo 17 Feb '25

17 Feb '25

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

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

15 Jan '25

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

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

13 Jan '25

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

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