April 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: rsnd: use inclusive language for DAIFMT mask
by Kuninori Morimoto 06 Apr '22

06 Apr '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> It is still using discriminatory terms on DAIFMT mask. This patch tidyup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a8fe0da7670..0c435bff3670 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -755,7 +755,7 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); /* set clock master for audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { case SND_SOC_DAIFMT_CBP_CFP: rdai->clk_master = 0; break; -- 2.25.1

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[PATCH] ASoC: mediatek: Fix error handling in mt8173_max98090_dev_probe
by Miaoqian Lin 06 Apr '22

06 Apr '22

Call of_node_put(platform_node) to avoid refcount leak in the error path. Fixes: 94319ba10eca ("ASoC: mediatek: Use platform_of_node for machine drivers") Fixes: 493433785df0 ("ASoC: mediatek: mt8173: fix device_node leak") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mediatek/mt8173/mt8173-max98090.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index 4cb90da89262..58778cd2e61b 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -167,7 +167,8 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) if (!codec_node) { dev_err(&pdev->dev, "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; + ret = -EINVAL; + goto put_platform_node; } for_each_card_prelinks(card, i, dai_link) { if (dai_link->codecs->name) @@ -179,6 +180,8 @@ static int mt8173_max98090_dev_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); of_node_put(codec_node); + +put_platform_node: of_node_put(platform_node); return ret; } -- 2.17.1

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[PATCH 0/5] ASoC: Add a driver for the Cirrus Logic CS35L45 Smart Amplifier
by Richard Fitzgerald 06 Apr '22

06 Apr '22

This adds basic audio support for the Cirrus Logic CS35L45 amplifier. The first two patches add two generic helpers to ASoC, and patch 3 is a kunit test for patch 2. James Schulman (1): ASoC: cs35l45: Add driver for Cirrus Logic CS35L45 Smart Amp Richard Fitzgerald (4): ASoC: soc.h: Add SOC_SINGLE_S_TLV() macro ASoC: soc-utils: Add helper to calculate BCLK from TDM info ASoC: soc-utils: Add kunit test for snd_soc_tdm_params_to_bclk() ASoC: dt-bindings: cs35l45: Cirrus Logic CS35L45 Smart Amp .../bindings/sound/cirrus,cs35l45.yaml | 75 ++ MAINTAINERS | 2 + include/dt-bindings/sound/cs35l45.h | 20 + include/sound/soc.h | 4 + sound/soc/Kconfig | 9 +- sound/soc/Makefile | 5 + sound/soc/codecs/Kconfig | 30 + sound/soc/codecs/Makefile | 8 + sound/soc/codecs/cs35l45-i2c.c | 73 ++ sound/soc/codecs/cs35l45-spi.c | 72 ++ sound/soc/codecs/cs35l45-tables.c | 202 +++++ sound/soc/codecs/cs35l45.c | 689 ++++++++++++++++++ sound/soc/codecs/cs35l45.h | 213 ++++++ sound/soc/soc-utils-test.c | 186 +++++ sound/soc/soc-utils.c | 45 ++ 15 files changed, 1632 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml create mode 100644 include/dt-bindings/sound/cs35l45.h create mode 100644 sound/soc/codecs/cs35l45-i2c.c create mode 100644 sound/soc/codecs/cs35l45-spi.c create mode 100644 sound/soc/codecs/cs35l45-tables.c create mode 100644 sound/soc/codecs/cs35l45.c create mode 100644 sound/soc/codecs/cs35l45.h create mode 100644 sound/soc/soc-utils-test.c -- 2.30.2

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[PATCH] ALSA: hda/tegra: Fix hda Jack detection
by Mohan Kumar 06 Apr '22

06 Apr '22

Tegra HDA Jack detection logic doesn't work when the HDACODEC in runtime suspended state as unsol event won't be triggered during D3 state. As pulseaudio server in userspace rely on the jack mixer control status to show the audio devices in gui and any display sink device hotplug event during D3 state will never updates the jack status which will result in no audio device option available in userspace settings. The possible option available to resolve this issue for multiple tegra platforms is to use Jack polling method for every 5 seconds. Also to make Jack detection work seamlessly the Jack worker thread needs to run continuously after HDA sound card registered irrespective of whether HDMI sink device connected or not, but the Jack state update call happens only when Codec is not powered on. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 43 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 43 insertions(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 2347d0304f93..c92938a47b65 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -28,6 +28,7 @@ #include <sound/hda_codec.h> #include "hda_controller.h" +#include "hda_jack.h" /* Defines for Nvidia Tegra HDA support */ #define HDA_BAR0 0x8000 @@ -67,6 +68,7 @@ * is used to update the GCAP register to workaround the issue. */ #define TEGRA194_NUM_SDO_LINES 4 +#define JACKPOLL_INTERVAL msecs_to_jiffies(5000) struct hda_tegra_soc { bool has_hda2codec_2x_reset; @@ -82,6 +84,7 @@ struct hda_tegra { unsigned int nclocks; void __iomem *regs; struct work_struct probe_work; + struct delayed_work jack_work; const struct hda_tegra_soc *soc; }; @@ -127,8 +130,11 @@ static void hda_tegra_init(struct hda_tegra *hda) static int __maybe_unused hda_tegra_suspend(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); int rc; + cancel_delayed_work_sync(&hda->jack_work); rc = pm_runtime_force_suspend(dev); if (rc < 0) return rc; @@ -140,6 +146,8 @@ static int __maybe_unused hda_tegra_suspend(struct device *dev) static int __maybe_unused hda_tegra_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); int rc; rc = pm_runtime_force_resume(dev); @@ -147,6 +155,8 @@ static int __maybe_unused hda_tegra_resume(struct device *dev) return rc; snd_power_change_state(card, SNDRV_CTL_POWER_D0); + schedule_delayed_work(&hda->jack_work, JACKPOLL_INTERVAL); + return 0; } @@ -209,6 +219,29 @@ static const struct dev_pm_ops hda_tegra_pm = { NULL) }; +static void hda_tegra_jack_work(struct work_struct *work) +{ + struct hda_tegra *hda = + container_of(work, struct hda_tegra, jack_work.work); + struct azx *chip = &hda->chip; + struct hda_codec *codec; + + if (!chip->running) + return; + + list_for_each_codec(codec, &chip->bus) { + if (snd_hdac_is_power_on(&codec->core)) + continue; + + snd_hda_power_up_pm(codec); + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_poll_all(codec); + snd_hda_power_down_pm(codec); + } + + schedule_delayed_work(&hda->jack_work, JACKPOLL_INTERVAL); +} + static int hda_tegra_dev_disconnect(struct snd_device *device) { struct azx *chip = device->device_data; @@ -226,6 +259,7 @@ static int hda_tegra_dev_free(struct snd_device *device) struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); cancel_work_sync(&hda->probe_work); + cancel_delayed_work_sync(&hda->jack_work); if (azx_bus(chip)->chip_init) { azx_stop_all_streams(chip); azx_stop_chip(chip); @@ -428,6 +462,7 @@ static int hda_tegra_create(struct snd_card *card, chip->snoop = true; INIT_WORK(&hda->probe_work, hda_tegra_probe_work); + INIT_DELAYED_WORK(&hda->jack_work, hda_tegra_jack_work); err = azx_bus_init(chip, NULL); if (err < 0) @@ -574,13 +609,18 @@ static void hda_tegra_probe_work(struct work_struct *work) out_free: pm_runtime_put(hda->dev); + schedule_delayed_work(&hda->jack_work, JACKPOLL_INTERVAL); return; /* no error return from async probe */ } static int hda_tegra_remove(struct platform_device *pdev) { + struct snd_card *card = dev_get_drvdata(&pdev->dev); + struct azx *chip = card->private_data; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); int ret; + cancel_delayed_work_sync(&hda->jack_work); ret = snd_card_free(dev_get_drvdata(&pdev->dev)); pm_runtime_disable(&pdev->dev); @@ -591,10 +631,13 @@ static void hda_tegra_shutdown(struct platform_device *pdev) { struct snd_card *card = dev_get_drvdata(&pdev->dev); struct azx *chip; + struct hda_tegra *hda; if (!card) return; chip = card->private_data; + hda = container_of(chip, struct hda_tegra, chip); + cancel_delayed_work_sync(&hda->jack_work); if (chip && chip->running) azx_stop_chip(chip); } -- 2.17.1

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[PATCH v4] arm64: dts: imx8mn-bsh-smm-s2pro: Add tlv320aic31xx audio card node
by Ariel D'Alessandro 06 Apr '22

06 Apr '22

BSH SystemMaster (SMM) S2 PRO board comes with an audio card based on tlv320aic31xx family codec. The audio card exposes two playback devices, one of them using the EASRC (Enhanced Asynchronous Sample Rate Converter) module. Note that this would require SDMA and EASRC firmware in order to work. Signed-off-by: Ariel D'Alessandro <ariel.dalessandro(a)collabora.com> Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- .../dts/freescale/imx8mn-bsh-smm-s2pro.dts | 90 +++++++++++++++++++ 1 file changed, 90 insertions(+) diff --git a/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts b/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts index c6a8ed6745c1..fbbb3367037b 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts +++ b/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts @@ -7,6 +7,7 @@ /dts-v1/; #include "imx8mn-bsh-smm-s2-common.dtsi" +#include <dt-bindings/sound/tlv320aic31xx.h> / { model = "BSH SMM S2 PRO"; @@ -16,6 +17,65 @@ memory@40000000 { device_type = "memory"; reg = <0x0 0x40000000 0x0 0x20000000>; }; + + sound-tlv320aic31xx { + compatible = "fsl,imx-audio-tlv320aic31xx"; + model = "tlv320aic31xx-hifi"; + audio-cpu = <&sai3>; + audio-codec = <&tlv320dac3101>; + audio-asrc = <&easrc>; + audio-routing = + "Ext Spk", "SPL", + "Ext Spk", "SPR"; + mclk-id = <PLL_CLKIN_BCLK>; + }; + + vdd_input: vdd_input { + compatible = "regulator-fixed"; + regulator-name = "vdd_input"; + regulator-min-microvolt = <5000000>; + regulator-max-microvolt = <5000000>; + }; +}; + +&easrc { + fsl,asrc-rate = <48000>; + fsl,asrc-format = <10>; + status = "okay"; +}; + +&i2c2 { + clock-frequency = <400000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2c2>; + status = "okay"; + + tlv320dac3101: audio-codec@18 { + compatible = "ti,tlv320dac3101"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_dac_rst>; + reg = <0x18>; + #sound-dai-cells = <0>; + HPVDD-supply = <&buck4_reg>; + SPRVDD-supply = <&vdd_input>; + SPLVDD-supply = <&vdd_input>; + AVDD-supply = <&buck4_reg>; + IOVDD-supply = <&buck4_reg>; + DVDD-supply = <&buck5_reg>; + reset-gpios = <&gpio1 6 GPIO_ACTIVE_LOW>; + ai31xx-micbias-vg = <MICBIAS_AVDDV>; + clocks = <&clk IMX8MN_CLK_SAI3_ROOT>; + }; +}; + +&sai3 { + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai3>; + assigned-clocks = <&clk IMX8MN_CLK_SAI3>; + assigned-clock-parents = <&clk IMX8MN_AUDIO_PLL1_OUT>; + assigned-clock-rates = <24576000>; + fsl,sai-mclk-direction-output; + status = "okay"; }; /* eMMC */ @@ -30,6 +90,36 @@ &usdhc1 { }; &iomuxc { + pinctrl_dac_rst: dacrstgrp { + fsl,pins = < + MX8MN_IOMUXC_GPIO1_IO06_GPIO1_IO6 0x19 /* DAC_RST */ + >; + }; + + pinctrl_espi2: espi2grp { + fsl,pins = < + MX8MN_IOMUXC_ECSPI2_SCLK_ECSPI2_SCLK 0x082 + MX8MN_IOMUXC_ECSPI2_MOSI_ECSPI2_MOSI 0x082 + MX8MN_IOMUXC_ECSPI2_MISO_ECSPI2_MISO 0x082 + MX8MN_IOMUXC_ECSPI2_SS0_ECSPI2_SS0 0x040 + >; + }; + + pinctrl_i2c2: i2c2grp { + fsl,pins = < + MX8MN_IOMUXC_I2C2_SCL_I2C2_SCL 0x400000c3 + MX8MN_IOMUXC_I2C2_SDA_I2C2_SDA 0x400000c3 + >; + }; + + pinctrl_sai3: sai3grp { + fsl,pins = < + MX8MN_IOMUXC_SAI3_TXFS_SAI3_TX_SYNC 0xd6 + MX8MN_IOMUXC_SAI3_TXC_SAI3_TX_BCLK 0xd6 + MX8MN_IOMUXC_SAI3_TXD_SAI3_TX_DATA0 0xd6 + >; + }; + pinctrl_usdhc1: usdhc1grp { fsl,pins = < MX8MN_IOMUXC_SD1_CLK_USDHC1_CLK 0x40000090 -- 2.34.1

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[PATCH v3 0/3] Make headphone work on Huawei Matebook D15
by Mauro Carvalho Chehab 05 Apr '22

05 Apr '22

At Huawei Matebook D15 two different GPIOs are used to control the output: - gpio0 controls the speaker output; - gpio1 controls the headphone output. Changing both at the same time cause spurious events that are mis-interpreted as input events, causing troubles on apps. So, a delay is needed before turning on such gpios. With this patch, plugging a headphone causes a jack event to trigger the speaker supply, powering down the speaker and powering up the headphone output. Removing the headphone also triggers the power supply, powering up the speaker and powering down the headphone. --- v3: - add a patch changing GPIO quirk speaker naming. Patch 2 got rebased on the top of it. Mauro Carvalho Chehab (2): ASoC: Intel: sof_es8336: support a separate gpio to control headphone ASoC: Intel: sof_es8336: Huawei Matebook D15 uses a headphone gpio Pierre-Louis Bossart (1): ASoC: Intel: sof_es8336: simplify speaker gpio naming sound/soc/intel/boards/sof_es8336.c | 97 +++++++++++++++++++++-------- 1 file changed, 72 insertions(+), 25 deletions(-) -- 2.35.1

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[PATCH] ALSA: echoaudio: remove redundant assignment to variable i
by Colin Ian King 05 Apr '22

05 Apr '22

The variable i is being assigned a value that is never read, it is being re-assigned in the following for-loop. The assignment is redundant and can be removed. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/pci/echoaudio/midi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 7be5c3327b16..47b2c023ee3d 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -124,7 +124,6 @@ static int midi_service_irq(struct echoaudio *chip) return 0; /* Get the MIDI data from the comm page */ - i = 1; received = 0; for (i = 1; i <= count; i++) { /* Get the MIDI byte */ -- 2.35.1

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[PATCH v2] ALSA: hda/i915 - skip acomp init if no matching display
by Kai Vehmanen 05 Apr '22

05 Apr '22

In systems with only a discrete i915 GPU, the acomp init will always timeout for the PCH HDA controller instance. Avoid the timeout by checking the PCI device hierarchy whether any display class PCI device can be found on the system, and at the same level as the HDA PCI device. If found, proceed with the acomp init, which will wait until i915 probe is complete and component binding can proceed. If no matching display device is found, the audio component bind can be safely skipped. The bind timeout will still be hit if the display is present in the system, but i915 driver does not bind to it by configuration choice or probe error. In this case the 60sec timeout will be hit. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Acked-by: Lucas De Marchi <lucas.demarchi(a)intel.com> --- sound/hda/hdac_i915.c | 29 +++++++++++++++++++---------- 1 file changed, 19 insertions(+), 10 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index efe810af28c5..48b8ed752b69 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -116,16 +116,25 @@ static int i915_component_master_match(struct device *dev, int subcomponent, return 0; } -/* check whether intel graphics is present */ -static bool i915_gfx_present(void) +/* check whether Intel graphics is present and reachable */ +static int i915_gfx_present(struct pci_dev *hdac_pci) { - static const struct pci_device_id ids[] = { - { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_ANY_ID), - .class = PCI_BASE_CLASS_DISPLAY << 16, - .class_mask = 0xff << 16 }, - {} - }; - return pci_dev_present(ids); + unsigned int class = PCI_BASE_CLASS_DISPLAY << 16; + struct pci_dev *display_dev = NULL; + bool match = false; + + do { + display_dev = pci_get_class(class, display_dev); + + if (display_dev && display_dev->vendor == PCI_VENDOR_ID_INTEL && + connectivity_check(display_dev, hdac_pci)) + match = true; + + pci_dev_put(display_dev); + + } while (!match && display_dev); + + return match; } /** @@ -145,7 +154,7 @@ int snd_hdac_i915_init(struct hdac_bus *bus) struct drm_audio_component *acomp; int err; - if (!i915_gfx_present()) + if (!i915_gfx_present(to_pci_dev(bus->dev))) return -ENODEV; err = snd_hdac_acomp_init(bus, NULL, base-commit: bfa1e1a62c8bdbe3d8c915fbb7a078dc783b2ee8 -- 2.35.1

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[PATCH v2] ALSA: emu10k1: Stop using iommu_present()
by Robin Murphy 05 Apr '22

05 Apr '22

iommu_get_domain_for_dev() is already perfectly happy to return NULL if the given device has no IOMMU. Drop the unnecessary check in favour of just handling that condition appropriately. Signed-off-by: Robin Murphy <robin.murphy(a)arm.com> --- v2: Get "!domain" condition right sound/pci/emu10k1/emu10k1_main.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 86cc1ca025e4..3880f359e688 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1751,11 +1751,8 @@ static void snd_emu10k1_detect_iommu(struct snd_emu10k1 *emu) emu->iommu_workaround = false; - if (!iommu_present(emu->card->dev->bus)) - return; - domain = iommu_get_domain_for_dev(emu->card->dev); - if (domain && domain->type == IOMMU_DOMAIN_IDENTITY) + if (!domain || domain->type == IOMMU_DOMAIN_IDENTITY) return; dev_notice(emu->card->dev, -- 2.28.0.dirty

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[PATCH 02/11] ALSA: usb-audio: Fix undefined behavior due to shift overflowing the constant
by Borislav Petkov 05 Apr '22

05 Apr '22

From: Borislav Petkov <bp(a)suse.de> Fix: sound/usb/midi.c: In function ‘snd_usbmidi_out_endpoint_create’: sound/usb/midi.c:1389:2: error: case label does not reduce to an integer constant case USB_ID(0xfc08, 0x0101): /* Unknown vendor Cable */ ^~~~ See https://lore.kernel.org/r/YkwQ6%2BtIH8GQpuct@zn.tnic for the gory details as to why it triggers with older gccs only. Signed-off-by: Borislav Petkov <bp(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/usb/usbaudio.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 167834133b9b..2b3dd55e8ee0 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -8,7 +8,7 @@ */ /* handling of USB vendor/product ID pairs as 32-bit numbers */ -#define USB_ID(vendor, product) (((vendor) << 16) | (product)) +#define USB_ID(vendor, product) ((((unsigned int)vendor) << 16) | (product)) #define USB_ID_VENDOR(id) ((id) >> 16) #define USB_ID_PRODUCT(id) ((u16)(id)) -- 2.35.1

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