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

[PATCH] ASoC: dt-binding: convert amlogic,g12a-tohdmitx to dt-schema
by Neil Armstrong 06 Jun '24

06 Jun '24

Convert text bindings to dt-schema format for the Amlogic TX HDMI control glue. Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- .../bindings/sound/amlogic,g12a-tohdmitx.txt | 58 ---------------------- .../bindings/sound/amlogic,g12a-tohdmitx.yaml | 56 +++++++++++++++++++++ 2 files changed, 56 insertions(+), 58 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt deleted file mode 100644 index 4e8cd7eb7cec..000000000000 --- a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt +++ /dev/null @@ -1,58 +0,0 @@ -* Amlogic HDMI Tx control glue - -Required properties: -- compatible: "amlogic,g12a-tohdmitx" or - "amlogic,sm1-tohdmitx" -- reg: physical base address of the controller and length of memory - mapped region. -- #sound-dai-cells: should be 1. -- resets: phandle to the dedicated reset line of the hdmitx glue. - -Example on the S905X2 SoC: - -tohdmitx: audio-controller@744 { - compatible = "amlogic,g12a-tohdmitx"; - reg = <0x0 0x744 0x0 0x4>; - #sound-dai-cells = <1>; - resets = <&clkc_audio AUD_RESET_TOHDMITX>; -}; - -Example of an 'amlogic,axg-sound-card': - -sound { - compatible = "amlogic,axg-sound-card"; - -[...] - - dai-link-x { - sound-dai = <&tdmif_a>; - dai-format = "i2s"; - dai-tdm-slot-tx-mask-0 = <1 1>; - - codec-0 { - sound-dai = <&tohdmitx TOHDMITX_I2S_IN_A>; - }; - - codec-1 { - sound-dai = <&external_dac>; - }; - }; - - dai-link-y { - sound-dai = <&tdmif_c>; - dai-format = "i2s"; - dai-tdm-slot-tx-mask-0 = <1 1>; - - codec { - sound-dai = <&tohdmitx TOHDMITX_I2S_IN_C>; - }; - }; - - dai-link-z { - sound-dai = <&tohdmitx TOHDMITX_I2S_OUT>; - - codec { - sound-dai = <&hdmi_tx>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml new file mode 100644 index 000000000000..34e9de1366f1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml @@ -0,0 +1,56 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,g12a-tohdmitx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic G12a HDMI TX Control Glue + +maintainers: + - Jerome Brunet <jbrunet(a)baylibre.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 1 + + compatible: + oneOf: + - items: + - const: amlogic,g12a-tohdmitx + - items: + - enum: + - amlogic,sm1-tohdmitx + - const: amlogic,g12a-tohdmitx + + reg: + maxItems: 1 + + resets: + maxItems: 1 + + sound-name-prefix: true + +required: + - "#sound-dai-cells" + - compatible + - reg + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + tohdmitx: audio-controller@744 { + compatible = "amlogic,g12a-tohdmitx"; + reg = <0x744 0x4>; + #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOHDMITX>; + }; --- base-commit: c3f38fa61af77b49866b006939479069cd451173 change-id: 20240605-topic-amlogic-upstream-bindings-convert-g12a-tohdmitx-d85095861d88 Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

2 3

[PATCH] ALSA: hda/realtek: Remove Framework Laptop 16 from quirks
by Dustin L. Howett 06 Jun '24

06 Jun '24

The Framework Laptop 16 does not have a combination headphone/headset 3.5mm jack; however, applying the pincfg from the Laptop 13 (nid=0x19) erroneously informs hda that the node is present. Fixes: 8804fa04a492 ("ALSA: hda/realtek: Add Framework laptop 16 to quirks") Signed-off-by: Dustin L. Howett <dustin(a)howett.net> --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e3c0b9d5552d..127f5933abd7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10605,7 +10605,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0xf111, 0x0005, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 --- base-commit: 7b44d5381e541de3da3cee2e948456b250f41f25 change-id: 20240605-alsa-hda-realtek-remove-framework-laptop-16-from-quirks-dc60267cb5e7 Best regards, -- Dustin L. Howett <dustin(a)howett.net>

6 5

[PATCH v4 0/2] ASoC: fsl_xcvr: Support i.MX95 platform
by Shengjiu Wang 06 Jun '24

06 Jun '24

On i.MX95 wakeup domain, there is one instance of Audio XCVR supporting SPDIF mode with a connection to the Audio XCVR physical interface. changes in v4: - refine the constarint for 'clocks' according to Rob's comments changes in v3: - refine the constraint for 'resets', 'clocks', 'interrupts' in document. changes in v2: - Merge patch 1&2, 3&4 from v1 together. - Add more comments in commit message - Add constaint for clocks used on i.mx95 - Add 'select SND_SOC_FSL_UTILS' for compiling issue. Shengjiu Wang (2): ASoC: dt-bindings: fsl,xcvr: Add compatible string for i.MX95 ASoC: fsl_xcvr: Add support for i.MX95 platform .../devicetree/bindings/sound/fsl,xcvr.yaml | 32 ++++- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_xcvr.c | 128 ++++++++++++------ sound/soc/fsl/fsl_xcvr.h | 91 +++++++++++++ 4 files changed, 208 insertions(+), 44 deletions(-) -- 2.34.1

3 4

[PATCH] dt-bindings: sound: tlv320adc3xxx: Fix incorrect GPIO description
by Ricard Wanderlof 06 Jun '24

06 Jun '24

Fix the description for the ti,dmdin-gpio1 and ti,dmclk-gpio2 properties to correctly describe that when configured as general purpose outputs (ADC3XXX_GPIO_GPO), the pins are available via the GPIO framework. Signed-off-by: Ricard Wanderlof <ricard.wanderlof(a)axis.com> --- Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml index ede14ca2c07a..8ac741f4cd56 100644 --- a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -58,8 +58,8 @@ properties: description: | Configuration for DMDIN/GPIO1 pin. - When ADC3XXX_GPIO_GPO is configured, this causes corresponding the - ALSA control "GPIOx Output" to appear, as a switch control. + When ADC3XXX_GPIO_GPO is selected, the pin may be controlled via the + GPIO framework, as pin number 0 on the device. ti,dmclk-gpio2: $ref: /schemas/types.yaml#/definitions/uint32 @@ -76,8 +76,8 @@ properties: description: | Configuration for DMCLK/GPIO2 pin. - When ADC3XXX_GPIO_GPO is configured, this causes corresponding the - ALSA control "GPIOx Output" to appear, as a switch control. + When ADC3XXX_GPIO_GPO is selected, the pin may be controlled via the + GPIO framework, as pin number 1 on the device. Note that there is currently no support for reading the GPIO pins as inputs. --- base-commit: 47d09270d7776e46858a161f94b735640b2fb056 change-id: 20240528-tlv320adc3xxx-dt-gpio-fix-a3384e435148 Best regards, -- -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 3

[PATCH 00/11] ASoC: simple-card: sync support
by Kuninori Morimoto 06 Jun '24

06 Jun '24

Hi Mark We have simle-card / audio-graph / audio-graph2, basically these supports same feature but is using different DT style. Because we are using 3 drivers, some feature was added to one driver, but other drivers doesn't have it. This patch set try to sync it on these 3 drivers. Kuninori Morimoto (11): ASoC: simple-card-utils: remove both playback/capture_only check ASoC: audio-graph-card2: add ep_to_port() / port_to_ports() ASoC: audio-graph-card2: remove ports node name check ASoC: audio-graph-card2: expand dai_link property part ASoC: audio-graph-card2: merge graph_parse_mclk_fs() into graph_link_init() ASoC: audio-graph-card: add ep_to_port() / port_to_ports() ASoC: audio-graph-card: remove ports node name check ASoC: audio-graph-card: enable playback/capture_only property ASoC: audio-graph-card: merge graph_parse_mclk_fs() into graph_link_init() ASoC: simple-audio-card: enable playback/capture_only property ASoC: simple-audio-card: merge simple_parse_mclk_fs() into simple_link_init() include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 95 ++++++++------ sound/soc/generic/audio-graph-card2.c | 171 +++++++++++++++----------- sound/soc/generic/simple-card-utils.c | 23 ++-- sound/soc/generic/simple-card.c | 56 +++++---- 5 files changed, 198 insertions(+), 149 deletions(-) -- 2.43.0

2 12

[PATCH v2 0/2] ASoC: fsl_xcvr: Support i.MX95 platform
by Shengjiu Wang 06 Jun '24

06 Jun '24

On i.MX95 wakeup domain, there is one instance of Audio XCVR supporting SPDIF mode with a connection to the Audio XCVR physical interface. changes in v2: - Merge patch 1&2, 3&4 from v1 together. - Add more comments in commit message - Add constaint for clocks used on i.mx95 - Add 'select SND_SOC_FSL_UTILS' for compiling issue. Shengjiu Wang (2): ASoC: dt-bindings: fsl,xcvr: Add compatible string for i.MX95 ASoC: fsl_xcvr: Add support for i.MX95 platform .../devicetree/bindings/sound/fsl,xcvr.yaml | 55 ++++++-- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_xcvr.c | 128 ++++++++++++------ sound/soc/fsl/fsl_xcvr.h | 91 +++++++++++++ 4 files changed, 222 insertions(+), 53 deletions(-) -- 2.34.1

4 5

[PATCH] ASoC: cs35l56: Use scope-based resource management in cs35l56_try_get_broken_sdca_spkid_gpio()
by Markus Elfring 05 Jun '24

05 Jun '24

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Wed, 5 Jun 2024 22:02:09 +0200 Scope-based resource management became supported also for another programming interface by contributions of Jonathan Cameron on 2024-02-17. See also the commit 59ed5e2d505bf5f9b4af64d0021cd0c96aec1f7c ("device property: Add cleanup.h based fwnode_handle_put() scope based cleanup."). * Thus use the attribute “__free(fwnode_handle)”. * Reduce the scope for the local variable “af01_fwnode”. * Omit explicit fwnode_handle_put() calls accordingly. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- See also the commit fbd741f0993203d07b2b6562d68d1e5e4745b59b ("ASoC: cs35l56: fix usages of device_get_named_child_node()"). sound/soc/codecs/cs35l56.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 758dfdf9d3ea..db41fc42dcac 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1345,13 +1345,13 @@ static void cs35l56_acpi_dev_release_driver_gpios(void *adev) static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l56) { - struct fwnode_handle *af01_fwnode; const union acpi_object *obj; struct gpio_desc *desc; int ret; /* Find the SDCA node containing the GpioIo */ - af01_fwnode = device_get_named_child_node(cs35l56->base.dev, "AF01"); + struct fwnode_handle *af01_fwnode __free(fwnode_handle) + = device_get_named_child_node(cs35l56->base.dev, "AF01"); if (!af01_fwnode) { dev_dbg(cs35l56->base.dev, "No AF01 node\n"); return -ENOENT; @@ -1361,7 +1361,6 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 "spk-id-gpios", ACPI_TYPE_PACKAGE, &obj); if (ret) { dev_dbg(cs35l56->base.dev, "Could not get spk-id-gpios package: %d\n", ret); - fwnode_handle_put(af01_fwnode); return -ENOENT; } @@ -1369,7 +1368,6 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 if (obj->package.count != 4) { dev_warn(cs35l56->base.dev, "Unexpected spk-id element count %d\n", obj->package.count); - fwnode_handle_put(af01_fwnode); return -ENOENT; } @@ -1383,26 +1381,21 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 * ACPI_COMPANION(). */ ret = acpi_dev_add_driver_gpios(adev, cs35l56_af01_spkid_gpios_mapping); - if (ret) { - fwnode_handle_put(af01_fwnode); + if (ret) return dev_err_probe(cs35l56->base.dev, ret, "Failed to add gpio mapping to AF01\n"); - } ret = devm_add_action_or_reset(cs35l56->base.dev, cs35l56_acpi_dev_release_driver_gpios, adev); - if (ret) { - fwnode_handle_put(af01_fwnode); + if (ret) return ret; - } dev_dbg(cs35l56->base.dev, "Added spk-id-gpios mapping to AF01\n"); } desc = fwnode_gpiod_get_index(af01_fwnode, "spk-id", 0, GPIOD_IN, NULL); if (IS_ERR(desc)) { - fwnode_handle_put(af01_fwnode); ret = PTR_ERR(desc); return dev_err_probe(cs35l56->base.dev, ret, "Get GPIO from AF01 failed\n"); } @@ -1411,13 +1404,10 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 gpiod_put(desc); if (ret < 0) { - fwnode_handle_put(af01_fwnode); dev_err_probe(cs35l56->base.dev, ret, "Error reading spk-id GPIO\n"); return ret; } - fwnode_handle_put(af01_fwnode); - dev_info(cs35l56->base.dev, "Got spk-id from AF01\n"); return ret; -- 2.45.1

1 0

[PATCH v2 0/4] soundwire: bus: suppress probe deferral errors
by Johan Hovold 05 Jun '24

05 Jun '24

The soundwire bus code is incorrectly logging probe deferrals as errors. This series addresses that, cleans up the soundwire bus logging and drops the unused soudwire driver name field. Johan Changes in v2 - drop probe error message completely - drop soundwire driver name field - cleanup probe warning messages - drop probe debug message Johan Hovold (4): soundwire: bus: suppress probe deferral errors soundwire: bus: drop unused driver name field soundwire: bus: clean up probe warnings soundwire: bus: drop redundant probe debug message drivers/soundwire/bus_type.c | 21 ++++----------------- include/linux/soundwire/sdw.h | 2 -- 2 files changed, 4 insertions(+), 19 deletions(-) -- 2.44.1

3 16

[PATCH] ASoC: atmel: atmel-classd: Re-add dai_link->platform to fix card init
by Andrei Simion 05 Jun '24

05 Jun '24

The removed dai_link->platform component cause a fail which is exposed at runtime. (ex: when a sound tool is used) This patch re-adds the dai_link->platform component to have a full card registered. Before this patch: :~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: CLASSD [CLASSD], device 0: CLASSD PCM snd-soc-dummy-dai-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 :~$ speaker-test -t sine speaker-test 1.2.6 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Sine wave rate is 440.0000Hz Playback open error: -22,Invalid argument After this patch which restores the platform component: :~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: CLASSD [CLASSD], device 0: CLASSD PCM snd-soc-dummy-dai-0 [CLASSD PCM snd-soc-dummy-dai-0] Subdevices: 1/1 Subdevice #0: subdevice #0 -> Resolve the playback error. Fixes: 2f650f87c03c ("ASoC: atmel: remove unnecessary dai_link->platform") Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- sound/soc/atmel/atmel-classd.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c index 6aed1ee443b4..ba314b279919 100644 --- a/sound/soc/atmel/atmel-classd.c +++ b/sound/soc/atmel/atmel-classd.c @@ -473,19 +473,22 @@ static int atmel_classd_asoc_card_init(struct device *dev, if (!dai_link) return -ENOMEM; - comp = devm_kzalloc(dev, sizeof(*comp), GFP_KERNEL); + comp = devm_kzalloc(dev, 2 * sizeof(*comp), GFP_KERNEL); if (!comp) return -ENOMEM; - dai_link->cpus = comp; + dai_link->cpus = &comp[0]; dai_link->codecs = &snd_soc_dummy_dlc; + dai_link->platforms = &comp[1]; dai_link->num_cpus = 1; dai_link->num_codecs = 1; + dai_link->num_platforms = 1; dai_link->name = "CLASSD"; dai_link->stream_name = "CLASSD PCM"; dai_link->cpus->dai_name = dev_name(dev); + dai_link->platforms->name = dev_name(dev); card->dai_link = dai_link; card->num_links = 1; -- 2.34.1

4 3

[PATCH] ALSA: hda/realtek: Add quirk for IdeaPad 330-17IKB 81DM instead of Yoga Duet 7 13ITL6
by Airat Makhmutov 05 Jun '24

05 Jun '24

From: Ajrat Makhmutov <rauty(a)altlinux.org> Headset microphone do not work out of the box with this laptop. This quirk fixes it. Based on the alsa-info https://alsa-project.org/db/?f=268601dbdb76f9a4c18d71e210091ffed4bf2818 that Zihao Wang sent me and the amixer data from linux-hardware.org for both devices, Zihao Wang specified the wrong subsystem id in his patch: https://lore.kernel.org/all/20220424084120.74125-1-wzhd@ustc.edu/. --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index aa76d1c88589..5ec95a7903b8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10502,7 +10502,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), - SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3820, "IdeaPad 330-17IKB 81DM", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), -- 2.33.8

1 0