July 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: dapm: don't call pm_runtime_* on card device
by Tzung-Bi Shih 24 Jul '20

24 Jul '20

runtime_usage of sound card has been observed to grow without bound. For example: $ cat /sys/devices/platform/sound/power/runtime_usage 46 $ sox -n -t s16 -r 48000 -c 2 - synth 1 sine 440 vol 0.1 | \ aplay -q -D hw:0,0 -f S16_LE -r 48000 -c 2 $ cat /sys/devices/platform/sound/power/runtime_usage 52 Commit 4e872a46823c ("ASoC: dapm: Don't force card bias level to be updated") stops to force update bias_level on card. If card doesn't provide set_bias_level callback, the snd_soc_dapm_set_bias_level() is equivalent to NOP for card device. As a result, dapm_pre_sequence_async() doesn't change the bias_level of card device correctly. Thus, pm_runtime_get_sync() would be called in dapm_pre_sequence_async() without symmetric pm_runtime_put() in dapm_post_sequence_async(). Don't call pm_runtime_* on card device. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) Hi, I'm not very sure if the patch makes sense. I was considering: (1) update the bias_level of card device correctly (2) skip pm_runtime_* for card device Per commit message on 4e872a46823c, I believe there is some context I'm not aware of it. Thus, the patch adopts (2). diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5076299abf37..3273161e2787 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1798,7 +1798,7 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie) /* If we're off and we're not supposed to go into STANDBY */ if (d->bias_level == SND_SOC_BIAS_OFF && d->target_bias_level != SND_SOC_BIAS_OFF) { - if (d->dev) + if (d->dev && cookie) pm_runtime_get_sync(d->dev); ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY); @@ -1845,7 +1845,7 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) dev_err(d->dev, "ASoC: Failed to turn off bias: %d\n", ret); - if (d->dev) + if (d->dev && cookie) pm_runtime_put(d->dev); } -- 2.28.0.rc0.142.g3c755180ce-goog

2 6

[PATCH v1] ALSA: hda/realtek: typo_fix: enable headset mic of ASUS ROG Zephyrus G14(GA401) series with ALC289
by Armas Spann 24 Jul '20

24 Jul '20

This patch fixes a small typo I accidently submitted with the initial patch. The board should be named GA401 not G401. Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b2d8e56390a..2dd7150ff6da 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6152,7 +6152,7 @@ enum { ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, - ALC289_FIXUP_ASUS_G401, + ALC289_FIXUP_ASUS_GA401, ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; @@ -7363,7 +7363,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, - [ALC289_FIXUP_ASUS_G401] = { + [ALC289_FIXUP_ASUS_GA401] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x03a11020 }, /* headset mic with jack detect */ @@ -7561,7 +7561,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), - SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), + SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

2 1

[PATCH v1] ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G15(GA502) series with ALC289
by Armas Spann 24 Jul '20

24 Jul '20

This patch adds support for headset mic to the ASUS ROG Zephyrus G15(GA502) notebook series by adding the corresponding vendor/pci_device id, as well as adding a new fixup for the used realtek ALC289. The fixup stets the correct pin to get the headset mic correctly recognized on audio-jack. Signed-off-by: Armas Spann <zappel(a)retarded.farm> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b2d8e56390a..a2c48a7f4594 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6153,6 +6153,7 @@ enum { ALC269VC_FIXUP_ACER_HEADSET_MIC, ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE, ALC289_FIXUP_ASUS_G401, + ALC289_FIXUP_ASUS_GA502, ALC256_FIXUP_ACER_MIC_NO_PRESENCE, }; @@ -7370,6 +7371,13 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC289_FIXUP_ASUS_GA502] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + }, [ALC256_FIXUP_ACER_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7561,6 +7569,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_G401), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), -- 2.27.0

2 1

[PATCH] dt-bindings: sound: convert Everest ES8316 binding to yaml
by Katsuhiro Suzuki 24 Jul '20

24 Jul '20

This patch converts Everest Semiconductor ES8316 low power audio CODEC binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- .../bindings/sound/everest,es8316.txt | 23 --------- .../bindings/sound/everest,es8316.yaml | 49 +++++++++++++++++++ 2 files changed, 49 insertions(+), 23 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.txt b/Documentation/devicetree/bindings/sound/everest,es8316.txt deleted file mode 100644 index 1bf03c5f2af4..000000000000 --- a/Documentation/devicetree/bindings/sound/everest,es8316.txt +++ /dev/null @@ -1,23 +0,0 @@ -Everest ES8316 audio CODEC - -This device supports both I2C and SPI. - -Required properties: - - - compatible : should be "everest,es8316" - - reg : the I2C address of the device for I2C - -Optional properties: - - - clocks : a list of phandle, should contain entries for clock-names - - clock-names : should include as follows: - "mclk" : master clock (MCLK) of the device - -Example: - -es8316: codec@11 { - compatible = "everest,es8316"; - reg = <0x11>; - clocks = <&clks 10>; - clock-names = "mclk"; -}; diff --git a/Documentation/devicetree/bindings/sound/everest,es8316.yaml b/Documentation/devicetree/bindings/sound/everest,es8316.yaml new file mode 100644 index 000000000000..b713404dac4f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8316.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8316.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8316 audio CODEC + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + compatible: + const: everest,es8316 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + es8316: codec@11 { + compatible = "everest,es8316"; + reg = <0x11>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + }; + }; -- 2.27.0

2 2

[PATCH] ASoC: soc-component: don't report of_xlate_dai_name failures
by Jerome Brunet 24 Jul '20

24 Jul '20

With commit e2329eeba45f ("ASoC: soc-component: add soc_component_err()") every error different for ENOTSUPP or EPROBE_DEFER will log an error. However, as explained in snd_soc_get_dai_name(), this callback may error to indicate that the DAI is not matched by the component tested. If the device provides other components, those may still match. Logging an error in this case is misleading. Don't use soc_component_ret() in snd_soc_component_of_xlate_dai_name() to avoid spamming the log. Fixes: e2329eeba45f ("ASoC: soc-component: add soc_component_err()") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- If we wish to report an error when no name is found for dai phandle, I think it would be more approriate to it with with "struct device *" of the card, which is the one actually failing to get what it needs. These user of snd_soc_get_dai_name() are already reporting the error: * sound/soc/samsung/tm2_wm5110.c * sound/soc/rockchip/rk3288_hdmi_analog.c These could be silently failing: * Simple cards * Amlogic's cards * Samsumg odroid card sound/soc/soc-component.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 9565a0dd7cb6..fb6479b22d26 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -344,13 +344,15 @@ int snd_soc_component_of_xlate_dai_name(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name) { - int ret = -ENOTSUPP; - if (component->driver->of_xlate_dai_name) - ret = component->driver->of_xlate_dai_name(component, - args, dai_name); - - return soc_component_ret(component, ret); + return component->driver->of_xlate_dai_name(component, + args, dai_name); + /* + * Don't use soc_component_ret here because we may not want to report + * the error just yet. If a device has more than one component, the + * first may not match and we don't want spam the log with this. + */ + return -ENOTSUPP; } void snd_soc_component_setup_regmap(struct snd_soc_component *component) -- 2.25.4

2 1

[PATCH 0/7] ASoC: qcom: Use qcom_snd_parse_of() for apq8016_sbc
by Stephan Gerhold 24 Jul '20

24 Jul '20

At the moment we have two separate functions to parse the sound card properties from the device tree: qcom_snd_parse_of() for DPCM and apq8016_sbc_parse_of() without DPCM. These functions are almost identical except for a few minor differences. This patch set extends qcom_snd_parse_of() to handle links without DPCM, so that we can use one common function for all (qcom) machine drivers. Stephan Gerhold (7): ASoC: qcom: Use devm for resource management ASoC: qcom: common: Use snd_soc_dai_link_set_capabilities() ASoC: q6afe: Remove unused q6afe_is_rx_port() function ASoC: qcom: common: Support parsing links without DPCM ASoC: qcom: common: Parse properties with "qcom," prefix ASoC: qcom: apq8016_sbc: Use qcom_snd_parse_of() ASoC: qcom: common: Avoid printing errors for -EPROBE_DEFER sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/apq8016_sbc.c | 120 ++++------------------------------- sound/soc/qcom/apq8096.c | 28 +------- sound/soc/qcom/common.c | 58 ++++++++++------- sound/soc/qcom/qdsp6/q6afe.c | 8 --- sound/soc/qcom/qdsp6/q6afe.h | 1 - sound/soc/qcom/sdm845.c | 40 ++---------- 7 files changed, 59 insertions(+), 197 deletions(-) -- 2.27.0

3 9

[PATCH] soundwire: master: enable pm runtime
by Bard Liao 24 Jul '20

24 Jul '20

We should enable pm runtime. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/master.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/master.c b/drivers/soundwire/master.c index 5f0b2189defe..3488bb824e84 100644 --- a/drivers/soundwire/master.c +++ b/drivers/soundwire/master.c @@ -154,6 +154,7 @@ int sdw_master_device_add(struct sdw_bus *bus, struct device *parent, bus->dev = &md->dev; bus->md = md; + pm_runtime_enable(&bus->md->dev); device_register_err: return ret; } @@ -166,6 +167,7 @@ int sdw_master_device_add(struct sdw_bus *bus, struct device *parent, */ int sdw_master_device_del(struct sdw_bus *bus) { + pm_runtime_disable(&bus->md->dev); device_unregister(bus->dev); return 0; -- 2.17.1

3 2

[PATCH 2/2] ASoC: max98390: Update dsm init sequence and condition.
by Steve Lee 24 Jul '20

24 Jul '20

Modify dsm_init sequence and dsm param bin check condition. - Move dsm_init() to after amp init setting to make sure dsm init is last setting. - dsm param bin check condition changed for extended register setting. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 44ffebac15ad..ff5cc9bbec29 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -790,7 +790,7 @@ static int max98390_dsm_init(struct snd_soc_component *component) param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8; param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8; if (param_size > MAX98390_DSM_PARAM_MAX_SIZE || - param_start_addr < DSM_STBASS_HPF_B0_BYTE0 || + param_start_addr < MAX98390_IRQ_CTRL || fw->size < param_size + MAX98390_DSM_PAYLOAD_OFFSET) { dev_err(component->dev, "param fw is invalid.\n"); @@ -864,11 +864,11 @@ static int max98390_probe(struct snd_soc_component *component) regmap_write(max98390->regmap, MAX98390_SOFTWARE_RESET, 0x01); /* Sleep reset settle time */ msleep(20); - /* Update dsm bin param */ - max98390_dsm_init(component); /* Amp init setting */ max98390_init_regs(component); + /* Update dsm bin param */ + max98390_dsm_init(component); /* Dsm Setting */ if (max98390->ref_rdc_value) { -- 2.17.1

1 0

[PATCH 1/2] ASoC: max98390: Fix dac event dapm mixer.
by Steve Lee 24 Jul '20

24 Jul '20

Global EN register guide to off before AMP_EN register when amp disable sequence. - remove AMP_EN control before max98390_dac_event call Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 3e8094241645..44ffebac15ad 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -678,7 +678,7 @@ static const struct snd_kcontrol_new max98390_dai_controls = static const struct snd_soc_dapm_widget max98390_dapm_widgets[] = { SND_SOC_DAPM_DAC_E("Amp Enable", "HiFi Playback", - MAX98390_R203A_AMP_EN, 0, 0, max98390_dac_event, + SND_SOC_NOPM, 0, 0, max98390_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MUX("DAI Sel Mux", SND_SOC_NOPM, 0, 0, &max98390_dai_controls), -- 2.17.1

1 0

[PATCH 1/2] ALSA: hda/realtek: Fix headset mic on Loongson platform
by Kaige Li 24 Jul '20

24 Jul '20

Add pin quirks to enable use of the headset mic on Loongson platform. Signed-off-by: Kaige Li <likaige(a)loongson.cn> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c7e191..b227be3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6041,6 +6041,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269_FIXUP_LOONGSON_HDA, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -6381,6 +6382,14 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC269_FIXUP_LOONGSON_HDA] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x02214c40 }, /* Front Mic */ + { 0x15, 0x01014030 }, /* Rear Mic */ + { } + }, + }, [ALC269_FIXUP_DELL4_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7654,6 +7663,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), + SND_PCI_QUIRK(0x10ec, 0x0269, "Loongson HDA", ALC269_FIXUP_LOONGSON_HDA), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), -- 2.1.0

2 5