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[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

[PATCH] ASoC: Intel: boards: eve: Fix DMIC records zero
by vamshi.krishna.gopal＠intel.com 24 Jul '20

24 Jul '20

From: Brent Lu <brent.lu(a)intel.com> Add a dapm route to provide ssp mclk/sclk early for DMIC on SSP0(rt5514) and Headset on SSP1(rt5663) since sclk for both codecs are different the struct now defines SSP0 and SSP1 mclk , sclk separately This change ensures the DMIC PCM port will not return all-zero data Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 150 ++++++++++++------ 1 file changed, 102 insertions(+), 48 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index b34cf6cf1139..584e4f9cedc2 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -53,8 +53,10 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; - struct clk *mclk; - struct clk *sclk; + struct clk *ssp0_mclk; + struct clk *ssp0_sclk; + struct clk *ssp1_mclk; + struct clk *ssp1_sclk; }; enum { @@ -77,13 +79,31 @@ static const struct snd_kcontrol_new kabylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event, int ssp_num) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + struct clk *mclk, *sclk; + unsigned long sclk_rate; int ret = 0; + switch (ssp_num) { + case 0: + mclk = priv->ssp0_mclk; + sclk = priv->ssp0_sclk; + sclk_rate = 6144000; + break; + case 1: + mclk = priv->ssp1_mclk; + sclk = priv->ssp1_sclk; + sclk_rate = 3072000; + break; + default: + dev_err(card->dev, "Invalid ssp_num %d\n", ssp_num); + return -EINVAL; + } + /* * MCLK/SCLK need to be ON early for a successful synchronization of * codec internal clock. And the clocks are turned off during @@ -91,38 +111,46 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, */ switch (event) { case SND_SOC_DAPM_PRE_PMU: + if (__clk_is_enabled(mclk)) + return 0; + /* Enable MCLK */ - ret = clk_set_rate(priv->mclk, 24000000); + ret = clk_set_rate(mclk, 24000000); if (ret < 0) { - dev_err(card->dev, "Can't set rate for mclk, err: %d\n", - ret); + dev_err(card->dev, "Can't set rate for ssp%d_mclk, err: %d\n", + ssp_num, ret); return ret; } - ret = clk_prepare_enable(priv->mclk); + ret = clk_prepare_enable(mclk); if (ret < 0) { - dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + dev_err(card->dev, "Can't enable ssp%d_mclk, err: %d\n", + ssp_num, ret); return ret; } /* Enable SCLK */ - ret = clk_set_rate(priv->sclk, 3072000); + ret = clk_set_rate(sclk, sclk_rate); if (ret < 0) { - dev_err(card->dev, "Can't set rate for sclk, err: %d\n", - ret); - clk_disable_unprepare(priv->mclk); + dev_err(card->dev, "Can't set rate for ssp%d_sclk, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(mclk); return ret; } - ret = clk_prepare_enable(priv->sclk); + ret = clk_prepare_enable(sclk); if (ret < 0) { - dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); - clk_disable_unprepare(priv->mclk); + dev_err(card->dev, "Can't enable ssp%d_sclk, err: %d\n", + ssp_num, ret); + clk_disable_unprepare(mclk); } break; case SND_SOC_DAPM_POST_PMD: - clk_disable_unprepare(priv->mclk); - clk_disable_unprepare(priv->sclk); + if (!__clk_is_enabled(mclk)) + return 0; + + clk_disable_unprepare(mclk); + clk_disable_unprepare(sclk); break; default: return 0; @@ -131,6 +159,18 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return 0; } +static int platform_clock_control_ssp0(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 0); +} + +static int platform_clock_control_ssp1(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + return platform_clock_control(w, k, event, 1); +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -139,15 +179,17 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_SUPPLY("Platform Clock SSP0", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp0, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Platform Clock SSP1", SND_SOC_NOPM, 0, 0, + platform_clock_control_ssp1, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ - { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "Platform Clock SSP1" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -156,7 +198,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock SSP1" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -180,6 +222,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "ssp0 Rx", NULL, "Right HiFi Capture" }, /* DMIC */ + { "DMIC", NULL, "Platform Clock SSP0" }, { "DMIC1L", NULL, "DMIC" }, { "DMIC1R", NULL, "DMIC" }, { "DMIC2L", NULL, "DMIC" }, @@ -666,7 +709,7 @@ static int kabylake_set_bias_level(struct snd_soc_card *card, if (!component || strcmp(component->name, RT5514_DEV_NAME)) return 0; - if (IS_ERR(priv->mclk)) + if (IS_ERR(priv->ssp0_mclk)) return 0; /* @@ -678,17 +721,17 @@ static int kabylake_set_bias_level(struct snd_soc_card *card, case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_ON) { dev_dbg(card->dev, "Disable mclk"); - clk_disable_unprepare(priv->mclk); + clk_disable_unprepare(priv->ssp0_mclk); } else { dev_dbg(card->dev, "Enable mclk"); - ret = clk_set_rate(priv->mclk, 24000000); + ret = clk_set_rate(priv->ssp0_mclk, 24000000); if (ret) { dev_err(card->dev, "Can't set rate for mclk, err: %d\n", ret); return ret; } - ret = clk_prepare_enable(priv->mclk); + ret = clk_prepare_enable(priv->ssp0_mclk); if (ret) { dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); @@ -757,6 +800,29 @@ static struct snd_soc_card kabylake_audio_card = { .late_probe = kabylake_card_late_probe, }; +static int kabylake_audio_clk_get(struct device *dev, const char *id, + struct clk **clk) +{ + int ret = 0; + + if (!clk) + return -EINVAL; + + *clk = devm_clk_get(dev, id); + if (IS_ERR(*clk)) { + ret = PTR_ERR(*clk); + if (ret == -ENOENT) { + dev_info(dev, "Failed to get %s, defer probe\n", id); + return -EPROBE_DEFER; + } + + dev_err(dev, "Failed to get %s with err:%d\n", id, ret); + return ret; + } + + return ret; +} + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; @@ -777,33 +843,21 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; - ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_mclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_mclk", &ctx->ssp0_mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp0_sclk", &ctx->ssp0_sclk); + if (ret != 0) return ret; - } - ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_mclk", &ctx->ssp1_mclk); + if (ret != 0) + return ret; - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); + ret = kabylake_audio_clk_get(&pdev->dev, "ssp1_sclk", &ctx->ssp1_sclk); + if (ret != 0) return ret; - } return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.17.1

3 2

Alsa or alsamixer muting itself
by GitHub issues - opened 24 Jul '20

24 Jul '20

alsa-project/alsa-lib issue #74 was opened from allelopath: Alsa or alsamixer mutes the master track, after 10 or so minutes or on restart Dell Optiplex 7020 Desktop journalctrl attached [journalctl.txt](https://github.com/alsa-project/alsa-lib/files/4969120/jour… $ lspci 00:00.0 Host bridge: Intel Corporation 4th Gen Core Processor DRAM Controller (rev 06) 00:02.0 VGA compatible controller: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor Integrated Graphics Controller (rev 06) 00:03.0 Audio device: Intel Corporation Xeon E3-1200 v3/4th Gen Core Processor HD Audio Controller (rev 06) 00:14.0 USB controller: Intel Corporation 8 Series/C220 Series Chipset Family USB xHCI (rev 04) 00:16.0 Communication controller: Intel Corporation 8 Series/C220 Series Chipset Family MEI Controller #1 (rev 04) 00:16.3 Serial controller: Intel Corporation 8 Series/C220 Series Chipset Family KT Controller (rev 04) 00:19.0 Ethernet controller: Intel Corporation Ethernet Connection I217-LM (rev 04) 00:1a.0 USB controller: Intel Corporation 8 Series/C220 Series Chipset Family USB EHCI #2 (rev 04) 00:1b.0 Audio device: Intel Corporation 8 Series/C220 Series Chipset High Definition Audio Controller (rev 04) 00:1d.0 USB controller: Intel Corporation 8 Series/C220 Series Chipset Family USB EHCI #1 (rev 04) 00:1f.0 ISA bridge: Intel Corporation Q87 Express LPC Controller (rev 04) 00:1f.2 SATA controller: Intel Corporation 8 Series/C220 Series Chipset Family 6-port SATA Controller 1 [AHCI mode] (rev 04) 00:1f.3 SMBus: Intel Corporation 8 Series/C220 Series Chipset Family SMBus Controller (rev 04) Issue URL : https://github.com/alsa-project/alsa-lib/issues/74 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: tegra: Fix build error due to 64-by-32 division
by Sameer Pujar 23 Jul '20

23 Jul '20

Build errors are seen on 32-bit platforms because of a plain 64-by-32 division. For example, following build erros were reported. "ERROR: modpost: "__udivdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined!" "ERROR: modpost: "__divdi3" [sound/soc/tegra/snd-soc-tegra210-dmic.ko] undefined!" This can be fixed by using div_u64() helper from 'math64.h' header. Fixes: 8c8ff982e9e2 ("ASoC: tegra: Add Tegra210 based DMIC driver") Reported-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/tegra/tegra210_dmic.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c index ff6fd65..d682414 100644 --- a/sound/soc/tegra/tegra210_dmic.c +++ b/sound/soc/tegra/tegra210_dmic.c @@ -6,6 +6,7 @@ #include <linux/clk.h> #include <linux/device.h> +#include <linux/math64.h> #include <linux/module.h> #include <linux/of_device.h> #include <linux/platform_device.h> @@ -129,7 +130,7 @@ static int tegra210_dmic_hw_params(struct snd_pcm_substream *substream, * Boost Gain Volume control has 100x factor. */ if (dmic->boost_gain) - gain_q23 = (gain_q23 * dmic->boost_gain) / 100; + gain_q23 = div_u64(gain_q23 * dmic->boost_gain, 100); regmap_write(dmic->regmap, TEGRA210_DMIC_LP_FILTER_GAIN, (unsigned int)gain_q23); -- 2.7.4

3 2

[PATCH v2 1/2] dt-bindings: tas2562: Convert the tas2562 binding to yaml
by Dan Murphy 23 Jul '20

23 Jul '20

Convert the TAS2562 text file to yaml format. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v2 - Updated the shutdown-gpio to shutdown-gpios and fixed licensing to be GPL-2.0-only. .../devicetree/bindings/sound/tas2562.yaml | 69 +++++++++++++++++++ 1 file changed, 69 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2562.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..8d75a798740b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2562.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2562 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shut-down-gpios: + description: GPIO used to control the state of the device. + deprecated: true + + shutdown-gpios: + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; + -- 2.27.0

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