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

[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

2 4

[PATCH] ASoC: fsl_esai: add IRQF_SHARED for devm_request_irq
by Shengjiu Wang 23 Jul '20

23 Jul '20

ESAI interfaces may share same interrupt line with EDMA on some platforms (e.g. i.MX8QXP, i.MX8QM). Add IRQF_SHARED flag to allow sharing the irq among several devices Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index b8fbd7ba94af..4ae36099ae82 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1012,7 +1012,7 @@ static int fsl_esai_probe(struct platform_device *pdev) if (irq < 0) return irq; - ret = devm_request_irq(&pdev->dev, irq, esai_isr, 0, + ret = devm_request_irq(&pdev->dev, irq, esai_isr, IRQF_SHARED, esai_priv->name, esai_priv); if (ret) { dev_err(&pdev->dev, "failed to claim irq %u\n", irq); -- 2.27.0

3 2

[PATCH] ASoC: omap: Replace HTTP links with HTTPS ones
by Alexander A. Klimov 23 Jul '20

23 Jul '20

Rationale: Reduces attack surface on kernel devs opening the links for MITM as HTTPS traffic is much harder to manipulate. Deterministic algorithm: For each file: If not .svg: For each line: If doesn't contain `\bxmlns\b`: For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`: If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`: If both the HTTP and HTTPS versions return 200 OK and serve the same content: Replace HTTP with HTTPS. Signed-off-by: Alexander A. Klimov <grandmaster(a)al2klimov.de> --- Continuing my work started at 93431e0607e5. See also: git log --oneline '--author=Alexander A. Klimov <grandmaster(a)al2klimov.de>' v5.7..master If there are any URLs to be removed completely or at least not (just) HTTPSified: Just clearly say so and I'll *undo my change*. See also: https://lkml.org/lkml/2020/6/27/64 If there are any valid, but yet not changed URLs: See: https://lkml.org/lkml/2020/6/26/837 If you apply the patch, please let me know. sound/soc/ti/omap-hdmi.c | 2 +- sound/soc/ti/omap-twl4030.c | 2 +- sound/soc/ti/sdma-pcm.c | 2 +- sound/soc/ti/sdma-pcm.h | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c index def2a0ce8886..3328c02f93c7 100644 --- a/sound/soc/ti/omap-hdmi.c +++ b/sound/soc/ti/omap-hdmi.c @@ -2,7 +2,7 @@ /* * omap-hdmi-audio.c -- OMAP4+ DSS HDMI audio support library * - * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com * * Author: Jyri Sarha <jsarha(a)ti.com> */ diff --git a/sound/soc/ti/omap-twl4030.c b/sound/soc/ti/omap-twl4030.c index 92dbe2c67290..4b408cec7a90 100644 --- a/sound/soc/ti/omap-twl4030.c +++ b/sound/soc/ti/omap-twl4030.c @@ -2,7 +2,7 @@ /* * omap-twl4030.c -- SoC audio for TI SoC based boards with twl4030 codec * - * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2012 Texas Instruments Incorporated - https://www.ti.com * All rights reserved. * * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c index 2b0bc234e1b6..9e7691103f05 100644 --- a/sound/soc/ti/sdma-pcm.c +++ b/sound/soc/ti/sdma-pcm.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 /* - * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> */ diff --git a/sound/soc/ti/sdma-pcm.h b/sound/soc/ti/sdma-pcm.h index cb0627c8dd34..c19efb4c043d 100644 --- a/sound/soc/ti/sdma-pcm.h +++ b/sound/soc/ti/sdma-pcm.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* - * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> */ -- 2.27.0

2 1

[PATCH v2] dt-bindings: sound: convert ROHM BD28623 amplifier binding to yaml
by Katsuhiro Suzuki 23 Jul '20

23 Jul '20

This patch converts ROHM BD28623UMV class D speaker amplifier binding to DT schema. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changes in v2: - add additionalProperties:false - add Rob's Reviewed-by --- .../bindings/sound/rohm,bd28623.txt | 29 -------- .../bindings/sound/rohm,bd28623.yaml | 67 +++++++++++++++++++ 2 files changed, 67 insertions(+), 29 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.yaml diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt b/Documentation/devicetree/bindings/sound/rohm,bd28623.txt deleted file mode 100644 index d84557c2686e..000000000000 --- a/Documentation/devicetree/bindings/sound/rohm,bd28623.txt +++ /dev/null @@ -1,29 +0,0 @@ -ROHM BD28623MUV Class D speaker amplifier for digital input - -This codec does not have any control buses such as I2C, it detect format and -rate of I2S signal automatically. It has two signals that can be connected -to GPIOs: reset and mute. - -Required properties: -- compatible : should be "rohm,bd28623" -- #sound-dai-cells: should be 0. -- VCCA-supply : regulator phandle for the VCCA supply -- VCCP1-supply : regulator phandle for the VCCP1 supply -- VCCP2-supply : regulator phandle for the VCCP2 supply - -Optional properties: -- reset-gpios : GPIO specifier for the active low reset line -- mute-gpios : GPIO specifier for the active low mute line - -Example: - - codec { - compatible = "rohm,bd28623"; - #sound-dai-cells = <0>; - - VCCA-supply = <&vcc_reg>; - VCCP1-supply = <&vcc_reg>; - VCCP2-supply = <&vcc_reg>; - reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; - mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; - }; diff --git a/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml new file mode 100644 index 000000000000..859ce64da152 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rohm,bd28623.yaml @@ -0,0 +1,67 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/rohm,bd28623.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ROHM BD28623MUV Class D speaker amplifier for digital input + +description: + This codec does not have any control buses such as I2C, it detect + format and rate of I2S signal automatically. It has two signals + that can be connected to GPIOs reset and mute. + +maintainers: + - Katsuhiro Suzuki <katsuhiro(a)katsuster.net> + +properties: + compatible: + const: rohm,bd28623 + + "#sound-dai-cells": + const: 0 + + VCCA-supply: + description: + regulator phandle for the VCCA (for analog) power supply + + VCCP1-supply: + description: + regulator phandle for the VCCP1 (for ch1) power supply + + VCCP2-supply: + description: + regulator phandle for the VCCP2 (for ch2) power supply + + reset-gpios: + maxItems: 1 + description: + GPIO specifier for the active low reset line + + mute-gpios: + maxItems: 1 + description: + GPIO specifier for the active low mute line + +required: + - compatible + - VCCA-supply + - VCCP1-supply + - VCCP2-supply + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + codec { + compatible = "rohm,bd28623"; + #sound-dai-cells = <0>; + + VCCA-supply = <&vcc_reg>; + VCCP1-supply = <&vcc_reg>; + VCCP2-supply = <&vcc_reg>; + reset-gpios = <&gpio 0 GPIO_ACTIVE_LOW>; + mute-gpios = <&gpio 1 GPIO_ACTIVE_LOW>; + }; -- 2.27.0

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[PATCH 1/2] ASoC: fsl-asoc-card: Support configuring dai fmt from DT
by Shengjiu Wang 23 Jul '20

23 Jul '20

Support same propeties as simple card for configuring fmt from DT. In order to make this change compatible with old DT, these properties are optional. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index ee80d02b56c6..4848ba61d083 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -531,11 +531,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct device_node *cpu_np, *codec_np, *asrc_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; + struct device_node *bitclkmaster = NULL; + struct device_node *framemaster = NULL; struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct device *codec_dev = NULL; const char *codec_dai_name; const char *codec_dev_name; + unsigned int daifmt; u32 width; int ret; @@ -667,6 +670,31 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto asrc_fail; } + /* Format info from DT is optional. */ + daifmt = snd_soc_of_parse_daifmt(np, NULL, + &bitclkmaster, &framemaster); + daifmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + if (bitclkmaster || framemaster) { + if (codec_np == bitclkmaster) + daifmt |= (codec_np == framemaster) ? + SND_SOC_DAIFMT_CBM_CFM : SND_SOC_DAIFMT_CBM_CFS; + else + daifmt |= (codec_np == framemaster) ? + SND_SOC_DAIFMT_CBS_CFM : SND_SOC_DAIFMT_CBS_CFS; + + /* Override dai_fmt with value from DT */ + priv->dai_fmt = daifmt; + } + + /* Change direction according to format */ + if (priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) { + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_IN; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_IN; + } + + of_node_put(bitclkmaster); + of_node_put(framemaster); + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { dev_err(&pdev->dev, "failed to find codec device\n"); ret = -EPROBE_DEFER; -- 2.27.0

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[PATCH 00/29] ASoC: add asoc_substream_to_rtd() macro
by Kuninori Morimoto 23 Jul '20

23 Jul '20

Hi Mark Many ASoC drivers are getting rtd from substream by rtd = substream->private_data OTOH, we have snd_pcm_substream_chip() macro for it. #define snd_pcm_substream_chip(substream) ((substream)->private_data) But, both are not understandable for reader. This patch adds new asoc_substream_to_rtd() which is easy to understand. These are not important, but for readable code. Kuninori Morimoto (29): ASoC: soc-xxx: add asoc_substream_to_rtd() ASoC: ux500: use asoc_substream_to_rtd() ASoC: ti: use asoc_substream_to_rtd() ASoC: tegra: use asoc_substream_to_rtd() ASoC: sunxi: use asoc_substream_to_rtd() ASoC: stm: use asoc_substream_to_rtd() ASoC: sof: use asoc_substream_to_rtd() ASoC: sh: use asoc_substream_to_rtd() ASoC: samsung: use asoc_substream_to_rtd() ASoC: pxa: use asoc_substream_to_rtd() ASoC: cirrus: use asoc_substream_to_rtd() ASoC: rockchip: use asoc_substream_to_rtd() ASoC: amd: use asoc_substream_to_rtd() ASoC: fsl: use asoc_substream_to_rtd() ASoC: mediatek: use asoc_substream_to_rtd() ASoC: atmel: use asoc_substream_to_rtd() ASoC: qcom: use asoc_substream_to_rtd() ASoC: dwc: use asoc_substream_to_rtd() ASoC: intel: use asoc_substream_to_rtd() ASoC: meson: use asoc_substream_to_rtd() ASoC: au1x: use asoc_substream_to_rtd() ASoC: bcm: use asoc_substream_to_rtd() ASoC: codecs: use asoc_substream_to_rtd() ASoC: generic: use asoc_substream_to_rtd() ASoC: sprd: use asoc_substream_to_rtd() ASoC: kirkwood: use asoc_substream_to_rtd() ASoC: xtensa: use asoc_substream_to_rtd() ASoC: mxs: use asoc_substream_to_rtd() ASoC: uniphier: use asoc_substream_to_rtd() include/sound/soc.h | 2 + sound/soc/amd/acp-da7219-max98357a.c | 12 ++-- sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/acp-rt5645.c | 2 +- sound/soc/amd/acp3x-rt5682-max9836.c | 8 +-- sound/soc/amd/raven/acp3x-i2s.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 +- sound/soc/atmel/atmel-classd.c | 8 +-- sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel-pdmic.c | 10 +-- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 12 ++-- sound/soc/bcm/cygnus-pcm.c | 16 ++--- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/snappercl15.c | 2 +- sound/soc/codecs/rt5677-spi.c | 4 +- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_spdif.c | 10 +-- sound/soc/fsl/fsl_ssi.c | 8 +-- sound/soc/fsl/imx-audmix.c | 6 +- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 8 +-- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- sound/soc/generic/simple-card-utils.c | 6 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 16 ++--- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 4 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 4 +- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 +-- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 4 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 4 +- sound/soc/intel/boards/sof_pcm512x.c | 4 +- sound/soc/intel/boards/sof_rt5682.c | 4 +- sound/soc/intel/boards/sof_sdw_rt1308.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/haswell/sst-haswell-pcm.c | 12 ++-- sound/soc/intel/keembay/kmb_platform.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 8 +-- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 12 ++-- .../mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 2 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 2 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 4 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 4 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 8 +-- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 12 ++-- sound/soc/meson/axg-card.c | 2 +- sound/soc/meson/gx-card.c | 2 +- sound/soc/meson/meson-card-utils.c | 2 +- sound/soc/meson/meson-codec-glue.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/brownstone.c | 2 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 2 +- sound/soc/pxa/imote2.c | 2 +- sound/soc/pxa/magician.c | 6 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/pxa/z2.c | 2 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/qcom/apq8096.c | 2 +- sound/soc/qcom/lpass-platform.c | 14 ++--- sound/soc/qcom/qdsp6/q6asm-dai.c | 6 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sdm845.c | 14 ++--- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 2 +- sound/soc/rockchip/rk3399_gru_sound.c | 8 +-- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_max98090.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/samsung/aries_wm8994.c | 4 +- sound/soc/samsung/arndale.c | 4 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 2 +- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 8 +-- sound/soc/samsung/odroid.c | 6 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +- sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 6 +- sound/soc/samsung/tm2_wm5110.c | 8 +-- sound/soc/sh/dma-sh7760.c | 12 ++-- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 4 +- sound/soc/sh/rcar/core.c | 4 +- sound/soc/soc-component.c | 20 +++--- sound/soc/soc-dai.c | 8 +-- sound/soc/soc-dapm.c | 6 +- sound/soc/soc-generic-dmaengine-pcm.c | 4 +- sound/soc/soc-link.c | 12 ++-- sound/soc/soc-pcm.c | 62 +++++++++---------- sound/soc/soc-utils.c | 2 +- sound/soc/sof/intel/hda-dai.c | 10 +-- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sof/intel/hda-pcm.c | 2 +- sound/soc/sof/pcm.c | 18 +++--- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 12 ++-- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 12 ++-- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 2 +- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/davinci-evm.c | 6 +- sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/j721e-evm.c | 6 +- sound/soc/ti/n810.c | 4 +- sound/soc/ti/omap-abe-twl6040.c | 4 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-twl4030.c | 2 +- sound/soc/ti/omap3pandora.c | 2 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 4 +- sound/soc/uniphier/aio-dma.c | 6 +- sound/soc/ux500/mop500_ab8500.c | 8 +-- sound/soc/ux500/ux500_pcm.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 177 files changed, 397 insertions(+), 395 deletions(-) -- 2.25.1

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[PATCH] ASoC: ti: Replace HTTP links with HTTPS ones
by Alexander A. Klimov 23 Jul '20

23 Jul '20

Rationale: Reduces attack surface on kernel devs opening the links for MITM as HTTPS traffic is much harder to manipulate. Deterministic algorithm: For each file: If not .svg: For each line: If doesn't contain `\bxmlns\b`: For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`: If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`: If both the HTTP and HTTPS versions return 200 OK and serve the same content: Replace HTTP with HTTPS. Signed-off-by: Alexander A. Klimov <grandmaster(a)al2klimov.de> --- Continuing my work started at 93431e0607e5. See also: git log --oneline '--author=Alexander A. Klimov <grandmaster(a)al2klimov.de>' v5.7..master If there are any URLs to be removed completely or at least not (just) HTTPSified: Just clearly say so and I'll *undo my change*. See also: https://lkml.org/lkml/2020/6/27/64 If there are any valid, but yet not changed URLs: See: https://lkml.org/lkml/2020/6/26/837 If you apply the patch, please let me know. sound/soc/ti/udma-pcm.c | 2 +- sound/soc/ti/udma-pcm.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/udma-pcm.c b/sound/soc/ti/udma-pcm.c index 39830caaaf7c..2ff0f518aba5 100644 --- a/sound/soc/ti/udma-pcm.c +++ b/sound/soc/ti/udma-pcm.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 /* - * Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com * Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> */ diff --git a/sound/soc/ti/udma-pcm.h b/sound/soc/ti/udma-pcm.h index 54111e7312c1..9ed588fd79b9 100644 --- a/sound/soc/ti/udma-pcm.h +++ b/sound/soc/ti/udma-pcm.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ /* - * Copyright (C) 2018 Texas Instruments Incorporated - http://www.ti.com + * Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com */ #ifndef __UDMA_PCM_H__ -- 2.27.0

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