May 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] Asoc: qcom: Add four speaker support on MI2S secondary
by Srinivasa Rao Mandadapu 09 Jun '21

09 Jun '21

Add four speaker support on MI2S secondary block by using I2S SD1 line on gpio52 pin, and add channel map control support in the lpass-cpu audio driver. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 106 +++++++++++++++++++++++++++++++++- sound/soc/qcom/lpass-sc7180.c | 1 + sound/soc/qcom/lpass.h | 5 ++ 3 files changed, 111 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..88a038a107a7 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -28,6 +28,19 @@ #define LPASS_CPU_I2S_SD2_3_MASK GENMASK(3, 2) #define LPASS_CPU_I2S_SD0_1_2_MASK GENMASK(2, 0) #define LPASS_CPU_I2S_SD0_1_2_3_MASK GENMASK(3, 0) +#define LPASS_CHMAP_IDX_UNKNOWN -1 + +/* + * Channel maps for multi-channel playbacks on MI2S Secondary, up to 4 n_ch + */ +static struct snd_pcm_chmap_elem lpass_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) @@ -86,6 +99,9 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); return ret; } + if (drvdata->chmap_info && drvdata->chmap_info->max_channels == 4) + drvdata->chmap_info->chmap = lpass_chmaps; + return 0; } @@ -96,6 +112,7 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -224,9 +241,14 @@ static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, ret); return ret; } - if (channels >= 2) + if (channels >= 2) { ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_STEREO); + if (channels == 4) + drvdata->chmap_idx = 1; + else + drvdata->chmap_idx = 0; + } else ret = regmap_fields_write(i2sctl->spkmono, id, LPAIF_I2SCTL_SPKMONO_MONO); @@ -324,6 +346,84 @@ const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); +static int lpass_cpu_chmap_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned const char *map; + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + ucontrol->value.integer.value[i] = 0; + else + ucontrol->value.integer.value[i] = map[i]; + } + + return 0; +} + +static int lpass_cpu_chmap_ctl_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int i; + struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); + struct lpass_data *drvdata = info->private_data; + unsigned char *map; + + if (info->max_channels == 2) + map = &lpass_chmaps[0].map[0]; + else if (info->max_channels == 4) + map = &lpass_chmaps[1].map[0]; + + for (i = 0; i < info->max_channels; i++) { + if (drvdata->chmap_idx == LPASS_CHMAP_IDX_UNKNOWN) + map[i] = 0; + else + map[i] = ucontrol->value.integer.value[i]; + } + + return 0; +} + +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + int ret; + + struct snd_soc_dai_driver *drv = dai->driver; + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + + ret = snd_pcm_add_chmap_ctls(rtd->pcm, SNDRV_PCM_STREAM_PLAYBACK, + snd_pcm_alt_chmaps, drv->playback.channels_max, 0, + &drvdata->chmap_info); + if (ret < 0) + return ret; + + + /* + * override handlers + */ + drvdata->chmap_info->private_data = drvdata; + drvdata->chmap_info->kctl->get = lpass_cpu_chmap_ctl_get; + drvdata->chmap_info->kctl->put = lpass_cpu_chmap_ctl_set; + + /* + * default chmap supported is stereo + */ + drvdata->chmap_info->chmap = lpass_chmaps; + drvdata->chmap_idx = LPASS_CHMAP_IDX_UNKNOWN; + + return 0; +} +EXPORT_SYMBOL_GPL(lpass_cpu_pcm_new); + int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); @@ -856,6 +956,10 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); return PTR_ERR(drvdata->mi2s_bit_clk[dai_id]); } + if (drvdata->mi2s_playback_sd_mode[dai_id] == LPAIF_I2SCTL_MODE_QUAD01) { + variant->dai_driver[dai_id].playback.channels_min = 4; + variant->dai_driver[dai_id].playback.channels_max = 4; + } } /* Allocation for i2sctl regmap fields */ diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 8c168d3c589e..77a556b27cf0 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -58,6 +58,7 @@ static struct snd_soc_dai_driver sc7180_lpass_cpu_dai_driver[] = { }, .probe = &asoc_qcom_lpass_cpu_dai_probe, .ops = &asoc_qcom_lpass_cpu_dai_ops, + .pcm_new = lpass_cpu_pcm_new, }, { .id = LPASS_DP_RX, .name = "Hdmi", diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 83b2e08ade06..fb82591d550c 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -76,6 +76,9 @@ struct lpass_data { /* regmap backed by the low-power audio interface (LPAIF) registers */ struct regmap *lpaif_map; struct regmap *hdmiif_map; + struct snd_pcm_chmap *chmap_info; + + int chmap_idx; /* interrupts from the low-power audio interface (LPAIF) */ int lpaif_irq; @@ -259,5 +262,7 @@ void asoc_qcom_lpass_cpu_platform_shutdown(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +int lpass_cpu_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); #endif /* __LPASS_H__ */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 2

alsamixer: use background color instead of COLOR_BLACK
by GitHub pull_request - opened 08 Jun '21

08 Jun '21

alsa-project/alsa-utils pull request #77 was opened from r-burns: Hi! I have a cosmetic "improvement" I'd like to share. Scare quotes because this is possibly opinionated, but I'd still like to know what you think. I think on typical terminals, COLOR_BLACK is the same or quite similar to the background color, so this does not make much of a difference. But on "solarized" color schemes and similar, the COLOR_BLACK may be quite different from the background color, so in my opinion it looks odd to have the alsamixer panel stand out this way. For example, I'm using the "Nord" color scheme on a pitch black #000000 background, and here's how the mixer looks with/without this patch (blank terminal, htop, and ncmpcpp also provided for comparison): ![image](https://user-images.githubusercontent.com/52847440/107131140-dbf46500-6888-11eb-938a-9a3258bf7184.png) As you can see, the current alsamixer in the center looks a bit strange and sticks out against my chosen background color and the other TUIs I'm using. Htop and ncmpcpp agree on using the builtin `-1` background color for their own backgrounds, but alsamixer uses COLOR_BLACK which stands out from the actual background. The alsamixer on the bottom right is using this patch, and blends in seamlessly. Again, this is just my opinion! Feel free to close if this is wrong or superfluous. Nonetheless, I thought I'd share this patch, in case anyone else feels the same way. Request URL : https://github.com/alsa-project/alsa-utils/pull/77 Patch URL : https://github.com/alsa-project/alsa-utils/pull/77.patch Repository URL: https://github.com/alsa-project/alsa-utils

2 1

[PATCH v3 0/7] ASoC: adds new .auto_selectable_formats support
by Kuninori Morimoto 07 Jun '21

07 Jun '21

Hi Mark These are v3 of "ASoC: adds new .get_fmt support", but renamed Subject. This is a little bit challenging patch-set. The idea/code is almost same as v1 / v2. v3 has "priority" support. We need to set dai_link->dai_fmt to select CPU/Codec settings, and it is selected by Sound Card Driver, today. Because of it, Sound Card user need to know both CPU / Codec available dai_fmt, and needs to select it. For example simple-card / audio-graph case, it is selected by "format" and "bitclock/frame-master/inversion" on DT. But, it can be automatically selected if both CPU and Codec drivers indicate it to ALSA SoC Framework, somehow. By this patch, dai_fmt can be automatically selected from each driver if both CPU / Codec driver had .auto_selectable_formats. Automatically selectable *field* is depends on each drivers. For example, some driver want to select format "automatically", but want to select other fields "manually", because of complex limitation. Or other example, in case of both CPU and Codec are possible to be clock provider, but the quality was different. In these case, user need/want to *manually* select each fields from Sound Card driver. It uses Sound Card specified fields preferentially, and try to select non-specific fields from CPU and Codec driver settings if driver had .auto_selectable_formats. In other words, we can select all dai_fmt via Sound Card driver same as before. Select dai_fmt 100% automatically is very difficult and will be very complex, but select automatically some fields only is very easy, I guess. This patch-set is based on such assumption. v1 -> v2 - Add more detail explanation on git-log, code, comment. - Possible to be Clock/Frame provider is depends on driver's situation. v2 -> v3 - has priority - tidyup function explanation for snd_soc_dai_get_fmt() - Each driver don't try to have SND_SOC_DAIFMT_CBx_CFx to avoid confusion Link: https://lore.kernel.org/r/871rb3hypy.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871racbx0w.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (7): ASoC: soc-core: move snd_soc_runtime_set_dai_fmt() to upside ASoC: soc-core: add snd_soc_runtime_get_dai_fmt() ASoC: ak4613: add .auto_selectable_formats support ASoC: pcm3168a: add .auto_selectable_formats support ASoC: rsnd: add .auto_selectable_formats support ASoC: fsi: add .auto_selectable_formats support ASoC: hdmi-codec: add .auto_selectable_formats support include/sound/soc-dai.h | 55 +++++++ sound/soc/codecs/ak4613.c | 11 ++ sound/soc/codecs/hdmi-codec.c | 21 +++ sound/soc/codecs/pcm3168a.c | 26 +++ sound/soc/sh/fsi.c | 15 ++ sound/soc/sh/rcar/core.c | 31 +++- sound/soc/soc-core.c | 288 ++++++++++++++++++++++++++-------- sound/soc/soc-dai.c | 63 ++++++++ sound/soc/soc-utils.c | 29 ++++ 9 files changed, 475 insertions(+), 64 deletions(-) -- 2.25.1

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[PATCH][next] ASoC: ti: davinci-mcasp: Fix fall-through warning for Clang
by Gustavo A. R. Silva 04 Jun '21

04 Jun '21

In preparation to enable -Wimplicit-fallthrough for Clang, fix a warning by explicitly adding a goto statement instead of letting the code fall through to the next case. Link: https://github.com/KSPP/linux/issues/115 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- JFYI: We had thousands of these sorts of warnings and now we are down to just 25 in linux-next. This is one of those last remaining warnings. sound/soc/ti/davinci-mcasp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index b94220306d1a..587967720135 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2317,6 +2317,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) break; default: dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret); + goto err; case -EPROBE_DEFER: goto err; } -- 2.27.0

4 3

[PATCH] ASoC: snd-soc-lpass requires REGMAP_MMIO
by Bertrand Jacquin 04 Jun '21

04 Jun '21

With CONFIG_SND_SOC_LPASS_RX_MACRO=m and CONFIG_REGMAP_MMIO undefined, build fails with the following error make -f /var/tmp/portage/sys-kernel/stable-sources-5.12.8/work/linux-5.12.8-stable/scripts/Makefile.modpost sed 's/\.ko$/\.o/' modules.order | scripts/mod/modpost -E -o modules-only.symvers -i vmlinux.symvers -T - ERROR: modpost: "__devm_regmap_init_mmio_clk" [sound/soc/codecs/snd-soc-lpass-rx-macro.ko] undefined! This does also apply to other Qualcomm Macro LPASS all making call to devm_regmap_init_mmio() --- sound/soc/codecs/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1c87b42606c9..f5614f47f44d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1824,18 +1824,22 @@ config SND_SOC_TPA6130A2 config SND_SOC_LPASS_WSA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm WSA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_VA_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm VA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_RX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm RX Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_TX_MACRO depends on COMMON_CLK + select REGMAP_MMIO tristate "Qualcomm TX Macro in LPASS(Low Power Audio SubSystem)" endmenu

2 2

[PATCH] ASoC: img: Fix PM reference leak in img_i2s_in_probe()
by Yufen Yu 03 Jun '21

03 Jun '21

pm_runtime_get_sync will increment pm usage counter even it failed. Forgetting to putting operation will result in reference leak here. Fix it by replacing it with pm_runtime_resume_and_get to keep usage counter balanced. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yufen Yu <yuyufen(a)huawei.com> --- sound/soc/img/img-i2s-in.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index 0843235d73c9..fd3432a1d6ab 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -464,7 +464,7 @@ static int img_i2s_in_probe(struct platform_device *pdev) if (ret) goto err_pm_disable; } - ret = pm_runtime_get_sync(&pdev->dev); + ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) goto err_suspend; -- 2.25.4

2 1

[PATCH v2] ASoC: dt-bindings: Convert imx-audmux binding to json schema
by Oleksij Rempel 03 Jun '21

03 Jun '21

Convert the imx-audmux binding to DT schema format using json-schema Signed-off-by: Oleksij Rempel <o.rempel(a)pengutronix.de> Reviewed-by: Rob Herring <robh(a)kernel.org> --- changes v2: - fix yamllint warnings .../devicetree/bindings/sound/imx-audmux.txt | 28 ----- .../devicetree/bindings/sound/imx-audmux.yaml | 119 ++++++++++++++++++ 2 files changed, 119 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt deleted file mode 100644 index 2db4dcbee1b9..000000000000 --- a/Documentation/devicetree/bindings/sound/imx-audmux.txt +++ /dev/null @@ -1,28 +0,0 @@ -Freescale Digital Audio Mux (AUDMUX) device - -Required properties: - - - compatible : "fsl,imx21-audmux" for AUDMUX version firstly used - on i.MX21, or "fsl,imx31-audmux" for the version - firstly used on i.MX31. - - - reg : Should contain AUDMUX registers location and length. - -An initial configuration can be setup using child nodes. - -Required properties of optional child nodes: - - - fsl,audmux-port : Integer of the audmux port that is configured by this - child node. - - - fsl,port-config : List of configuration options for the specific port. - For imx31-audmux and above, it is a list of tuples - <ptcr pdcr>. For imx21-audmux it is a list of pcr - values. - -Example: - -audmux@21d8000 { - compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; - reg = <0x021d8000 0x4000>; -}; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.yaml b/Documentation/devicetree/bindings/sound/imx-audmux.yaml new file mode 100644 index 000000000000..dab45c310670 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audmux.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audmux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Digital Audio Mux device + +maintainers: + - Oleksij Rempel <o.rempel(a)pengutronix.de> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx27-audmux + - const: fsl,imx21-audmux + - items: + - enum: + - fsl,imx25-audmux + - fsl,imx35-audmux + - fsl,imx50-audmux + - fsl,imx51-audmux + - fsl,imx53-audmux + - fsl,imx6q-audmux + - fsl,imx6sl-audmux + - fsl,imx6sll-audmux + - fsl,imx6sx-audmux + - const: fsl,imx31-audmux + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: audmux + +patternProperties: + "^mux-[0-9a-z]*$": + type: object + properties: + fsl,audmux-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Integer of the audmux port that is configured by this child node + + fsl,port-config: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples ptcr pdcr. + For imx21-audmux it is a list of pcr values. + + required: + - fsl,audmux-port + - fsl,port-config + + additionalProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + }; + - | + audmux@10016000 { + compatible = "fsl,imx27-audmux", "fsl,imx21-audmux"; + reg = <0x10016000 0x1000>; + clocks = <&clks 1>; + clock-names = "audmux"; + + mux-ssi0 { + fsl,audmux-port = <0>; + fsl,port-config = <0xcb205000>; + }; + + mux-pins4 { + fsl,audmux-port = <2>; + fsl,port-config = <0x00001000>; + }; + }; + - | + #include <dt-bindings/sound/fsl-imx-audmux.h> + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_audmux>; + + mux-ssi1 { + fsl,audmux-port = <0>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN 0 + IMX_AUDMUX_V2_PTCR_TFSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TCSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TFSDIR 0 + IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + mux-pins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0) + 0 IMX_AUDMUX_V2_PDCR_TXRXEN + >; + }; + }; -- 2.29.2

2 1

[PATCH] ASoC: tas2562: Fix TDM_CFG0_SAMPRATE values
by Richard Weinberger 03 Jun '21

03 Jun '21

TAS2562_TDM_CFG0_SAMPRATE_MASK starts at bit 1, not 0. So all values need to be left shifted by 1. Signed-off-by: Richard Weinberger <richard(a)nod.at> --- sound/soc/codecs/tas2562.h | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 81866aeb3fbf..55b2a1f52ca3 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -57,13 +57,13 @@ #define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) #define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) #define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) -#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 -#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 -#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 -#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 -#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 -#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 -#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ (0x0 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ (0x1 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ (0x2 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ (0x3 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ (0x4 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ (0x5 << 1) +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ (0x6 << 1) #define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) -- 2.26.2

2 1

[PATCH] ASoC: meson: gx-card: fix sound-dai dt schema
by Jerome Brunet 03 Jun '21

03 Jun '21

There is a fair amount of warnings when running 'make dtbs_check' with amlogic,gx-sound-card.yaml. Ex: arch/arm64/boot/dts/amlogic/meson-gxm-q200.dt.yaml: sound: dai-link-0:sound-dai:0:1: missing phandle tag in 0 arch/arm64/boot/dts/amlogic/meson-gxm-q200.dt.yaml: sound: dai-link-0:sound-dai:0:2: missing phandle tag in 0 arch/arm64/boot/dts/amlogic/meson-gxm-q200.dt.yaml: sound: dai-link-0:sound-dai:0: [66, 0, 0] is too long The reason is that the sound-dai phandle provided has cells, and in such case the schema should use 'phandle-array' instead of 'phandle'. Fixes: fd00366b8e41 ("ASoC: meson: gx: add sound card dt-binding documentation") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- .../devicetree/bindings/sound/amlogic,gx-sound-card.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index db61f0731a20..2e35aeaa8781 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -57,7 +57,7 @@ patternProperties: rate sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array description: phandle of the CPU DAI patternProperties: @@ -71,7 +71,7 @@ patternProperties: properties: sound-dai: - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle-array description: phandle of the codec DAI required: -- 2.32.0.rc0

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[PATCH v2 0/4] ASoC: codecs: wcd934x: add Headset and button detection support
by Srinivas Kandagatla 03 Jun '21

03 Jun '21

This patchset adds support to MBHC(Multi Button Headset Control) block found in Qualcomm WCD codecs. MBHC support headset type detection, both Mechanical and electrical insert/removal detection along with 8 buttons detection, Over current interrupts on HPHL/R, Impedance Measurements on HPHL/R. Eventhough MBHC block supports things like OverCurrent detection, Currently its reported as a kernel debug message. Should this be reported as an uevent to userspace? like the way USB reports? Any suggestions? First patch adds a common mbhc driver and the second one wcd934x specific driver changes along with sdm845 soundcard related changes. Common wcd-mbhc-v2 driver should be reusable across multiple codecs like WCD9335, WCD934x, WCD937x and WCD938x. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. This patchset thanks, srini Changes since v1: - updated buttons-vthreshold bindings from review comments for wcd938x Srinivas Kandagatla (4): ASoC: dt-bindings: wcd934x: add bindings for Headset Button detection ASoC: codecs: wcd: add multi button Headset detection support ASoC: codecs: wcd934x: add mbhc support ASoC: qcom: sdm845: add jack support for WCD934x .../bindings/sound/qcom,wcd934x.yaml | 30 + include/linux/mfd/wcd934x/registers.h | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-mbhc-v2.c | 1468 +++++++++++++++++ sound/soc/codecs/wcd-mbhc-v2.h | 340 ++++ sound/soc/codecs/wcd934x.c | 884 +++++++++- sound/soc/qcom/sdm845.c | 8 + 8 files changed, 2778 insertions(+), 15 deletions(-) create mode 100644 sound/soc/codecs/wcd-mbhc-v2.c create mode 100644 sound/soc/codecs/wcd-mbhc-v2.h -- 2.21.0

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