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[PATCH v3] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 12 Jul '23

12 Jul '23

This configuration supports JSL boards which implement ALC5650 dual I2S interface codec. Two DAI links are added: AIF1 (on codec side) for headphone and AIF2 for speakers. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/sof_rt5682.c | 80 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 3 files changed, 93 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f472f603ab75..1fe830af2b84 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -475,7 +475,7 @@ endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC if SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_SOF_RT5682_MACH - tristate "SOF with rt5682 codec in I2S Mode" + tristate "SOF with rt5650/rt5682 codec in I2S Mode" depends on I2C && ACPI depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ @@ -485,6 +485,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P + select SND_SOC_RT5645 select SND_SOC_RT5682_I2C select SND_SOC_RT5682S select SND_SOC_DMIC @@ -494,7 +495,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_REALTEK_COMMON help This adds support for ASoC machine driver for SOF platforms - with rt5682 codec. + with rt5650 or rt5682 codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..b4f07bdcf8b4 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,6 +22,7 @@ #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/rt5645.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" @@ -60,6 +61,7 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) #define SOF_RT1019_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_RT5650_HEADPHONE_CODEC_PRESENT BIT(27) /* Default: MCLK on, MCLK 19.2M, SSP0 */ @@ -305,6 +307,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; + int extra_jack_data; int ret; /* need to enable ASRC function for 24MHz mclk rate */ @@ -315,7 +318,16 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) RT5682S_DA_STEREO1_FILTER | RT5682S_AD_STEREO1_FILTER, RT5682S_CLK_SEL_I2S1_ASRC); - else + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(component, + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | RT5682_AD_STEREO1_FILTER, @@ -365,7 +377,12 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); + + if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + extra_jack_data = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra_jack_data); + } else + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); @@ -402,6 +419,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_MCLK; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_MCLK; else pll_source = RT5682_PLL1_S_MCLK; @@ -422,6 +441,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, } else { if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_BCLK1; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_BCLK1; else pll_source = RT5682_PLL1_S_BCLK1; @@ -431,6 +452,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -559,11 +583,30 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "SPOL" }, + { "Right Spk", NULL, "SPOR" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, + ARRAY_SIZE(rt5650_spk_dapm_routes)); + if (ret) + dev_err(rtd->dev, "fail to add dapm routes, ret=%d\n", ret); + + return ret; +} + static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -614,6 +657,17 @@ static struct snd_soc_dai_link_component rt5682s_component[] = { } }; +static struct snd_soc_dai_link_component rt5650_components[] = { + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif1", + }, + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif2", + } +}; + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -652,6 +706,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { links[id].codecs = rt5682s_component; links[id].num_codecs = ARRAY_SIZE(rt5682s_component); + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[0]; + links[id].num_codecs = 1; } else { links[id].codecs = rt5682_component; links[id].num_codecs = ARRAY_SIZE(rt5682_component); @@ -804,6 +861,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[1]; + links[id].num_codecs = 1; + links[id].init = rt5650_spk_init; + links[id].ops = &sof_rt5682_ops; } else { max_98357a_dai_link(&links[id]); } @@ -890,6 +952,12 @@ static int sof_audio_probe(struct platform_device *pdev) /* Detect the headset codec variant */ if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + else if (acpi_dev_present("10EC5650", NULL, -1)) { + sof_rt5682_quirk |= SOF_RT5650_HEADPHONE_CODEC_PRESENT; + + sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", + GFP_KERNEL); + } if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; @@ -1178,6 +1246,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "jsl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index f5c7e1bbded0..36ea2bacbe7d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -34,6 +34,11 @@ static const struct snd_soc_acpi_codecs mx98360a_spk = { .codecs = {"MX98360A"} }; +static struct snd_soc_acpi_codecs rt5650_spk = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -98,6 +103,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "jsl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt5650_spk, + .sof_tplg_filename = "sof-jsl-rt5650-dual.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

3 4

[PATCH v2] ASoC: dt-bindings: Update maintainer email id
by Rohit kumar 12 Jul '23

12 Jul '23

Updated my mail id to latest quicinc id. Acked-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Signed-off-by: Rohit kumar <quic_rohkumar(a)quicinc.com> --- v2: Updated commit header .../devicetree/bindings/sound/google,sc7180-trogdor.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 666a95ac22c8..ba5b7728cf33 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Google SC7180-Trogdor ASoC sound card driver maintainers: - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> - Cheng-Yi Chiang <cychiang(a)chromium.org> description: diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index 6cc8f86c7531..3a559bd07a79 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -8,7 +8,7 @@ title: Qualcomm Technologies Inc. LPASS CPU dai driver maintainers: - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> description: | Qualcomm Technologies Inc. SOC Low-Power Audio SubSystem (LPASS) that consist -- 2.25.1

3 3

[PATCH v1] ASoC: dt-bindings: Add tas2505 audio amplifier
by Kevin Lu 12 Jul '23

12 Jul '23

Create tas2505.yaml for tas2505. Signed-off-by: Kevin Lu <luminlong(a)139.com> --- .../devicetree/bindings/sound/tas2505.yaml | 72 +++++++++++++++++++ 1 file changed, 72 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2505.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2505.yaml b/Documentation/devicetree/bindings/sound/tas2505.yaml new file mode 100644 index 0000000..6cc3fe4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2505.yaml @@ -0,0 +1,72 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2505.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2505 Mono Audio Amplifier + +maintainers: + - Kevin Lu <kevin-lu(a)ti.com> + +description: | + The TAS2505 is a mono Class-D speaker amp that supports both Digital + and Analog inputs. The device is ideal for automotive instrument cluster, + emergency call (eCall), and telematics applications. Direct I2S input + removes the need for an external DAC in the audio signal path. + An on-chip PLL provides the high speed clock needed by the DSP. + The volume level is register controlled. + + Specifications about the audio amplifier can be found at: + https://www.ti.com/lit/gpn/tas2505 + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,tas2505 + + reg: + maxItems: 1 + description: | + I2C address of the device can be set to 0x18. + + reset-gpios: + maxItems: 1 + description: GPIO used to reset the device. + + clock-source: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM clock sourse. + enum: + - 0 # Select MLCK + - 1 # Select BLCK + + '#sound-dai-cells': + # The codec has a single DAI, the #sound-dai-cells=<1>; case is left in for backward + # compatibility but is deprecated. + enum: [0, 1] + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@18 { + compatible = "ti,tas2505"; + reg = <0x18>; + #sound-dai-cells = <0>; + reset-gpios = <&gpio4 21 GPIO_ACTIVE_LOW>; + clock-source = <0>; + }; + }; -- 2.34.1

3 2

[PATCH] ASoC: dt-bindings: nau8821: Convert to dtschema
by Seven Lee 12 Jul '23

12 Jul '23

Convert the NAU8821 audio CODEC bindings to DT schema. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 55 -------- .../bindings/sound/nuvoton,nau8821.yaml | 120 ++++++++++++++++++ 2 files changed, 120 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt deleted file mode 100644 index 7c84e7c7327a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8821.txt +++ /dev/null @@ -1,55 +0,0 @@ -Nuvoton NAU88L21 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8821" - - - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. - - nuvoton,key_enable: Headset button detection switch. - -Example: - - headset: nau8821@1b { - compatible = "nuvoton,nau8821"; - reg = <0x1b>; - interrupt-parent = <&gpio>; - interrupts = <23 IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,dmic-clk-threshold = 3072000; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml new file mode 100644 index 000000000000..728223d20876 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -0,0 +1,120 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8821.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU88L21 audio codec + +maintainers: + - Seven Lee <wtli(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8821 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: Pull-up JKDET pin. If set then JKDET pin is pull up, + otherwise pull down. + type: boolean + + nuvoton,jkdet-polarity: + description: JKDET pin polarity. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,micbias-voltage: + description: MICBIAS output level select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # VDDA + - 1 # VDDA * 1 + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,vref-impedance: + description: VMID Tie-off impedance select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # open(default) + - 1 # 25KOhms + - 2 # 125KOhms + - 3 # 2.5KOhms + default: 2 + + nuvoton,jack-insert-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,dmic-clk-threshold: + description: DMIC clock speed expected value. Unit is Hz. + $ref: /schemas/types.yaml#/definitions/uint32 + default: 3072000 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1b { + #sound-dai-cells = <0>; + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,micbias-voltage = <6>; + nuvoton,vref-impedance = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <0>; + nuvoton,dmic-clk-threshold = <3072000>; + }; + }; -- 2.25.1

3 2

[PATCH v1] ALSA: hda/tas2781: fix 'fw_entry' is used uninitialized
by Shenghao Ding 12 Jul '23

12 Jul '23

Fix 'fw_entry' is used uninitialized. Signed-off-by: Shenghao Ding <13916275206(a)139.com> Changes in v1: | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307070922.doxi1HFc-lkp@intel.com/ --- sound/soc/codecs/tas2781-fmwlib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index eb55abae0d7b..0b08e24ed23a 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1908,7 +1908,7 @@ int tas2781_load_calibration(void *context, char *file_name, { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *)context; struct tasdevice *tasdev = &(tas_priv->tasdevice[i]); - const struct firmware *fw_entry; + const struct firmware *fw_entry = NULL; struct tasdevice_fw *tas_fmw; struct firmware fmw; int offset = 0; -- 2.34.1

1 0

[PATCH v2] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 12 Jul '23

12 Jul '23

This configuration supports JSL boards which implement ALC5650 dual I2S interface codec. Two DAI links are added: AIF1 (on codec side) for headphone and AIF2 for speakers. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/sof_rt5682.c | 81 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 3 files changed, 93 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f472f603ab75..1fe830af2b84 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -475,7 +475,7 @@ endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC if SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_SOF_RT5682_MACH - tristate "SOF with rt5682 codec in I2S Mode" + tristate "SOF with rt5650/rt5682 codec in I2S Mode" depends on I2C && ACPI depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ @@ -485,6 +485,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P + select SND_SOC_RT5645 select SND_SOC_RT5682_I2C select SND_SOC_RT5682S select SND_SOC_DMIC @@ -494,7 +495,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_REALTEK_COMMON help This adds support for ASoC machine driver for SOF platforms - with rt5682 codec. + with rt5650 or rt5682 codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..53451c1b67f8 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,6 +22,7 @@ #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/rt5645.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" @@ -60,6 +61,7 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) #define SOF_RT1019_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_RT5650_HEADPHONE_CODEC_PRESENT BIT(27) /* Default: MCLK on, MCLK 19.2M, SSP0 */ @@ -305,7 +307,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; - int ret; + int ret, extra; /* need to enable ASRC function for 24MHz mclk rate */ if ((sof_rt5682_quirk & SOF_RT5682_MCLK_EN) && @@ -315,7 +317,16 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) RT5682S_DA_STEREO1_FILTER | RT5682S_AD_STEREO1_FILTER, RT5682S_CLK_SEL_I2S1_ASRC); - else + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(component, + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | RT5682_AD_STEREO1_FILTER, @@ -365,7 +376,12 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); + + if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + extra = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra); + } else + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); @@ -402,6 +418,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_MCLK; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_MCLK; else pll_source = RT5682_PLL1_S_MCLK; @@ -422,6 +440,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, } else { if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_BCLK1; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_BCLK1; else pll_source = RT5682_PLL1_S_BCLK1; @@ -431,6 +451,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -559,11 +582,30 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "SPOL" }, + { "Right Spk", NULL, "SPOR" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, + ARRAY_SIZE(rt5650_spk_dapm_routes)); + if (ret) + dev_err(rtd->dev, "fail to add dapm routes, ret=%d\n", ret); + + return ret; +} + static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -614,6 +656,17 @@ static struct snd_soc_dai_link_component rt5682s_component[] = { } }; +static struct snd_soc_dai_link_component rt5650_components[] = { + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif1", + }, + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif2", + } +}; + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -652,6 +705,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { links[id].codecs = rt5682s_component; links[id].num_codecs = ARRAY_SIZE(rt5682s_component); + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[0]; + links[id].num_codecs = 1; } else { links[id].codecs = rt5682_component; links[id].num_codecs = ARRAY_SIZE(rt5682_component); @@ -804,6 +860,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[1]; + links[id].num_codecs = 1; + links[id].init = rt5650_spk_init; + links[id].ops = &sof_rt5682_ops; } else { max_98357a_dai_link(&links[id]); } @@ -890,6 +951,12 @@ static int sof_audio_probe(struct platform_device *pdev) /* Detect the headset codec variant */ if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + else if (acpi_dev_present("10EC5650", NULL, -1)) { + sof_rt5682_quirk |= SOF_RT5650_HEADPHONE_CODEC_PRESENT; + + sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", + GFP_KERNEL); + } if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; @@ -1178,6 +1245,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "jsl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index f5c7e1bbded0..36ea2bacbe7d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -34,6 +34,11 @@ static const struct snd_soc_acpi_codecs mx98360a_spk = { .codecs = {"MX98360A"} }; +static struct snd_soc_acpi_codecs rt5650_spk = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -98,6 +103,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "jsl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt5650_spk, + .sof_tplg_filename = "sof-jsl-rt5650-dual.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

3 2

[PATCH] ASoC: amd: ps: Fix extraneous error messages
by Mario Limonciello 12 Jul '23

12 Jul '23

On Pink Sardine machines that don't have SdW controllers in use, the property `mipi-sdw-manager-list` won't exist. There is no point in showing an error to a user when this situation is encountered. Furthermore if the machine doesn't have a DMIC connected to the ACP, there may be no platform devices created either. Downgrade the associated message to debug. Fixes: d1351c30ac8a6 ("ASoC: amd: ps: create platform devices based on acp config") Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 5b46dc8573f86..4af3c3665387d 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -257,7 +257,7 @@ static int sdw_amd_scan_controller(struct device *dev) &sdw_manager_bitmap, 1); if (ret) { - dev_err(dev, "Failed to read mipi-sdw-manager-list: %d\n", ret); + dev_dbg(dev, "Failed to read mipi-sdw-manager-list: %d\n", ret); return -EINVAL; } count = hweight32(sdw_manager_bitmap); @@ -641,7 +641,7 @@ static int snd_acp63_probe(struct pci_dev *pci, ret = get_acp63_device_config(val, pci, adata); /* ACP PCI driver probe should be continued even PDM or SoundWire Devices are not found */ if (ret) { - dev_err(&pci->dev, "get acp device config failed:%d\n", ret); + dev_dbg(&pci->dev, "get acp device config failed:%d\n", ret); goto skip_pdev_creation; } ret = create_acp63_platform_devs(pci, adata, addr); base-commit: 8689f4f2ea561dd080118eeb05c0255ac9542905 -- 2.34.1

2 1

[PATCH] ASoC: fsl_sai: Revert "ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode"
by Fabio Estevam 12 Jul '23

12 Jul '23

From: Fabio Estevam <festevam(a)denx.de> This reverts commit ff87d619ac180444db297f043962a5c325ded47b. Andreas reports that on an i.MX8MP-based system where MCLK needs to be used as an input, the MCLK pin is actually an output, despite not having the 'fsl,sai-mclk-direction-output' property present in the devicetree. This is caused by commit ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") that sets FSL_SAI_MCTL_MCLK_EN unconditionally for imx8mm/8mn/8mp/93, causing the MCLK to always be configured as output. FSL_SAI_MCTL_MCLK_EN corresponds to the MOE (MCLK Output Enable) bit of register MCR and the drivers sets it when the 'fsl,sai-mclk-direction-output' devicetree property is present. Revert the commit to allow SAI to use MCLK as input as well. Cc: stable(a)vger.kernel.org Fixes: ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") Reported-by: Andreas Henriksson <andreas(a)fatal.se> Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- sound/soc/fsl/fsl_sai.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5e09f634c61b..54b4bf3744c6 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -507,12 +507,6 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) savediv / 2 - 1); } - if (sai->soc_data->max_register >= FSL_SAI_MCTL) { - /* SAI is in master mode at this point, so enable MCLK */ - regmap_update_bits(sai->regmap, FSL_SAI_MCTL, - FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); - } - return 0; } -- 2.34.1

3 2

[PATCH] ASoC: qcom: q6apm: do not close GPR port before closing graph
by Srinivas Kandagatla 11 Jul '23

11 Jul '23

Closing GPR port before graph close can result in un handled notifications from DSP, this results in spam of errors from GPR driver as there is no one to handle these notification at that point in time. Fix this by closing GPR port after graph close is finished. Fixes: 5477518b8a0e ("ASoC: qdsp6: audioreach: add q6apm support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c index 7bfac9492ab5..5d44d07acd69 100644 --- a/sound/soc/qcom/qdsp6/q6apm.c +++ b/sound/soc/qcom/qdsp6/q6apm.c @@ -511,6 +511,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) switch (hdr->opcode) { case DATA_CMD_RSP_WR_SH_MEM_EP_DATA_BUFFER_DONE_V2: + if (!graph->ar_graph) + break; client_event = APM_CLIENT_EVENT_DATA_WRITE_DONE; mutex_lock(&graph->lock); token = hdr->token & APM_WRITE_TOKEN_MASK; @@ -544,6 +546,8 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op) wake_up(&graph->cmd_wait); break; case DATA_CMD_RSP_RD_SH_MEM_EP_DATA_BUFFER_V2: + if (!graph->ar_graph) + break; client_event = APM_CLIENT_EVENT_DATA_READ_DONE; mutex_lock(&graph->lock); rd_done = data->payload; @@ -649,8 +653,9 @@ int q6apm_graph_close(struct q6apm_graph *graph) { struct audioreach_graph *ar_graph = graph->ar_graph; - gpr_free_port(graph->port); + graph->ar_graph = NULL; kref_put(&ar_graph->refcount, q6apm_put_audioreach_graph); + gpr_free_port(graph->port); kfree(graph); return 0; -- 2.25.1

2 1

[PATCH 0/8] ASoC/soundwire/qdsp6/wcd: fix leaks and probe deferral
by Johan Hovold 11 Jul '23

11 Jul '23

I've been hitting a race during boot which breaks probe of the sound card on the Lenovo ThinkPad X13s as I've previously reported here: https://lore.kernel.org/all/ZIHMMFtuDtvdpFAZ@hovoldconsulting.com/ The immediate issue appeared to be a probe deferral that was turned into a hard failure, but addressing that in itself only made things worse as it exposed further bugs. I was hoping someone more familiar with the code in question would look into this, but as this affects users of the X13s and breaks audio on my machine every fifth boot or so, I decided to investigate it myself. As expected, the Qualcomm codec drivers are broken and specifically leak resources on component remove, which in turn breaks sound card probe deferrals. The source of the deferral itself appears to be legitimate and was simply due to some audio component not yet having been registered due to random changes in timing during boot. These issues can most easily be reproduced by simply blacklisting the q6apm_dai module and loading it manually after boot. The sound card probe deferral also exposes a bug in the soundwire subsystem, which uses completion structures for signalling that a device has been enumerated on the bus and initialised. The way this is implemented prevents reprobed codec drivers from learning that the soundwire devices are still attached, which causes probe to fail. Included are also two patches that suppresses error messages on component probe deferral to avoid spamming the logs during boot. These patches should preferably all go through the ASoC tree even if merging the soundwire fix separately also works. Note the ASoC tree already has the following related fixes: https://lore.kernel.org/lkml/20230630120318.6571-1-johan+linaro@kernel.org/ https://lore.kernel.org/lkml/20230630142717.5314-1-johan+linaro@kernel.org/ https://lore.kernel.org/lkml/20230701094723.29379-1-johan+linaro@kernel.org/ https://lore.kernel.org/lkml/20230703124701.11734-1-johan+linaro@kernel.org/ Johan Johan Hovold (8): soundwire: fix enumeration completion ASoC: qdsp6: audioreach: fix topology probe deferral ASoC: codecs: wcd938x: fix missing clsh ctrl error handling ASoC: codecs: wcd938x: fix resource leaks on component remove ASoC: codecs: wcd934x: fix resource leaks on component remove ASoC: codecs: wcd-mbhc-v2: fix resource leaks on component remove ASoC: topology: suppress probe deferral errors ASoC: core: suppress probe deferral errors drivers/soundwire/bus.c | 8 ++--- sound/soc/codecs/wcd-mbhc-v2.c | 57 ++++++++++++++++++++++--------- sound/soc/codecs/wcd934x.c | 12 +++++++ sound/soc/codecs/wcd938x.c | 59 +++++++++++++++++++++++++++++---- sound/soc/qcom/qdsp6/topology.c | 4 +-- sound/soc/soc-core.c | 6 ++-- sound/soc/soc-topology.c | 10 ++++-- 7 files changed, 122 insertions(+), 34 deletions(-) -- 2.39.3

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