May 2023 - Alsa-devel - mailman.alsa-project.org

[ASoC: da7219: Add Jack insertion detection polarity selection 1/2] ASoC: dt-bindings: dialog,da7219: convert to dtschema
by David Rau 23 May '23

23 May '23

- Convert Dialog DA7219 bindings to DT schema format. - Remove unused `dlg,ldo-lvl` property. Signed-off-by: David Rau <David.Rau.opensource(a)dm.renesas.com> --- .../devicetree/bindings/sound/da7219.txt | 112 --------- .../bindings/sound/dialog,da7219.yaml | 230 ++++++++++++++++++ 2 files changed, 230 insertions(+), 112 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 Documentation/devicetree/bindings/sound/dialog,da7219.yaml diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt deleted file mode 100644 index add1caf26ac2..000000000000 --- a/Documentation/devicetree/bindings/sound/da7219.txt +++ /dev/null @@ -1,112 +0,0 @@ -Dialog Semiconductor DA7219 Audio Codec bindings - -DA7219 is an audio codec with advanced accessory detect features. - -====== - -Required properties: -- compatible : Should be "dlg,da7219" -- reg: Specifies the I2C slave address - -- interrupts : IRQ line info for DA7219. - (See Documentation/devicetree/bindings/interrupt-controller/interrupts.txt for - further information relating to interrupt properties) - -- VDD-supply: VDD power supply for the device -- VDDMIC-supply: VDDMIC power supply for the device -- VDDIO-supply: VDDIO power supply for the device - (See Documentation/devicetree/bindings/regulator/regulator.txt for further - information relating to regulators) - -Optional properties: -- interrupt-names : Name associated with interrupt line. Should be "wakeup" if - interrupt is to be used to wake system, otherwise "irq" should be used. -- wakeup-source: Flag to indicate this device can wake system (suspend/resume). - -- #clock-cells : Should be set to '<1>', two clock sources provided; -- clock-output-names : Names given for DAI clock outputs (WCLK & BCLK); - -- clocks : phandle and clock specifier for codec MCLK. -- clock-names : Clock name string for 'clocks' attribute, should be "mclk". - -- dlg,micbias-lvl : Voltage (mV) for Mic Bias - [<1600>, <1800>, <2000>, <2200>, <2400>, <2600>] -- dlg,mic-amp-in-sel : Mic input source type - ["diff", "se_p", "se_n"] - -Deprecated properties: -- dlg,ldo-lvl : Required internal LDO voltage (mV) level for digital engine - (LDO unavailable in production HW so property no longer required). - -====== - -Child node - 'da7219_aad': - -Optional properties: -- dlg,micbias-pulse-lvl : Mic bias higher voltage pulse level (mV). - [<2800>, <2900>] -- dlg,micbias-pulse-time : Mic bias higher voltage pulse duration (ms) -- dlg,btn-cfg : Periodic button press measurements for 4-pole jack (ms) - [<2>, <5>, <10>, <50>, <100>, <200>, <500>] -- dlg,mic-det-thr : Impedance threshold for mic detection measurement (Ohms) - [<200>, <500>, <750>, <1000>] -- dlg,jack-ins-deb : Debounce time for jack insertion (ms) - [<5>, <10>, <20>, <50>, <100>, <200>, <500>, <1000>] -- dlg,jack-det-rate: Jack type detection latency (3/4 pole) - ["32ms_64ms", "64ms_128ms", "128ms_256ms", "256ms_512ms"] -- dlg,jack-rem-deb : Debounce time for jack removal (ms) - [<1>, <5>, <10>, <20>] -- dlg,a-d-btn-thr : Impedance threshold between buttons A and D - [0x0 - 0xFF] -- dlg,d-b-btn-thr : Impedance threshold between buttons D and B - [0x0 - 0xFF] -- dlg,b-c-btn-thr : Impedance threshold between buttons B and C - [0x0 - 0xFF] -- dlg,c-mic-btn-thr : Impedance threshold between button C and Mic - [0x0 - 0xFF] -- dlg,btn-avg : Number of 8-bit readings for averaged button measurement - [<1>, <2>, <4>, <8>] -- dlg,adc-1bit-rpt : Repeat count for 1-bit button measurement - [<1>, <2>, <4>, <8>] - -====== - -Example: - - codec: da7219@1a { - compatible = "dlg,da7219"; - reg = <0x1a>; - - interrupt-parent = <&gpio6>; - interrupts = <11 IRQ_TYPE_LEVEL_LOW>; - - VDD-supply = <&reg_audio>; - VDDMIC-supply = <&reg_audio>; - VDDIO-supply = <&reg_audio>; - - #clock-cells = <1>; - clock-output-names = "dai-wclk", "dai-bclk"; - - clocks = <&clks 201>; - clock-names = "mclk"; - - dlg,ldo-lvl = <1200>; - dlg,micbias-lvl = <2600>; - dlg,mic-amp-in-sel = "diff"; - - da7219_aad { - dlg,btn-cfg = <50>; - dlg,mic-det-thr = <500>; - dlg,jack-ins-deb = <20>; - dlg,jack-det-rate = "32ms_64ms"; - dlg,jack-rem-deb = <1>; - - dlg,a-d-btn-thr = <0xa>; - dlg,d-b-btn-thr = <0x16>; - dlg,b-c-btn-thr = <0x21>; - dlg,c-mic-btn-thr = <0x3E>; - - dlg,btn-avg = <4>; - dlg,adc-1bit-rpt = <1>; - }; - }; diff --git a/Documentation/devicetree/bindings/sound/dialog,da7219.yaml b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml new file mode 100644 index 000000000000..9160f68696dd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dialog,da7219.yaml @@ -0,0 +1,230 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dialog,da7219.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dialog Semiconductor DA7219 Audio Codec + +maintainers: + - David Rau <David.Rau.opensource(a)dm.renesas.com> + +description: + The DA7219 is an ultra low-power audio codec with + in-built advanced accessory detection (AAD) for mobile + computing and accessory applications, which supports + sample rates up to 96 kHz at 24-bit resolution. + +properties: + compatible: + const: dlg,da7219 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + VDD-supply: + description: + VDD power supply for the device. + + VDDMIC-supply: + description: + VDDMIC power supply for the device. + + VDDIO-supply: + description: + VDDIO power supply for the device. + + interrupt-names: + description: + Should be "wakeup" if interrupt is to be used to wake system, + otherwise "irq" should be used. + enum: + - wakeup + - irq + + wakeup-source: + type: boolean + description: + Flag to indicate this device can wake system (suspend/resume). + + "#clock-cells": + const: 1 + + clock-output-names: + minItems: 2 + maxItems: 2 + description: + Name given for DAI WCLK and BCLK outputs. + + clocks: + maxItems: 1 + description: + phandle and clock specifier for codec MCLK. + + clock-names: + const: mclk + + dlg,micbias-lvl: + enum: [1600, 1800, 2000, 2200, 2400, 2600] + description: + Voltage (mV) for Mic Bias. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-amp-in-sel: + enum: ["diff", "se_p", "se_n"] + description: + Mic input source type. + + diff - Differential. + + se_p - MIC_P. + Positive differential analog microphone input. + + se_n - MIC_N. + Negative differential analog microphone input. + $ref: /schemas/types.yaml#/definitions/string + + da7219_aad: + type: object + description: + Configuration of advanced accessory detection. + properties: + dlg,micbias-pulse-lvl: + enum: [2800, 2900] + description: + Mic bias higher voltage pulse level (mV). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,micbias-pulse-time: + description: + Mic bias higher voltage pulse duration (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + + dlg,btn-cfg: + enum: [2, 5, 10, 50, 100, 200, 500] + description: + Periodic button press measurements for 4-pole jack (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,mic-det-thr: + enum: [200, 500, 750, 1000] + description: + Impedance threshold for mic detection measurement (Ohms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-ins-deb: + enum: [5, 10, 20, 50, 100, 200, 500, 1000] + description: + Debounce time for jack insertion (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,jack-det-rate: + enum: ["32_64", "64_128", "128_256", "256_512"] + description: + Jack type (3/4 pole) detection latency (ms). + $ref: /schemas/types.yaml#/definitions/string + + dlg,jack-rem-deb: + enum: [1, 5, 10, 20] + description: + Debounce time for jack removal (ms). + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,a-d-btn-thr: + description: + Impedance threshold between buttons A and D. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,d-b-btn-thr: + description: + Impedance threshold between buttons D and B. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,b-c-btn-thr: + description: + Impedance threshold between buttons B and C. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,c-mic-btn-thr: + description: + Impedance threshold between button C and Mic. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 255 + + dlg,btn-avg: + enum: [1, 2, 4, 8] + description: + Number of 8-bit readings for averaged button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + + dlg,adc-1bit-rpt: + enum: [1, 2, 4, 8] + description: + Repeat count for 1-bit button measurement. + $ref: /schemas/types.yaml#/definitions/uint32 + +required: + - compatible + - reg + - interrupts + - VDD-supply + - VDDMIC-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec: da7219@1a { + compatible = "dlg,da7219"; + reg = <0x1a>; + + interrupt-parent = <&gpio6>; + interrupts = <11 IRQ_TYPE_LEVEL_LOW>; + + VDD-supply = <&vdd_reg>; + VDDMIC-supply = <&vddmic_reg>; + VDDIO-supply = <&vddio_reg>; + + #clock-cells = <1>; + clock-output-names = "da7219-dai-wclk", "da7219-dai-bclk"; + + clocks = <&clks 201>; + clock-names = "mclk"; + + dlg,micbias-lvl = <2600>; + dlg,mic-amp-in-sel = "diff"; + + da7219_aad { + dlg,btn-cfg = <50>; + dlg,mic-det-thr = <500>; + dlg,jack-ins-deb = <20>; + dlg,jack-det-rate = "32_64"; + dlg,jack-rem-deb = <1>; + + dlg,a-d-btn-thr = <0xa>; + dlg,d-b-btn-thr = <0x16>; + dlg,b-c-btn-thr = <0x21>; + dlg,c-mic-btn-thr = <0x3E>; + + dlg,btn-avg = <4>; + dlg,adc-1bit-rpt = <1>; + }; + }; + }; -- 2.17.1

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[PATCH] ASoC: SOF: amd: Add pci revision id check
by Venkata Prasad Potturu 23 May '23

23 May '23

Add pci revision id check for renoir and rembrandt platforms. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp.h | 3 +++ sound/soc/sof/amd/pci-rmb.c | 3 +++ sound/soc/sof/amd/pci-rn.c | 3 +++ 3 files changed, 9 insertions(+) diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 1c535cc6c3a9..dc624f727aa3 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -55,6 +55,9 @@ #define ACP_DSP_TO_HOST_IRQ 0x04 +#define ACP_RN_PCI_ID 0x01 +#define ACP_RMB_PCI_ID 0x6F + #define HOST_BRIDGE_CZN 0x1630 #define HOST_BRIDGE_RMB 0x14B5 #define ACP_SHA_STAT 0x8000 diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index eaf70ea6e556..58b3092425f1 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -65,6 +65,9 @@ static int acp_pci_rmb_probe(struct pci_dev *pci, const struct pci_device_id *pc { unsigned int flag; + if (pci->revision != ACP_RMB_PCI_ID) + return -ENODEV; + flag = snd_amd_acp_find_config(pci); if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) return -ENODEV; diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 4809cb644152..7409e21ce5aa 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -65,6 +65,9 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci { unsigned int flag; + if (pci->revision != ACP_RN_PCI_ID) + return -ENODEV; + flag = snd_amd_acp_find_config(pci); if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) return -ENODEV; -- 2.25.1

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[PATCH] ASoC: nau8825: Add pre-charge actions for input
by David Lin 23 May '23

23 May '23

Adding pre-charge actions to make FEPGA power stable faster. It improve the recording quality at the beginning. Thus, it is also meaningfully to decrease the final adc delay time. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 30 ++++++++++++++++++++++++++++-- sound/soc/codecs/nau8825.h | 7 +++++++ 2 files changed, 35 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 775c8e0cb09e..cc3e18207c42 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -911,6 +911,32 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) } } +static int nau8825_fepga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct nau8825 *nau8825 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8825->regmap, NAU8825_REG_FEPGA, + NAU8825_ACDC_CTRL_MASK, + NAU8825_ACDC_VREF_MICP | NAU8825_ACDC_VREF_MICN); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_DISCHRG_EN, NAU8825_DISCHRG_EN); + msleep(40); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_DISCHRG_EN, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FEPGA, + NAU8825_ACDC_CTRL_MASK, 0); + break; + default: + break; + } + + return 0; +} + static int nau8825_adc_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1127,8 +1153,8 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC"), SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), - SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, - NULL, 0), + SND_SOC_DAPM_PGA_E("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0, nau8825_fepga_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, nau8825_adc_event, SND_SOC_DAPM_POST_PMU | diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 44b62bc3880f..38ce052aed50 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -442,10 +442,17 @@ /* BOOST (0x76) */ #define NAU8825_PRECHARGE_DIS (1 << 13) #define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_DISCHRG_EN (1 << 11) #define NAU8825_HP_BOOST_DIS (1 << 9) #define NAU8825_HP_BOOST_G_DIS (1 << 8) #define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) +/* FEPGA (0x77) */ +#define NAU8825_ACDC_CTRL_SFT 14 +#define NAU8825_ACDC_CTRL_MASK (0x3 << NAU8825_ACDC_CTRL_SFT) +#define NAU8825_ACDC_VREF_MICP (0x1 << NAU8825_ACDC_CTRL_SFT) +#define NAU8825_ACDC_VREF_MICN (0x2 << NAU8825_ACDC_CTRL_SFT) + /* POWER_UP_CONTROL (0x7f) */ #define NAU8825_POWERUP_INTEGR_R (1 << 5) #define NAU8825_POWERUP_INTEGR_L (1 << 4) -- 2.25.1

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[PATCH] ASoC: dt-bindings: rt1016: Convert to dtschema
by Anup Sharma 23 May '23

23 May '23

Convert the RT1016 Stereo Audio Amplifier bindings to DT schema Signed-off-by: Anup Sharma <anupnewsmail(a)gmail.com> --- .../bindings/sound/realtek,rt1016.yaml | 40 +++++++++++++++++++ .../devicetree/bindings/sound/rt1016.txt | 17 -------- 2 files changed, 40 insertions(+), 17 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1016.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt1016.txt diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml new file mode 100644 index 000000000000..5287e9c9197e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt1016.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt1016.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Reaktek RT1016 Stereo Class D Audio Amplifier + +maintainers: + - oder_chiou(a)realtek.com + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt1016 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1a { + compatible = "realtek,rt1016"; + reg = <0x1a>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rt1016.txt b/Documentation/devicetree/bindings/sound/rt1016.txt deleted file mode 100644 index 2310f8ff259b..000000000000 --- a/Documentation/devicetree/bindings/sound/rt1016.txt +++ /dev/null @@ -1,17 +0,0 @@ -RT1016 Stereo Class D Audio Amplifier - -This device supports I2C only. - -Required properties: - -- compatible : "realtek,rt1016". - -- reg : The I2C address of the device. - - -Example: - -rt1016: codec@1a { - compatible = "realtek,rt1016"; - reg = <0x1a>; -}; -- 2.34.1

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[PATCH alsa-lib v2 00/10] Add MIDI 2.0 support
by Takashi Iwai 23 May '23

23 May '23

This is a revised series of patches for alsa-lib to add the support of MIDI 2.0 features. Since v1 [*], two more control API helpers to obtain UMP EP / block info are added and the CTL protocol version is bumped. Takashi [*] http://lore.kernel.org/r/20230520070021.1301-1-tiwai@suse.de === Takashi Iwai (10): uapi: Update rawmidi API to 2.0.3 uapi: Update control API to 2.0.9 rawmidi: Add UMP ioctl support ump: Add initial support control: Add UMP device query support control: Add UMP Endpoint and Block info query support ump: Add helpers to parse / set UMP packet data ump: Add helpers for handling SysEx data uapi: Update asequencer.h definitions for 1.0.3 seq: Add UMP support configure.ac | 1 + include/Makefile.am | 2 +- include/control.h | 3 + include/local.h | 5 + include/rawmidi.h | 3 + include/seq.h | 44 ++ include/seq_event.h | 42 +- include/seqmid.h | 24 ++ include/sound/uapi/asequencer.h | 91 ++++- include/sound/uapi/asound.h | 62 ++- include/ump.h | 119 ++++++ include/ump_msg.h | 598 +++++++++++++++++++++++++++ src/Versions.in | 26 ++ src/control/control.c | 38 ++ src/control/control_hw.c | 29 ++ src/control/control_local.h | 3 + src/rawmidi/Makefile.am | 5 +- src/rawmidi/rawmidi.c | 19 + src/rawmidi/rawmidi_hw.c | 36 +- src/rawmidi/rawmidi_local.h | 7 + src/rawmidi/ump.c | 702 ++++++++++++++++++++++++++++++++ src/rawmidi/ump_local.h | 10 + src/seq/seq.c | 410 +++++++++++++++++-- src/seq/seq_hw.c | 72 +++- src/seq/seq_local.h | 6 +- src/seq/seqmid.c | 38 ++ 26 files changed, 2316 insertions(+), 79 deletions(-) create mode 100644 include/ump.h create mode 100644 include/ump_msg.h create mode 100644 src/rawmidi/ump.c create mode 100644 src/rawmidi/ump_local.h -- 2.35.3

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[PATCH 1/2] ALSA: emu10k1: introduce higher-level voice manipulation functions
by Oswald Buddenhagen 23 May '23

23 May '23

This adds snd_emu10k1_pcm_init_{voices,extra_voice}() and snd_emu10k1_playback_{un,}mute_voices() to slightly abstract by voice function and potential stereo property. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/pci/emu10k1/emupcm.c | 70 +++++++++++++++++++++++++++++--------- 1 file changed, 53 insertions(+), 17 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 0036593cca7c..65af94d08b47 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -312,6 +312,30 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, spin_unlock_irqrestore(&emu->reg_lock, flags); } +static void snd_emu10k1_pcm_init_voices(struct snd_emu10k1 *emu, + struct snd_emu10k1_voice *evoice, + bool w_16, bool stereo, + unsigned int start_addr, + unsigned int end_addr, + struct snd_emu10k1_pcm_mixer *mix) +{ + snd_emu10k1_pcm_init_voice(emu, 1, 0, evoice, w_16, stereo, + start_addr, end_addr, mix); + if (stereo) + snd_emu10k1_pcm_init_voice(emu, 0, 0, evoice + 1, w_16, true, + start_addr, end_addr, mix); +} + +static void snd_emu10k1_pcm_init_extra_voice(struct snd_emu10k1 *emu, + struct snd_emu10k1_voice *evoice, + bool w_16, + unsigned int start_addr, + unsigned int end_addr) +{ + snd_emu10k1_pcm_init_voice(emu, 1, 1, evoice, w_16, false, + start_addr, end_addr, NULL); +} + static int snd_emu10k1_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { @@ -393,18 +417,15 @@ static int snd_emu10k1_playback_prepare(struct snd_pcm_substream *substream) start_addr = epcm->start_addr >> w_16; end_addr = start_addr + runtime->period_size; - snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra, w_16, false, - start_addr, end_addr, NULL); + snd_emu10k1_pcm_init_extra_voice(emu, epcm->extra, w_16, + start_addr, end_addr); start_addr >>= stereo; epcm->ccca_start_addr = start_addr; end_addr = start_addr + runtime->buffer_size; - snd_emu10k1_pcm_init_voice(emu, 1, 0, epcm->voices[0], w_16, stereo, - start_addr, end_addr, - &emu->pcm_mixer[substream->number]); - if (stereo) - snd_emu10k1_pcm_init_voice(emu, 0, 0, epcm->voices[0] + 1, w_16, true, - start_addr, end_addr, - &emu->pcm_mixer[substream->number]); + snd_emu10k1_pcm_init_voices(emu, epcm->voices[0], w_16, stereo, + start_addr, end_addr, + &emu->pcm_mixer[substream->number]); + return 0; } @@ -421,8 +442,8 @@ static int snd_emu10k1_efx_playback_prepare(struct snd_pcm_substream *substream) channel_size = runtime->buffer_size; - snd_emu10k1_pcm_init_voice(emu, 1, 1, epcm->extra, true, false, - start_addr, start_addr + (channel_size / 2), NULL); + snd_emu10k1_pcm_init_extra_voice(emu, epcm->extra, true, + start_addr, start_addr + (channel_size / 2)); epcm->ccca_start_addr = start_addr; for (i = 0; i < NUM_EFX_PLAYBACK; i++) { @@ -598,12 +619,31 @@ static void snd_emu10k1_playback_unmute_voice(struct snd_emu10k1 *emu, snd_emu10k1_playback_commit_volume(emu, evoice, vattn); } +static void snd_emu10k1_playback_unmute_voices(struct snd_emu10k1 *emu, + struct snd_emu10k1_voice *evoice, + bool stereo, + struct snd_emu10k1_pcm_mixer *mix) +{ + snd_emu10k1_playback_unmute_voice(emu, evoice, stereo, true, mix); + if (stereo) + snd_emu10k1_playback_unmute_voice(emu, evoice + 1, true, false, mix); +} + static void snd_emu10k1_playback_mute_voice(struct snd_emu10k1 *emu, struct snd_emu10k1_voice *evoice) { snd_emu10k1_playback_commit_volume(emu, evoice, 0); } +static void snd_emu10k1_playback_mute_voices(struct snd_emu10k1 *emu, + struct snd_emu10k1_voice *evoice, + bool stereo) +{ + snd_emu10k1_playback_mute_voice(emu, evoice); + if (stereo) + snd_emu10k1_playback_mute_voice(emu, evoice + 1); +} + static void snd_emu10k1_playback_commit_pitch(struct snd_emu10k1 *emu, u32 voice, u32 pitch_target) { @@ -680,22 +720,18 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: mix = &emu->pcm_mixer[substream->number]; - snd_emu10k1_playback_unmute_voice(emu, epcm->voices[0], stereo, true, mix); - if (stereo) - snd_emu10k1_playback_unmute_voice(emu, epcm->voices[0] + 1, true, false, mix); + snd_emu10k1_playback_unmute_voices(emu, epcm->voices[0], stereo, mix); snd_emu10k1_playback_set_running(emu, epcm); snd_emu10k1_playback_trigger_voice(emu, epcm->voices[0]); snd_emu10k1_playback_trigger_voice(emu, epcm->extra); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: snd_emu10k1_playback_stop_voice(emu, epcm->voices[0]); snd_emu10k1_playback_stop_voice(emu, epcm->extra); snd_emu10k1_playback_set_stopped(emu, epcm); - snd_emu10k1_playback_mute_voice(emu, epcm->voices[0]); - if (stereo) - snd_emu10k1_playback_mute_voice(emu, epcm->voices[0] + 1); + snd_emu10k1_playback_mute_voices(emu, epcm->voices[0], stereo); break; default: result = -EINVAL; -- 2.40.0.152.g15d061e6df

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[PATCH v1 1/3] ASoC: jz4740-i2s: Add support for X1000 SoC
by Aidan MacDonald 23 May '23

23 May '23

The X1000's AIC is similar to the AIC found on other Ingenic SoCs. It has symmetric playback/capture rates like the JZ4740, but more flexible clocking when outputting the system or bit clocks. Signed-off-by: Aidan MacDonald <aidanmacdonald.0x0(a)gmail.com> --- sound/soc/jz4740/jz4740-i2s.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 6d9cfe0a5041..7cb563bb8b09 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -329,6 +329,14 @@ static const struct i2s_soc_info jz4760_i2s_soc_info = { .field_i2sdiv_playback = REG_FIELD(JZ_REG_AIC_CLK_DIV, 0, 3), }; +static const struct i2s_soc_info x1000_i2s_soc_info = { + .dai = &jz4740_i2s_dai, + .field_rx_fifo_thresh = REG_FIELD(JZ_REG_AIC_CONF, 24, 27), + .field_tx_fifo_thresh = REG_FIELD(JZ_REG_AIC_CONF, 16, 20), + .field_i2sdiv_capture = REG_FIELD(JZ_REG_AIC_CLK_DIV, 0, 8), + .field_i2sdiv_playback = REG_FIELD(JZ_REG_AIC_CLK_DIV, 0, 8), +}; + static struct snd_soc_dai_driver jz4770_i2s_dai = { .probe = jz4740_i2s_dai_probe, .playback = { @@ -440,6 +448,7 @@ static const struct of_device_id jz4740_of_matches[] = { { .compatible = "ingenic,jz4760-i2s", .data = &jz4760_i2s_soc_info }, { .compatible = "ingenic,jz4770-i2s", .data = &jz4770_i2s_soc_info }, { .compatible = "ingenic,jz4780-i2s", .data = &jz4780_i2s_soc_info }, + { .compatible = "ingenic,x1000-i2s", .data = &x1000_i2s_soc_info }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, jz4740_of_matches); -- 2.39.2

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[PATCH 0/5] ASoC: topology: Clean up error messages handling
by Amadeusz Sławiński 22 May '23

22 May '23

Series of patches cleaning up error messages when loading topology. In few places instead of logging in place of failure message is logged in caller. Additionally there are places where both caller and failing function log error, leading to unnecessary logs. Clean all of the above up. Amadeusz Sławiński (5): ASoC: topology: Log control load errors in soc_tplg_control_load() ASoC: topology: Remove redundant logs ASoC: topology: Do not split message string on multiple lines ASoC: topology: Remove redundant log ASoC: topology: Remove redundant log sound/soc/soc-topology.c | 63 ++++++++++++---------------------------- 1 file changed, 19 insertions(+), 44 deletions(-) -- 2.34.1

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[PATCH 0/5] ASoC: SOF: ipc4: Querry CPC value from firmware's manifest
by Peter Ujfalusi 22 May '23

22 May '23

Hi, Hi, The MOD_INIT_INSTANCE message contains a CPC (Cycles Per Chunk/processing unit) parameter. This CPC value is used by the firmware to calculate the total cycles needed by the enabled module instances and based on this it can decide to set the frequency of the DSP core(s). The manifest section of the firmware image contains a module configuration section, where a per module table of configurations are listed with measured CPC values as triplet of IBS/IBS/CPC (Input/Output buffer size - corresponding to the selected audio format). In case the CPC value is 0 (missing from the manifest or the configuration cannot be matched) the firmware will force the DSP cores to maximum speed to avoid audio glitches due to starvation. In these cases the kernel will print a warning message to let the SOF developers know about the gap and provide information to correct it with a firmware update. Regards, Peter --- Peter Ujfalusi (5): ASoC: SOF: ipc4-loader: Drop unused bss_size from struct sof_ipc4_fw_module ASoC: SOF: ipc4-loader: Save a pointer to fm_config in sof_ipc4_fw_module ASoC: SOF: ipc4-topology: Rename sof_ipc4_update_pipeline_mem_usage() to be generic ASoC: SOF: ipc4-topology: Do not use the CPC value from topology ASoC: SOF: ipc4-loader/topology: Query the CPC value from manifest sound/soc/sof/ipc4-loader.c | 72 ++++++++++++++++++++++++++++++++--- sound/soc/sof/ipc4-priv.h | 10 ++++- sound/soc/sof/ipc4-topology.c | 33 +++++++++------- 3 files changed, 94 insertions(+), 21 deletions(-) -- 2.40.1

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[PATCH] ASoC: dt-bindings: tlv320aic32x4: Fix supply names
by David Epping 22 May '23

22 May '23

The term "-supply" is a suffix to regulator names. Signed-off-by: David Epping <david.epping(a)missinglinkelectronics.com> --- Documentation/devicetree/bindings/sound/tlv320aic32x4.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt index f59125bc79d1..0b4e21bde5bc 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt @@ -8,7 +8,7 @@ Required properties: "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 "ti,tas2505" TAS2505, TAS2521 - reg: I2C slave address - - supply-*: Required supply regulators are: + - *-supply: Required supply regulators are: "iov" - digital IO power supply "ldoin" - LDO power supply "dv" - Digital core power supply -- 2.17.1

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