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/linux-6.12.1/Documentation/devicetree/bindings/net/
Dlantiq,pef2256.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Herve Codina <herve.codina@bootlin.com>
20 - const: lantiq,pef2256
27 - description: Master Clock
28 - description: System Clock Receive
29 - description: System Clock Transmit
31 clock-names:
33 - const: mclk
[all …]
/linux-6.12.1/Documentation/devicetree/bindings/sound/
Drenesas,idt821034.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Herve Codina <herve.codina@bootlin.com>
13 The IDT821034 codec is a four channel PCM codec with onchip filters and
16 The time-slots used by the codec must be set and so, the properties
17 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and
18 'dai-tdm-slot-rx-mask' must be present in the ALSA sound card node for
19 sub-nodes that involve the codec. The codec uses one 8bit time-slot per
21 'dai-tdm-tdm-slot-with' must be set to 8.
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Dinfineon,peb2466.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Herve Codina <herve.codina@bootlin.com>
13 The Infineon PEB2466 codec is a programmable DSP-based four channels codec
14 with filters capabilities.
16 The time-slots used by the codec must be set and so, the properties
17 'dai-tdm-slot-num', 'dai-tdm-slot-width', 'dai-tdm-slot-tx-mask' and
18 'dai-tdm-slot-rx-mask' must be present in the sound card node for sub-nodes
19 that involve the codec. The codec uses one 8bit time-slot per channel.
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Damlogic,axg-sound-card.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
4 $id: http://devicetree.org/schemas/sound/amlogic,axg-sound-card.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Jerome Brunet <jbrunet@baylibre.com>
13 - $ref: sound-card-common.yaml#
17 const: amlogic,axg-sound-card
19 audio-aux-devs:
20 $ref: /schemas/types.yaml#/definitions/phandle-array
23 audio-widgets:
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Dqcom,q6dsp-lpass-ports.yaml1 # SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause)
3 ---
4 $id: http://devicetree.org/schemas/sound/qcom,q6dsp-lpass-ports.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
18 - qcom,q6afe-dais
20 '#sound-dai-cells':
23 '#address-cells':
26 '#size-cells':
31 '^dai@[0-9]+$':
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Dst,sti-asoc-card.txt3 The sti ASoC Sound Card can be used, for all sti SoCs using internal sti-sas
8 Documentation/devicetree/bindings/sound/simple-card.yaml.
10 1) sti-uniperiph-dai: audio dai device.
11 ---------------------------------------
14 - compatible: "st,stih407-uni-player-hdmi", "st,stih407-uni-player-pcm-out",
15 "st,stih407-uni-player-dac", "st,stih407-uni-player-spdif",
16 "st,stih407-uni-reader-pcm_in", "st,stih407-uni-reader-hdmi",
18 - st,syscfg: phandle to boot-device system configuration registers
20 - clock-names: name of the clocks listed in clocks property in the same order
22 - reg: CPU DAI IP Base address and size entries, listed in same
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Dsimple-card.yaml1 # SPDX-License-Identifier: GPL-2.0
3 ---
4 $id: http://devicetree.org/schemas/sound/simple-card.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
14 frame-master:
15 description: Indicates dai-link frame master.
18 bitclock-master:
19 description: Indicates dai-link bit clock master
22 frame-inversion:
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Dti,tas2781.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
2 # Copyright (C) 2022 - 2023 Texas Instruments Incorporated
4 ---
6 $schema: http://devicetree.org/meta-schemas/core.yaml#
11 - Shenghao Ding <shenghao-ding@ti.com>
14 The TAS2563/TAS2781 is a mono, digital input Class-D audio
16 small loudspeakers. An integrated on-chip DSP supports Texas
28 ti,tas2563: 6.1-W Boosted Class-D Audio Amplifier With Integrated
29 DSP and IV Sense, 16/20/24/32bit stereo I2S or multichannel TDM.
31 ti,tas2781: 24-V Class-D Amplifier with Real Time Integrated Speaker
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/linux-6.12.1/arch/arm64/boot/dts/amlogic/
Dmeson-g12b-bananapi-cm4-cm4io.dts1 // SPDX-License-Identifier: (GPL-2.0+ OR MIT)
6 /dts-v1/;
8 #include "meson-g12b-bananapi-cm4.dtsi"
9 #include <dt-bindings/input/input.h>
10 #include <dt-bindings/leds/common.h>
11 #include <dt-bindings/sound/meson-g12a-tohdmitx.h>
14 compatible = "bananapi,bpi-cm4io", "bananapi,bpi-cm4", "amlogic,a311d", "amlogic,g12b";
15 model = "BananaPi BPI-CM4IO Baseboard with BPI-CM4 Module";
23 adc-keys {
24 compatible = "adc-keys";
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Dmeson-g12b-bananapi-cm4-mnt-reform2.dts1 // SPDX-License-Identifier: (GPL-2.0+ OR MIT)
7 /dts-v1/;
9 #include "meson-g12b-bananapi-cm4.dtsi"
10 #include <dt-bindings/input/input.h>
11 #include <dt-bindings/leds/common.h>
12 #include <dt-bindings/sound/meson-g12a-tohdmitx.h>
15 model = "MNT Reform 2 with BPI-CM4 Module";
16 compatible = "mntre,reform2-cm4", "bananapi,bpi-cm4", "amlogic,a311d", "amlogic,g12b";
17 chassis-type = "laptop";
25 hdmi_connector: hdmi-connector {
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/linux-6.12.1/sound/soc/qcom/qdsp6/
Dq6afe-dai.c1 // SPDX-License-Identifier: GPL-2.0
2 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
5 #include <dt-bindings/sound/qcom,q6afe.h>
15 #include "q6dsp-lpass-ports.h"
16 #include "q6dsp-common.h"
39 struct snd_soc_dai *dai) in q6slim_hw_params() argument
42 struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); in q6slim_hw_params()
43 struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim; in q6slim_hw_params()
45 slim->sample_rate = params_rate(params); in q6slim_hw_params()
50 slim->bit_width = 16; in q6slim_hw_params()
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/linux-6.12.1/sound/soc/
Dsoc-utils.c1 // SPDX-License-Identifier: GPL-2.0+
3 // soc-util.c -- ALSA SoC Audio Layer utility functions
57 * snd_soc_tdm_params_to_bclk - calculate bclk from params and tdm slot info.
59 * Calculate the bclk from the params sample rate, the tdm slot count and the
60 * tdm slot width. Optionally round-up the slot count to a given multiple.
63 * If tdm_width == 0: use params_width() as the slot width.
64 * If tdm_slots == 0: use params_channels() as the slot count.
66 * If slot_multiple > 1 the slot count (or params_channels() if tdm_slots == 0)
75 * @tdm_width: Width in bits of the tdm slots. Must be >= 0.
76 * @tdm_slots: Number of tdm slots per frame. Must be >= 0.
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Dsoc-dai.c1 // SPDX-License-Identifier: GPL-2.0
3 // soc-dai.c
10 #include <sound/soc-dai.h>
11 #include <sound/soc-link.h>
13 #define soc_dai_ret(dai, ret) _soc_dai_ret(dai, __func__, ret) argument
14 static inline int _soc_dai_ret(const struct snd_soc_dai *dai, in _soc_dai_ret() argument
23 case -EPROBE_DEFER: in _soc_dai_ret()
24 case -ENOTSUPP: in _soc_dai_ret()
27 dev_err(dai->dev, in _soc_dai_ret()
29 func, dai->name, ret); in _soc_dai_ret()
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/linux-6.12.1/sound/soc/meson/
Daxg-tdm-interface.c1 // SPDX-License-Identifier: (GPL-2.0 OR MIT)
11 #include <sound/soc-dai.h>
13 #include "axg-tdm.h"
38 int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, in axg_tdm_set_tdm_slots() argument
42 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); in axg_tdm_set_tdm_slots()
43 struct axg_tdm_stream *tx = snd_soc_dai_dma_data_get_playback(dai); in axg_tdm_set_tdm_slots()
44 struct axg_tdm_stream *rx = snd_soc_dai_dma_data_get_capture(dai); in axg_tdm_set_tdm_slots()
51 /* We should at least have a slot for a valid interface */ in axg_tdm_set_tdm_slots()
53 dev_err(dai->dev, "interface has no slot\n"); in axg_tdm_set_tdm_slots()
54 return -EINVAL; in axg_tdm_set_tdm_slots()
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/linux-6.12.1/arch/arm/boot/dts/st/
Dstm32mp15xx-dhcom-pdk2.dtsi1 // SPDX-License-Identifier: GPL-2.0+ OR BSD-3-Clause
3 * Copyright (C) 2019-2020 Marek Vasut <marex@denx.de>
6 #include <dt-bindings/input/input.h>
7 #include <dt-bindings/pwm/pwm.h>
17 stdout-path = "serial0:115200n8";
20 clk_ext_audio_codec: clock-codec {
21 compatible = "fixed-clock";
22 #clock-cells = <0>;
23 clock-frequency = <24000000>;
26 display_bl: display-bl {
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/linux-6.12.1/sound/soc/fsl/
Dimx-card.c1 // SPDX-License-Identifier: GPL-2.0+
2 // Copyright 2017-2021 NXP
14 #include <sound/soc-dapm.h>
45 * Mapping TDM mode and frame width
54 * struct imx_card_plat_data - specific info for codecs
57 * @tdm_fs_mul: ratio of mclk/fs for tdm mode
59 * @support_tdm_rates: supported sample rate for tdm mode
61 * @support_tdm_channels: supported channels for tdm mode
88 * struct dai_link_data - specific info for dai link
90 * @slots: slot number
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/linux-6.12.1/sound/soc/codecs/
Dtas5720.c1 // SPDX-License-Identifier: GPL-2.0-only
3 * tas5720.c - ALSA SoC Texas Instruments TAS5720 Mono Audio Amplifier
5 * Copyright (C)2015-2016 Texas Instruments Incorporated - https://www.ti.com
22 #include <sound/soc-dapm.h>
37 "dvdd", /* Digital power supply. Connect to 3.3-V supply. */
38 "pvdd", /* Class-D amp and analog power supply (connected). */
55 struct snd_soc_dai *dai) in tas5720_hw_params() argument
57 struct snd_soc_component *component = dai->component; in tas5720_hw_params()
72 dev_err(component->dev, "unsupported sample rate: %u\n", rate); in tas5720_hw_params()
73 return -EINVAL; in tas5720_hw_params()
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Drtq9128.c1 // SPDX-License-Identifier: GPL-2.0-only
104 return -EINVAL; in rtq9128_i2c_write()
108 return i2c_smbus_write_i2c_block_data(i2c, reg, rg_size, data + count - rg_size); in rtq9128_i2c_write()
123 return -EINVAL; in rtq9128_i2c_read()
131 return -EIO; in rtq9128_i2c_read()
133 memset(val_buf, 0, val_size - rg_size); in rtq9128_i2c_read()
134 memcpy(val_buf + val_size - rg_size, data_tmp, rg_size); in rtq9128_i2c_read()
218 static const DECLARE_TLV_DB_SCALE(dig_tlv, -10375, 25, 0);
221 0, 3, TLV_DB_SCALE_ITEM(-600, 600, 0),
289 struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); in rtq9128_dac_power_event()
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Dssm3515.c1 // SPDX-License-Identifier: GPL-2.0-only OR MIT
101 // The specced range is -71.25...24.00 dB with step size of 0.375 dB,
102 // and a mute item below that. This is represented by -71.62...24.00 dB
103 // with the mute item mapped onto the low end.
104 static DECLARE_TLV_DB_MINMAX_MUTE(ssm3515_dac_volume, -7162, 2400);
141 dev_err(component->dev, "device reports:%s%s%s%s%s%s%s\n", in ssm3515_read_faults()
145 FIELD_GET(SSM3515_STATUS_AMP_OC, ret) ? " amp over-current fault" : "", in ssm3515_read_faults()
161 /* Disable the 'master power-down' */ in ssm3515_probe()
170 static int ssm3515_mute(struct snd_soc_dai *dai, int mute, int direction) in ssm3515_mute() argument
174 ret = snd_soc_component_update_bits(dai->component, in ssm3515_mute()
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Dtas6424.c1 // SPDX-License-Identifier: GPL-2.0
3 * ALSA SoC Texas Instruments TAS6424 Quad-Channel Audio Amplifier
5 * Copyright (C) 2016-2017 Texas Instruments Incorporated - https://www.ti.com/
23 #include <sound/soc-dapm.h>
32 "dvdd", /* Digital power supply. Connect to 3.3-V supply. */
34 "pvdd", /* Class-D amp output FETs supply. */
52 * DAC digital volumes. From -103.5 to 24 dB in 0.5 dB steps. Note that
53 * setting the gain below -100 dB (register value <0x7) is effectively a MUTE
56 static DECLARE_TLV_DB_SCALE(dac_tlv, -10350, 50, 0);
74 struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); in tas6424_dac_event()
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/linux-6.12.1/sound/soc/sh/
Dssi.c1 // SPDX-License-Identifier: GPL-2.0
15 * unit, and its pins are shared with the AC97 unit, among others.
23 * useful to support TDM mode codecs like the AD1939 which have a
24 * fixed TDM slot size, regardless of sample resolution.
62 #define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg)))
86 * track usage of the SSI; it is simplex-only so prevent attempts of
90 struct snd_soc_dai *dai) in ssi_startup() argument
92 struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; in ssi_startup()
93 if (ssi->inuse) { in ssi_startup()
95 return -EBUSY; in ssi_startup()
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/linux-6.12.1/sound/soc/atmel/
Dmchp-i2s-mcc.c1 // SPDX-License-Identifier: GPL-2.0
3 // Driver for Microchip I2S Multi-channel controller
29 * ---- I2S Controller Register map ----
75 * ---- Control Register (Write-only) ----
86 * ---- Mode Register A (Read/Write) ----
135 /* Number of TDM Channels - 1 */
138 ((((ch) - 1) << 13) & MCHP_I2SMCC_MRA_NBCHAN_MASK)
145 /* TDM Frame Synchronization */
163 /* Slot Width */
164 /* 0: slot is 32 bits wide for DATALENGTH = 18/20/24 bits. */
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/linux-6.12.1/sound/soc/sti/
Duniperif_reader.c1 // SPDX-License-Identifier: GPL-2.0-only
12 #define UNIPERIF_READER_I2S_IN 0 /* reader id connected to I2S/TDM TX bus */
41 * stream lock to avoid race condition with trigger callback.
49 spin_lock(&reader->irq_lock); in uni_reader_irq_handler()
50 if (!reader->substream) in uni_reader_irq_handler()
53 snd_pcm_stream_lock(reader->substream); in uni_reader_irq_handler()
54 if (reader->state == UNIPERIF_STATE_STOPPED) { in uni_reader_irq_handler()
56 dev_warn(reader->dev, "unexpected IRQ\n"); in uni_reader_irq_handler()
66 dev_err(reader->dev, "FIFO error detected\n"); in uni_reader_irq_handler()
68 snd_pcm_stop(reader->substream, SNDRV_PCM_STATE_XRUN); in uni_reader_irq_handler()
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/linux-6.12.1/sound/soc/intel/boards/
Dcht_bsw_rt5645.c1 // SPDX-License-Identifier: GPL-2.0-only
3 * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
4 * Cherrytrail and Braswell, with RT5645 codec.
25 #include <sound/soc-acpi.h>
27 #include "../atom/sst-atom-controls.h"
28 #include "../common/soc-intel-quirks.h"
31 #define CHT_CODEC_DAI1 "rt5645-aif1"
32 #define CHT_CODEC_DAI2 "rt5645-aif2"
69 struct snd_soc_dapm_context *dapm = w->dapm; in platform_clock_control()
70 struct snd_soc_card *card = dapm->card; in platform_clock_control()
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/linux-6.12.1/sound/soc/generic/
Dsimple-card-utils.c1 // SPDX-License-Identifier: GPL-2.0
3 // simple-card-utils.c
7 #include <dt-bindings/sound/audio-graph.h>
21 int val = -EINVAL; in simple_util_get_sample_fmt()
35 if (!strcmp(data->convert_sample_format, in simple_util_get_sample_fmt()
72 snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-rate"); in simple_util_parse_convert()
73 of_property_read_u32(np, prop, &data->convert_rate); in simple_util_parse_convert()
76 snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-channels"); in simple_util_parse_convert()
77 of_property_read_u32(np, prop, &data->convert_channels); in simple_util_parse_convert()
80 snprintf(prop, sizeof(prop), "%s%s", prefix, "convert-sample-format"); in simple_util_parse_convert()
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