| /kernel/linux/linux-5.10/Documentation/devicetree/bindings/sound/ |
| D | amlogic,axg-sound-card.txt | 5 - compatible: "amlogic,axg-sound-card"
6 - model : User specified audio sound card name, one string
10 - audio-aux-devs : List of phandles pointing to auxiliary devices
11 - audio-widgets : Please refer to widgets.txt.
12 - audio-routing : A list of the connections between audio components.
16 - dai-link: Container for dai-link level properties and the CODEC
17 sub-nodes. There should be at least one (and probably more)
20 Required dai-link properties:
22 - sound-dai: phandle and port of the CPU DAI.
24 Required TDM Backend dai-link properties:
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| D | qcom,q6afe.txt | 7 - compatible:
10 Definition: must be "qcom,q6afe-v<MAJOR-NUMBER>.<MINOR-NUMBER>"
16 "dais" subnode of the AFE node. It represents afe dais, each afe dai is a
17 subnode of "dais" representing board specific dai setup.
18 "dais" node should have following properties followed by dai children.
20 - compatible:
23 Definition: must be "qcom,q6afe-dais"
25 - #sound-dai-cells
30 - #address-cells
35 - #size-cells
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| D | st,sti-asoc-card.txt | 3 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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| D | simple-card.yaml | 1 # 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.
16 $ref: /schemas/types.yaml#/definitions/phandle-array
19 bitclock-master:
20 description: Indicates dai-link bit clock master
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| D | audio-graph-card.txt | 3 Audio Graph Card specifies audio DAI connections of SoC <-> codec.
8 see ${LINUX}/Documentation/devicetree/bindings/sound/simple-card.yaml
10 Below are same as Simple-Card.
12 - label
13 - widgets
14 - routing
15 - dai-format
16 - frame-master
17 - bitclock-master
18 - bitclock-inversion
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| D | renesas,rsnd.txt | 1 Renesas R-Car sound
7 Renesas R-Car and RZ/G sound is constructed from below modules
11 - SRC : Sampling Rate Converter
12 - CMD
13 - CTU : Channel Transfer Unit
14 - MIX : Mixer
15 - DVC : Digital Volume and Mute Function
25 Multi channel is supported by Multi-SSI, or TDM-SSI.
27 Multi-SSI : 6ch case, you can use stereo x 3 SSI
28 TDM-SSI : 6ch case, you can use TDM
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| /kernel/linux/linux-6.6/Documentation/devicetree/bindings/sound/ |
| D | renesas,idt821034.yaml | 1 # 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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| D | amlogic,axg-sound-card.yaml | 1 # 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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| D | infineon,peb2466.yaml | 1 # 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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| D | qcom,q6dsp-lpass-ports.yaml | 1 # 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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| D | st,sti-asoc-card.txt | 3 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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| D | simple-card.yaml | 1 # 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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| /kernel/linux/linux-6.6/arch/arm64/boot/dts/amlogic/ |
| D | meson-g12b-bananapi-cm4-cm4io.dts | 1 // 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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| /kernel/linux/linux-6.6/sound/soc/qcom/qdsp6/ |
| D | q6afe-dai.c | 1 // SPDX-License-Identifier: GPL-2.0
2 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
14 #include "q6dsp-lpass-ports.h"
15 #include "q6dsp-common.h"
38 struct snd_soc_dai *dai) in q6slim_hw_params() argument
41 struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); in q6slim_hw_params()
42 struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim; in q6slim_hw_params()
44 slim->sample_rate = params_rate(params); in q6slim_hw_params()
49 slim->bit_width = 16; in q6slim_hw_params()
52 slim->bit_width = 24; in q6slim_hw_params()
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| /kernel/linux/linux-6.6/sound/soc/ |
| D | soc-utils.c | 1 // 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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| D | soc-dai.c | 1 // 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(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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| /kernel/linux/linux-5.10/sound/soc/meson/ |
| D | axg-tdm-interface.c | 1 // SPDX-License-Identifier: (GPL-2.0 OR MIT)
11 #include <sound/soc-dai.h>
13 #include "axg-tdm.h"
35 int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, in axg_tdm_set_tdm_slots() argument
39 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); in axg_tdm_set_tdm_slots()
41 dai->playback_dma_data; in axg_tdm_set_tdm_slots()
43 dai->capture_dma_data; in axg_tdm_set_tdm_slots()
50 /* We should at least have a slot for a valid interface */ in axg_tdm_set_tdm_slots()
52 dev_err(dai->dev, "interface has no slot\n"); in axg_tdm_set_tdm_slots()
53 return -EINVAL; in axg_tdm_set_tdm_slots()
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| /kernel/linux/linux-6.6/sound/soc/meson/ |
| D | axg-tdm-interface.c | 1 // SPDX-License-Identifier: (GPL-2.0 OR MIT)
11 #include <sound/soc-dai.h>
13 #include "axg-tdm.h"
35 int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, in axg_tdm_set_tdm_slots() argument
39 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); in axg_tdm_set_tdm_slots()
40 struct axg_tdm_stream *tx = snd_soc_dai_dma_data_get_playback(dai); in axg_tdm_set_tdm_slots()
41 struct axg_tdm_stream *rx = snd_soc_dai_dma_data_get_capture(dai); in axg_tdm_set_tdm_slots()
48 /* We should at least have a slot for a valid interface */ in axg_tdm_set_tdm_slots()
50 dev_err(dai->dev, "interface has no slot\n"); in axg_tdm_set_tdm_slots()
51 return -EINVAL; in axg_tdm_set_tdm_slots()
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| /kernel/linux/linux-5.10/sound/soc/qcom/qdsp6/ |
| D | q6afe-dai.c | 1 // SPDX-License-Identifier: GPL-2.0
2 // Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
18 .stream_name = pre" TDM"#num" Playback", \
39 .stream_name = pre" TDM"#num" Capture", \
119 struct snd_soc_dai *dai) in q6slim_hw_params() argument
122 struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev); in q6slim_hw_params()
123 struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim; in q6slim_hw_params()
125 slim->sample_rate = params_rate(params); in q6slim_hw_params()
130 slim->bit_width = 16; in q6slim_hw_params()
133 slim->bit_width = 24; in q6slim_hw_params()
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| /kernel/linux/linux-6.6/arch/arm/boot/dts/st/ |
| D | stm32mp15xx-dhcom-pdk2.dtsi | 1 // 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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| /kernel/linux/linux-5.10/arch/arm/boot/dts/ |
| D | stm32mp15xx-dhcom-pdk2.dtsi | 1 // 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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| /kernel/linux/linux-6.6/sound/soc/fsl/ |
| D | imx-card.c | 1 // SPDX-License-Identifier: GPL-2.0+
2 // Copyright 2017-2021 NXP
15 #include <sound/soc-dapm.h>
46 * Mapping TDM mode and frame width
55 * struct imx_card_plat_data - specific info for codecs
58 * @tdm_fs_mul: ratio of mclk/fs for tdm mode
60 * @support_tdm_rates: supported sample rate for tdm mode
62 * @support_tdm_channels: supported channels for tdm mode
89 * struct dai_link_data - specific info for dai link
91 * @slots: slot number
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| /kernel/linux/linux-5.10/sound/soc/codecs/ |
| D | tas5720.c | 1 // 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
23 #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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| /kernel/linux/linux-6.6/sound/soc/codecs/ |
| D | tas5720.c | 1 // 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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| /kernel/linux/linux-5.10/sound/soc/sh/ |
| D | ssi.c | 1 // 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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