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/openbmc/linux/Documentation/devicetree/bindings/sound/
H A Daudio-graph-port.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
4 $id: http://devicetree.org/schemas/sound/audio-graph-port.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
15 port-base:
16 $ref: /schemas/graph.yaml#/$defs/port-base
18 convert-rate:
19 $ref: /schemas/sound/dai-params.yaml#/$defs/dai-sample-rate
20 convert-channels:
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H A Ddai-params.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
4 $id: http://devicetree.org/schemas/sound/dai-params.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
7 title: Digital Audio Interface (DAI) Stream Parameters
10 - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
16 dai-channels:
17 description: Number of audio channels used by DAI
22 dai-sample-format:
23 description: Audio sample format used by DAI
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H A Daudio-graph.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
4 $id: http://devicetree.org/schemas/sound/audio-graph.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
14 $ref: /schemas/types.yaml#/definitions/phandle-array
25 $ref: /schemas/types.yaml#/definitions/non-unique-string-array
32 $ref: /schemas/types.yaml#/definitions/non-unique-string-array
33 convert-rate:
34 $ref: /schemas/sound/dai-params.yaml#/$defs/dai-sample-rate
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H A Dqcom,lpass-tx-macro.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
4 $id: http://devicetree.org/schemas/sound/qcom,lpass-tx-macro.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
15 - qcom,sc7280-lpass-tx-macro
16 - qcom,sm8250-lpass-tx-macro
17 - qcom,sm8450-lpass-tx-macro
18 - qcom,sm8550-lpass-tx-macro
19 - qcom,sc8280xp-lpass-tx-macro
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H A Dqcom,lpass-va-macro.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
4 $id: http://devicetree.org/schemas/sound/qcom,lpass-va-macro.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
15 - qcom,sc7280-lpass-va-macro
16 - qcom,sm8250-lpass-va-macro
17 - qcom,sm8450-lpass-va-macro
18 - qcom,sm8550-lpass-va-macro
19 - qcom,sc8280xp-lpass-va-macro
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H A Dqcom,lpass-wsa-macro.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
4 $id: http://devicetree.org/schemas/sound/qcom,lpass-wsa-macro.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
15 - qcom,sc7280-lpass-wsa-macro
16 - qcom,sm8250-lpass-wsa-macro
17 - qcom,sm8450-lpass-wsa-macro
18 - qcom,sm8550-lpass-wsa-macro
19 - qcom,sc8280xp-lpass-wsa-macro
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H A Doption,gtm601.yaml1 # SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause
3 ---
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 - kernel@puri.sm
13 This device has no configuration interface. The sample rate and channels are
19 - description: Broadmobi BM818 (48Khz stereo)
21 - const: broadmobi,bm818
22 - const: option,gtm601
23 - description: GTM601 (8kHz mono)
26 '#sound-dai-cells':
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H A Dnvidia,tegra186-asrc.yaml1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
3 ---
4 $id: http://devicetree.org/schemas/sound/nvidia,tegra186-asrc.yaml#
5 $schema: http://devicetree.org/meta-schemas/core.yaml#
10 Asynchronous Sample Rate Converter (ASRC) converts the sampling frequency
12 wide range of sample rate ratios (freq_in/freq_out) from 1:24 to 24:1.
16 It supports sample rate conversions in the range of 8 to 192 kHz and
21 - Jon Hunter <jonathanh@nvidia.com>
22 - Mohan Kumar <mkumard@nvidia.com>
23 - Sameer Pujar <spujar@nvidia.com>
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/openbmc/linux/Documentation/sound/soc/
H A Ddai.rst2 ASoC Digital Audio Interface (DAI)
5 ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
13 now also popular in many portable devices. This DAI has a RESET line and time
26 I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and
30 usually varies depending on the sample rate and the master system clock
31 (SYSCLK). LRCLK is the same as the sample rate. A few devices support separate
33 different sample rates.
35 I2S has several different operating modes:-
45 MSB is transmitted sample size BCLKs before LRC transition.
53 receive the audio data. Bit clock usually varies depending on sample rate
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H A Dclocking.rst10 ------------
15 audio playback and capture sample rates.
22 DAI Clocks
23 ----------
28 The DAI also has a frame clock to signal the start of each audio frame. This
30 runs at exactly the sample rate (LRC = Rate).
32 Bit Clock can be generated as follows:-
34 - BCLK = MCLK / x, or
35 - BCLK = LRC * x, or
36 - BCLK = LRC * Channels * Word Size
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/openbmc/linux/sound/soc/tegra/
H A Dtegra_audio_graph_card.c1 // SPDX-License-Identifier: GPL-2.0-only
3 // tegra_audio_graph_card.c - Audio Graph based Tegra Machine Driver
5 // Copyright (c) 2020-2021 NVIDIA CORPORATION. All rights reserved.
13 #include <sound/soc-dai.h>
22 * Sample rates multiple of 8000 Hz and below are supported:
28 * Sample rates multiple of 11025 Hz and below are supported:
48 static bool need_clk_update(struct snd_soc_dai *dai) in need_clk_update() argument
50 if (snd_soc_dai_is_dummy(dai) || in need_clk_update()
51 !dai->driver->ops || in need_clk_update()
52 !dai->driver->name) in need_clk_update()
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/openbmc/linux/sound/soc/meson/
H A 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()
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()
57 iface->slots = slots; in axg_tdm_set_tdm_slots()
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H A Daxg-pdm.c1 // SPDX-License-Identifier: (GPL-2.0 OR MIT)
12 #include <sound/soc-dai.h>
53 #define PDM_CHAN_CTRL_POINTER_MAX ((1 << PDM_CHAN_CTRL_POINTER_WIDTH) - 1)
126 struct snd_soc_dai *dai) in axg_pdm_trigger() argument
128 struct axg_pdm *priv = snd_soc_dai_get_drvdata(dai); in axg_pdm_trigger()
134 axg_pdm_enable(priv->map); in axg_pdm_trigger()
140 axg_pdm_disable(priv->map); in axg_pdm_trigger()
144 return -EINVAL; in axg_pdm_trigger()
150 const struct axg_pdm_filters *filters = priv->cfg->filters; in axg_pdm_get_os()
151 unsigned int os = filters->hcic.ds; in axg_pdm_get_os()
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/openbmc/linux/sound/soc/codecs/
H A Dwm8524.c1 // SPDX-License-Identifier: GPL-2.0-only
3 * wm8524.c -- WM8524 ALSA SoC Audio driver
60 struct snd_soc_dai *dai) in wm8524_startup() argument
62 struct snd_soc_component *component = dai->component; in wm8524_startup()
65 /* The set of sample rates that can be supported depends on the in wm8524_startup()
66 * MCLK supplied to the CODEC - enforce this. in wm8524_startup()
68 if (!wm8524->sysclk) { in wm8524_startup()
69 dev_err(component->dev, in wm8524_startup()
71 return -EINVAL; in wm8524_startup()
74 snd_pcm_hw_constraint_list(substream->runtime, 0, in wm8524_startup()
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H A Dcs4270.c6 * Copyright 2007-2009 Freescale Semiconductor, Inc. This file is licensed
15 * - Software mode is supported. Stand-alone mode is not supported.
16 * - Only I2C is supported, not SPI
17 * - Support for master and slave mode
18 * - The machine driver's 'startup' function must call
20 * - Only I2S and left-justified modes are supported
21 * - Power management is supported
51 #define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1)
101 /* Power-on default values for the registers
103 * This array contains the power-on default values of the registers, with the
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H A Duda1334.c1 // SPDX-License-Identifier: GPL-2.0-only
3 // uda1334.c -- UDA1334 ALSA SoC Audio driver
47 int deemph = ucontrol->value.integer.value[0]; in uda1334_put_deemph()
50 return -EINVAL; in uda1334_put_deemph()
52 gpiod_set_value_cansleep(uda1334->deemph, deemph); in uda1334_put_deemph()
64 ret = gpiod_get_value_cansleep(uda1334->deemph); in uda1334_get_deemph()
66 return -EINVAL; in uda1334_get_deemph()
68 ucontrol->value.integer.value[0] = ret; in uda1334_get_deemph()
91 struct snd_soc_dai *dai) in uda1334_startup() argument
93 struct snd_soc_component *component = dai->component; in uda1334_startup()
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H A Dsi476x.c1 // SPDX-License-Identifier: GPL-2.0-only
3 * sound/soc/codecs/si476x.c -- Codec driver for SI476X chips
21 #include <linux/mfd/si476x-core.h>
68 struct si476x_core *core = i2c_mfd_cell_to_core(codec_dai->dev); in si476x_codec_set_dai_fmt()
73 return -EINVAL; in si476x_codec_set_dai_fmt()
92 return -EINVAL; in si476x_codec_set_dai_fmt()
105 return -EINVAL; in si476x_codec_set_dai_fmt()
125 return -EINVAL; in si476x_codec_set_dai_fmt()
129 return -EINVAL; in si476x_codec_set_dai_fmt()
134 err = snd_soc_component_update_bits(codec_dai->component, SI476X_DIGITAL_IO_OUTPUT_FORMAT, in si476x_codec_set_dai_fmt()
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H A Dcs4271.c1 // SPDX-License-Identifier: GPL-2.0-or-later
9 * The data format accepted is I2S or left-justified.
132 * Default CS4271 power-up configuration
133 * Array contains non-existing in hw register at address 0
161 /* Current sample rate for de-emphasis control */
162 int rate; member
177 SND_SOC_DAPM_OUTPUT("AOUTA-"),
179 SND_SOC_DAPM_OUTPUT("AOUTB-"),
187 { "AOUTA-", NULL, "Playback" },
189 { "AOUTB-", NULL, "Playback" },
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/openbmc/linux/sound/soc/fsl/
H A Dfsl_spdif.c1 // SPDX-License-Identifier: GPL-2.0
3 // Freescale S/PDIF ALSA SoC Digital Audio Interface (DAI) driver
27 #include "imx-pcm.h"
47 #define RX_SAMPLE_RATE_KCONTROL "RX Sample Rate"
54 * so the driver shouldn't set root clock rate
99 * struct fsl_spdif_priv - Freescale SPDIF private data
102 * @cpu_dai_drv: cpu dai driver
104 * @rxrate_kcontrol: kcontrol for RX Sample Rate
116 * @sysclk: system clock for rx clock rate measurement
122 * @pll8k_clk: PLL clock for the rate of multiply of 8kHz
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H A Dfsl_asrc.c1 // SPDX-License-Identifier: GPL-2.0
3 // Freescale ASRC ALSA SoC Digital Audio Interface (DAI) driver
11 #include <linux/dma-mapping.h>
14 #include <linux/dma/imx-dma.h>
26 dev_err(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
29 dev_dbg(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
32 dev_warn(&asrc->pdev->dev, "Pair %c: " fmt, 'A' + index, ##__VA_ARGS__)
127 static bool fsl_asrc_divider_avail(int clk_rate, int rate, int *div) in fsl_asrc_divider_avail() argument
135 if (clk_rate == 0 || rate == 0) in fsl_asrc_divider_avail()
139 rem = do_div(n, rate); in fsl_asrc_divider_avail()
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/openbmc/linux/sound/soc/sh/
H A Dfsi.c1 // SPDX-License-Identifier: GPL-2.0
3 // Fifo-attached Serial Interface (FSI) support for SH7724
12 #include <linux/dma-mapping.h>
137 * A : sample widtht 16bit setting
138 * B : sample widtht 24bit setting
160 * period/frame/sample image
166 * |<-------------------- period--------------------->|
169 * ||<----- frame ----->|<------ frame ----->| ... |
170 * |+--------------------+--------------------+- ... |
171 * ||[ sample ][ sample ]|[ sample ][ sample ]| ... |
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/openbmc/linux/sound/soc/ti/
H A Ddavinci-i2s.c1 // SPDX-License-Identifier: GPL-2.0-only
9 * based on davinci-mcasp.c DT support
31 #include "edma-pcm.h"
32 #include "davinci-i2s.h"
34 #define DRV_NAME "davinci-i2s"
39 * - This driver supports the "Audio Serial Port" (ASP),
42 * - But it labels it a "Multi-channel Buffered Serial Port"
44 * backward-compatible, possibly explaining that confusion.
46 * - OMAP chips have a controller called McBSP, which is
49 * - Newer DaVinci chips have a controller called McASP,
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/openbmc/linux/sound/soc/sti/
H A Duniperif_player.c1 // SPDX-License-Identifier: GPL-2.0-only
17 * Some hardware-related definitions
27 #define UNIPERIF_PLAYER_CLK_ADJ_MIN -999999
33 * integrate DAI_CPU capability in term of rate and supported channels
68 spin_lock(&player->irq_lock); in uni_player_irq_handler()
69 if (!player->substream) in uni_player_irq_handler()
72 snd_pcm_stream_lock(player->substream); in uni_player_irq_handler()
73 if (player->state == UNIPERIF_STATE_STOPPED) in uni_player_irq_handler()
82 dev_err(player->dev, "FIFO underflow error detected\n"); in uni_player_irq_handler()
85 if (player->underflow_enabled) { in uni_player_irq_handler()
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/openbmc/linux/sound/soc/mediatek/mt8186/
H A Dmt8186-dai-hw-gain.c1 // SPDX-License-Identifier: GPL-2.0
3 // MediaTek ALSA SoC Audio DAI HW Gain Control
9 #include "mt8186-afe-common.h"
10 #include "mt8186-interconnection.h"
15 /* dai component */
40 struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); in mtk_hw_gain_event()
45 dev_dbg(cmpnt->dev, "%s(), name %s, event 0x%x\n", in mtk_hw_gain_event()
46 __func__, w->name, event); in mtk_hw_gain_event()
50 if (strcmp(w->name, HW_GAIN_1_EN_W_NAME) == 0) { in mtk_hw_gain_event()
59 regmap_update_bits(afe->regmap, gain_cur, AFE_GAIN1_CUR_MASK_SFT, 0); in mtk_hw_gain_event()
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/openbmc/linux/sound/soc/mediatek/mt6797/
H A Dmt6797-dai-adda.c1 // SPDX-License-Identifier: GPL-2.0
3 // MediaTek ALSA SoC Audio DAI ADDA Control
10 #include "mt6797-afe-common.h"
11 #include "mt6797-interconnection.h"
12 #include "mt6797-reg.h"
39 unsigned int rate) in adda_dl_rate_transform() argument
41 switch (rate) { in adda_dl_rate_transform()
65 dev_warn(afe->dev, "%s(), rate %d invalid, use 48kHz!!!\n", in adda_dl_rate_transform()
66 __func__, rate); in adda_dl_rate_transform()
72 unsigned int rate) in adda_ul_rate_transform() argument
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