| /kernel/linux/linux-6.6/Documentation/devicetree/bindings/sound/ |
| D | audio-graph-port.yaml | 1 # 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: [all …]
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| D | dai-params.yaml | 1 # 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 [all …]
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| D | audio-graph.yaml | 1 # 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 [all …]
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| D | qcom,lpass-tx-macro.yaml | 1 # 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 [all …]
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| D | qcom,lpass-va-macro.yaml | 1 # 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 [all …]
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| D | qcom,lpass-wsa-macro.yaml | 1 # 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 [all …]
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| /kernel/linux/linux-6.6/Documentation/sound/soc/ |
| D | dai.rst | 2 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 [all …]
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| D | clocking.rst | 10 ------------ 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 [all …]
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| /kernel/linux/linux-5.10/Documentation/sound/soc/ |
| D | dai.rst | 2 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 [all …]
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| D | clocking.rst | 10 ------------ 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 [all …]
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| /kernel/linux/linux-6.6/sound/soc/tegra/ |
| D | tegra_audio_graph_card.c | 1 // 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() [all …]
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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" 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() [all …]
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| D | axg-pdm.c | 1 // 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() [all …]
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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() 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() 56 iface->slots = slots; in axg_tdm_set_tdm_slots() [all …]
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| D | axg-pdm.c | 1 // 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() [all …]
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| /kernel/linux/linux-6.6/sound/soc/codecs/ |
| D | wm8524.c | 1 // 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() [all …]
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| D | cs4270.c | 6 * 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 [all …]
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| D | uda1334.c | 1 // 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() [all …]
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| /kernel/linux/linux-5.10/sound/soc/codecs/ |
| D | wm8524.c | 1 // 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() [all …]
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| D | cs4270.c | 6 * 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 36 * The codec isn't really big-endian or little-endian, since the I2S 60 #define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) 110 /* Power-on default values for the registers [all …]
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| D | uda1334.c | 1 // 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() [all …]
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| /kernel/linux/linux-5.10/sound/soc/ti/ |
| D | davinci-i2s.c | 1 // 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, [all …]
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| /kernel/linux/linux-5.10/sound/soc/sh/ |
| D | fsi.c | 1 // 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 ]| ... | [all …]
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| /kernel/linux/linux-6.6/sound/soc/sh/ |
| D | fsi.c | 1 // 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 ]| ... | [all …]
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| /kernel/linux/linux-6.6/sound/soc/ti/ |
| D | davinci-i2s.c | 1 // 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, [all …]
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