dai\-sample\-rate (full) in projects: linux

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Searched +full:dai +full:- +full:sample +full:- +full:rate (Results 1 – 25 of 138) sorted by relevance

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/linux/Documentation/sound/soc/
H A 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 …]
/linux/Documentation/devicetree/bindings/sound/
H A 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> 15 convert-channels: 16 description: Number of audio channels used by DAI 21 convert-sample-format: 22 description: Audio sample format used by DAI [all …]`
H A D	nvidia,tegra186-asrc.yaml	1 # 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> [all …]
H A D	option,gtm601.yaml	`1 # 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': [all …]`
H A D	davinci-mcbsp.yaml	`1 # SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) 3 --- 4 $id: http://devicetree.org/schemas/sound/davinci-mcbsp.yaml# 5 $schema: http://devicetree.org/meta-schemas/core.yaml# 10 - Bastien Curutchet <bastien.curutchet@bootlin.com> 13 - $ref: dai-common.yaml# 18 - ti,da850-mcbsp 23 - description: CFG registers 24 - description: data registers 26 reg-names: [all …]`
/linux/sound/soc/tegra/
H A D	tegra_audio_graph_card.c	1 // SPDX-License-Identifier: GPL-2.0-only 2 // SPDX-FileCopyrightText: Copyright (c) 2020-2025 NVIDIA CORPORATION. All rights reserved. 4 // tegra_audio_graph_card.c - Audio Graph based Tegra Machine Driver 12 #include <sound/soc-dai.h> 21 * Sample rates multiple of 8000 Hz and below are supported: 27 * Sample rates multiple of 11025 Hz and below are supported: 47 static bool need_clk_update(struct snd_soc_dai *dai) in need_clk_update() argument 49 if (snd_soc_dai_is_dummy(dai) \|\| in need_clk_update() 50 !dai->driver->ops \|\| in need_clk_update() 51 !dai->driver->name) in need_clk_update() [all …]
/linux/sound/soc/codecs/
H A 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() 68 if (wm8524->sysclk) in wm8524_startup() 69 snd_pcm_hw_constraint_list(substream->runtime, 0, in wm8524_startup() 71 &wm8524->rate_constraint); in wm8524_startup() 73 gpiod_set_value_cansleep(wm8524->mute, 1); in wm8524_startup() 79 struct snd_soc_dai *dai) in wm8524_shutdown() argument [all …]
H A 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 …]
H A D	si476x.c	1 // 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() [all …]
H A D	cs48l32.c	1 // SPDX-License-Identifier: GPL-2.0-only 5 // Copyright (C) 2016-2018, 2020, 2022, 2025 Cirrus Logic, Inc. and 8 #include <dt-bindings/sound/cs48l32.h> 32 #include <sound/soc-component.h> 33 #include <sound/soc-dai.h> 34 #include <sound/soc-dapm.h> 39 static const char * const cs48l32_core_supplies[] = { "vdd-a", "vdd-io" }; 193 static const DECLARE_TLV_DB_SCALE(cs48l32_eq_tlv, -1200, 100, 0); 194 static const DECLARE_TLV_DB_SCALE(cs48l32_digital_tlv, -6400, 50, 0); 195 static const DECLARE_TLV_DB_SCALE(cs48l32_noise_tlv, -10800, 600, 0); [all …]
/linux/sound/soc/fsl/
H A D	fsl_spdif.c	1 // SPDX-License-Identifier: GPL-2.0 3 // Freescale S/PDIF ALSA SoC Digital Audio Interface (DAI) driver 25 #include "imx-pcm.h" 45 #define RX_SAMPLE_RATE_KCONTROL "RX Sample Rate" 52 * so the driver shouldn't set root clock rate 97 * struct fsl_spdif_priv - Freescale SPDIF private data 100 * @cpu_dai_drv: cpu dai driver 102 * @rxrate_kcontrol: kcontrol for RX Sample Rate 114 * @sysclk: system clock for rx clock rate measurement 120 * @pll8k_clk: PLL clock for the rate of multiply of 8kHz [all …]
H A D	fsl_asrc.c	1 // 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() [all …]
H A D	fsl-asoc-card.c	`1 // SPDX-License-Identifier: GPL-2.0 23 #include "imx-audmux.h" 33 #define DRIVER_NAME "fsl-asoc-card" 40 /* Default DAI format without Master and Slave flag / 44 struct codec_priv - CODEC private data 46 * @mclk_freq: Clock rate of MCLK 47 * @free_freq: Clock rate of MCLK for hw_free() 62 * struct cpu_priv - CPU private data 66 * @sysclk_ratio: SYSCLK ratio on sample rate 82 * struct fsl_asoc_card_priv - Freescale Generic ASOC card private data [all …]`
/linux/sound/soc/meson/
H A 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 …]
H A D	axg-spdifin.c	1 // SPDX-License-Identifier: (GPL-2.0 OR MIT) 12 #include <sound/soc-dai.h> 61 * It would have been nice to check the actual rate against the sample rate 78 unsigned int stat, mode, rate = 0; in axg_spdifin_get_rate() local 80 regmap_read(priv->map, SPDIFIN_STAT0, &stat); in axg_spdifin_get_rate() 90 rate = priv->conf->mode_rates[mode]; in axg_spdifin_get_rate() 92 return rate; in axg_spdifin_get_rate() 96 struct snd_soc_dai dai) in axg_spdifin_prepare() argument 98 struct axg_spdifin priv = snd_soc_dai_get_drvdata(dai); in axg_spdifin_prepare() 101 regmap_update_bits(priv->map, SPDIFIN_CTRL0, in axg_spdifin_prepare() [all …]
/linux/sound/soc/sti/
H A D	uniperif_player.c	1 // 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() [all …]
/linux/sound/soc/mediatek/mt8186/
H A D	mt8186-dai-hw-gain.c	1 // 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() 59 regmap_update_bits(afe->regmap, gain_cur, AFE_GAIN1_CUR_MASK_SFT, 0); in mtk_hw_gain_event() 62 regmap_update_bits(afe->regmap, gain_con1, GAIN1_TARGET_MASK_SFT, 0); in mtk_hw_gain_event() [all …]
/linux/sound/soc/sunxi/
H A D	sun4i-i2s.c	1 // SPDX-License-Identifier: GPL-2.0-or-later 7 * Maxime Ripard <maxime.ripard@free-electrons.com> 22 #include <sound/soc-dai.h> 85 #define SUN4I_I2S_CHAN_SEL(num_chan) (((num_chan) - 1) << 0) 88 #define SUN4I_I2S_TX_CHAN_MAP(chan, sample) ((sample) << (chan << 2)) argument 93 /* Defines required for sun8i-h3 support */ 106 #define SUN8I_I2S_FMT0_LRCK_PERIOD(period) ((period - 1) << 8) 119 #define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) ((chan - 1) << 4) 121 #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(chan) (chan - 1) 128 #define SUN8I_I2S_TX_CHAN_EN(num_chan) (((1 << num_chan) - 1) << 4) [all …]
/linux/sound/soc/mediatek/mt6797/
H A D	mt6797-dai-adda.c	`1 // 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" 13 #include "../common/mtk-dai-adda-common.h" 15 /* dai component / 55 struct snd_soc_component cmpnt = snd_soc_dapm_to_component(w->dapm); in mtk_adda_ul_event() 58 dev_dbg(afe->dev, "%s(), name %s, event 0x%x\n", in mtk_adda_ul_event() 59 __func__, w->name, event); in mtk_adda_ul_event() [all …]`
/linux/sound/soc/pxa/
H A D	pxa-ssp.c	1 // SPDX-License-Identifier: GPL-2.0-or-later 3 * pxa-ssp.c -- ALSA Soc Audio Layer 10 * o Test network mode for > 16bit sample size 30 #include <sound/pxa2xx-lib.h> 33 #include "pxa-ssp.h" 55 dev_dbg(ssp->dev, "SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x\n", in dump_registers() 59 dev_dbg(ssp->dev, "SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x\n", in dump_registers() 67 dma->addr_width = width4 ? DMA_SLAVE_BUSWIDTH_4_BYTES : in pxa_ssp_set_dma_params() 69 dma->maxburst = 16; in pxa_ssp_set_dma_params() 70 dma->addr = ssp->phys_base + SSDR; in pxa_ssp_set_dma_params() [all …]
/linux/sound/soc/mediatek/mt8183/
H A D	mt8183-dai-adda.c	1 // SPDX-License-Identifier: GPL-2.0 3 // MediaTek ALSA SoC Audio DAI ADDA Control 10 #include "mt8183-afe-common.h" 11 #include "mt8183-interconnection.h" 12 #include "mt8183-reg.h" 13 #include "../common/mtk-dai-adda-common.h" 22 /* dai component / 62 struct snd_soc_component cmpnt = snd_soc_dapm_to_component(w->dapm); in mtk_adda_ul_event() 64 struct mt8183_afe_private *afe_priv = afe->platform_priv; in mtk_adda_ul_event() 66 dev_dbg(afe->dev, "%s(), name %s, event 0x%x\n", in mtk_adda_ul_event() [all …]
/linux/sound/soc/atmel/
H A D	mchp-i2s-mcc.c	1 // 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) ---- 127 /* x sample transmitted when underrun / 128 #define MCHP_I2SMCC_MRA_TXSAME_ZERO (0 << 11) / Zero sample / 129 #define MCHP_I2SMCC_MRA_TXSAME_PREVIOUS (1 << 11) / Previous sample / 135 / Number of TDM Channels - 1 */ 138 ((((ch) - 1) << 13) & MCHP_I2SMCC_MRA_NBCHAN_MASK) [all …]
H A D	mchp-spdifrx.c	1 // SPDX-License-Identifier: GPL-2.0 21 * ---- S/PDIF Receiver Controller Register map ---- 45 * ---- Control Register (Write-only) ---- 50 * ---- Mode Register (Read/Write) ---- 60 (0 << 1) /* Load sample regardless of validity bit value / 62 (1 << 1) / Load sample only if validity bit is 0 / 74 / Sample Data Width / 77 (((6 - (width) / 4) << 4) & SPDIFRX_MR_DATAWIDTH_MASK) 95 ---- Interrupt Enable/Disable/Mask/Status Register (Write/Read-only) ---- 113 * ---- Receiver Status Register (Read/Write) ---- [all …]
/linux/sound/soc/
H A 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_ argument 14 _soc_dai_ret(const struct snd_soc_dai * dai,const char * func,int ret) _soc_dai_ret() argument 25 soc_dai_mark_push(dai,substream,tgt) global() argument 26 soc_dai_mark_pop(dai,tgt) global() argument 27 soc_dai_mark_match(dai,substream,tgt) global() argument 38 snd_soc_dai_set_sysclk(struct snd_soc_dai * dai,int clk_id,unsigned int freq,int dir) snd_soc_dai_set_sysclk() argument 64 snd_soc_dai_set_clkdiv(struct snd_soc_dai * dai,int div_id,int div) snd_soc_dai_set_clkdiv() argument 87 snd_soc_dai_set_pll(struct snd_soc_dai * dai,int pll_id,int source,unsigned int freq_in,unsigned int freq_out) snd_soc_dai_set_pll() argument 111 snd_soc_dai_set_bclk_ratio(struct snd_soc_dai * dai,unsigned int ratio) snd_soc_dai_set_bclk_ratio() argument 125 struct snd_soc_dai dai; snd_soc_dai_get_fmt_max_priority() local 155 snd_soc_dai_get_fmt(const struct snd_soc_dai dai,int priority) snd_soc_dai_get_fmt() argument 193 snd_soc_dai_set_fmt(struct snd_soc_dai * dai,unsigned int fmt) snd_soc_dai_set_fmt() argument 251 snd_soc_dai_set_tdm_slot(struct snd_soc_dai * dai,unsigned int tx_mask,unsigned int rx_mask,int slots,int slot_width) snd_soc_dai_set_tdm_slot() argument 296 snd_soc_dai_set_channel_map(struct snd_soc_dai * dai,unsigned int tx_num,const unsigned int * tx_slot,unsigned int rx_num,const unsigned int * rx_slot) snd_soc_dai_set_channel_map() argument 320 snd_soc_dai_get_channel_map(const struct snd_soc_dai * dai,unsigned int * tx_num,unsigned int * tx_slot,unsigned int * rx_num,unsigned int * rx_slot) snd_soc_dai_get_channel_map() argument 341 snd_soc_dai_set_tristate(struct snd_soc_dai * dai,int tristate) snd_soc_dai_set_tristate() argument 353 snd_soc_dai_prepare(struct snd_soc_dai * dai,struct snd_pcm_substream * substream) snd_soc_dai_prepare() argument 369 snd_soc_dai_mute_is_ctrled_at_trigger(struct snd_soc_dai * dai) snd_soc_dai_mute_is_ctrled_at_trigger() argument 385 snd_soc_dai_digital_mute(struct snd_soc_dai * dai,int mute,int direction) snd_soc_dai_digital_mute() argument 404 snd_soc_dai_hw_params(struct snd_soc_dai * dai,struct snd_pcm_substream * substream,struct snd_pcm_hw_params * params) snd_soc_dai_hw_params() argument 421 snd_soc_dai_hw_free(struct snd_soc_dai * dai,struct snd_pcm_substream * substream,int rollback) snd_soc_dai_hw_free() argument 436 snd_soc_dai_startup(struct snd_soc_dai * dai,struct snd_pcm_substream * substream) snd_soc_dai_startup() argument 455 snd_soc_dai_shutdown(struct snd_soc_dai * dai,struct snd_pcm_substream * substream,int rollback) snd_soc_dai_shutdown() argument 473 snd_soc_dai_compress_new(struct snd_soc_dai * dai,struct snd_soc_pcm_runtime * rtd) snd_soc_dai_compress_new() argument 488 snd_soc_dai_stream_valid(const struct snd_soc_dai * dai,int dir) snd_soc_dai_stream_valid() argument 496 snd_soc_dai_action(struct snd_soc_dai * dai,int stream,int action) snd_soc_dai_action() argument 507 snd_soc_dai_active(const struct snd_soc_dai * dai) snd_soc_dai_active() argument 521 struct snd_soc_dai dai; snd_soc_pcm_dai_probe() local 547 struct snd_soc_dai dai; snd_soc_pcm_dai_remove() local 572 struct snd_soc_dai dai; snd_soc_pcm_dai_new() local 590 struct snd_soc_dai dai; snd_soc_pcm_dai_prepare() local 602 soc_dai_trigger(struct snd_soc_dai * dai,struct snd_pcm_substream * substream,int cmd) soc_dai_trigger() argument 621 struct snd_soc_dai dai; snd_soc_pcm_dai_trigger() local 664 struct snd_soc_dai dai; snd_soc_pcm_dai_delay() local 687 snd_soc_dai_compr_startup(struct snd_soc_dai * dai,struct snd_compr_stream * cstream) snd_soc_dai_compr_startup() argument 704 snd_soc_dai_compr_shutdown(struct snd_soc_dai * dai,struct snd_compr_stream * cstream,int rollback) snd_soc_dai_compr_shutdown() argument 720 snd_soc_dai_compr_trigger(struct snd_soc_dai * dai,struct snd_compr_stream * cstream,int cmd) snd_soc_dai_compr_trigger() argument 733 snd_soc_dai_compr_set_params(struct snd_soc_dai * dai,struct snd_compr_stream * cstream,struct snd_compr_params * params) snd_soc_dai_compr_set_params() argument 747 snd_soc_dai_compr_get_params(struct snd_soc_dai * dai,struct snd_compr_stream * cstream,struct snd_codec * params) snd_soc_dai_compr_get_params() argument 761 snd_soc_dai_compr_ack(struct snd_soc_dai * dai,struct snd_compr_stream * cstream,size_t bytes) snd_soc_dai_compr_ack() argument 775 snd_soc_dai_compr_pointer(struct snd_soc_dai * dai,struct snd_compr_stream * cstream,struct snd_compr_tstamp * tstamp) snd_soc_dai_compr_pointer() argument 789 snd_soc_dai_compr_set_metadata(struct snd_soc_dai * dai,struct snd_compr_stream * cstream,struct snd_compr_metadata * metadata) snd_soc_dai_compr_set_metadata() argument 803 snd_soc_dai_compr_get_metadata(struct snd_soc_dai * dai,struct snd_compr_stream * cstream,struct snd_compr_metadata * metadata) snd_soc_dai_compr_get_metadata() argument [all...]
/linux/sound/soc/mxs/
H A D	mxs-saif.c	1 // SPDX-License-Identifier: GPL-2.0-or-later 11 #include <linux/dma-mapping.h> 13 #include <linux/clk-provider.h> 22 #include "mxs-saif.h" 32 * For MXS, two SAIF modules are instantiated on-chip. 34 * mode simultaneously if they are connected to different off-chip codecs. 37 * This also means that both SAIFs must operate at the same sample rate. 42 * and operating its master to generate the proper clock rate for it. 43 * The master id is provided in mach-specific layer according to different 54 saif->mclk = freq; in mxs_saif_set_dai_sysclk() [all …]

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