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

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Searched +full:dai +full:- +full:sample +full:- +full:format (Results 1 – 25 of 104) sorted by relevance

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/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	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: 17 - $ref: /schemas/graph.yaml#/$defs/port-base 18 - $ref: /schemas/sound/dai-params.yaml# 20 mclk-fs: 21 $ref: simple-card.yaml#/definitions/mclk-fs [all …]`
/linux/sound/soc/fsl/
H A D	mpc5200_psc_i2s.c	1 // SPDX-License-Identifier: GPL-2.0-only 4 // ALSA SoC Digital Audio Interface (DAI) driver 21 * PSC_I2S_RATES: sample rates supported by the I2S 24 * which means the codec determines the sample rate. Therefore, we tell 38 struct snd_soc_dai *dai) in psc_i2s_hw_params() argument 44 dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" in psc_i2s_hw_params() 64 dev_dbg(psc_dma->dev, "invalid format\n"); in psc_i2s_hw_params() 65 return -EINVAL; in psc_i2s_hw_params() 67 out_be32(&psc_dma->psc_regs->sicr, psc_dma->sicr \| mode); in psc_i2s_hw_params() 90 dev_dbg(psc_dma->dev, "psc_i2s_set_sysclk(cpu_dai=%p, dir=%i)\n", in psc_i2s_set_sysclk() [all …]
H A D	fsl_qmc_audio.c	1 // SPDX-License-Identifier: GPL-2.0 10 #include <linux/dma-mapping.h> 60 struct snd_card card = rtd->card->snd_card; 63 ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); 67 snd_pcm_set_managed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, card->dev, in qmc_audio_pcm_construct() 88 struct snd_pcm_runtime runtime = substream->runtim in qmc_audio_access_is_interleaved() 394 qmc_dai_get_index(struct snd_soc_dai * dai) qmc_dai_get_index() argument 401 qmc_dai_get_data(struct snd_soc_dai * dai) qmc_dai_get_data() argument 422 snd_pcm_format_t format = params_format(params); qmc_dai_hw_rule_channels_by_format() local 472 snd_pcm_format_t format; qmc_dai_hw_rule_format_by_channels() local 595 qmc_dai_startup(struct snd_pcm_substream * substream,struct snd_soc_dai * dai) qmc_dai_startup() argument 615 qmc_dai_hw_params(struct snd_pcm_substream * substream,struct snd_pcm_hw_params * params,struct snd_soc_dai * dai) qmc_dai_hw_params() argument 663 qmc_dai_trigger(struct snd_pcm_substream * substream,int cmd,struct snd_soc_dai * dai) qmc_dai_trigger() argument 750 snd_pcm_format_t format; qmc_audio_formats() local [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 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 83 * @dai_link: DAI link structure including normal one and DPCM link 91 * @sample_rate: Current sample rate [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__) 159 * fsl_asrc_sel_proc - Select the pre-processing and post-processing options 160 * @inrate: input sample rate 161 * @outrate: output sample rate [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 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() 87 player->state = UNIPERIF_STATE_UNDERFLOW; in uni_player_irq_handler() [all …]
H A D	uniperif_reader.c	1 // SPDX-License-Identifier: GPL-2.0-only 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() 74 snd_pcm_stream_unlock(reader->substream); in uni_reader_irq_handler() 76 spin_unlock(&reader->irq_lock); in uni_reader_irq_handler() [all …]
/linux/sound/soc/codecs/
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() 70 u16 format = 0; in si476x_codec_set_dai_fmt() local 73 return -EINVAL; in si476x_codec_set_dai_fmt() 77 format \|= SI476X_DAUDIO_MODE_DSP_A; in si476x_codec_set_dai_fmt() 80 format \|= SI476X_DAUDIO_MODE_DSP_B; in si476x_codec_set_dai_fmt() 83 format \|= SI476X_DAUDIO_MODE_I2S; in si476x_codec_set_dai_fmt() 86 format \|= SI476X_DAUDIO_MODE_RIGHT_J; in si476x_codec_set_dai_fmt() [all …]
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	tas5086.c	1 // SPDX-License-Identifier: GPL-2.0-or-later 8 * - implement DAPM and input muxing 9 * - implement modulation limit 10 * - implement non-default PWM start 13 * because the registers are of unequal size, and multi-byte registers 18 * it doesn't matter because the entire map can be accessed as 8-bit 21 * routines have to be open-coded. 70 #define TAS5086_CHANNEL_VOL(X) (0x08 + (X)) /* Channel 1-6 volume / 88 Default TAS5086 power-up configuration 172 size = tas5086_register_size(&client->dev, reg); in tas5086_reg_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/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 …]
H A D	sun8i-codec.c	1 // SPDX-License-Identifier: GPL-2.0-or-later 6 * (C) Copyright 2010-2016 9 * Mylène Josserand <mylene.josserand@free-electrons.com> 27 #include <sound/soc-dapm.h> 251 ret = clk_prepare_enable(scodec->clk_bus); in sun8i_codec_runtime_resume() 257 regcache_cache_only(scodec->regmap, false); in sun8i_codec_runtime_resume() 259 ret = regcache_sync(scodec->regmap); in sun8i_codec_runtime_resume() 272 regcache_cache_only(scodec->regmap, true); in sun8i_codec_runtime_suspend() 273 regcache_mark_dirty(scodec->regmap); in sun8i_codec_runtime_suspend() 275 clk_disable_unprepare(scodec->clk_bus); in sun8i_codec_runtime_suspend() [all …]
/linux/sound/soc/pxa/
H A D	mmp-sspa.c	1 // SPDX-License-Identifier: GPL-2.0-or-later 3 * linux/sound/soc/pxa/mmp-sspa.c 4 * Base on pxa2xx-ssp.c 23 #include <sound/pxa2xx-lib.h> 25 #include "mmp-sspa.h" 47 unsigned int sspa_sp = sspa->sp; in mmp_sspa_tx_enable() 52 __raw_writel(sspa_sp, sspa->tx_base + SSPA_SP); in mmp_sspa_tx_enable() 57 unsigned int sspa_sp = sspa->sp; in mmp_sspa_tx_disable() 62 __raw_writel(sspa_sp, sspa->tx_base + SSPA_SP); in mmp_sspa_tx_disable() 67 unsigned int sspa_sp = sspa->sp; in mmp_sspa_rx_enable() [all …]
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-tdm.c	1 // SPDX-License-Identifier: GPL-2.0 3 // MediaTek ALSA SoC Audio DAI TDM Control 10 #include "mt8183-afe-clk.h" 11 #include "mt8183-afe-common.h" 12 #include "mt8183-interconnection.h" 13 #include "mt8183-reg.h" 22 int mclk_multiple; /* according to sample rate */ 72 static unsigned int get_hdmi_wlen(snd_pcm_format_t format) in get_hdmi_wlen() argument 74 return snd_pcm_format_physical_width(format) <= 16 ? in get_hdmi_wlen() 78 static unsigned int get_tdm_wlen(snd_pcm_format_t format) in get_tdm_wlen() argument [all …]
/linux/sound/soc/sof/
H A D	ipc4-topology.c	`1 // SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) 14 #include <sound/intel-nhlt.h> 15 #include "sof-priv.h" 16 #include "sof-audio.h" 17 #include "ipc4-priv.h" 18 #include "ipc4-topology.h" 188 [SOF_DAI_TOKENS] = {"DAI tokens", dai_tokens, ARRAY_SIZE(dai_tokens)}, 196 [SOF_IN_AUDIO_FORMAT_TOKENS] = {"IPC4 Input Audio format tokens", 198 [SOF_OUT_AUDIO_FORMAT_TOKENS] = {"IPC4 Output Audio format tokens", 204 [SOF_AUDIO_FMT_NUM_TOKENS] = {"IPC4 Audio format number tokens", [all …]`
H A D	ipc4-topology.h	1 /* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) / 15 #define SOF_IPC4_FW_PAGE(x) ((((x) + BIT(12) - 1) & ~(BIT(12) - 1)) >> 12) 16 #define SOF_IPC4_FW_ROUNDUP(x) (((x) + BIT(6) - 1) & (~(BIT(6) - 1))) 22 LL domain - Low latency domain 23 * DP domain - Data processing domain 44 /* IPC4 sample types / 53 / Node index and mask applicable for host copier and ALH/HDA type DAI copiers / 59 / Node ID for SSP type DAI copiers / 62 / Node ID for DMIC type DAI copiers / 80 The base of multi-gateways. Multi-gateways addressing starts from [all …]
/linux/sound/soc/mediatek/mt8192/
H A D	mt8192-dai-tdm.c	1 // SPDX-License-Identifier: GPL-2.0 3 // MediaTek ALSA SoC Audio DAI TDM Control 11 #include "mt8192-afe-clk.h" 12 #include "mt8192-afe-common.h" 13 #include "mt8192-afe-gpio.h" 14 #include "mt8192-interconnection.h" 24 int mclk_multiple; /* according to sample rate */ 70 static unsigned int get_tdm_wlen(snd_pcm_format_t format) in get_tdm_wlen() argument 72 return snd_pcm_format_physical_width(format) <= 16 ? in get_tdm_wlen() 76 static unsigned int get_tdm_channel_bck(snd_pcm_format_t format) in get_tdm_channel_bck() argument [all …]
/linux/arch/arm/boot/dts/cirrus/
H A D	ep93xx-edb9302.dts	`1 // SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause 5 /dts-v1/; 9 #address-cells = <1>; 10 #size-cells = <1>; 27 compatible = "audio-graph-card2"; 33 compatible = "gpio-leds"; 34 led-0 { 37 linux,default-trigger = "heartbeat"; 41 led-1 { 55 compatible = "cfi-flash"; [all …]`
/linux/sound/soc/atmel/
H A D	mchp-spdiftx.c	1 // SPDX-License-Identifier: GPL-2.0 22 * ---- S/PDIF Transmitter Controller Register map ---- 39 * ---- Control Register (Write-only) ---- 45 * ---- Mode Register (Read/Write) ---- 73 /* Valid Bits per Sample / 88 / Bytes per Sample / 92 ---- Interrupt Enable/Disable/Mask/Status Register (Write/Read-only) ---- 205 regmap_read(dev->regmap, SPDIFTX_MR, &mr); in mchp_spdiftx_is_running() 211 struct mchp_spdiftx_mixer_control *ctrl = &dev->control; in mchp_spdiftx_channel_status_write() 215 for (i = 0; i < ARRAY_SIZE(ctrl->ch_stat) / 4; i++) { in mchp_spdiftx_channel_status_write() [all …]
/linux/sound/soc/mediatek/mt8186/
H A D	mt8186-dai-tdm.c	1 // SPDX-License-Identifier: GPL-2.0 3 // MediaTek ALSA SoC Audio DAI TDM Control 11 #include "mt8186-afe-clk.h" 12 #include "mt8186-afe-common.h" 13 #include "mt8186-afe-gpio.h" 14 #include "mt8186-interconnection.h" 27 unsigned int mclk_multiple; /* according to sample rate / 59 static unsigned int get_tdm_lrck_width(snd_pcm_format_t format, in get_tdm_lrck_width() argument 65 return snd_pcm_format_physical_width(format) - 1; in get_tdm_lrck_width() 103 struct snd_soc_component cmpnt = snd_soc_dapm_to_component(w->dapm); in mtk_tdm_en_event() [all …]
/linux/sound/soc/ti/
H A D	omap-mcbsp.c	1 // SPDX-License-Identifier: GPL-2.0-only 3 * omap-mcbsp.c -- OMAP ALSA SoC DAI driver using McBSP port 23 #include "omap-mcbsp-priv.h" 24 #include "omap-mcbsp.h" 25 #include "sdma-pcm.h" 40 dev_dbg(mcbsp->dev, "** McBSP%d regs **\n", mcbsp->id); in omap_mcbsp_dump_reg() 41 dev_dbg(mcbsp->dev, "DRR2: 0x%04x\n", MCBSP_READ(mcbsp, DRR2)); in omap_mcbsp_dump_reg() 42 dev_dbg(mcbsp->dev, "DRR1: 0x%04x\n", MCBSP_READ(mcbsp, DRR1)); in omap_mcbsp_dump_reg() 43 dev_dbg(mcbsp->dev, "DXR2: 0x%04x\n", MCBSP_READ(mcbsp, DXR2)); in omap_mcbsp_dump_reg() 44 dev_dbg(mcbsp->dev, "DXR1: 0x%04x\n", MCBSP_READ(mcbsp, DXR1)); in omap_mcbsp_dump_reg() [all …]

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