mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
synced 2024-12-08 17:26:40 +07:00
be2500b835
Add DAI format definition for PDM interfaces. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
232 lines
7.0 KiB
C
232 lines
7.0 KiB
C
/*
|
|
* linux/sound/soc-dai.h -- ALSA SoC Layer
|
|
*
|
|
* Copyright: 2005-2008 Wolfson Microelectronics. PLC.
|
|
*
|
|
* This program is free software; you can redistribute it and/or modify
|
|
* it under the terms of the GNU General Public License version 2 as
|
|
* published by the Free Software Foundation.
|
|
*
|
|
* Digital Audio Interface (DAI) API.
|
|
*/
|
|
|
|
#ifndef __LINUX_SND_SOC_DAI_H
|
|
#define __LINUX_SND_SOC_DAI_H
|
|
|
|
|
|
#include <linux/list.h>
|
|
|
|
struct snd_pcm_substream;
|
|
|
|
/*
|
|
* DAI hardware audio formats.
|
|
*
|
|
* Describes the physical PCM data formating and clocking. Add new formats
|
|
* to the end.
|
|
*/
|
|
#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */
|
|
#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */
|
|
#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */
|
|
#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */
|
|
#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */
|
|
#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
|
|
#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */
|
|
|
|
/* left and right justified also known as MSB and LSB respectively */
|
|
#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
|
|
#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
|
|
|
|
/*
|
|
* DAI Clock gating.
|
|
*
|
|
* DAI bit clocks can be be gated (disabled) when the DAI is not
|
|
* sending or receiving PCM data in a frame. This can be used to save power.
|
|
*/
|
|
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
|
|
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
|
|
|
|
/*
|
|
* DAI hardware signal inversions.
|
|
*
|
|
* Specifies whether the DAI can also support inverted clocks for the specified
|
|
* format.
|
|
*/
|
|
#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */
|
|
#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */
|
|
#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */
|
|
#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */
|
|
|
|
/*
|
|
* DAI hardware clock masters.
|
|
*
|
|
* This is wrt the codec, the inverse is true for the interface
|
|
* i.e. if the codec is clk and FRM master then the interface is
|
|
* clk and frame slave.
|
|
*/
|
|
#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */
|
|
#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */
|
|
#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */
|
|
#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */
|
|
|
|
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
|
|
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
|
|
#define SND_SOC_DAIFMT_INV_MASK 0x0f00
|
|
#define SND_SOC_DAIFMT_MASTER_MASK 0xf000
|
|
|
|
/*
|
|
* Master Clock Directions
|
|
*/
|
|
#define SND_SOC_CLOCK_IN 0
|
|
#define SND_SOC_CLOCK_OUT 1
|
|
|
|
#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
|
|
SNDRV_PCM_FMTBIT_S16_LE |\
|
|
SNDRV_PCM_FMTBIT_S16_BE |\
|
|
SNDRV_PCM_FMTBIT_S20_3LE |\
|
|
SNDRV_PCM_FMTBIT_S20_3BE |\
|
|
SNDRV_PCM_FMTBIT_S24_3LE |\
|
|
SNDRV_PCM_FMTBIT_S24_3BE |\
|
|
SNDRV_PCM_FMTBIT_S32_LE |\
|
|
SNDRV_PCM_FMTBIT_S32_BE)
|
|
|
|
struct snd_soc_dai_ops;
|
|
struct snd_soc_dai;
|
|
struct snd_ac97_bus_ops;
|
|
|
|
/* Digital Audio Interface registration */
|
|
int snd_soc_register_dai(struct snd_soc_dai *dai);
|
|
void snd_soc_unregister_dai(struct snd_soc_dai *dai);
|
|
int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
|
|
void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
|
|
|
|
/* Digital Audio Interface clocking API.*/
|
|
int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
|
|
unsigned int freq, int dir);
|
|
|
|
int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
|
|
int div_id, int div);
|
|
|
|
int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
|
|
int pll_id, int source, unsigned int freq_in, unsigned int freq_out);
|
|
|
|
/* Digital Audio interface formatting */
|
|
int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
|
|
|
|
int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
|
|
unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
|
|
|
|
int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
|
|
unsigned int tx_num, unsigned int *tx_slot,
|
|
unsigned int rx_num, unsigned int *rx_slot);
|
|
|
|
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
|
|
|
|
/* Digital Audio Interface mute */
|
|
int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
|
|
|
|
/*
|
|
* Digital Audio Interface.
|
|
*
|
|
* Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
|
|
* operations and capabilities. Codec and platform drivers will register this
|
|
* structure for every DAI they have.
|
|
*
|
|
* This structure covers the clocking, formating and ALSA operations for each
|
|
* interface.
|
|
*/
|
|
struct snd_soc_dai_ops {
|
|
/*
|
|
* DAI clocking configuration, all optional.
|
|
* Called by soc_card drivers, normally in their hw_params.
|
|
*/
|
|
int (*set_sysclk)(struct snd_soc_dai *dai,
|
|
int clk_id, unsigned int freq, int dir);
|
|
int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source,
|
|
unsigned int freq_in, unsigned int freq_out);
|
|
int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
|
|
|
|
/*
|
|
* DAI format configuration
|
|
* Called by soc_card drivers, normally in their hw_params.
|
|
*/
|
|
int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
|
|
int (*set_tdm_slot)(struct snd_soc_dai *dai,
|
|
unsigned int tx_mask, unsigned int rx_mask,
|
|
int slots, int slot_width);
|
|
int (*set_channel_map)(struct snd_soc_dai *dai,
|
|
unsigned int tx_num, unsigned int *tx_slot,
|
|
unsigned int rx_num, unsigned int *rx_slot);
|
|
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
|
|
|
|
/*
|
|
* DAI digital mute - optional.
|
|
* Called by soc-core to minimise any pops.
|
|
*/
|
|
int (*digital_mute)(struct snd_soc_dai *dai, int mute);
|
|
|
|
/*
|
|
* ALSA PCM audio operations - all optional.
|
|
* Called by soc-core during audio PCM operations.
|
|
*/
|
|
int (*startup)(struct snd_pcm_substream *,
|
|
struct snd_soc_dai *);
|
|
void (*shutdown)(struct snd_pcm_substream *,
|
|
struct snd_soc_dai *);
|
|
int (*hw_params)(struct snd_pcm_substream *,
|
|
struct snd_pcm_hw_params *, struct snd_soc_dai *);
|
|
int (*hw_free)(struct snd_pcm_substream *,
|
|
struct snd_soc_dai *);
|
|
int (*prepare)(struct snd_pcm_substream *,
|
|
struct snd_soc_dai *);
|
|
int (*trigger)(struct snd_pcm_substream *, int,
|
|
struct snd_soc_dai *);
|
|
};
|
|
|
|
/*
|
|
* Digital Audio Interface runtime data.
|
|
*
|
|
* Holds runtime data for a DAI.
|
|
*/
|
|
struct snd_soc_dai {
|
|
/* DAI description */
|
|
char *name;
|
|
unsigned int id;
|
|
int ac97_control;
|
|
|
|
struct device *dev;
|
|
void *ac97_pdata; /* platform_data for the ac97 codec */
|
|
|
|
/* DAI callbacks */
|
|
int (*probe)(struct platform_device *pdev,
|
|
struct snd_soc_dai *dai);
|
|
void (*remove)(struct platform_device *pdev,
|
|
struct snd_soc_dai *dai);
|
|
int (*suspend)(struct snd_soc_dai *dai);
|
|
int (*resume)(struct snd_soc_dai *dai);
|
|
|
|
/* ops */
|
|
struct snd_soc_dai_ops *ops;
|
|
|
|
/* DAI capabilities */
|
|
struct snd_soc_pcm_stream capture;
|
|
struct snd_soc_pcm_stream playback;
|
|
unsigned int symmetric_rates:1;
|
|
|
|
/* DAI runtime info */
|
|
struct snd_pcm_runtime *runtime;
|
|
struct snd_soc_codec *codec;
|
|
unsigned int active;
|
|
unsigned char pop_wait:1;
|
|
void *dma_data;
|
|
|
|
/* DAI private data */
|
|
void *private_data;
|
|
|
|
/* parent platform */
|
|
struct snd_soc_platform *platform;
|
|
|
|
struct list_head list;
|
|
};
|
|
|
|
#endif
|