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/*
* 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 */
/* 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 not 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 Left/Right Clocks.
*
* Specifies whether the DAI can support different samples for similtanious
* playback and capture. This usually requires a seperate physical frame
* clock for playback and capture.
*/
#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
/*
* TDM
*
* Time Division Multiplexing. Allows PCM data to be multplexed with other
* data on the DAI.
*/
#define SND_SOC_DAIFMT_TDM (1 << 6)
/*
* 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
struct snd_soc_dai_ops;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
/* 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, 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 mask, int slots);
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 it's ALSA, DAI and AC97
* operations an capabilities. Codec and platfom drivers will register a this
* structure for every DAI they have.
*
* This structure covers the clocking, formating and ALSA operations for each
* interface a
*/
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, 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 mask, int slots);
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;
/* 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 platform_device *pdev,
struct snd_soc_dai *dai);
int (*resume)(struct platform_device *pdev,
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;
/* 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 codec/platform */
union {
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
};
struct list_head list;
};
#endif
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