/*
 * 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;
struct snd_soc_dapm_widget;
struct snd_compr_stream;

/*
 * DAI hardware audio formats.
 *
 * Describes the physical PCM data formating and clocking. Add new formats
 * to the end.
 */
#define SND_SOC_DAIFMT_I2S              1 /* I2S mode */
#define SND_SOC_DAIFMT_RIGHT_J          2 /* Right Justified mode */
#define SND_SOC_DAIFMT_LEFT_J           3 /* Left Justified mode */
#define SND_SOC_DAIFMT_DSP_A            4 /* L data MSB after FRM LRC */
#define SND_SOC_DAIFMT_DSP_B            5 /* L data MSB during FRM LRC */
#define SND_SOC_DAIFMT_AC97             6 /* AC97 */
#define SND_SOC_DAIFMT_PDM              7 /* Pulse density modulation */
#define SND_SOC_DAIFMT_SPDIF            8 /* SPDIF */

/* 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             (1 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED            (0 << 4) /* clock is gated */

/*
 * DAI hardware signal polarity.
 *
 * Specifies whether the DAI can also support inverted clocks for the specified
 * format.
 *
 * BCLK:
 * - "normal" polarity means signal is available at rising edge of BCLK
 * - "inverted" polarity means signal is available at falling edge of BCLK
 *
 * FSYNC "normal" polarity depends on the frame format:
 * - I2S: frame consists of left then right channel data. Left channel starts
 *      with falling FSYNC edge, right channel starts with rising FSYNC edge.
 * - Left/Right Justified: frame consists of left then right channel data.
 *      Left channel starts with rising FSYNC edge, right channel starts with
 *      falling FSYNC edge.
 * - DSP A/B: Frame starts with rising FSYNC edge.
 * - AC97: Frame starts with rising FSYNC edge.
 *
 * "Negative" FSYNC polarity is the one opposite of "normal" polarity.
 */
#define SND_SOC_DAIFMT_NB_NF            (0 << 8) /* normal bit clock + frame */
#define SND_SOC_DAIFMT_NB_IF            (2 << 8) /* normal BCLK + inv FRM */
#define SND_SOC_DAIFMT_IB_NF            (3 << 8) /* invert BCLK + nor FRM */
#define SND_SOC_DAIFMT_IB_IF            (4 << 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          (1 << 12) /* codec clk & FRM master */
#define SND_SOC_DAIFMT_CBS_CFM          (2 << 12) /* codec clk slave & FRM master */
#define SND_SOC_DAIFMT_CBM_CFS          (3 << 12) /* codec clk master & frame slave */
#define SND_SOC_DAIFMT_CBS_CFS          (4 << 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_driver;
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, int source, unsigned int freq_in, unsigned int freq_out);

int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio);

/* 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,
                             int direction);

int snd_soc_dai_is_dummy(struct snd_soc_dai *dai);

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);
        int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio);

        /*
         * 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 (*xlate_tdm_slot_mask)(unsigned int slots,
                unsigned int *tx_mask, unsigned int *rx_mask);
        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);
        int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);

        /*
         * 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 *);
        /*
         * NOTE: Commands passed to the trigger function are not necessarily
         * compatible with the current state of the dai. For example this
         * sequence of commands is possible: START STOP STOP.
         * So do not unconditionally use refcounting functions in the trigger
         * function, e.g. clk_enable/disable.
         */
        int (*trigger)(struct snd_pcm_substream *, int,
                struct snd_soc_dai *);
        int (*bespoke_trigger)(struct snd_pcm_substream *, int,
                struct snd_soc_dai *);
        /*
         * For hardware based FIFO caused delay reporting.
         * Optional.
         */
        snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
                struct snd_soc_dai *);
};

/*
 * Digital Audio Interface Driver.
 *
 * 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_driver {
        /* DAI description */
        const char *name;
        unsigned int id;
        unsigned int base;
        struct snd_soc_dobj dobj;

        /* DAI driver callbacks */
        int (*probe)(struct snd_soc_dai *dai);
        int (*remove)(struct snd_soc_dai *dai);
        int (*suspend)(struct snd_soc_dai *dai);
        int (*resume)(struct snd_soc_dai *dai);
        /* compress dai */
        int (*compress_new)(struct snd_soc_pcm_runtime *rtd, int num);
        /* DAI is also used for the control bus */
        bool bus_control;

        /* ops */
        const 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;
        unsigned int symmetric_channels:1;
        unsigned int symmetric_samplebits:1;

        /* probe ordering - for components with runtime dependencies */
        int probe_order;
        int remove_order;
};

/*
 * Digital Audio Interface runtime data.
 *
 * Holds runtime data for a DAI.
 */
struct snd_soc_dai {
        const char *name;
        int id;
        struct device *dev;

        /* driver ops */
        struct snd_soc_dai_driver *driver;

        /* DAI runtime info */
        unsigned int capture_active:1;          /* stream is in use */
        unsigned int playback_active:1;         /* stream is in use */
        unsigned int symmetric_rates:1;
        unsigned int symmetric_channels:1;
        unsigned int symmetric_samplebits:1;
        unsigned int active;
        unsigned char probed:1;

        struct snd_soc_dapm_widget *playback_widget;
        struct snd_soc_dapm_widget *capture_widget;

        /* DAI DMA data */
        void *playback_dma_data;
        void *capture_dma_data;

        /* Symmetry data - only valid if symmetry is being enforced */
        unsigned int rate;
        unsigned int channels;
        unsigned int sample_bits;

        /* parent platform/codec */
        struct snd_soc_codec *codec;
        struct snd_soc_component *component;

        /* CODEC TDM slot masks and params (for fixup) */
        unsigned int tx_mask;
        unsigned int rx_mask;

        struct list_head list;
};

static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
                                             const struct snd_pcm_substream *ss)
{
        return (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
                dai->playback_dma_data : dai->capture_dma_data;
}

static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai,
                                            const struct snd_pcm_substream *ss,
                                            void *data)
{
        if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK)
                dai->playback_dma_data = data;
        else
                dai->capture_dma_data = data;
}

static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai,
                                             void *playback, void *capture)
{
        dai->playback_dma_data = playback;
        dai->capture_dma_data = capture;
}

static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai,
                void *data)
{
        dev_set_drvdata(dai->dev, data);
}

static inline void *snd_soc_dai_get_drvdata(struct snd_soc_dai *dai)
{
        return dev_get_drvdata(dai->dev);
}

#endif