| /* | 
 |  * 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 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, 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_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, 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_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 |