blob: 99f65fb16ac5fdff8d735876f6e69b910721a5d0 [file] [log] [blame]
/*
* 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 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 */
/* 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 (2 << 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 (1 << 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)
/*
* DAI channels - for mapping to widgets.
* Indexes correspond to WAV format for convenience.
*/
#define SND_SOC_DAI_CHAN(chan) (1 << chan)
#define SND_SOC_DAI_CHAN_MASK(channels) ((1UL << channels) - 1)
#define SND_SOC_DAI_CHAN_LEFT (1 << 0)
#define SND_SOC_DAI_CHAN_RIGHT (1 << 1)
#define SND_SOC_DAI_CHAN_STEREO \
(SND_SOC_DAI_CHAN_LEFT | SND_SOC_DAI_CHAN_RIGHT)
#define SND_SOC_DAI_CHAN_MONO SND_SOC_DAI_CHAN_LEFT
#define SND_SOC_DAI_CHAN_FRONT_LEFT SND_SOC_DAI_CHAN_LEFT
#define SND_SOC_DAI_CHAN_FRONT_RIGHT SND_SOC_DAI_CHAN_RIGHT
#define SND_SOC_DAI_CHAN_CENTER (1 << 2)
#define SND_SOC_DAI_CHAN_LFE (1 << 3)
#define SND_SOC_DAI_CHAN_BACK_LEFT (1 << 4)
#define SND_SOC_DAI_CHAN_BACK_RIGHT (1 << 5)
#define SND_SOC_DAI_CHAN_FRONT_LEFT_CENTER (1 << 6)
#define SND_SOC_DAI_CHAN_FRONT_RIGHT_CENTER (1 << 7)
#define SND_SOC_DAI_CHAN_BACK_CENTER (1 << 8)
#define SND_SOC_DAI_CHAN_SIDE_LEFT (1 << 9)
#define SND_SOC_DAI_CHAN_SIDE_RIGHT (1 << 10)
#define SND_SOC_DAI_CHAN_LEFT_HEIGHT (1 << 12)
#define SND_SOC_DAI_CHAN_RIGHT_HEIGHT (1 << 14)
struct snd_soc_dai_driver;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
/* Digital Audio Interface registration */
int snd_soc_register_dai(struct device *dev,
struct snd_soc_dai_driver *dai_drv);
void snd_soc_unregister_dai(struct device *dev);
int snd_soc_register_dais(struct device *dev,
struct snd_soc_dai_driver *dai_drv, size_t count);
void snd_soc_unregister_dais(struct device *dev, 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);
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 *);
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 *);
};
/*
* Maps widgets to channels and DAIs.
*
* Used by the DAI driver to map connected widgets and supported channel
* masks.
*
* Also can be used by the DAI device to configure the channel to widget
* map at hw_params() prior to DAPM walk at prepare().
*
*/
struct snd_soc_dai_widget {
const char *name;
u32 channel_map;
};
/*
* 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;
int ac97_control;
/* 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);
/* 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;
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
struct snd_soc_dai_widget *widgets;
int num_widgets;
};
/*
* Digital Audio Interface runtime data.
*
* Holds runtime data for a DAI.
*/
struct snd_soc_dai {
const char *name;
int id;
struct device *dev;
void *ac97_pdata; /* platform_data for the ac97 codec */
/* 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;
struct snd_pcm_runtime *runtime;
unsigned int active;
unsigned char pop_wait:1;
unsigned char probed:1;
/* DAI DMA data */
void *playback_dma_data;
void *capture_dma_data;
/* Symmetry data - only valid if symmetry is being enforced */
unsigned int rate;
/* parent platform/codec */
union {
struct snd_soc_platform *platform;
struct snd_soc_codec *codec;
};
struct snd_soc_card *card;
struct list_head list;
struct list_head card_list;
struct snd_soc_dai_widget *widgets;
};
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_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);
}
/* Backend DAI PCM ops */
static inline int snd_soc_dai_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
int ret = 0;
mutex_lock(&rtd->pcm_mutex);
if (dai->driver->ops->startup)
ret = dai->driver->ops->startup(substream, dai);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dai->playback_active++;
else
dai->capture_active++;
dai->active++;
mutex_unlock(&rtd->pcm_mutex);
return ret;
}
static inline void snd_soc_dai_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
mutex_lock(&rtd->pcm_mutex);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
dai->playback_active--;
else
dai->capture_active--;
dai->active--;
if (dai->driver->ops->shutdown)
dai->driver->ops->shutdown(substream, dai);
mutex_unlock(&rtd->pcm_mutex);
}
static inline int snd_soc_dai_hw_params(struct snd_pcm_substream * substream,
struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
int ret = 0;
mutex_lock(&rtd->pcm_mutex);
if (dai->driver->ops->hw_params)
ret = dai->driver->ops->hw_params(substream, hw_params, dai);
mutex_unlock(&rtd->pcm_mutex);
return ret;
}
static inline int snd_soc_dai_hw_free(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
int ret = 0;
mutex_lock(&rtd->pcm_mutex);
if (dai->driver->ops->hw_free)
ret = dai->driver->ops->hw_free(substream, dai);
mutex_unlock(&rtd->pcm_mutex);
return ret;
}
static inline int snd_soc_dai_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
int ret = 0;
mutex_lock(&rtd->pcm_mutex);
if (dai->driver->ops->prepare)
ret = dai->driver->ops->prepare(substream, dai);
mutex_unlock(&rtd->pcm_mutex);
return ret;
}
static inline int snd_soc_dai_trigger(struct snd_pcm_substream *substream,
int cmd, struct snd_soc_dai *dai)
{
if (dai->driver->ops->trigger)
return dai->driver->ops->trigger(substream, cmd, dai);
return 0;
}
#endif