--- /dev/null
+/*
+ * 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 */
+
+/* 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 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 (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;
+
+ /* 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 */
+ bool compress_dai;
+ /* 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
Code Review / kvmfornfv.git / blobdiff