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-rw-r--r--include/sound/soc-dai.h209
-rw-r--r--include/sound/soc.h148
2 files changed, 211 insertions, 146 deletions
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
new file mode 100644
index 000000000000..08b8f7025c64
--- /dev/null
+++ b/include/sound/soc-dai.h
@@ -0,0 +1,209 @@
+/*
+ * 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);
+};
+
+/*
+ * Digital Audio Interface runtime data.
+ *
+ * Holds runtime data for a DAI.
+ */
+struct snd_soc_dai {
+ /* DAI description */
+ char *name;
+ unsigned int id;
+ unsigned char type;
+
+ /* 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_ops ops;
+ struct snd_soc_dai_ops dai_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
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3be17b3c650c..e4465f73aa46 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -151,76 +151,6 @@ enum snd_soc_bias_level {
#define SND_SOC_DAI_PCM 0x4
#define SND_SOC_DAI_AC97_BUS 0x8 /* for custom i.e. non ac97_codec.c */
-/*
- * DAI hardware audio formats
- */
-#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 or LRC */
-#define SND_SOC_DAIFMT_DSP_B 4 /* L data msb during FRM or LRC */
-#define SND_SOC_DAIFMT_AC97 5 /* AC97 */
-
-#define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J
-#define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J
-
-/*
- * DAI Gating
- */
-#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
-#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated when not Tx/Rx */
-
-/*
- * DAI Sync
- * Synchronous LR (Left Right) clocks and Frame signals.
- */
-#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
-#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
-
-/*
- * TDM
- */
-#define SND_SOC_DAIFMT_TDM (1 << 6)
-
-/*
- * DAI hardware signal inversions
- */
-#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bclk + frm */
-#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
-
-/*
- * AC97 codec ID's bitmask
- */
-#define SND_SOC_DAI_AC97_ID0 (1 << 0)
-#define SND_SOC_DAI_AC97_ID1 (1 << 1)
-#define SND_SOC_DAI_AC97_ID2 (1 << 2)
-#define SND_SOC_DAI_AC97_ID3 (1 << 3)
-
struct snd_soc_device;
struct snd_soc_pcm_stream;
struct snd_soc_ops;
@@ -260,27 +190,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
struct snd_ac97_bus_ops *ops, int num);
void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
-/* 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);
-
/*
*Controls
*/
@@ -338,61 +247,6 @@ struct snd_soc_ops {
int (*trigger)(struct snd_pcm_substream *, int);
};
-/* ASoC DAI ops */
-struct snd_soc_dai_ops {
- /* DAI clocking configuration */
- 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 */
- 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);
-
- /* digital mute */
- int (*digital_mute)(struct snd_soc_dai *dai, int mute);
-};
-
-/* SoC DAI (Digital Audio Interface) */
-struct snd_soc_dai {
- /* DAI description */
- char *name;
- unsigned int id;
- unsigned char type;
-
- /* 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_ops ops;
- struct snd_soc_dai_ops dai_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;
-};
-
/* SoC Audio Codec */
struct snd_soc_codec {
char *name;
@@ -543,4 +397,6 @@ struct soc_enum {
void *dapm;
};
+#include <sound/soc-dai.h>
+
#endif