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-rw-r--r--include/sound/ac97_codec.h9
-rw-r--r--include/sound/sh_fsi.h83
-rw-r--r--include/sound/soc-dai.h40
-rw-r--r--include/sound/soc-dapm.h10
-rw-r--r--include/sound/soc.h49
-rw-r--r--include/sound/uda1380.h22
-rw-r--r--include/sound/wm8993.h44
7 files changed, 235 insertions, 22 deletions
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 251fc1cd5002..3dae3f799b9b 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -32,6 +32,9 @@
#include "control.h"
#include "info.h"
+/* maximum number of devices on the AC97 bus */
+#define AC97_BUS_MAX_DEVICES 4
+
/*
* AC'97 codec registers
*/
@@ -642,4 +645,10 @@ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime);
/* ad hoc AC97 device driver access */
extern struct bus_type ac97_bus_type;
+/* AC97 platform_data adding function */
+static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data)
+{
+ ac97->dev.platform_data = data;
+}
+
#endif /* __SOUND_AC97_CODEC_H */
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
new file mode 100644
index 000000000000..c0227361a876
--- /dev/null
+++ b/include/sound/sh_fsi.h
@@ -0,0 +1,83 @@
+#ifndef __SOUND_FSI_H
+#define __SOUND_FSI_H
+
+/*
+ * Fifo-attached Serial Interface (FSI) support for SH7724
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * 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.
+ */
+
+/* flags format
+
+ * 0xABCDEEFF
+ *
+ * A: channel size for TDM (input)
+ * B: channel size for TDM (ooutput)
+ * C: inversion
+ * D: mode
+ * E: input format
+ * F: output format
+ */
+
+#include <linux/clk.h>
+#include <sound/soc.h>
+
+/* TDM channel */
+#define SH_FSI_SET_CH_I(x) ((x & 0xF) << 28)
+#define SH_FSI_SET_CH_O(x) ((x & 0xF) << 24)
+
+#define SH_FSI_CH_IMASK 0xF0000000
+#define SH_FSI_CH_OMASK 0x0F000000
+#define SH_FSI_GET_CH_I(x) ((x & SH_FSI_CH_IMASK) >> 28)
+#define SH_FSI_GET_CH_O(x) ((x & SH_FSI_CH_OMASK) >> 24)
+
+/* clock inversion */
+#define SH_FSI_INVERSION_MASK 0x00F00000
+#define SH_FSI_LRM_INV (1 << 20)
+#define SH_FSI_BRM_INV (1 << 21)
+#define SH_FSI_LRS_INV (1 << 22)
+#define SH_FSI_BRS_INV (1 << 23)
+
+/* mode */
+#define SH_FSI_MODE_MASK 0x000F0000
+#define SH_FSI_IN_SLAVE_MODE (1 << 16) /* default master mode */
+#define SH_FSI_OUT_SLAVE_MODE (1 << 17) /* default master mode */
+
+/* DI format */
+#define SH_FSI_FMT_MASK 0x000000FF
+#define SH_FSI_IFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8)
+#define SH_FSI_OFMT(x) (((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0)
+#define SH_FSI_GET_IFMT(x) ((x >> 8) & SH_FSI_FMT_MASK)
+#define SH_FSI_GET_OFMT(x) ((x >> 0) & SH_FSI_FMT_MASK)
+
+#define SH_FSI_FMT_MONO (1 << 0)
+#define SH_FSI_FMT_MONO_DELAY (1 << 1)
+#define SH_FSI_FMT_PCM (1 << 2)
+#define SH_FSI_FMT_I2S (1 << 3)
+#define SH_FSI_FMT_TDM (1 << 4)
+#define SH_FSI_FMT_TDM_DELAY (1 << 5)
+
+#define SH_FSI_IFMT_TDM_CH(x) \
+ (SH_FSI_IFMT(TDM) | SH_FSI_SET_CH_I(x))
+#define SH_FSI_IFMT_TDM_DELAY_CH(x) \
+ (SH_FSI_IFMT(TDM_DELAY) | SH_FSI_SET_CH_I(x))
+
+#define SH_FSI_OFMT_TDM_CH(x) \
+ (SH_FSI_OFMT(TDM) | SH_FSI_SET_CH_O(x))
+#define SH_FSI_OFMT_TDM_DELAY_CH(x) \
+ (SH_FSI_OFMT(TDM_DELAY) | SH_FSI_SET_CH_O(x))
+
+struct sh_fsi_platform_info {
+ unsigned long porta_flags;
+ unsigned long portb_flags;
+};
+
+extern struct snd_soc_dai fsi_soc_dai[2];
+extern struct snd_soc_platform fsi_soc_platform;
+
+#endif /* __SOUND_FSI_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 352d7eee9b6d..97ca9af414dc 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -27,8 +27,8 @@ struct snd_pcm_substream;
#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_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 */
@@ -38,7 +38,7 @@ struct snd_pcm_substream;
/*
* DAI Clock gating.
*
- * DAI bit clocks can be be gated (disabled) when not the DAI is not
+ * 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 */
@@ -51,21 +51,21 @@ struct snd_pcm_substream;
* 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 */
+#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
+ * 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_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_CBS_CFS (3 << 12) /* codec clk & FRM slave */
#define SND_SOC_DAIFMT_FORMAT_MASK 0x000f
#define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0
@@ -78,7 +78,13 @@ struct snd_pcm_substream;
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
-#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
+#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)
@@ -106,7 +112,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
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);
+ 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);
@@ -116,12 +122,12 @@ 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
+ * 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 a
+ * interface.
*/
struct snd_soc_dai_ops {
/*
@@ -140,7 +146,8 @@ struct snd_soc_dai_ops {
*/
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);
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
/*
@@ -179,6 +186,7 @@ struct snd_soc_dai {
int ac97_control;
struct device *dev;
+ void *ac97_pdata; /* platform_data for the ac97 codec */
/* DAI callbacks */
int (*probe)(struct platform_device *pdev,
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index ec8a45f9a069..c1410e3191e3 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -137,6 +137,12 @@
.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD}
/* stream domain */
+#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
+ .reg = wreg, .shift = wshift, .invert = winvert }
+#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \
+{ .id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
+ .reg = wreg, .shift = wshift, .invert = winvert }
#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
@@ -279,9 +285,11 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
int event);
+void snd_soc_dapm_shutdown(struct snd_soc_device *socdev);
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
+void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec);
/* dapm audio pin control and status */
int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin);
@@ -311,6 +319,8 @@ enum snd_soc_dapm_type {
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
snd_soc_dapm_supply, /* power/clock supply */
+ snd_soc_dapm_aif_in, /* audio interface input */
+ snd_soc_dapm_aif_out, /* audio interface output */
};
/*
diff --git a/include/sound/soc.h b/include/sound/soc.h
index cf6111d72b17..475cb7ed6bec 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -135,6 +135,28 @@
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+ xhandler_get, xhandler_put, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+ .max = xmax, .invert = xinvert} }
+#define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\
+ xhandler_get, xhandler_put, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_2r, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+ .max = xmax, .invert = xinvert} }
#define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_bool_ext, \
@@ -183,14 +205,28 @@ struct snd_soc_jack_gpio;
#endif
typedef int (*hw_write_t)(void *,const char* ,int);
-typedef int (*hw_read_t)(void *,char* ,int);
extern struct snd_ac97_bus_ops soc_ac97_ops;
+enum snd_soc_control_type {
+ SND_SOC_CUSTOM,
+ SND_SOC_I2C,
+ SND_SOC_SPI,
+};
+
int snd_soc_register_platform(struct snd_soc_platform *platform);
void snd_soc_unregister_platform(struct snd_soc_platform *platform);
int snd_soc_register_codec(struct snd_soc_codec *codec);
void snd_soc_unregister_codec(struct snd_soc_codec *codec);
+int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg);
+int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
+ int addr_bits, int data_bits,
+ enum snd_soc_control_type control);
+
+#ifdef CONFIG_PM
+int snd_soc_suspend_device(struct device *dev);
+int snd_soc_resume_device(struct device *dev);
+#endif
/* pcm <-> DAI connect */
void snd_soc_free_pcms(struct snd_soc_device *socdev);
@@ -216,9 +252,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
/* codec register bit access */
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
- unsigned short mask, unsigned short value);
+ unsigned int mask, unsigned int value);
int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
- unsigned short mask, unsigned short value);
+ unsigned int mask, unsigned int value);
int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
struct snd_ac97_bus_ops *ops, int num);
@@ -356,8 +392,10 @@ struct snd_soc_codec {
int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
int (*display_register)(struct snd_soc_codec *, char *,
size_t, unsigned int);
+ int (*volatile_register)(unsigned int);
+ int (*readable_register)(unsigned int);
hw_write_t hw_write;
- hw_read_t hw_read;
+ unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
void *reg_cache;
short reg_cache_size;
short reg_cache_step;
@@ -369,8 +407,6 @@ struct snd_soc_codec {
enum snd_soc_bias_level bias_level;
enum snd_soc_bias_level suspend_bias_level;
struct delayed_work delayed_work;
- struct list_head up_list;
- struct list_head down_list;
/* codec DAI's */
struct snd_soc_dai *dai;
@@ -379,6 +415,7 @@ struct snd_soc_codec {
#ifdef CONFIG_DEBUG_FS
struct dentry *debugfs_reg;
struct dentry *debugfs_pop_time;
+ struct dentry *debugfs_dapm;
#endif
};
diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h
new file mode 100644
index 000000000000..381319c7000c
--- /dev/null
+++ b/include/sound/uda1380.h
@@ -0,0 +1,22 @@
+/*
+ * UDA1380 ALSA SoC Codec driver
+ *
+ * Copyright 2009 Philipp Zabel
+ *
+ * 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.
+ */
+
+#ifndef __UDA1380_H
+#define __UDA1380_H
+
+struct uda1380_platform_data {
+ int gpio_power;
+ int gpio_reset;
+ int dac_clk;
+#define UDA1380_DAC_CLK_SYSCLK 0
+#define UDA1380_DAC_CLK_WSPLL 1
+};
+
+#endif /* __UDA1380_H */
diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h
new file mode 100644
index 000000000000..9c661f2f8cda
--- /dev/null
+++ b/include/sound/wm8993.h
@@ -0,0 +1,44 @@
+/*
+ * linux/sound/wm8993.h -- Platform data for WM8993
+ *
+ * Copyright 2009 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.
+ */
+
+#ifndef __LINUX_SND_WM8993_H
+#define __LINUX_SND_WM8993_H
+
+/* Note that EQ1 only contains the enable/disable bit so will be
+ ignored but is included for simplicity.
+ */
+struct wm8993_retune_mobile_setting {
+ const char *name;
+ unsigned int rate;
+ u16 config[24];
+};
+
+struct wm8993_platform_data {
+ struct wm8993_retune_mobile_setting *retune_configs;
+ int num_retune_configs;
+
+ /* LINEOUT can be differential or single ended */
+ unsigned int lineout1_diff:1;
+ unsigned int lineout2_diff:1;
+
+ /* Common mode feedback */
+ unsigned int lineout1fb:1;
+ unsigned int lineout2fb:1;
+
+ /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
+ unsigned int micbias1_lvl:1;
+ unsigned int micbias2_lvl:1;
+
+ /* Jack detect threashold levels, see datasheet for values */
+ unsigned int jd_scthr:2;
+ unsigned int jd_thr:2;
+};
+
+#endif
generated by cgit v1.2.3 (git 2.0.4) at 2024-11-14 19:05:51 +0000