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authorTakashi Iwai <tiwai@suse.de>2008-12-25 11:40:25 +0100
committerTakashi Iwai <tiwai@suse.de>2008-12-25 11:40:25 +0100
commit5c8261e44eaebbc91f9fc1bbd3f3167e91a50a57 (patch)
tree6b932687edc73c07e544ccba3f0130fdb257d902 /include
parentfacef8685b3ff95c01c33d9d836401d0dd26211d (diff)
parent472346da9cc4231bec03ff2032e0d5fd4037232c (diff)
Merge branch 'topic/asoc' into to-push
Diffstat (limited to 'include')
-rw-r--r--include/linux/mfd/wm8350/audio.h38
-rw-r--r--include/sound/l3.h18
-rw-r--r--include/sound/s3c24xx_uda134x.h14
-rw-r--r--include/sound/soc-dai.h231
-rw-r--r--include/sound/soc-dapm.h2
-rw-r--r--include/sound/soc.h206
-rw-r--r--include/sound/uda134x.h26
7 files changed, 359 insertions, 176 deletions
diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h
index 217bb22ebb8e..af95a1d2f3a1 100644
--- a/include/linux/mfd/wm8350/audio.h
+++ b/include/linux/mfd/wm8350/audio.h
@@ -1,7 +1,7 @@
/*
* audio.h -- Audio Driver for Wolfson WM8350 PMIC
*
- * Copyright 2007 Wolfson Microelectronics PLC
+ * Copyright 2007, 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 as published by the
@@ -70,9 +70,9 @@
#define WM8350_CODEC_ISEL_0_5 3 /* x0.5 */
#define WM8350_VMID_OFF 0
-#define WM8350_VMID_500K 1
-#define WM8350_VMID_100K 2
-#define WM8350_VMID_10K 3
+#define WM8350_VMID_300K 1
+#define WM8350_VMID_50K 2
+#define WM8350_VMID_5K 3
/*
* R40 (0x28) - Clock Control 1
@@ -591,8 +591,38 @@
#define WM8350_IRQ_CODEC_MICSCD 41
#define WM8350_IRQ_CODEC_MICD 42
+/*
+ * WM8350 Platform data.
+ *
+ * This must be initialised per platform for best audio performance.
+ * Please see WM8350 datasheet for information.
+ */
+struct wm8350_audio_platform_data {
+ int vmid_discharge_msecs; /* VMID --> OFF discharge time */
+ int drain_msecs; /* OFF drain time */
+ int cap_discharge_msecs; /* Cap ON (from OFF) discharge time */
+ int vmid_charge_msecs; /* vmid power up time */
+ u32 vmid_s_curve:2; /* vmid enable s curve speed */
+ u32 dis_out4:2; /* out4 discharge speed */
+ u32 dis_out3:2; /* out3 discharge speed */
+ u32 dis_out2:2; /* out2 discharge speed */
+ u32 dis_out1:2; /* out1 discharge speed */
+ u32 vroi_out4:1; /* out4 tie off */
+ u32 vroi_out3:1; /* out3 tie off */
+ u32 vroi_out2:1; /* out2 tie off */
+ u32 vroi_out1:1; /* out1 tie off */
+ u32 vroi_enable:1; /* enable tie off */
+ u32 codec_current_on:2; /* current level ON */
+ u32 codec_current_standby:2; /* current level STANDBY */
+ u32 codec_current_charge:2; /* codec current @ vmid charge */
+};
+
+struct snd_soc_codec;
+
struct wm8350_codec {
struct platform_device *pdev;
+ struct snd_soc_codec *codec;
+ struct wm8350_audio_platform_data *platform_data;
};
#endif
diff --git a/include/sound/l3.h b/include/sound/l3.h
new file mode 100644
index 000000000000..423a08f0f1b0
--- /dev/null
+++ b/include/sound/l3.h
@@ -0,0 +1,18 @@
+#ifndef _L3_H_
+#define _L3_H_ 1
+
+struct l3_pins {
+ void (*setdat)(int);
+ void (*setclk)(int);
+ void (*setmode)(int);
+ int data_hold;
+ int data_setup;
+ int clock_high;
+ int mode_hold;
+ int mode;
+ int mode_setup;
+};
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
+
+#endif
diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h
new file mode 100644
index 000000000000..33df4cb909d3
--- /dev/null
+++ b/include/sound/s3c24xx_uda134x.h
@@ -0,0 +1,14 @@
+#ifndef _S3C24XX_UDA134X_H_
+#define _S3C24XX_UDA134X_H_ 1
+
+#include <sound/uda134x.h>
+
+struct s3c24xx_uda134x_platform_data {
+ int l3_clk;
+ int l3_mode;
+ int l3_data;
+ void (*power) (int);
+ int model;
+};
+
+#endif
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
new file mode 100644
index 000000000000..24247f763608
--- /dev/null
+++ b/include/sound/soc-dai.h
@@ -0,0 +1,231 @@
+/*
+ * 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 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 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);
+
+ /*
+ * 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;
+
+ /* 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;
+
+ /* 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-dapm.h b/include/sound/soc-dapm.h
index ca699a3017f3..7ee2f70ca42e 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -221,8 +221,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
int num);
/* dapm path setup */
-int __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
- const char *sink_name, const char *control_name, const char *src_name);
int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
void snd_soc_dapm_free(struct snd_soc_device *socdev);
int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 5e0189876afd..f86e455d3828 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -21,8 +21,6 @@
#include <sound/control.h>
#include <sound/ac97_codec.h>
-#define SND_SOC_VERSION "0.13.2"
-
/*
* Convenience kcontrol builders
*/
@@ -145,105 +143,31 @@ enum snd_soc_bias_level {
SND_SOC_BIAS_OFF,
};
-/*
- * Digital Audio Interface (DAI) types
- */
-#define SND_SOC_DAI_AC97 0x1
-#define SND_SOC_DAI_I2S 0x2
-#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;
struct snd_soc_dai_mode;
struct snd_soc_pcm_runtime;
struct snd_soc_dai;
+struct snd_soc_platform;
struct snd_soc_codec;
-struct snd_soc_machine_config;
struct soc_enum;
struct snd_soc_ac97_ops;
-struct snd_soc_clock_info;
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;
+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);
+
/* pcm <-> DAI connect */
void snd_soc_free_pcms(struct snd_soc_device *socdev);
int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
-int snd_soc_register_card(struct snd_soc_device *socdev);
+int snd_soc_init_card(struct snd_soc_device *socdev);
/* set runtime hw params */
int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
@@ -263,27 +187,6 @@ 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
*/
@@ -341,66 +244,14 @@ 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;
struct module *owner;
struct mutex mutex;
+ struct device *dev;
+
+ struct list_head list;
/* callbacks */
int (*set_bias_level)(struct snd_soc_codec *,
@@ -426,6 +277,7 @@ struct snd_soc_codec {
short reg_cache_step;
/* dapm */
+ u32 pop_time;
struct list_head dapm_widgets;
struct list_head dapm_paths;
enum snd_soc_bias_level bias_level;
@@ -435,6 +287,11 @@ struct snd_soc_codec {
/* codec DAI's */
struct snd_soc_dai *dai;
unsigned int num_dai;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_reg;
+ struct dentry *debugfs_pop_time;
+#endif
};
/* codec device */
@@ -448,13 +305,12 @@ struct snd_soc_codec_device {
/* SoC platform interface */
struct snd_soc_platform {
char *name;
+ struct list_head list;
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
- int (*suspend)(struct platform_device *pdev,
- struct snd_soc_dai *dai);
- int (*resume)(struct platform_device *pdev,
- struct snd_soc_dai *dai);
+ int (*suspend)(struct snd_soc_dai *dai);
+ int (*resume)(struct snd_soc_dai *dai);
/* pcm creation and destruction */
int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
@@ -484,9 +340,14 @@ struct snd_soc_dai_link {
struct snd_pcm *pcm;
};
-/* SoC machine */
-struct snd_soc_machine {
+/* SoC card */
+struct snd_soc_card {
char *name;
+ struct device *dev;
+
+ struct list_head list;
+
+ int instantiated;
int (*probe)(struct platform_device *pdev);
int (*remove)(struct platform_device *pdev);
@@ -499,23 +360,26 @@ struct snd_soc_machine {
int (*resume_post)(struct platform_device *pdev);
/* callbacks */
- int (*set_bias_level)(struct snd_soc_machine *,
+ int (*set_bias_level)(struct snd_soc_card *,
enum snd_soc_bias_level level);
/* CPU <--> Codec DAI links */
struct snd_soc_dai_link *dai_link;
int num_links;
+
+ struct snd_soc_device *socdev;
+
+ struct snd_soc_platform *platform;
+ struct delayed_work delayed_work;
+ struct work_struct deferred_resume_work;
};
/* SoC Device - the audio subsystem */
struct snd_soc_device {
struct device *dev;
- struct snd_soc_machine *machine;
- struct snd_soc_platform *platform;
+ struct snd_soc_card *card;
struct snd_soc_codec *codec;
struct snd_soc_codec_device *codec_dev;
- struct delayed_work delayed_work;
- struct work_struct deferred_resume_work;
void *codec_data;
};
@@ -542,4 +406,6 @@ struct soc_enum {
void *dapm;
};
+#include <sound/soc-dai.h>
+
#endif
diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h
new file mode 100644
index 000000000000..475ef8bb7dcd
--- /dev/null
+++ b/include/sound/uda134x.h
@@ -0,0 +1,26 @@
+/*
+ * uda134x.h -- UDA134x ALSA SoC Codec driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * 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 _UDA134X_H
+#define _UDA134X_H
+
+#include <sound/l3.h>
+
+struct uda134x_platform_data {
+ struct l3_pins l3;
+ void (*power) (int);
+ int model;
+#define UDA134X_UDA1340 1
+#define UDA134X_UDA1341 2
+#define UDA134X_UDA1344 3
+};
+
+#endif /* _UDA134X_H */