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-rw-r--r--include/sound/ac97_codec.h9
-rw-r--r--include/sound/asound.h2
-rw-r--r--include/sound/core.h51
-rw-r--r--include/sound/info.h4
-rw-r--r--include/sound/memalloc.h6
-rw-r--r--include/sound/pcm.h25
-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/tlv.h14
-rw-r--r--include/sound/uda1380.h22
-rw-r--r--include/sound/version.h2
-rw-r--r--include/sound/wm8993.h44
14 files changed, 298 insertions, 63 deletions
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 251fc1cd500..3dae3f799b9 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/asound.h b/include/sound/asound.h
index 82aed3f4753..1f57bb92eb5 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -138,7 +138,7 @@ struct snd_hwdep_dsp_image {
* *
*****************************************************************************/
-#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 9)
+#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 10)
typedef unsigned long snd_pcm_uframes_t;
typedef signed long snd_pcm_sframes_t;
diff --git a/include/sound/core.h b/include/sound/core.h
index 309cb9659a0..a61499c22b0 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -93,15 +93,6 @@ struct snd_device {
#define snd_device(n) list_entry(n, struct snd_device, list)
-/* monitor files for graceful shutdown (hotplug) */
-
-struct snd_monitor_file {
- struct file *file;
- const struct file_operations *disconnected_f_op;
- struct list_head shutdown_list; /* still need to shutdown */
- struct list_head list; /* link of monitor files */
-};
-
/* main structure for soundcard */
struct snd_card {
@@ -311,9 +302,7 @@ int snd_component_add(struct snd_card *card, const char *component);
int snd_card_file_add(struct snd_card *card, struct file *file);
int snd_card_file_remove(struct snd_card *card, struct file *file);
-#ifndef snd_card_set_dev
#define snd_card_set_dev(card, devptr) ((card)->dev = (devptr))
-#endif
/* device.c */
@@ -340,18 +329,17 @@ unsigned int snd_dma_pointer(unsigned long dma, unsigned int size);
struct resource;
void release_and_free_resource(struct resource *res);
-#ifdef CONFIG_SND_VERBOSE_PRINTK
-void snd_verbose_printk(const char *file, int line, const char *format, ...)
- __attribute__ ((format (printf, 3, 4)));
-#endif
-#if defined(CONFIG_SND_DEBUG) && defined(CONFIG_SND_VERBOSE_PRINTK)
-void snd_verbose_printd(const char *file, int line, const char *format, ...)
- __attribute__ ((format (printf, 3, 4)));
-#endif
-
/* --- */
-#ifdef CONFIG_SND_VERBOSE_PRINTK
+#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK)
+void __snd_printk(unsigned int level, const char *file, int line,
+ const char *format, ...)
+ __attribute__ ((format (printf, 4, 5)));
+#else
+#define __snd_printk(level, file, line, format, args...) \
+ printk(format, ##args)
+#endif
+
/**
* snd_printk - printk wrapper
* @fmt: format string
@@ -360,15 +348,9 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
* when configured with CONFIG_SND_VERBOSE_PRINTK.
*/
#define snd_printk(fmt, args...) \
- snd_verbose_printk(__FILE__, __LINE__, fmt ,##args)
-#else
-#define snd_printk(fmt, args...) \
- printk(fmt ,##args)
-#endif
+ __snd_printk(0, __FILE__, __LINE__, fmt, ##args)
#ifdef CONFIG_SND_DEBUG
-
-#ifdef CONFIG_SND_VERBOSE_PRINTK
/**
* snd_printd - debug printk
* @fmt: format string
@@ -377,11 +359,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
* Ignored when CONFIG_SND_DEBUG is not set.
*/
#define snd_printd(fmt, args...) \
- snd_verbose_printd(__FILE__, __LINE__, fmt ,##args)
-#else
-#define snd_printd(fmt, args...) \
- printk(fmt ,##args)
-#endif
+ __snd_printk(1, __FILE__, __LINE__, fmt, ##args)
/**
* snd_BUG - give a BUG warning message and stack trace
@@ -428,9 +406,10 @@ static inline int __snd_bug_on(int cond)
* Works like snd_printk() for debugging purposes.
* Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
*/
-#define snd_printdd(format, args...) snd_printk(format, ##args)
+#define snd_printdd(format, args...) \
+ __snd_printk(2, __FILE__, __LINE__, format, ##args)
#else
-#define snd_printdd(format, args...) /* nothing */
+#define snd_printdd(format, args...) do { } while (0)
#endif
@@ -438,12 +417,10 @@ static inline int __snd_bug_on(int cond)
/* for easier backward-porting */
#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE)
-#ifndef gameport_set_dev_parent
#define gameport_set_dev_parent(gp,xdev) ((gp)->dev.parent = (xdev))
#define gameport_set_port_data(gp,r) ((gp)->port_data = (r))
#define gameport_get_port_data(gp) (gp)->port_data
#endif
-#endif
/* PCI quirk list helper */
struct snd_pci_quirk {
diff --git a/include/sound/info.h b/include/sound/info.h
index 7c2ee1a21b0..112e8949e1a 100644
--- a/include/sound/info.h
+++ b/include/sound/info.h
@@ -110,13 +110,13 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer);
static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {}
#endif
-int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) \
+int snd_iprintf(struct snd_info_buffer *buffer, const char *fmt, ...) \
__attribute__ ((format (printf, 2, 3)));
int snd_info_init(void);
int snd_info_done(void);
int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len);
-char *snd_info_get_str(char *dest, char *src, int len);
+const char *snd_info_get_str(char *dest, const char *src, int len);
struct snd_info_entry *snd_info_create_module_entry(struct module *module,
const char *name,
struct snd_info_entry *parent);
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index 7ccce94a525..c4250621264 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -47,7 +47,11 @@ struct snd_dma_device {
#define SNDRV_DMA_TYPE_UNKNOWN 0 /* not defined */
#define SNDRV_DMA_TYPE_CONTINUOUS 1 /* continuous no-DMA memory */
#define SNDRV_DMA_TYPE_DEV 2 /* generic device continuous */
+#ifdef CONFIG_SND_DMA_SGBUF
#define SNDRV_DMA_TYPE_DEV_SG 3 /* generic device SG-buffer */
+#else
+#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */
+#endif
/*
* info for buffer allocation
@@ -60,6 +64,7 @@ struct snd_dma_buffer {
void *private_data; /* private for allocator; don't touch */
};
+#ifdef CONFIG_SND_DMA_SGBUF
/*
* Scatter-Gather generic device pages
*/
@@ -107,6 +112,7 @@ static inline void *snd_sgbuf_get_ptr(struct snd_sg_buf *sgbuf, size_t offset)
{
return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE;
}
+#endif /* CONFIG_SND_DMA_SGBUF */
/* allocate/release a buffer */
int snd_dma_alloc_pages(int type, struct device *dev, size_t size,
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 23893523dc8..de6d981de5d 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -902,6 +902,7 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm,
int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size);
int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream);
+#ifdef CONFIG_SND_DMA_SGBUF
/*
* SG-buffer handling
*/
@@ -927,6 +928,28 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream,
unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
unsigned int ofs, unsigned int size);
+#else /* !SND_DMA_SGBUF */
+/*
+ * fake using a continuous buffer
+ */
+static inline dma_addr_t
+snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs)
+{
+ return substream->runtime->dma_addr + ofs;
+}
+
+static inline void *
+snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs)
+{
+ return substream->runtime->dma_area + ofs;
+}
+
+#define snd_pcm_sgbuf_ops_page NULL
+
+#define snd_pcm_sgbuf_get_chunk_size(subs, ofs, size) (size)
+
+#endif /* SND_DMA_SGBUF */
+
/* handle mmap counter - PCM mmap callback should handle this counter properly */
static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area)
{
@@ -965,4 +988,6 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
#define PCM_RUNTIME_CHECK(sub) snd_BUG_ON(!(sub) || !(sub)->runtime)
+const char *snd_pcm_format_name(snd_pcm_format_t format);
+
#endif /* __SOUND_PCM_H */
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
new file mode 100644
index 00000000000..c0227361a87
--- /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 352d7eee9b6..97ca9af414d 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 ec8a45f9a06..c1410e3191e 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 cf6111d72b1..475cb7ed6be 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/tlv.h b/include/sound/tlv.h
index d136ea2181e..9fd5b19ccf5 100644
--- a/include/sound/tlv.h
+++ b/include/sound/tlv.h
@@ -35,6 +35,8 @@
#define SNDRV_CTL_TLVT_DB_SCALE 1 /* dB scale */
#define SNDRV_CTL_TLVT_DB_LINEAR 2 /* linear volume */
#define SNDRV_CTL_TLVT_DB_RANGE 3 /* dB range container */
+#define SNDRV_CTL_TLVT_DB_MINMAX 4 /* dB scale with min/max */
+#define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5 /* dB scale with min/max with mute */
#define TLV_DB_SCALE_ITEM(min, step, mute) \
SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int), \
@@ -42,6 +44,18 @@
#define DECLARE_TLV_DB_SCALE(name, min, step, mute) \
unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) }
+/* dB scale specified with min/max values instead of step */
+#define TLV_DB_MINMAX_ITEM(min_dB, max_dB) \
+ SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int), \
+ (min_dB), (max_dB)
+#define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) \
+ SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int), \
+ (min_dB), (max_dB)
+#define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \
+ unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) }
+#define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \
+ unsigned int name[] = { TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB) }
+
/* linear volume between min_dB and max_dB (.01dB unit) */
#define TLV_DB_LINEAR_ITEM(min_dB, max_dB) \
SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \
diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h
new file mode 100644
index 00000000000..381319c7000
--- /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/version.h b/include/sound/version.h
index 456f1359e1c..22939142dd2 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
/* include/version.h */
-#define CONFIG_SND_VERSION "1.0.20"
+#define CONFIG_SND_VERSION "1.0.21"
#define CONFIG_SND_DATE ""
diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h
new file mode 100644
index 00000000000..9c661f2f8cd
--- /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 v1.2.3-70-g09d2 (git 2.46.0) at 2024-11-21 09:36:44 +0000