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-rw-r--r--include/linux/tty.h4
-rw-r--r--include/sound/ac97_codec.h6
-rw-r--r--include/sound/soc-dai.h27
-rw-r--r--include/sound/soc.h23
-rw-r--r--include/sound/wm8993.h44
5 files changed, 90 insertions, 14 deletions
diff --git a/include/linux/tty.h b/include/linux/tty.h
index 1488d8c81aa..26af9816391 100644
--- a/include/linux/tty.h
+++ b/include/linux/tty.h
@@ -23,7 +23,7 @@
*/
#define NR_UNIX98_PTY_DEFAULT 4096 /* Default maximum for Unix98 ptys */
#define NR_UNIX98_PTY_MAX (1 << MINORBITS) /* Absolute limit */
-#define NR_LDISCS 19
+#define NR_LDISCS 20
/* line disciplines */
#define N_TTY 0
@@ -47,6 +47,8 @@
#define N_SLCAN 17 /* Serial / USB serial CAN Adaptors */
#define N_PPS 18 /* Pulse per Second */
+#define N_AMSDELTA 19 /* codec control over modem, board specific */
+
/*
* This character is the same as _POSIX_VDISABLE: it cannot be used as
* a c_cc[] character, but indicates that a particular special character
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 251fc1cd500..9b1c0985480 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -642,4 +642,10 @@ 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/soc-dai.h b/include/sound/soc-dai.h
index 352d7eee9b6..25d62ac53fc 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
@@ -116,12 +116,12 @@ 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 {
/*
@@ -179,6 +179,7 @@ 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.h b/include/sound/soc.h
index 4a5846e7247..dbb1702688c 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, \
@@ -372,6 +394,7 @@ struct snd_soc_codec {
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;
unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
void *reg_cache;
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 (git 2.39.1) at 2024-07-12 15:56:09 +0000