---

yaml
---
r: 157617
b: refs/heads/master
c: b9b5cc2
h: refs/heads/master
i:
  157615: a93aa72
v: v3

[refs]

@@ -1,2 +1,2 @@
 ---
-refs/heads/master: c12abc012e18b362204345c323536f228d65c4db
+refs/heads/master: b9b5cc26d0b3a9b361cc56c3a3b0d6f819b9195f

trunk/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

trunk/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 */

trunk/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,

trunk/include/sound/soc-dapm.h

@@ -279,6 +279,7 @@ 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);

trunk/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, \
@@ -191,6 +213,12 @@ 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);
+
+#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 +244,9 @@ 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);
@@ -356,6 +384,8 @@ 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;
 	void *reg_cache;
@@ -369,8 +399,6 @@ 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;

trunk/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 */

trunk/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