---

yaml
---
r: 113711
b: refs/heads/master
c: 4b33c76
h: refs/heads/master
i:
  113709: 3ea19f2
  113707: 476457f
  113703: e61c9b6
  113695: 1a5456a
v: v3

[refs]

@@ -1,2 +1,2 @@
 ---
-refs/heads/master: a7e54e6de3b01d9085202fdbf0110da425f4af38
+refs/heads/master: 4b33c7675d2b0d4a9cb4e38cd73aa1d940f9278d

trunk/include/sound/soc-dapm.h

@@ -240,7 +240,6 @@ int snd_soc_dapm_sys_add(struct device *dev);
 /* dapm audio pin control and status */
 int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin);
 int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin);
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin);
 int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin);
 int snd_soc_dapm_sync(struct snd_soc_codec *codec);
 

trunk/sound/oss/ac97_codec.c

@@ -30,7 +30,7 @@
  **************************************************************************
  *
  * History
- * May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
+ * May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *	Removed non existant WM9700
  *	Added support for WM9705, WM9708, WM9709, WM9710, WM9711
  *	WM9712 and WM9717

trunk/sound/pci/ac97/ac97_patch.c

@@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97)
 }
 
 /*
- * May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
+ * May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *  removed broken wolfson00 patch.
  *  added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717.
  */

trunk/sound/pci/hda/patch_sigmatel.c

@@ -322,8 +322,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = {
 	0x1a, 0x1b
 };
 
-static hda_nid_t stac92hd71bxx_dmux_nids[1] = {
-	0x1c,
+static hda_nid_t stac92hd71bxx_dmux_nids[2] = {
+	0x1c, 0x1d,
 };
 
 static hda_nid_t stac92hd71bxx_smux_nids[2] = {
@@ -861,20 +861,18 @@ static struct hda_verb stac92hd71bxx_core_init[] = {
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* connect headphone jack to dac1 */
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
 	/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 };
 
-#define HD_DISABLE_PORTF 3
+#define HD_DISABLE_PORTF 2
 static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 	/* start of config #1 */
 
 	/* connect port 0f to audio mixer */
 	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
 	/* unmute right and left channels for node 0x0f */
 	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* start of config #2 */
@@ -883,10 +881,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
 	/* connect headphone jack to dac1 */
 	{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* connect port 0d to audio mixer */
-	{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2},
-	/* unmute dac0 input in audio mixer */
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
 	/* unmute right and left channels for nodes 0x0a, 0xd */
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -1107,6 +1101,7 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
 
 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	STAC_INPUT_SOURCE(2),
+	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
 
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
@@ -1119,8 +1114,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
 	*/
 
-	HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x3, HDA_INPUT),
+
+	HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x3, HDA_INPUT),
+
+	HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),
+
+	HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT),
+	HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -1649,7 +1653,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 
 static unsigned int ref92hd71bxx_pin_configs[11] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
-	0x0181302e, 0x01114010, 0x01019020, 0x90a000f0,
+	0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
 	0x90a000f0, 0x01452050, 0x01452050,
 };
 
@@ -3000,7 +3004,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
 
 /* labels for amp mux outputs */
 static const char *stac92xx_amp_labels[3] = {
-	"Front Microphone", "Microphone", "Line In"
+	"Front Microphone", "Microphone", "Line In",
 };
 
 /* create amp out controls mux on capable codecs */
@@ -4327,6 +4331,16 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = {
 #endif
 };
 
+static struct hda_input_mux stac92hd71bxx_dmux = {
+	.num_items = 4,
+	.items = {
+		{ "Analog Inputs", 0x00 },
+		{ "Mixer", 0x01 },
+		{ "Digital Mic 1", 0x02 },
+		{ "Digital Mic 2", 0x03 },
+	}
+};
+
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
@@ -4341,6 +4355,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 	spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
 	spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
 	spec->pin_nids = stac92hd71bxx_pin_nids;
+	memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
+			sizeof(stac92hd71bxx_dmux));
 	spec->board_config = snd_hda_check_board_config(codec,
 							STAC_92HD71BXX_MODELS,
 							stac92hd71bxx_models,
@@ -4392,6 +4408,7 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 		/* no output amps */
 		spec->num_pwrs = 0;
 		spec->mixer = stac92hd71bxx_analog_mixer;
+		spec->dinput_mux = &spec->private_dimux;
 
 		/* disable VSW */
 		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
@@ -4409,12 +4426,13 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 		spec->num_pwrs = 0;
 		/* fallthru */
 	default:
+		spec->dinput_mux = &spec->private_dimux;
 		spec->mixer = stac92hd71bxx_analog_mixer;
 		spec->init = stac92hd71bxx_analog_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 	}
 
-	spec->aloopback_mask = 0x20;
+	spec->aloopback_mask = 0x50;
 	spec->aloopback_shift = 0;
 
 	if (spec->board_config > STAC_92HD71BXX_REF) {
@@ -4456,6 +4474,10 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 	spec->multiout.num_dacs = 1;
 	spec->multiout.hp_nid = 0x11;
 	spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
+	if (spec->dinput_mux)
+		spec->private_dimux.num_items +=
+			spec->num_dmics -
+				(ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1);
 
 	err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
 	if (!err) {

trunk/sound/soc/at91/Kconfig

@@ -8,3 +8,20 @@ config SND_AT91_SOC
 
 config SND_AT91_SOC_SSC
 	tristate
+
+config SND_AT91_SOC_ETI_B1_WM8731
+	tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards"
+	depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1)
+	select SND_AT91_SOC_SSC
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on WM8731-based
+	  Endrelia Technologies Inc ETI-B1 or ETI-C1 boards.
+
+config SND_AT91_SOC_ETI_SLAVE
+	bool "Run codec in slave Mode on Endrelia boards"
+	depends on SND_AT91_SOC_ETI_B1_WM8731
+	default n
+	help
+	  Say Y if you want to run with the AT91 SSC generating the BCLK
+	  and LRC signals on Endrelia boards.

trunk/sound/soc/at91/Makefile

@@ -4,3 +4,8 @@ snd-soc-at91-ssc-objs := at91-ssc.o
 
 obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
 obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
+
+# AT91 Machine Support
+snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o
+
+obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o

trunk/sound/soc/at91/at91-ssc.c

@@ -5,7 +5,7 @@
  *         Endrelia Technologies Inc.
  *
  * Based on pxa2xx Platform drivers by
- * Liam Girdwood <lrg@slimlogic.co.uk>
+ * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
  *
  *  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