From 4e01f54bfd3f423db8fd6c91c4f0471f18aa0c50 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 16 Apr 2009 08:53:34 +0200
Subject: ALSA: hda - Add Creative CA0110-IBG support

Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode.
In the HD-audio mode, no multiple streams are supported by just it
behaves like a normal HD-audio device.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/Kconfig        |  13 +
 sound/pci/hda/Makefile       |   4 +
 sound/pci/hda/hda_codec.c    |   1 +
 sound/pci/hda/hda_intel.c    |   5 +
 sound/pci/hda/patch_ca0110.c | 574 +++++++++++++++++++++++++++++++++++++++++++
 5 files changed, 597 insertions(+)
 create mode 100644 sound/pci/hda/patch_ca0110.c

(limited to 'sound/pci')

diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index eb2a19b894a..c710150d506 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -139,6 +139,19 @@ config SND_HDA_CODEC_CONEXANT
 	  snd-hda-codec-conexant.
 	  This module is automatically loaded at probing.
 
+config SND_HDA_CODEC_CA0110
+	bool "Build Creative CA0110-IBG codec support"
+	depends on SND_HDA_INTEL
+	default y
+	help
+	  Say Y here to include Creative CA0110-IBG codec support in
+	  snd-hda-intel driver, found on some Creative X-Fi cards.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-ca0110.
+	  This module is automatically loaded at probing.
+
 config SND_HDA_CODEC_CMEDIA
 	bool "Build C-Media HD-audio codec support"
 	default y
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index 50f9d096725..e3081d4586c 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -13,6 +13,7 @@ snd-hda-codec-analog-objs :=	patch_analog.o
 snd-hda-codec-idt-objs :=	patch_sigmatel.o
 snd-hda-codec-si3054-objs :=	patch_si3054.o
 snd-hda-codec-atihdmi-objs :=	patch_atihdmi.o
+snd-hda-codec-ca0110-objs :=	patch_ca0110.o
 snd-hda-codec-conexant-objs :=	patch_conexant.o
 snd-hda-codec-via-objs :=	patch_via.o
 snd-hda-codec-nvhdmi-objs :=	patch_nvhdmi.o
@@ -40,6 +41,9 @@ endif
 ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o
 endif
+ifdef CONFIG_SND_HDA_CODEC_CA0110
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
+endif
 ifdef CONFIG_SND_HDA_CODEC_CONEXANT
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-conexant.o
 endif
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index fd6e6f337d1..37f24ce7c3a 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -48,6 +48,7 @@ static struct hda_vendor_id hda_vendor_ids[] = {
 	{ 0x1095, "Silicon Image" },
 	{ 0x10de, "Nvidia" },
 	{ 0x10ec, "Realtek" },
+	{ 0x1102, "Creative" },
 	{ 0x1106, "VIA" },
 	{ 0x111d, "IDT" },
 	{ 0x11c1, "LSI" },
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 21e99cfa8c4..21a3092fad0 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2513,6 +2513,11 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA },
 	/* Teradici */
 	{ PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
+	/* Creative X-Fi (CA0110-IBG) */
+	{ PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_ANY_ID),
+	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
+	  .class_mask = 0xffffff,
+	  .driver_data = AZX_DRIVER_GENERIC },
 	/* AMD Generic, PCI class code and Vendor ID for HD Audio */
 	{ PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
new file mode 100644
index 00000000000..7ec41daa3f0
--- /dev/null
+++ b/sound/pci/hda/patch_ca0110.c
@@ -0,0 +1,574 @@
+/*
+ * HD audio interface patch for Creative X-Fi CA0110-IBG chip
+ *
+ * Copyright (c) 2008 Takashi Iwai <tiwai@suse.de>
+ *
+ *  This driver is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; either version 2 of the License, or
+ *  (at your option) any later version.
+ *
+ *  This driver is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+/*
+ */
+
+struct ca0110_spec {
+	struct auto_pin_cfg autocfg;
+	struct hda_multi_out multiout;
+	hda_nid_t out_pins[AUTO_CFG_MAX_OUTS];
+	hda_nid_t dacs[AUTO_CFG_MAX_OUTS];
+	hda_nid_t hp_dac;
+	hda_nid_t input_pins[AUTO_PIN_LAST];
+	hda_nid_t adcs[AUTO_PIN_LAST];
+	hda_nid_t dig_out;
+	hda_nid_t dig_in;
+	unsigned int num_inputs;
+	const char *input_labels[AUTO_PIN_LAST];
+	struct hda_pcm pcm_rec[2];	/* PCM information */
+};
+
+/*
+ * PCM callbacks
+ */
+static int ca0110_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				    struct hda_codec *codec,
+				    struct snd_pcm_substream *substream)
+{
+	struct ca0110_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+					     hinfo);
+}
+
+static int ca0110_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       unsigned int stream_tag,
+				       unsigned int format,
+				       struct snd_pcm_substream *substream)
+{
+	struct ca0110_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+						stream_tag, format, substream);
+}
+
+static int ca0110_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       struct snd_pcm_substream *substream)
+{
+	struct ca0110_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int ca0110_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					struct snd_pcm_substream *substream)
+{
+	struct ca0110_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int ca0110_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+					 struct hda_codec *codec,
+					 struct snd_pcm_substream *substream)
+{
+	struct ca0110_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int ca0110_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+					   struct hda_codec *codec,
+					   unsigned int stream_tag,
+					   unsigned int format,
+					   struct snd_pcm_substream *substream)
+{
+	struct ca0110_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+					     format, substream);
+}
+
+/*
+ * Analog capture
+ */
+static int ca0110_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      unsigned int stream_tag,
+				      unsigned int format,
+				      struct snd_pcm_substream *substream)
+{
+	struct ca0110_spec *spec = codec->spec;
+
+	snd_hda_codec_setup_stream(codec, spec->adcs[substream->number],
+				   stream_tag, 0, format);
+	return 0;
+}
+
+static int ca0110_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      struct snd_pcm_substream *substream)
+{
+	struct ca0110_spec *spec = codec->spec;
+
+	snd_hda_codec_cleanup_stream(codec, spec->adcs[substream->number]);
+	return 0;
+}
+
+/*
+ */
+
+static char *dirstr[2] = { "Playback", "Capture" };
+
+static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
+		       int chan, int dir)
+{
+	char namestr[44];
+	int type = dir ? HDA_INPUT : HDA_OUTPUT;
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type);
+	sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]);
+	return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+}
+
+static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx,
+		       int chan, int dir)
+{
+	char namestr[44];
+	int type = dir ? HDA_INPUT : HDA_OUTPUT;
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type);
+	sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]);
+	return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec));
+}
+
+#define add_out_switch(codec, nid, pfx)	_add_switch(codec, nid, pfx, 3, 0)
+#define add_out_volume(codec, nid, pfx)	_add_volume(codec, nid, pfx, 3, 0)
+#define add_in_switch(codec, nid, pfx)	_add_switch(codec, nid, pfx, 3, 1)
+#define add_in_volume(codec, nid, pfx)	_add_volume(codec, nid, pfx, 3, 1)
+#define add_mono_switch(codec, nid, pfx, chan) \
+	_add_switch(codec, nid, pfx, chan, 0)
+#define add_mono_volume(codec, nid, pfx, chan) \
+	_add_volume(codec, nid, pfx, chan, 0)
+
+static int ca0110_build_controls(struct hda_codec *codec)
+{
+	struct ca0110_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	static char *prefix[AUTO_CFG_MAX_OUTS] = {
+		"Front", "Surround", NULL, "Side", "Multi"
+	};
+	hda_nid_t mutenid;
+	int i, err;
+
+	for (i = 0; i < spec->multiout.num_dacs; i++) {
+		if (get_wcaps(codec, spec->out_pins[i]) & AC_WCAP_OUT_AMP)
+			mutenid = spec->out_pins[i];
+		else
+			mutenid = spec->multiout.dac_nids[i];
+		if (!prefix[i]) {
+			err = add_mono_switch(codec, mutenid,
+					      "Center", 1);
+			if (err < 0)
+				return err;
+			err = add_mono_switch(codec, mutenid,
+					      "LFE", 1);
+			if (err < 0)
+				return err;
+			err = add_mono_volume(codec, spec->multiout.dac_nids[i],
+					      "Center", 1);
+			if (err < 0)
+				return err;
+			err = add_mono_volume(codec, spec->multiout.dac_nids[i],
+					      "LFE", 1);
+			if (err < 0)
+				return err;
+		} else {
+			err = add_out_switch(codec, mutenid,
+					     prefix[i]);
+			if (err < 0)
+				return err;
+			err = add_out_volume(codec, spec->multiout.dac_nids[i],
+					 prefix[i]);
+			if (err < 0)
+				return err;
+		}
+	}
+	if (cfg->hp_outs) {
+		if (get_wcaps(codec, cfg->hp_pins[0]) & AC_WCAP_OUT_AMP)
+			mutenid = cfg->hp_pins[0];
+		else
+			mutenid = spec->multiout.dac_nids[i];
+
+		err = add_out_switch(codec, mutenid, "Headphone");
+		if (err < 0)
+			return err;
+		if (spec->hp_dac) {
+			err = add_out_volume(codec, spec->hp_dac, "Headphone");
+			if (err < 0)
+				return err;
+		}
+	}
+	for (i = 0; i < spec->num_inputs; i++) {
+		const char *label = spec->input_labels[i];
+		if (get_wcaps(codec, spec->input_pins[i]) & AC_WCAP_IN_AMP)
+			mutenid = spec->input_pins[i];
+		else
+			mutenid = spec->adcs[i];
+		err = add_in_switch(codec, mutenid, label);
+		if (err < 0)
+			return err;
+		err = add_in_volume(codec, spec->adcs[i], label);
+		if (err < 0)
+			return err;
+	}
+
+	if (spec->dig_out) {
+		err = snd_hda_create_spdif_out_ctls(codec, spec->dig_out);
+		if (err < 0)
+			return err;
+		err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
+		if (err < 0)
+			return err;
+		spec->multiout.share_spdif = 1;
+	}
+	if (spec->dig_in) {
+		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+		if (err < 0)
+			return err;
+		err = add_in_volume(codec, spec->dig_in, "IEC958");
+	}
+	return 0;
+}
+
+/*
+ */
+static struct hda_pcm_stream ca0110_pcm_analog_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 8,
+	.ops = {
+		.open = ca0110_playback_pcm_open,
+		.prepare = ca0110_playback_pcm_prepare,
+		.cleanup = ca0110_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream ca0110_pcm_analog_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.prepare = ca0110_capture_pcm_prepare,
+		.cleanup = ca0110_capture_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream ca0110_pcm_digital_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.open = ca0110_dig_playback_pcm_open,
+		.close = ca0110_dig_playback_pcm_close,
+		.prepare = ca0110_dig_playback_pcm_prepare
+	},
+};
+
+static struct hda_pcm_stream ca0110_pcm_digital_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static int ca0110_build_pcms(struct hda_codec *codec)
+{
+	struct ca0110_spec *spec = codec->spec;
+	struct hda_pcm *info = spec->pcm_rec;
+
+	codec->pcm_info = info;
+	codec->num_pcms = 0;
+
+	info->name = "CA0110 Analog";
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0110_pcm_analog_playback;
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0];
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+		spec->multiout.num_dacs * 2;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0110_pcm_analog_capture;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
+	codec->num_pcms++;
+
+	if (!spec->dig_out && !spec->dig_in)
+		return 0;
+
+	info++;
+	info->name = "CA0110 Digital";
+	info->pcm_type = HDA_PCM_TYPE_SPDIF;
+	if (spec->dig_out) {
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+			ca0110_pcm_digital_playback;
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dig_out;
+	}
+	if (spec->dig_in) {
+		info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+			ca0110_pcm_digital_capture;
+		info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+	}
+	codec->num_pcms++;
+
+	return 0;
+}
+
+static void init_output(struct hda_codec *codec, hda_nid_t pin, hda_nid_t dac)
+{
+	if (pin) {
+		snd_hda_codec_write(codec, pin, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+		if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
+			snd_hda_codec_write(codec, pin, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE,
+					    AMP_OUT_UNMUTE);
+	}
+	if (dac)
+		snd_hda_codec_write(codec, dac, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO);
+}
+
+static void init_input(struct hda_codec *codec, hda_nid_t pin, hda_nid_t adc)
+{
+	if (pin) {
+		snd_hda_codec_write(codec, pin, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80);
+		if (get_wcaps(codec, pin) & AC_WCAP_IN_AMP)
+			snd_hda_codec_write(codec, pin, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE,
+					    AMP_IN_UNMUTE(0));
+	}
+	if (adc)
+		snd_hda_codec_write(codec, adc, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    AMP_IN_UNMUTE(0));
+}
+
+static int ca0110_init(struct hda_codec *codec)
+{
+	struct ca0110_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+	for (i = 0; i < spec->multiout.num_dacs; i++)
+		init_output(codec, spec->out_pins[i],
+			    spec->multiout.dac_nids[i]);
+	init_output(codec, cfg->hp_pins[0], spec->hp_dac);
+	init_output(codec, cfg->dig_out_pins[0], spec->dig_out);
+
+	for (i = 0; i < spec->num_inputs; i++)
+		init_input(codec, spec->input_pins[i], spec->adcs[i]);
+	init_input(codec, cfg->dig_in_pin, spec->dig_in);
+	return 0;
+}
+
+static void ca0110_free(struct hda_codec *codec)
+{
+	kfree(codec->spec);
+}
+
+static struct hda_codec_ops ca0110_patch_ops = {
+	.build_controls = ca0110_build_controls,
+	.build_pcms = ca0110_build_pcms,
+	.init = ca0110_init,
+	.free = ca0110_free,
+};
+
+
+static void parse_line_outs(struct hda_codec *codec)
+{
+	struct ca0110_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i, n;
+	unsigned int def_conf;
+	hda_nid_t nid;
+
+	n = 0;
+	for (i = 0; i < cfg->line_outs; i++) {
+		nid = cfg->line_out_pins[i];
+		def_conf = snd_hda_codec_read(codec, nid, 0,
+					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		if (!def_conf)
+			continue; /* invalid pin */
+		if (snd_hda_get_connections(codec, nid, &spec->dacs[i], 1) != 1)
+			continue;
+		spec->out_pins[n++] = nid;
+	}
+	spec->multiout.dac_nids = spec->dacs;
+	spec->multiout.num_dacs = n;
+}
+
+static void parse_hp_out(struct hda_codec *codec)
+{
+	struct ca0110_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+	unsigned int def_conf;
+	hda_nid_t nid, dac;
+
+	if (!cfg->hp_outs)
+		return;
+	nid = cfg->hp_pins[0];
+	def_conf = snd_hda_codec_read(codec, nid, 0,
+				      AC_VERB_GET_CONFIG_DEFAULT, 0);
+	if (!def_conf) {
+		cfg->hp_outs = 0;
+		return;
+	}
+	if (snd_hda_get_connections(codec, nid, &dac, 1) != 1)
+		return;
+
+	for (i = 0; i < cfg->line_outs; i++)
+		if (dac == spec->dacs[i])
+			break;
+	if (i >= cfg->line_outs) {
+		spec->hp_dac = dac;
+		spec->multiout.hp_nid = dac;
+	}
+}
+
+static void parse_input(struct hda_codec *codec)
+{
+	struct ca0110_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t nid, pin;
+	int n, i, j;
+
+	n = 0;
+	nid = codec->start_nid;
+	for (i = 0; i < codec->num_nodes; i++, nid++) {
+		unsigned int wcaps = get_wcaps(codec, nid);
+		unsigned int type = (wcaps & AC_WCAP_TYPE) >>
+			AC_WCAP_TYPE_SHIFT;
+		if (type != AC_WID_AUD_IN)
+			continue;
+		if (snd_hda_get_connections(codec, nid, &pin, 1) != 1)
+			continue;
+		if (pin == cfg->dig_in_pin) {
+			spec->dig_in = nid;
+			continue;
+		}
+		for (j = 0; j < AUTO_PIN_LAST; j++)
+			if (cfg->input_pins[j] == pin)
+				break;
+		if (j >= AUTO_PIN_LAST)
+			continue;
+		spec->input_pins[n] = pin;
+		spec->input_labels[n] = auto_pin_cfg_labels[j];
+		spec->adcs[n] = nid;
+		n++;
+	}
+	spec->num_inputs = n;
+}
+
+static void parse_digital(struct hda_codec *codec)
+{
+	struct ca0110_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+
+	if (cfg->dig_outs &&
+	    snd_hda_get_connections(codec, cfg->dig_out_pins[0],
+				    &spec->dig_out, 1) == 1)
+		spec->multiout.dig_out_nid = cfg->dig_out_pins[0];
+}
+
+static int ca0110_parse_auto_config(struct hda_codec *codec)
+{
+	struct ca0110_spec *spec = codec->spec;
+	int err;
+
+	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+	if (err < 0)
+		return err;
+
+	parse_line_outs(codec);
+	parse_hp_out(codec);
+	parse_digital(codec);
+	parse_input(codec);
+	return 0;
+}
+
+
+int patch_ca0110(struct hda_codec *codec)
+{
+	struct ca0110_spec *spec;
+	int err;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+	codec->spec = spec;
+
+	codec->bus->needs_damn_long_delay = 1;
+
+	err = ca0110_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
+
+	codec->patch_ops = ca0110_patch_ops;
+
+	return 0;
+
+ error:
+	kfree(codec->spec);
+	codec->spec = NULL;
+	return err;
+}
+
+
+/*
+ * patch entries
+ */
+static struct hda_codec_preset snd_hda_preset_ca0110[] = {
+	{ .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 },
+	{ .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 },
+	{ .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 },
+	{} /* terminator */
+};
+
+MODULE_ALIAS("snd-hda-codec-id:1102000a");
+MODULE_ALIAS("snd-hda-codec-id:1102000b");
+MODULE_ALIAS("snd-hda-codec-id:1102000d");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec");
+
+static struct hda_codec_preset_list ca0110_list = {
+	.preset = snd_hda_preset_ca0110,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_ca0110_init(void)
+{
+	return snd_hda_add_codec_preset(&ca0110_list);
+}
+
+static void __exit patch_ca0110_exit(void)
+{
+	snd_hda_delete_codec_preset(&ca0110_list);
+}
+
+module_init(patch_ca0110_init)
+module_exit(patch_ca0110_exit)
-- 
cgit v1.2.3-18-g5258


From 18cb7109d3e83195b605ff2905981020e86f72ca Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 16 Apr 2009 10:22:24 +0200
Subject: ALSA: hda - Check strcpy length

Check the length to copy via strlen() beforehand to avoid the stack
corruption, or use strlcpy() to be safe in HD-audio codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c |  2 ++
 sound/pci/hda/hda_intel.c | 10 ++++++----
 2 files changed, 8 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 37f24ce7c3a..48f0cea7df1 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1431,6 +1431,8 @@ _snd_hda_find_mixer_ctl(struct hda_codec *codec,
 	memset(&id, 0, sizeof(id));
 	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
 	id.index = idx;
+	if (snd_BUG_ON(strlen(name) >= sizeof(id.name)))
+		return NULL;
 	strcpy(id.name, name);
 	return snd_ctl_find_id(codec->bus->card, &id);
 }
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 21a3092fad0..41db5d4da47 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1830,7 +1830,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec,
 			  &pcm);
 	if (err < 0)
 		return err;
-	strcpy(pcm->name, cpcm->name);
+	strlcpy(pcm->name, cpcm->name, sizeof(pcm->name));
 	apcm = kzalloc(sizeof(*apcm), GFP_KERNEL);
 	if (apcm == NULL)
 		return -ENOMEM;
@@ -2358,9 +2358,11 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	}
 
 	strcpy(card->driver, "HDA-Intel");
-	strcpy(card->shortname, driver_short_names[chip->driver_type]);
-	sprintf(card->longname, "%s at 0x%lx irq %i",
-		card->shortname, chip->addr, chip->irq);
+	strlcpy(card->shortname, driver_short_names[chip->driver_type],
+		sizeof(card->shortname));
+	snprintf(card->longname, sizeof(card->longname),
+		 "%s at 0x%lx irq %i",
+		 card->shortname, chip->addr, chip->irq);
 
 	*rchip = chip;
 	return 0;
-- 
cgit v1.2.3-18-g5258


From 67667263674663767ddf4250bab2437a00ee780e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Apr 2009 10:49:25 +0200
Subject: ALSA: hda - Fix channels_max setting for CA0110

Added the missing definition of max channels for CA0110, which resulted
in an error at opening PCM devices.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0110.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 7ec41daa3f0..9398d92f18b 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -309,7 +309,7 @@ static int ca0110_build_pcms(struct hda_codec *codec)
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ca0110_pcm_analog_playback;
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dacs[0];
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
-		spec->multiout.num_dacs * 2;
+		spec->multiout.max_channels;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ca0110_pcm_analog_capture;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adcs[0];
@@ -418,6 +418,7 @@ static void parse_line_outs(struct hda_codec *codec)
 	}
 	spec->multiout.dac_nids = spec->dacs;
 	spec->multiout.num_dacs = n;
+	spec->multiout.max_channels = n * 2;
 }
 
 static void parse_hp_out(struct hda_codec *codec)
-- 
cgit v1.2.3-18-g5258


From 7670dc41b51983b369f9adfb8694a580e7b0cef2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 20 Apr 2009 10:51:11 +0200
Subject: ALSA: hda - Use snd_hda_codec_get_pincfg() in patch_ca0110.c

Use the new function to reduce the access and allow the user setup
via sysfs, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_ca0110.c | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 9398d92f18b..392d108c355 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -408,8 +408,7 @@ static void parse_line_outs(struct hda_codec *codec)
 	n = 0;
 	for (i = 0; i < cfg->line_outs; i++) {
 		nid = cfg->line_out_pins[i];
-		def_conf = snd_hda_codec_read(codec, nid, 0,
-					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
 		if (!def_conf)
 			continue; /* invalid pin */
 		if (snd_hda_get_connections(codec, nid, &spec->dacs[i], 1) != 1)
@@ -432,8 +431,7 @@ static void parse_hp_out(struct hda_codec *codec)
 	if (!cfg->hp_outs)
 		return;
 	nid = cfg->hp_pins[0];
-	def_conf = snd_hda_codec_read(codec, nid, 0,
-				      AC_VERB_GET_CONFIG_DEFAULT, 0);
+	def_conf = snd_hda_codec_get_pincfg(codec, nid);
 	if (!def_conf) {
 		cfg->hp_outs = 0;
 		return;
-- 
cgit v1.2.3-18-g5258


From 92c7c8a7d6e03eb4c0a3c5888e35dbc45f24744c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 24 Mar 2009 07:32:14 +0100
Subject: ALSA: hda - Cache PCM and STREAM parameters queries

Cache quries for PCM and STREAM parameters as well as ampcap and
pincap sharing the hash table.  This will reduce the superfluous
access of the same codec verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 97 +++++++++++++++++++++++++++++------------------
 1 file changed, 61 insertions(+), 36 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a4e5e595211..3d8bf39e6d9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -1053,6 +1053,8 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream);
 /* FIXME: more better hash key? */
 #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24))
 #define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24))
+#define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24))
+#define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24))
 #define INFO_AMP_CAPS	(1<<0)
 #define INFO_AMP_VOL(ch)	(1 << (1 + (ch)))
 
@@ -1143,19 +1145,32 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 }
 EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps);
 
-u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
+static unsigned int
+query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key,
+		unsigned int (*func)(struct hda_codec *, hda_nid_t))
 {
 	struct hda_amp_info *info;
 
-	info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid));
+	info = get_alloc_amp_hash(codec, key);
 	if (!info)
 		return 0;
 	if (!info->head.val) {
-		info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
 		info->head.val |= INFO_AMP_CAPS;
+		info->amp_caps = func(codec, nid);
 	}
 	return info->amp_caps;
 }
+
+static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid)
+{
+	return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+}
+
+u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
+{
+	return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid),
+			       read_pin_cap);
+}
 EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
 
 /*
@@ -2538,6 +2553,41 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 }
 EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
 
+static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int val = 0;
+	if (nid != codec->afg &&
+	    (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD))
+		val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
+	if (!val || val == -1)
+		val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
+	if (!val || val == -1)
+		return 0;
+	return val;
+}
+
+static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid)
+{
+	return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid),
+			       get_pcm_param);
+}
+
+static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
+	if (!streams || streams == -1)
+		streams = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM);
+	if (!streams || streams == -1)
+		return 0;
+	return streams;
+}
+
+static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid)
+{
+	return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid),
+			       get_stream_param);
+}
+
 /**
  * snd_hda_query_supported_pcm - query the supported PCM rates and formats
  * @codec: the HDA codec
@@ -2556,15 +2606,8 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 {
 	unsigned int i, val, wcaps;
 
-	val = 0;
 	wcaps = get_wcaps(codec, nid);
-	if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) {
-		val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
-		if (val == -1)
-			return -EIO;
-	}
-	if (!val)
-		val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
+	val = query_pcm_param(codec, nid);
 
 	if (ratesp) {
 		u32 rates = 0;
@@ -2586,15 +2629,9 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 		u64 formats = 0;
 		unsigned int streams, bps;
 
-		streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
-		if (streams == -1)
+		streams = query_stream_param(codec, nid);
+		if (!streams)
 			return -EIO;
-		if (!streams) {
-			streams = snd_hda_param_read(codec, codec->afg,
-						     AC_PAR_STREAM);
-			if (streams == -1)
-				return -EIO;
-		}
 
 		bps = 0;
 		if (streams & AC_SUPFMT_PCM) {
@@ -2668,17 +2705,9 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
 	int i;
 	unsigned int val = 0, rate, stream;
 
-	if (nid != codec->afg &&
-	    (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) {
-		val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
-		if (val == -1)
-			return 0;
-	}
-	if (!val) {
-		val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
-		if (val == -1)
-			return 0;
-	}
+	val = query_pcm_param(codec, nid);
+	if (!val)
+		return 0;
 
 	rate = format & 0xff00;
 	for (i = 0; i < AC_PAR_PCM_RATE_BITS; i++)
@@ -2690,12 +2719,8 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
 	if (i >= AC_PAR_PCM_RATE_BITS)
 		return 0;
 
-	stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
-	if (stream == -1)
-		return 0;
-	if (!stream && nid != codec->afg)
-		stream = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM);
-	if (!stream || stream == -1)
+	stream = query_stream_param(codec, nid);
+	if (!stream)
 		return 0;
 
 	if (stream & AC_SUPFMT_PCM) {
-- 
cgit v1.2.3-18-g5258


From b613291fb21a2d74eb8323d97fe9aa5d281b306c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 24 Mar 2009 07:36:09 +0100
Subject: ALSA: hda - Retry codec-verbs at errors

The current error-recovery scheme for the codec communication errors
doesn't work always well.  Especially falling back to the
single-command mode causes the fatal problem on many systems.

In this patch, the problematic verb is re-issued again after the error
(even with polling mode) instead of the single-cmd mode.  The
single-cmd mode will be used only when specified via the command
option explicitly, mainly just for testing.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 17 +++++++++++++----
 sound/pci/hda/hda_codec.h |  1 +
 sound/pci/hda/hda_intel.c | 19 ++++++++++---------
 3 files changed, 24 insertions(+), 13 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3d8bf39e6d9..1736ccbebc7 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -174,14 +174,23 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
 				unsigned int verb, unsigned int parm)
 {
 	struct hda_bus *bus = codec->bus;
-	unsigned int res;
+	unsigned int cmd, res;
+	int repeated = 0;
 
-	res = make_codec_cmd(codec, nid, direct, verb, parm);
+	cmd = make_codec_cmd(codec, nid, direct, verb, parm);
 	snd_hda_power_up(codec);
 	mutex_lock(&bus->cmd_mutex);
-	if (!bus->ops.command(bus, res))
+ again:
+	if (!bus->ops.command(bus, cmd)) {
 		res = bus->ops.get_response(bus);
-	else
+		if (res == -1 && bus->rirb_error) {
+			if (repeated++ < 1) {
+				snd_printd(KERN_WARNING "hda_codec: "
+					   "Trying verb 0x%08x again\n", cmd);
+				goto again;
+			}
+		}
+	} else
 		res = (unsigned int)-1;
 	mutex_unlock(&bus->cmd_mutex);
 	snd_hda_power_down(codec);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 2fdecf4b0eb..cd8979c7670 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -623,6 +623,7 @@ struct hda_bus {
 	/* misc op flags */
 	unsigned int needs_damn_long_delay :1;
 	unsigned int shutdown :1;	/* being unloaded */
+	unsigned int rirb_error:1;	/* error in codec communication */
 };
 
 /*
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 30829ee920c..803b72098ed 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -604,6 +604,7 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus)
 		}
 		if (!chip->rirb.cmds) {
 			smp_rmb();
+			bus->rirb_error = 0;
 			return chip->rirb.res; /* the last value */
 		}
 		if (time_after(jiffies, timeout))
@@ -623,8 +624,10 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus)
 		chip->irq = -1;
 		pci_disable_msi(chip->pci);
 		chip->msi = 0;
-		if (azx_acquire_irq(chip, 1) < 0)
+		if (azx_acquire_irq(chip, 1) < 0) {
+			bus->rirb_error = 1;
 			return -1;
+		}
 		goto again;
 	}
 
@@ -644,14 +647,12 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus)
 		return -1;
 	}
 
-	snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, "
-		   "switching to single_cmd mode: last cmd=0x%08x\n",
-		   chip->last_cmd);
-	chip->rirb.rp = azx_readb(chip, RIRBWP);
-	chip->rirb.cmds = 0;
-	/* switch to single_cmd mode */
-	chip->single_cmd = 1;
-	azx_free_cmd_io(chip);
+	snd_printk(KERN_ERR "hda_intel: azx_get_response timeout (ERROR): "
+		   "last cmd=0x%08x\n", chip->last_cmd);
+	spin_lock_irq(&chip->reg_lock);
+	chip->rirb.cmds = 0; /* reset the index */
+	bus->rirb_error = 1;
+	spin_unlock_irq(&chip->reg_lock);
 	return -1;
 }
 
-- 
cgit v1.2.3-18-g5258


From 586be3fcf97eec22fbc0ef6d67e823706aea7167 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 24 Mar 2009 07:43:24 +0100
Subject: ALSA: hda - Add debug prints for Realtek auto-init

Added a couple of debug prints to show the checked id numbers in
alc_subsystem_id().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 82097790f6f..ee92c73df08 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1022,6 +1022,9 @@ static void alc_subsystem_id(struct hda_codec *codec,
 	if (codec->vendor_id == 0x10ec0260)
 		nid = 0x17;
 	ass = snd_hda_codec_get_pincfg(codec, nid);
+	snd_printd("realtek: No valid SSID, "
+		   "checking pincfg 0x%08x for NID 0x%x\n",
+		   nid, ass);
 	if (!(ass & 1) && !(ass & 0x100000))
 		return;
 	if ((ass >> 30) != 1)	/* no physical connection */
@@ -1036,6 +1039,8 @@ static void alc_subsystem_id(struct hda_codec *codec,
 	if (((ass >> 16) & 0xf) != tmp)
 		return;
 do_sku:
+	snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n",
+		   ass & 0xffff, codec->vendor_id);
 	/*
 	 * 0 : override
 	 * 1 :	Swap Jack
-- 
cgit v1.2.3-18-g5258


From a3b48c88f2d5a34c0e25aec0a3dab8069e5a9a72 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 21 Apr 2009 13:37:29 +0200
Subject: ALSA: hda - minor optimization in hda_set_power_state()

Check the target power-state before checking EAPD exception to reduce
unneeded verb executions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b649033a4c8..b91f6ed5cc5 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2348,7 +2348,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 		if (wcaps & AC_WCAP_POWER) {
 			unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
 				AC_WCAP_TYPE_SHIFT;
-			if (wid_type == AC_WID_PIN) {
+			if (power_state == AC_PWRST_D3 &&
+			    wid_type == AC_WID_PIN) {
 				unsigned int pincap;
 				/*
 				 * don't power down the widget if it controls
@@ -2360,7 +2361,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 						nid, 0,
 						AC_VERB_GET_EAPD_BTLENABLE, 0);
 					eapd &= 0x02;
-					if (power_state == AC_PWRST_D3 && eapd)
+					if (eapd)
 						continue;
 				}
 			}
-- 
cgit v1.2.3-18-g5258


From dfed0ef9b3ff9e37903920b6938ed33344ad0b3d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 21 Apr 2009 18:33:12 +0200
Subject: ALSA: hda - Fix a typo in debug print for realtek auto-detection

The NID and ASS numbers were swapped...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a6ec87a5c06..887712046c0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1026,7 +1026,7 @@ static void alc_subsystem_id(struct hda_codec *codec,
 	ass = snd_hda_codec_get_pincfg(codec, nid);
 	snd_printd("realtek: No valid SSID, "
 		   "checking pincfg 0x%08x for NID 0x%x\n",
-		   nid, ass);
+		   ass, nid);
 	if (!(ass & 1) && !(ass & 0x100000))
 		return;
 	if ((ass >> 30) != 1)	/* no physical connection */
-- 
cgit v1.2.3-18-g5258


From 4a79ba34cada6a5a4ee86ed53aa8a73ba1e6fc51 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Apr 2009 16:31:35 +0200
Subject: ALSA: hda - Add amp initialization for realtek auto mode

In the realtek auto-probing mode, the initialization of amp with
some magic COEF or EAPD verbs is applied only when the codec SSID
has valid values to satisfy the realtek's definition.
However, many devices don't provide in that way, thus the device
doesn't work as is.

This patch allows the same initialization code even if the SSID
doesn't pass the bit test.  Also, alc_subsystem_id() is changed
just to check and define the type, so that it's called in the
parser, instead of the initializer.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 239 +++++++++++++++++++++++++-----------------
 1 file changed, 145 insertions(+), 94 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 583603f449b..3a6306302c7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -253,6 +253,15 @@ enum {
 /* for GPIO Poll */
 #define GPIO_MASK	0x03
 
+/* extra amp-initialization sequence types */
+enum {
+	ALC_INIT_NONE,
+	ALC_INIT_DEFAULT,
+	ALC_INIT_GPIO1,
+	ALC_INIT_GPIO2,
+	ALC_INIT_GPIO3,
+};
+
 struct alc_spec {
 	/* codec parameterization */
 	struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
@@ -322,6 +331,7 @@ struct alc_spec {
 
 	/* other flags */
 	unsigned int no_analog :1; /* digital I/O only */
+	int init_amp;
 
 	/* for virtual master */
 	hda_nid_t vmaster_nid;
@@ -994,74 +1004,21 @@ static void alc888_coef_init(struct hda_codec *codec)
 				    AC_VERB_SET_PROC_COEF, 0x3030);
 }
 
-/* 32-bit subsystem ID for BIOS loading in HD Audio codec.
- *	31 ~ 16 :	Manufacture ID
- *	15 ~ 8	:	SKU ID
- *	7  ~ 0	:	Assembly ID
- *	port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36
- */
-static void alc_subsystem_id(struct hda_codec *codec,
-			     unsigned int porta, unsigned int porte,
-			     unsigned int portd)
+static void alc_auto_init_amp(struct hda_codec *codec, int type)
 {
-	unsigned int ass, tmp, i;
-	unsigned nid;
-	struct alc_spec *spec = codec->spec;
-
-	ass = codec->subsystem_id & 0xffff;
-	if ((ass != codec->bus->pci->subsystem_device) && (ass & 1))
-		goto do_sku;
-
-	/*
-	 * 31~30	: port conetcivity
-	 * 29~21	: reserve
-	 * 20		: PCBEEP input
-	 * 19~16	: Check sum (15:1)
-	 * 15~1		: Custom
-	 * 0		: override
-	*/
-	nid = 0x1d;
-	if (codec->vendor_id == 0x10ec0260)
-		nid = 0x17;
-	ass = snd_hda_codec_get_pincfg(codec, nid);
-	snd_printd("realtek: No valid SSID, "
-		   "checking pincfg 0x%08x for NID 0x%x\n",
-		   ass, nid);
-	if (!(ass & 1) && !(ass & 0x100000))
-		return;
-	if ((ass >> 30) != 1)	/* no physical connection */
-		return;
+	unsigned int tmp;
 
-	/* check sum */
-	tmp = 0;
-	for (i = 1; i < 16; i++) {
-		if ((ass >> i) & 1)
-			tmp++;
-	}
-	if (((ass >> 16) & 0xf) != tmp)
-		return;
-do_sku:
-	snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n",
-		   ass & 0xffff, codec->vendor_id);
-	/*
-	 * 0 : override
-	 * 1 :	Swap Jack
-	 * 2 : 0 --> Desktop, 1 --> Laptop
-	 * 3~5 : External Amplifier control
-	 * 7~6 : Reserved
-	*/
-	tmp = (ass & 0x38) >> 3;	/* external Amp control */
-	switch (tmp) {
-	case 1:
+	switch (type) {
+	case ALC_INIT_GPIO1:
 		snd_hda_sequence_write(codec, alc_gpio1_init_verbs);
 		break;
-	case 3:
+	case ALC_INIT_GPIO2:
 		snd_hda_sequence_write(codec, alc_gpio2_init_verbs);
 		break;
-	case 7:
+	case ALC_INIT_GPIO3:
 		snd_hda_sequence_write(codec, alc_gpio3_init_verbs);
 		break;
-	case 5:	/* set EAPD output high */
+	case ALC_INIT_DEFAULT:
 		switch (codec->vendor_id) {
 		case 0x10ec0260:
 			snd_hda_codec_write(codec, 0x0f, 0,
@@ -1115,7 +1072,7 @@ do_sku:
 					    tmp | 0x2010);
 			break;
 		case 0x10ec0888:
-			/*alc888_coef_init(codec);*/ /* called in alc_init() */
+			alc888_coef_init(codec);
 			break;
 		case 0x10ec0267:
 		case 0x10ec0268:
@@ -1130,7 +1087,104 @@ do_sku:
 					    tmp | 0x3000);
 			break;
 		}
-	default:
+		break;
+	}
+}
+
+static void alc_init_auto_hp(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (!spec->autocfg.hp_pins[0])
+		return;
+
+	if (!spec->autocfg.speaker_pins[0]) {
+		if (spec->autocfg.line_out_pins[0])
+			spec->autocfg.speaker_pins[0] =
+				spec->autocfg.line_out_pins[0];
+		else
+			return;
+	}
+
+	snd_hda_codec_write_cache(codec, spec->autocfg.hp_pins[0], 0,
+				  AC_VERB_SET_UNSOLICITED_ENABLE,
+				  AC_USRSP_EN | ALC880_HP_EVENT);
+	spec->unsol_event = alc_sku_unsol_event;
+}
+
+/* check subsystem ID and set up device-specific initialization;
+ * return 1 if initialized, 0 if invalid SSID
+ */
+/* 32-bit subsystem ID for BIOS loading in HD Audio codec.
+ *	31 ~ 16 :	Manufacture ID
+ *	15 ~ 8	:	SKU ID
+ *	7  ~ 0	:	Assembly ID
+ *	port-A --> pin 39/41, port-E --> pin 14/15, port-D --> pin 35/36
+ */
+static int alc_subsystem_id(struct hda_codec *codec,
+			    hda_nid_t porta, hda_nid_t porte,
+			    hda_nid_t portd)
+{
+	unsigned int ass, tmp, i;
+	unsigned nid;
+	struct alc_spec *spec = codec->spec;
+
+	ass = codec->subsystem_id & 0xffff;
+	if ((ass != codec->bus->pci->subsystem_device) && (ass & 1))
+		goto do_sku;
+
+	/* invalid SSID, check the special NID pin defcfg instead */
+	/*
+	 * 31~30	: port conetcivity
+	 * 29~21	: reserve
+	 * 20		: PCBEEP input
+	 * 19~16	: Check sum (15:1)
+	 * 15~1		: Custom
+	 * 0		: override
+	*/
+	nid = 0x1d;
+	if (codec->vendor_id == 0x10ec0260)
+		nid = 0x17;
+	ass = snd_hda_codec_get_pincfg(codec, nid);
+	snd_printd("realtek: No valid SSID, "
+		   "checking pincfg 0x%08x for NID 0x%x\n",
+		   nid, ass);
+	if (!(ass & 1) && !(ass & 0x100000))
+		return 0;
+	if ((ass >> 30) != 1)	/* no physical connection */
+		return 0;
+
+	/* check sum */
+	tmp = 0;
+	for (i = 1; i < 16; i++) {
+		if ((ass >> i) & 1)
+			tmp++;
+	}
+	if (((ass >> 16) & 0xf) != tmp)
+		return 0;
+do_sku:
+	snd_printd("realtek: Enabling init ASM_ID=0x%04x CODEC_ID=%08x\n",
+		   ass & 0xffff, codec->vendor_id);
+	/*
+	 * 0 : override
+	 * 1 :	Swap Jack
+	 * 2 : 0 --> Desktop, 1 --> Laptop
+	 * 3~5 : External Amplifier control
+	 * 7~6 : Reserved
+	*/
+	tmp = (ass & 0x38) >> 3;	/* external Amp control */
+	switch (tmp) {
+	case 1:
+		spec->init_amp = ALC_INIT_GPIO1;
+		break;
+	case 3:
+		spec->init_amp = ALC_INIT_GPIO2;
+		break;
+	case 7:
+		spec->init_amp = ALC_INIT_GPIO3;
+		break;
+	case 5:
+		spec->init_amp = ALC_INIT_DEFAULT;
 		break;
 	}
 
@@ -1138,7 +1192,7 @@ do_sku:
 	 * when the external headphone out jack is plugged"
 	 */
 	if (!(ass & 0x8000))
-		return;
+		return 1;
 	/*
 	 * 10~8 : Jack location
 	 * 12~11: Headphone out -> 00: PortA, 01: PortE, 02: PortD, 03: Resvered
@@ -1146,14 +1200,6 @@ do_sku:
 	 * 15   : 1 --> enable the function "Mute internal speaker
 	 *	        when the external headphone out jack is plugged"
 	 */
-	if (!spec->autocfg.speaker_pins[0]) {
-		if (spec->autocfg.line_out_pins[0])
-			spec->autocfg.speaker_pins[0] =
-				spec->autocfg.line_out_pins[0];
-		else
-			return;
-	}
-
 	if (!spec->autocfg.hp_pins[0]) {
 		tmp = (ass >> 11) & 0x3;	/* HP to chassis */
 		if (tmp == 0)
@@ -1163,23 +1209,23 @@ do_sku:
 		else if (tmp == 2)
 			spec->autocfg.hp_pins[0] = portd;
 		else
-			return;
+			return 1;
 	}
-	if (spec->autocfg.hp_pins[0])
-		snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0,
-			AC_VERB_SET_UNSOLICITED_ENABLE,
-			AC_USRSP_EN | ALC880_HP_EVENT);
 
-#if 0 /* it's broken in some acses -- temporarily disabled */
-	if (spec->autocfg.input_pins[AUTO_PIN_MIC] &&
-		spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC])
-		snd_hda_codec_write(codec,
-			spec->autocfg.input_pins[AUTO_PIN_MIC], 0,
-			AC_VERB_SET_UNSOLICITED_ENABLE,
-			AC_USRSP_EN | ALC880_MIC_EVENT);
-#endif /* disabled */
+	alc_init_auto_hp(codec);
+	return 1;
+}
 
-	spec->unsol_event = alc_sku_unsol_event;
+static void alc_ssid_check(struct hda_codec *codec,
+			   hda_nid_t porta, hda_nid_t porte, hda_nid_t portd)
+{
+	if (!alc_subsystem_id(codec, porta, porte, portd)) {
+		struct alc_spec *spec = codec->spec;
+		snd_printd("realtek: "
+			   "Enable default setup for auto mode as fallback\n");
+		spec->init_amp = ALC_INIT_DEFAULT;
+		alc_init_auto_hp(codec);
+	}
 }
 
 /*
@@ -2923,8 +2969,7 @@ static int alc_init(struct hda_codec *codec)
 	unsigned int i;
 
 	alc_fix_pll(codec);
-	if (codec->vendor_id == 0x10ec0888)
-		alc888_coef_init(codec);
+	alc_auto_init_amp(codec, spec->init_amp);
 
 	for (i = 0; i < spec->num_init_verbs; i++)
 		snd_hda_sequence_write(codec, spec->init_verbs[i]);
@@ -4198,7 +4243,6 @@ static void alc880_auto_init_multi_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int i;
 
-	alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
 	for (i = 0; i < spec->autocfg.line_outs; i++) {
 		hda_nid_t nid = spec->autocfg.line_out_pins[i];
 		int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -4303,6 +4347,8 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux[0];
 
+	alc_ssid_check(codec, 0x15, 0x1b, 0x14);
+
 	return 1;
 }
 
@@ -5678,7 +5724,6 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	hda_nid_t nid;
 
-	alc_subsystem_id(codec, 0x10, 0x15, 0x0f);
 	nid = spec->autocfg.line_out_pins[0];
 	if (nid) {
 		int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -5788,6 +5833,8 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 	spec->num_mux_defs = 1;
 	spec->input_mux = &spec->private_imux[0];
 
+	alc_ssid_check(codec, 0x10, 0x15, 0x0f);
+
 	return 1;
 }
 
@@ -7013,7 +7060,6 @@ static void alc882_auto_init_multi_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int i;
 
-	alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
 	for (i = 0; i <= HDA_SIDE; i++) {
 		hda_nid_t nid = spec->autocfg.line_out_pins[i];
 		int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -9154,7 +9200,6 @@ static void alc883_auto_init_multi_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int i;
 
-	alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
 	for (i = 0; i <= HDA_SIDE; i++) {
 		hda_nid_t nid = spec->autocfg.line_out_pins[i];
 		int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -9317,6 +9362,7 @@ static int patch_alc883(struct hda_codec *codec)
 		if (!spec->capsrc_nids)
 			spec->capsrc_nids = alc883_capsrc_nids;
 		spec->capture_style = CAPT_MIX; /* matrix-style capture */
+		spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */
 		break;
 	case 0x10ec0889:
 		spec->stream_name_analog = "ALC889 Analog";
@@ -10842,6 +10888,8 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
+	alc_ssid_check(codec, 0x15, 0x14, 0x1b);
+
 	return 1;
 }
 
@@ -13925,7 +13973,6 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int i;
 
-	alc_subsystem_id(codec, 0x0e, 0x0f, 0x0b);
 	for (i = 0; i < spec->autocfg.line_outs; i++) {
 		hda_nid_t nid = spec->autocfg.line_out_pins[i];
 		int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -14008,6 +14055,8 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
 	set_capture_mixer(spec);
 
+	alc_ssid_check(codec, 0x0e, 0x0f, 0x0b);
+
 	return 1;
 }
 
@@ -14889,7 +14938,6 @@ static void alc861vd_auto_init_multi_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int i;
 
-	alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
 	for (i = 0; i <= HDA_SIDE; i++) {
 		hda_nid_t nid = spec->autocfg.line_out_pins[i];
 		int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -15107,6 +15155,8 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
+	alc_ssid_check(codec, 0x15, 0x1b, 0x14);
+
 	return 1;
 }
 
@@ -16931,7 +16981,6 @@ static void alc662_auto_init_multi_out(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 	int i;
 
-	alc_subsystem_id(codec, 0x15, 0x1b, 0x14);
 	for (i = 0; i <= HDA_SIDE; i++) {
 		hda_nid_t nid = spec->autocfg.line_out_pins[i];
 		int pin_type = get_pin_type(spec->autocfg.line_out_type);
@@ -17028,6 +17077,8 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
+	alc_ssid_check(codec, 0x15, 0x1b, 0x14);
+
 	return 1;
 }
 
-- 
cgit v1.2.3-18-g5258


From 2a2ed0dfc9ec44a899c7d4672f73f2c045099118 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Apr 2009 13:01:26 +0200
Subject: ALSA: hda - Don't enable auto-mute but for speakers in
 patch_realtek.c

Enable auto-muting in model=auto only for devices with HP and speakers.
For devices with HP and line-outs, don't enable the auto-muting.

Also, add a debug print to show the auto-mute feature.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3a6306302c7..96475dc95fb 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1099,13 +1099,16 @@ static void alc_init_auto_hp(struct hda_codec *codec)
 		return;
 
 	if (!spec->autocfg.speaker_pins[0]) {
-		if (spec->autocfg.line_out_pins[0])
+		if (spec->autocfg.line_out_pins[0] &&
+		    spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
 			spec->autocfg.speaker_pins[0] =
 				spec->autocfg.line_out_pins[0];
 		else
 			return;
 	}
 
+	snd_printdd("realtek: Enable HP auto-muting on NID 0x%x\n",
+		    spec->autocfg.hp_pins[0]);
 	snd_hda_codec_write_cache(codec, spec->autocfg.hp_pins[0], 0,
 				  AC_VERB_SET_UNSOLICITED_ENABLE,
 				  AC_USRSP_EN | ALC880_HP_EVENT);
-- 
cgit v1.2.3-18-g5258


From cb6605c1e4d2a2eaffdde433fbfe1567ca688458 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Apr 2009 13:03:19 +0200
Subject: ALSA: hda - Fix a typo in patch_realtek.c again

The commmit dfed0ef9b3ff9e37903920b6938ed33344ad0b3d was reverted
accidentally by the merge of auto-detection fix patch.

Fixed again now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 96475dc95fb..3e7207b927c 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1151,7 +1151,7 @@ static int alc_subsystem_id(struct hda_codec *codec,
 	ass = snd_hda_codec_get_pincfg(codec, nid);
 	snd_printd("realtek: No valid SSID, "
 		   "checking pincfg 0x%08x for NID 0x%x\n",
-		   nid, ass);
+		   ass, nid);
 	if (!(ass & 1) && !(ass & 0x100000))
 		return 0;
 	if ((ass >> 30) != 1)	/* no physical connection */
-- 
cgit v1.2.3-18-g5258


From 514bf54cd8c7f172816d3c003a6d022e9165a29b Mon Sep 17 00:00:00 2001
From: James Gardiner <renidragsemaj@yahoo.com>
Date: Sun, 3 May 2009 04:00:44 -0400
Subject: ALSA: hda - Addition for HP dv4-1222nr laptop support

Signed-off-by: James Gardiner <renidragsemaj@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 44 ++++++++++++++++++++++++++++++++++++------
 1 file changed, 38 insertions(+), 6 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 917bc5d3ac2..76487de33c8 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -100,6 +100,7 @@ enum {
 	STAC_HP_M4,
 	STAC_HP_DV5,
 	STAC_HP_HDX,
+	STAC_HP_DV4_1222NR,
 	STAC_92HD71BXX_MODELS
 };
 
@@ -1836,6 +1837,7 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
 	[STAC_HP_M4]		= NULL,
 	[STAC_HP_DV5]		= NULL,
 	[STAC_HP_HDX]           = NULL,
+	[STAC_HP_DV4_1222NR]	= NULL,
 };
 
 static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
@@ -1847,6 +1849,7 @@ static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
 	[STAC_HP_M4] = "hp-m4",
 	[STAC_HP_DV5] = "hp-dv5",
 	[STAC_HP_HDX] = "hp-hdx",
+	[STAC_HP_DV4_1222NR] = "hp-dv4-1222nr",
 };
 
 static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
@@ -1855,6 +1858,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "DFI LanParty", STAC_92HD71BXX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
 		      "DFI LanParty", STAC_92HD71BXX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fb,
+		      "HP dv4-1222nr", STAC_HP_DV4_1222NR),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080,
 		      "HP", STAC_HP_DV5),
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
@@ -4520,27 +4525,38 @@ static int stac92xx_resume(struct hda_codec *codec)
 	return 0;
 }
 
-
 /*
- * using power check for controlling mute led of HP HDX notebooks
+ * using power check for controlling mute led of HP notebooks
  * check for mute state only on Speakers (nid = 0x10)
  *
  * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise
  * the LED is NOT working properly !
+ *
+ * Changed name to reflect that it now works for any designated
+ * model, not just HP HDX.
  */
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec,
+static int stac92xx_hp_check_power_status(struct hda_codec *codec,
 					      hda_nid_t nid)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	unsigned int gpio_bit = 0; /* gets rid of compiler warning */
+
+	switch (spec->board_config) {
+	case STAC_HP_DV4_1222NR:
+		gpio_bit = 0x01;
+		break;
+	case STAC_HP_HDX:
+		gpio_bit = 0x08;
+	}
 
 	if (nid == 0x10) {
 		if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
 		    HDA_AMP_MUTE)
-			spec->gpio_data &= ~0x08;  /* orange */
+			spec->gpio_data &= ~gpio_bit;  /* orange */
 		else
-			spec->gpio_data |= 0x08;   /* white */
+			spec->gpio_data |= gpio_bit;   /* white */
 
 		stac_gpio_set(codec, spec->gpio_mask,
 			      spec->gpio_dir,
@@ -5219,6 +5235,22 @@ again:
 		spec->num_smuxes = 0;
 		spec->num_dmuxes = 1;
 		break;
+	case STAC_HP_DV4_1222NR:
+		spec->num_dmics = 1;
+		/* I don't know if it needs 1 or 2 smuxes - will wait for
+		 * bug reports to fix if needed
+		 */
+		spec->num_smuxes = 1;
+		spec->num_dmuxes = 1;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+		/* This controls MUTE LED */
+		spec->gpio_mask |= 0x01;
+		spec->gpio_dir  |= 0x01;
+		spec->gpio_data |= 0x01;
+		codec->patch_ops.check_power_status =
+		    stac92xx_hp_check_power_status;
+#endif
+		/* fallthrough */
 	case STAC_HP_DV5:
 		snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010);
 		stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
@@ -5239,7 +5271,7 @@ again:
 
 		/* register check_power_status callback. */
 		codec->patch_ops.check_power_status =
-		    stac92xx_hp_hdx_check_power_status;
+		    stac92xx_hp_check_power_status;
 #endif	
 		break;
 	};
-- 
cgit v1.2.3-18-g5258


From 41d5545d23d2ccf4d34725094dccebd37f15c1c4 Mon Sep 17 00:00:00 2001
From: Kacper Szczesniak <kacper@qwe.pl>
Date: Thu, 7 May 2009 12:47:43 +0200
Subject: ALSA: hda - Add support for MacBook 5.1 (Aluminium)

Signed-off-by: Kacper Szczesniak <kacper@qwe.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 73 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 73 insertions(+)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3e7207b927c..b9495342c92 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -205,6 +205,7 @@ enum {
 	ALC882_ASUS_A7M,
 	ALC885_MACPRO,
 	ALC885_MBP3,
+	ALC885_MB5,
 	ALC885_IMAC24,
 	ALC882_AUTO,
 	ALC882_MODEL_LAST,
@@ -6164,6 +6165,16 @@ static struct hda_input_mux alc882_capture_source = {
 		{ "CD", 0x4 },
 	},
 };
+
+static struct hda_input_mux mb5_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
 /*
  * 2ch mode
  */
@@ -6293,6 +6304,20 @@ static struct snd_kcontrol_new alc885_mbp3_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
 	{ } /* end */
 };
+
+static struct snd_kcontrol_new alc885_mb5_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("Front Playback Switch", 0x0d, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE   ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE  ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x19, 0x00, HDA_INPUT),
+	{ } /* end */
+};
 static struct snd_kcontrol_new alc882_w2jc_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -6520,6 +6545,38 @@ static struct hda_verb alc882_macpro_init_verbs[] = {
 	{ }
 };
 
+/* Macbook 5,1 */
+static struct hda_verb alc885_mb5_init_verbs[] = {
+	/* Front mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* LineOut mixer */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Front Pin: output 0 (0x0d) */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x01},
+	/* HP Pin: output 0 (0x0c) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Front Mic pin: input vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line In pin */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	{ }
+};
+
 /* Macbook Pro rev3 */
 static struct hda_verb alc885_mbp3_init_verbs[] = {
 	/* Front mixer: unmute input/output amp left and right (volume = 0) */
@@ -6864,6 +6921,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
 	[ALC882_ASUS_A7J]	= "asus-a7j",
 	[ALC882_ASUS_A7M]	= "asus-a7m",
 	[ALC885_MACPRO]		= "macpro",
+	[ALC885_MB5]		= "mb5",
 	[ALC885_MBP3]		= "mbp3",
 	[ALC885_IMAC24]		= "imac24",
 	[ALC882_AUTO]		= "auto",
@@ -6944,6 +7002,18 @@ static struct alc_config_preset alc882_presets[] = {
 		.unsol_event = alc885_mbp3_unsol_event,
 		.init_hook = alc885_mbp3_automute,
 	},
+	[ALC885_MB5] = {
+		.mixers = { alc885_mb5_mixer },
+		.init_verbs = { alc885_mb5_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.channel_mode = alc885_mbp_6ch_modes,
+		.num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes),
+		.input_mux = &mb5_capture_source,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+	},
 	[ALC885_MACPRO] = {
 		.mixers = { alc882_macpro_mixer },
 		.init_verbs = { alc882_macpro_init_verbs },
@@ -7249,6 +7319,9 @@ static int patch_alc882(struct hda_codec *codec)
 		case 0x106b3800: /* MacbookPro4,1 - latter revision */
 			board_config = ALC885_MBP3;
 			break;
+		case 0x106b3f00: /* Macbook 5,1 */
+			board_config = ALC885_MB5;
+			break;
 		default:
 			/* ALC889A is handled better as ALC888-compatible */
 			if (codec->revision_id == 0x100101 ||
-- 
cgit v1.2.3-18-g5258


From 9da29271bea5d831d745f3ceb7f6f6b2def13a5b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 7 May 2009 16:31:14 +0200
Subject: ALSA: hda - Fix secondary SPDIF on VT1708S and VT1702 codecs

VIA VT1708S and VT1702 codecs can have two SPDIF outputs.  One of them
should have been handled as the extra digital out, but it's not
properly accessed.

This patch fixes the handling of the secondary SPDIF on these codecs
with the slave dig-out as found in patch_sigmatel.c.  This makes the
use of such a device easier (for normal users).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_via.c | 111 +++++++++++++++++++++-------------------------
 1 file changed, 51 insertions(+), 60 deletions(-)

(limited to 'sound/pci')

diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index b25a5cc637d..8e004fb6961 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -205,7 +205,7 @@ struct via_spec {
 
 	/* playback */
 	struct hda_multi_out multiout;
-	hda_nid_t extra_dig_out_nid;
+	hda_nid_t slave_dig_outs[2];
 
 	/* capture */
 	unsigned int num_adc_nids;
@@ -731,21 +731,6 @@ static int via_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
 	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
 }
 
-/* setup SPDIF output stream */
-static void setup_dig_playback_stream(struct hda_codec *codec, hda_nid_t nid,
-				 unsigned int stream_tag, unsigned int format)
-{
-	/* turn off SPDIF once; otherwise the IEC958 bits won't be updated */
-	if (codec->spdif_ctls & AC_DIG1_ENABLE)
-		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
-				    codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff);
-	snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format);
-	/* turn on again (if needed) */
-	if (codec->spdif_ctls & AC_DIG1_ENABLE)
-		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
-				    codec->spdif_ctls & 0xff);
-}
-
 static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					struct hda_codec *codec,
 					unsigned int stream_tag,
@@ -753,19 +738,16 @@ static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					struct snd_pcm_substream *substream)
 {
 	struct via_spec *spec = codec->spec;
-	hda_nid_t nid;
-
-	/* 1st or 2nd S/PDIF */
-	if (substream->number == 0)
-		nid = spec->multiout.dig_out_nid;
-	else if (substream->number == 1)
-		nid = spec->extra_dig_out_nid;
-	else
-		return -1;
+	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout,
+					     stream_tag, format, substream);
+}
 
-	mutex_lock(&codec->spdif_mutex);
-	setup_dig_playback_stream(codec, nid, stream_tag, format);
-	mutex_unlock(&codec->spdif_mutex);
+static int via_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					struct snd_pcm_substream *substream)
+{
+	struct via_spec *spec = codec->spec;
+	snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
 	return 0;
 }
 
@@ -842,7 +824,8 @@ static struct hda_pcm_stream vt1708_pcm_digital_playback = {
 	.ops = {
 		.open = via_dig_playback_pcm_open,
 		.close = via_dig_playback_pcm_close,
-		.prepare = via_dig_playback_pcm_prepare
+		.prepare = via_dig_playback_pcm_prepare,
+		.cleanup = via_dig_playback_pcm_cleanup
 	},
 };
 
@@ -874,13 +857,6 @@ static int via_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 		spec->multiout.share_spdif = 1;
-
-		if (spec->extra_dig_out_nid) {
-			err = snd_hda_create_spdif_out_ctls(codec,
-						    spec->extra_dig_out_nid);
-			if (err < 0)
-				return err;
-		}
 	}
 	if (spec->dig_in_nid) {
 		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
@@ -1013,10 +989,6 @@ static void via_unsol_event(struct hda_codec *codec,
 		via_gpio_control(codec);
 }
 
-static hda_nid_t slave_dig_outs[] = {
-	0,
-};
-
 static int via_init(struct hda_codec *codec)
 {
 	str