From d1409ae4cecb4af260759bdfdf88fafca23a9940 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 17 Dec 2009 15:01:31 +0100
Subject: ALSA: hda - Fix NULL dereference in kctl-NID mapping in
 patch_realtek.c

capsrc_nids can be NULL, and adc_nids should be taken as fallback.

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

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 36556b10357..012435212e5 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2517,7 +2517,10 @@ static int alc_build_controls(struct hda_codec *codec)
 	if (!kctl)
 		kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
 	for (i = 0; kctl && i < kctl->count; i++) {
-		err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids,
+		hda_nid_t *nids = spec->capsrc_nids;
+		if (!nids)
+			nids = spec->adc_nids;
+		err = snd_hda_add_nids(codec, kctl, i, nids,
 				       spec->input_mux->num_items);
 		if (err < 0)
 			return err;
-- 
cgit v1.2.3


From 0c2fd1bf4c6cb6095d8b3088d285167e66c12147 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 18 Dec 2009 16:41:39 +0100
Subject: ALSA: hda - Check class to identify Nvidia controller chips

Instead of listing all individual PCI IDs, check the matching with
the PCI class together with the vendor id for Nvidia.
This simplifies the pci id entries.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 30 ++++--------------------------
 1 file changed, 4 insertions(+), 26 deletions(-)

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 9b56f937913..93eaf4fc39b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2694,32 +2694,10 @@ static struct pci_device_id azx_ids[] = {
 	/* ULI M5461 */
 	{ PCI_DEVICE(0x10b9, 0x5461), .driver_data = AZX_DRIVER_ULI },
 	/* NVIDIA MCP */
-	{ PCI_DEVICE(0x10de, 0x026c), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0371), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x03e4), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x03f0), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x044a), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0777), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x07fc), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x07fd), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0ac0), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0be2), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0be3), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0be4), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(PCI_VENDOR_ID_NVIDIA, PCI_ANY_ID),
+	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
+	  .class_mask = 0xffffff,
+	  .driver_data = AZX_DRIVER_NVIDIA },
 	/* Teradici */
 	{ PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
 	/* Creative X-Fi (CA0110-IBG) */
-- 
cgit v1.2.3


From 75d1aeb9d6899b10420d10284e8ea894b2794224 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 22 Dec 2009 11:56:32 +0100
Subject: ALSA: hda - Add Bass Speaker switch for HP dv7

The bass speaker is controlled via GPIO5.

Tested-by: Wael Nasreddine <mla@nasreddine.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 57 ++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 57 insertions(+)

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 0bafea9d510..a4526d00804 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5402,6 +5402,54 @@ static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec,
 		return 0;
 }
 
+/* HP dv7 bass switch - GPIO5 */
+#define stac_hp_bass_gpio_info	snd_ctl_boolean_mono_info
+static int stac_hp_bass_gpio_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+	ucontrol->value.integer.value[0] = !!(spec->gpio_data & 0x20);
+	return 0;
+}
+
+static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct sigmatel_spec *spec = codec->spec;
+	unsigned int gpio_data;
+
+	gpio_data = (spec->gpio_data & ~0x20) |
+		(ucontrol->value.integer.value[0] ? 0x20 : 0);
+	if (gpio_data == spec->gpio_data)
+		return 0;
+	spec->gpio_data = gpio_data;
+	stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data);
+	return 1;
+}
+
+static struct snd_kcontrol_new stac_hp_bass_sw_ctrl = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.info = stac_hp_bass_gpio_info,
+	.get = stac_hp_bass_gpio_get,
+	.put = stac_hp_bass_gpio_put,
+};
+
+static int stac_add_hp_bass_switch(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+
+	if (!stac_control_new(spec, &stac_hp_bass_sw_ctrl,
+			      "Bass Speaker Playback Switch", 0))
+		return -ENOMEM;
+
+	spec->gpio_mask |= 0x20;
+	spec->gpio_dir |= 0x20;
+	spec->gpio_data |= 0x20;
+	return 0;
+}
+
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
@@ -5642,6 +5690,15 @@ again:
 		return err;
 	}
 
+	/* enable bass on HP dv7 */
+	if (spec->board_config == STAC_HP_DV5) {
+		unsigned int cap;
+		cap = snd_hda_param_read(codec, 0x1, AC_PAR_GPIO_CAP);
+		cap &= AC_GPIO_IO_COUNT;
+		if (cap >= 6)
+			stac_add_hp_bass_switch(codec);
+	}
+
 	codec->proc_widget_hook = stac92hd7x_proc_hook;
 
 	return 0;
-- 
cgit v1.2.3


From 21949f00a022e090a7e8bc9a01dfca88273c6146 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 23 Dec 2009 08:31:59 +0100
Subject: ALSA: hda - Fix NID association for capture mixers

Fix the wrong implementation of NID <-> kctl mapping for capture mixers
introduced by the ocmmit 5b0cb1d850c26893b1468b3a519433a1b7a176be.
So far, the driver returns an error at probe.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c     | 26 --------------------------
 sound/pci/hda/hda_local.h     |  2 --
 sound/pci/hda/patch_analog.c  |  3 +--
 sound/pci/hda/patch_cirrus.c  | 12 ++++++++----
 sound/pci/hda/patch_cmedia.c  |  3 +--
 sound/pci/hda/patch_realtek.c |  3 +--
 sound/pci/hda/patch_via.c     |  3 +--
 7 files changed, 12 insertions(+), 40 deletions(-)

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index c848ec0f085..29c90d748c9 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3537,32 +3537,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew)
 }
 EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls);
 
-/**
- * snd_hda_add_nids - assign nids to controls from the array
- * @codec: the HDA codec
- * @kctl: struct snd_kcontrol
- * @index: index to kctl
- * @nids: the array of hda_nid_t
- * @size: count of hda_nid_t items
- *
- * This helper function assigns NIDs in the given array to a control element.
- *
- * Returns 0 if successful, or a negative error code.
- */
-int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl,
-		     unsigned int index, hda_nid_t *nids, unsigned int size)
-{
-	int err;
-
-	for ( ; size > 0; size--, nids++) {
-		err = snd_hda_add_nid(codec, kctl, index, *nids);
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-EXPORT_SYMBOL_HDA(snd_hda_add_nids);
-
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 				unsigned int power_state);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index d505d052972..7cee364976f 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -343,8 +343,6 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec,
                                const struct snd_pci_quirk *tbl);
 int snd_hda_add_new_ctls(struct hda_codec *codec,
 			 struct snd_kcontrol_new *knew);
-int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl,
-		     unsigned int index, hda_nid_t *nids, unsigned int size);
 
 /*
  * unsolicited event handler
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 92b72d4f398..45ee352df32 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -244,8 +244,7 @@ static int ad198x_build_controls(struct hda_codec *codec)
 	if (!kctl)
 		kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
 	for (i = 0; kctl && i < kctl->count; i++) {
-		err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids,
-				       spec->input_mux->num_items);
+		err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]);
 		if (err < 0)
 			return err;
 	}
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 093cfbb55e9..7de782a5b8f 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -753,6 +753,7 @@ static int build_input(struct hda_codec *codec)
 	spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol);
 	for (i = 0; i < 2; i++) {
 		struct snd_kcontrol *kctl;
+		int n;
 		if (!spec->capture_bind[i])
 			return -ENOMEM;
 		kctl = snd_ctl_new1(&cs_capture_ctls[i], codec);
@@ -762,10 +763,13 @@ static int build_input(struct hda_codec *codec)
 		err = snd_hda_ctl_add(codec, 0, kctl);
 		if (err < 0)
 			return err;
-		err = snd_hda_add_nids(codec, kctl, 0, spec->adc_nid,
-				       spec->num_inputs);
-		if (err < 0)
-			return err;
+		for (n = 0; n < AUTO_PIN_LAST; n++) {
+			if (!spec->adc_nid[n])
+				continue;
+			err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[i]);
+			if (err < 0)
+				return err;
+		}
 	}
 	
 	if (spec->num_inputs > 1 && !spec->mic_detect) {
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index cc1c22370a6..ff60908f455 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -345,8 +345,7 @@ static int cmi9880_build_controls(struct hda_codec *codec)
 	/* assign Capture Source enums to NID */
 	kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
 	for (i = 0; kctl && i < kctl->count; i++) {
-		err = snd_hda_add_nids(codec, kctl, i, spec->adc_nids,
-				       spec->input_mux->num_items);
+		err = snd_hda_add_nid(codec, kctl, i, spec->adc_nids[i]);
 		if (err < 0)
 			return err;
 	}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e7cdc6a7d61..a4519901498 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2551,8 +2551,7 @@ static int alc_build_controls(struct hda_codec *codec)
 		hda_nid_t *nids = spec->capsrc_nids;
 		if (!nids)
 			nids = spec->adc_nids;
-		err = snd_hda_add_nids(codec, kctl, i, nids,
-				       spec->input_mux->num_items);
+		err = snd_hda_add_nid(codec, kctl, i, nids[i]);
 		if (err < 0)
 			return err;
 	}
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index de4839e4676..9ddc37300f6 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1907,8 +1907,7 @@ static int via_build_controls(struct hda_codec *codec)
 	/* assign Capture Source enums to NID */
 	kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
 	for (i = 0; kctl && i < kctl->count; i++) {
-		err = snd_hda_add_nids(codec, kctl, i, spec->mux_nids,
-				       spec->input_mux->num_items);
+		err = snd_hda_add_nid(codec, kctl, i, spec->mux_nids[i]);
 		if (err < 0)
 			return err;
 	}
-- 
cgit v1.2.3


From ef18beded8ddbaafdf4914bab209f77e60ae3a18 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Fri, 25 Dec 2009 13:14:27 +0800
Subject: ALSA: hda - HDMI sticky stream tag support

When we run the following commands in turn (with
CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0),

	speaker-test -Dhw:0,3 -c2 -twav  # HDMI
	speaker-test -Dhw:0,0 -c2 -twav  # Analog

The second command will produce sound in the analog lineout _as well as_
HDMI sink. The root cause is, device 0 "reuses" the same stream tag that
was used by device 3, and the "intelhdmi - sticky stream id" patch leaves
the HDMI codec in a functional state. So the HDMI codec happily accepts
the audio samples which reuse its stream tag.

The proposed solution is to remember the last device each azx_dev was
assigned to, and prefer to
1) reuse the azx_dev (and hence the stream tag) the HDMI codec last used
2) or assign a never-used azx_dev for HDMI

With this patch and the above two speaker-test commands,
HDMI codec will use stream tag 8 and Analog codec will use 5.

The stream tag used by HDMI codec won't be reused by others, as long
as we don't run out of the 4 playback azx_dev's. The legacy Analog
codec will continue to use stream tag 5 because its device id is 0
(this is a bit tricky).

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 21 +++++++++++++++------
 1 file changed, 15 insertions(+), 6 deletions(-)

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index ff8ad46cc50..ec9c348336c 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -356,6 +356,7 @@ struct azx_dev {
 					 */
 	unsigned char stream_tag;	/* assigned stream */
 	unsigned char index;		/* stream index */
+	int device;			/* last device number assigned to */
 
 	unsigned int opened :1;
 	unsigned int running :1;
@@ -1441,10 +1442,13 @@ static int __devinit azx_codec_configure(struct azx *chip)
  */
 
 /* assign a stream for the PCM */
-static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
+static inline struct azx_dev *
+azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream)
 {
 	int dev, i, nums;
-	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+	struct azx_dev *res = NULL;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		dev = chip->playback_index_offset;
 		nums = chip->playback_streams;
 	} else {
@@ -1453,10 +1457,15 @@ static inline struct azx_dev *azx_assign_device(struct azx *chip, int stream)
 	}
 	for (i = 0; i < nums; i++, dev++)
 		if (!chip->azx_dev[dev].opened) {
-			chip->azx_dev[dev].opened = 1;
-			return &chip->azx_dev[dev];
+			res = &chip->azx_dev[dev];
+			if (res->device == substream->pcm->device)
+				break;
 		}
-	return NULL;
+	if (res) {
+		res->opened = 1;
+		res->device = substream->pcm->device;
+	}
+	return res;
 }
 
 /* release the assigned stream */
@@ -1505,7 +1514,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	int err;
 
 	mutex_lock(&chip->open_mutex);
-	azx_dev = azx_assign_device(chip, substream->stream);
+	azx_dev = azx_assign_device(chip, substream);
 	if (azx_dev == NULL) {
 		mutex_unlock(&chip->open_mutex);
 		return -EBUSY;
-- 
cgit v1.2.3


From 043958e602ac2cbf918c0dab1e4e2a7f9751ebf6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 26 Dec 2009 10:36:12 +0100
Subject: ALSA: hda - Add more hints for GPIO setup of IDT/STAC codecs

gpio_led, gpio_led_polarity and gpio_mute are added now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 14 ++++++++++++++
 1 file changed, 14 insertions(+)

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 247be19e17b..69dd5a4e52f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4184,9 +4184,23 @@ static void stac_store_hints(struct hda_codec *codec)
 	p = snd_hda_get_hint(codec, "eapd_mask");
 	if (p)
 		spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
+	p = snd_hda_get_hint(codec, "gpio_mute");
+	if (p)
+		spec->gpio_mute = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
 	val = snd_hda_get_bool_hint(codec, "eapd_switch");
 	if (val >= 0)
 		spec->eapd_switch = val;
+	p = snd_hda_get_hint(codec, "gpio_led_polarity");
+	if (p)
+		spec->gpio_led_polarity = simple_strtoul(p, NULL, 0);
+	p = snd_hda_get_hint(codec, "gpio_led");
+	if (p) {
+		spec->gpio_led = simple_strtoul(p, NULL, 0);
+		spec->gpio_mask |= spec->gpio_led;
+		spec->gpio_dir |= spec->gpio_led;
+		if (spec->gpio_led_polarity)
+			spec->gpio_data |= spec->gpio_led;
+	}
 }
 
 static int stac92xx_init(struct hda_codec *codec)
-- 
cgit v1.2.3


From 92ee6162c48fab24f0676969f0f147fc12f8f21c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 27 Dec 2009 11:18:59 +0100
Subject: ALSA: hda - Add snd_hda_shutup_pins() helper function

Add a common helper function for clearing pin controls before suspend.
Use the pincfg array instead of looking through all widget tree.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 19 +++++++++++++++++++
 sound/pci/hda/hda_codec.h      |  1 +
 sound/pci/hda/patch_sigmatel.c | 12 +-----------
 3 files changed, 21 insertions(+), 11 deletions(-)

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b3554df740f..94ae69f2092 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -899,6 +899,25 @@ static void restore_pincfgs(struct hda_codec *codec)
 	}
 }
 
+/**
+ * snd_hda_shutup_pins - Shut up all pins
+ * @codec: the HDA codec
+ *
+ * Clear all pin controls to shup up before suspend for avoiding click noise.
+ * The controls aren't cached so that they can be resumed properly.
+ */
+void snd_hda_shutup_pins(struct hda_codec *codec)
+{
+	int i;
+	for (i = 0; i < codec->init_pins.used; i++) {
+		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+		/* use read here for syncing after issuing each verb */
+		snd_hda_codec_read(codec, pin->nid, 0,
+				   AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+	}
+}
+EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
+
 static void init_hda_cache(struct hda_cache_rec *cache,
 			   unsigned int record_size);
 static void free_hda_cache(struct hda_cache_rec *cache);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 0d08ad5bd89..11c4aa8ee99 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -898,6 +898,7 @@ int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid,
 			     unsigned int cfg);
 int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
 		       hda_nid_t nid, unsigned int cfg); /* for hwdep */
+void snd_hda_shutup_pins(struct hda_codec *codec);
 
 /*
  * Mixer
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 69dd5a4e52f..dc1d9f12457 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4385,18 +4385,8 @@ static void stac92xx_free_kctls(struct hda_codec *codec)
 static void stac92xx_shutup(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	int i;
-	hda_nid_t nid;
 
-	/* reset each pin before powering down DAC/ADC to avoid click noise */
-	nid = codec->start_nid;
-	for (i = 0; i < codec->num_nodes; i++, nid++) {
-		unsigned int wcaps = get_wcaps(codec, nid);
-		unsigned int wid_type = get_wcaps_type(wcaps);
-		if (wid_type == AC_WID_PIN)
-			snd_hda_codec_read(codec, nid, 0,
-				AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
-	}
+	snd_hda_shutup_pins(codec);
 
 	if (spec->eapd_mask)
 		stac_gpio_set(codec, spec->gpio_mask,
-- 
cgit v1.2.3


From a4e09aa3cf592d9f084ff4ceb216be40c4c265dc Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 27 Dec 2009 11:22:24 +0100
Subject: ALSA: hda - Fix click noises at suspend/free with Realtek codecs

Call snd_hda_shutup_pins() at suspend and free for avoiding click noises.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 7 +++++++
 1 file changed, 7 insertions(+)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6361e6b3c9c..cd6d139b4fd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -3693,6 +3693,11 @@ static int alc_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+static inline void alc_shutup(struct hda_codec *codec)
+{
+	snd_hda_shutup_pins(codec);
+}
+
 static void alc_free_kctls(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -3713,6 +3718,7 @@ static void alc_free(struct hda_codec *codec)
 	if (!spec)
 		return;
 
+	alc_shutup(codec);
 	alc_free_kctls(codec);
 	kfree(spec);
 	snd_hda_detach_beep_device(codec);
@@ -3722,6 +3728,7 @@ static void alc_free(struct hda_codec *codec)
 static int alc_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	struct alc_spec *spec = codec->spec;
+	alc_shutup(codec);
 	if (spec && spec->power_hook)
 		spec->power_hook(codec, 0);
 	return 0;
-- 
cgit v1.2.3


From b82855a0d76ebda1cc14c00040560d77bfa042ce Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 27 Dec 2009 11:24:56 +0100
Subject: ALSA: hda - Add sanity check for storing the user-defined pin configs

Check whether the given NID is a pin widget before storing the
user-defined pin configs.

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

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 94ae69f2092..d02ea8926e7 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -824,6 +824,9 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
 	struct hda_pincfg *pin;
 	unsigned int oldcfg;
 
+	if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+		return -EINVAL;
+
 	oldcfg = snd_hda_codec_get_pincfg(codec, nid);
 	pin = look_up_pincfg(codec, list, nid);
 	if (!pin) {
-- 
cgit v1.2.3


From 014c41fce1bd5cec381e70fc6f58fdfc96cdaf69 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 27 Dec 2009 13:53:24 +0100
Subject: ALSA: hda - Use strict_strtoul()

Rewrite the codes to use strict_strtoul() instead of simple_strtoul().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_hwdep.c      |  7 ++++--
 sound/pci/hda/patch_sigmatel.c | 48 +++++++++++++++++++++++-------------------
 2 files changed, 31 insertions(+), 24 deletions(-)

diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 40ccb419b6e..b36919c0d36 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -293,8 +293,11 @@ static ssize_t type##_store(struct device *dev,			\
 {								\
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
 	struct hda_codec *codec = hwdep->private_data;		\
-	char *after;						\
-	codec->type = simple_strtoul(buf, &after, 0);		\
+	unsigned long val;					\
+	int err = strict_strtoul(buf, 0, &val);			\
+	if (err < 0)						\
+		return err;					\
+	codec->type = val;					\
 	return count;						\
 }
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index dc1d9f12457..e28c810bc00 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4159,43 +4159,47 @@ static void stac92xx_power_down(struct hda_codec *codec)
 static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid,
 				  int enable);
 
+static inline int get_int_hint(struct hda_codec *codec, const char *key,
+			       int *valp)
+{
+	const char *p;
+	p = snd_hda_get_hint(codec, key);
+	if (p) {
+		unsigned long val;
+		if (!strict_strtoul(p, 0, &val)) {
+			*valp = val;
+			return 1;
+		}
+	}
+	return 0;
+}
+
 /* override some hints from the hwdep entry */
 static void stac_store_hints(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	const char *p;
 	int val;
 
 	val = snd_hda_get_bool_hint(codec, "hp_detect");
 	if (val >= 0)
 		spec->hp_detect = val;
-	p = snd_hda_get_hint(codec, "gpio_mask");
-	if (p) {
-		spec->gpio_mask = simple_strtoul(p, NULL, 0);
+	if (get_int_hint(codec, "gpio_mask", &spec->gpio_mask)) {
 		spec->eapd_mask = spec->gpio_dir = spec->gpio_data =
 			spec->gpio_mask;
 	}
-	p = snd_hda_get_hint(codec, "gpio_dir");
-	if (p)
-		spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
-	p = snd_hda_get_hint(codec, "gpio_data");
-	if (p)
-		spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
-	p = snd_hda_get_hint(codec, "eapd_mask");
-	if (p)
-		spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
-	p = snd_hda_get_hint(codec, "gpio_mute");
-	if (p)
-		spec->gpio_mute = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
+	if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir))
+		spec->gpio_mask &= spec->gpio_mask;
+	if (get_int_hint(codec, "gpio_data", &spec->gpio_data))
+		spec->gpio_dir &= spec->gpio_mask;
+	if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask))
+		spec->eapd_mask &= spec->gpio_mask;
+	if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute))
+		spec->gpio_mute &= spec->gpio_mask;
 	val = snd_hda_get_bool_hint(codec, "eapd_switch");
 	if (val >= 0)
 		spec->eapd_switch = val;
-	p = snd_hda_get_hint(codec, "gpio_led_polarity");
-	if (p)
-		spec->gpio_led_polarity = simple_strtoul(p, NULL, 0);
-	p = snd_hda_get_hint(codec, "gpio_led");
-	if (p) {
-		spec->gpio_led = simple_strtoul(p, NULL, 0);
+	get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity);
+	if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) {
 		spec->gpio_mask |= spec->gpio_led;
 		spec->gpio_dir |= spec->gpio_led;
 		if (spec->gpio_led_polarity)
-- 
cgit v1.2.3


From ea52bf260ecbb175339af3178c15788df21b7516 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Sun, 27 Dec 2009 18:48:29 -0500
Subject: ALSA: hda: Add powerdown for Analog Devices HDA codecs

This patch ports powerdown fixes to AD198x. Currently we only turn off
Front and HP for suspend, but this is easily extended for additional
nids.

Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 68 ++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 68 insertions(+)

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 45ee352df32..cecd3c10899 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -441,6 +441,11 @@ static int ad198x_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+static inline void ad198x_shutup(struct hda_codec *codec)
+{
+	snd_hda_shutup_pins(codec);
+}
+
 static void ad198x_free_kctls(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec = codec->spec;
@@ -454,6 +459,46 @@ static void ad198x_free_kctls(struct hda_codec *codec)
 	snd_array_free(&spec->kctls);
 }
 
+static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front,
+				hda_nid_t hp)
+{
+	struct ad198x_spec *spec = codec->spec;
+	snd_hda_codec_write(codec, front, 0, AC_VERB_SET_EAPD_BTLENABLE,
+			    !spec->inv_eapd ? 0x00 : 0x02);
+	snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_EAPD_BTLENABLE,
+			    !spec->inv_eapd ? 0x00 : 0x02);
+}
+
+static void ad198x_power_eapd(struct hda_codec *codec)
+{
+	/* We currently only handle front, HP */
+	switch (codec->vendor_id) {
+	case 0x11d41882:
+	case 0x11d4882a:
+	case 0x11d41884:
+	case 0x11d41984:
+	case 0x11d41883:
+	case 0x11d4184a:
+	case 0x11d4194a:
+	case 0x11d4194b:
+		ad198x_power_eapd_write(codec, 0x12, 0x11);
+		break;
+	case 0x11d41981:
+	case 0x11d41983:
+		ad198x_power_eapd_write(codec, 0x05, 0x06);
+		break;
+	case 0x11d41986:
+		ad198x_power_eapd_write(codec, 0x1b, 0x1a);
+		break;
+	case 0x11d41988:
+	case 0x11d4198b:
+	case 0x11d4989a:
+	case 0x11d4989b:
+		ad198x_power_eapd_write(codec, 0x29, 0x22);
+		break;
+	}
+}
+
 static void ad198x_free(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec = codec->spec;
@@ -461,11 +506,29 @@ static void ad198x_free(struct hda_codec *codec)
 	if (!spec)
 		return;
 
+	ad198x_shutup(codec);
 	ad198x_free_kctls(codec);
 	kfree(spec);
 	snd_hda_detach_beep_device(codec);
 }
 
+#ifdef SND_HDA_NEEDS_RESUME
+static int ad198x_suspend(struct hda_codec *codec, pm_message_t state)
+{
+	ad198x_shutup(codec);
+	ad198x_power_eapd(codec);
+	return 0;
+}
+
+static int ad198x_resume(struct hda_codec *codec)
+{
+	ad198x_init(codec);
+	snd_hda_codec_resume_amp(codec);
+	snd_hda_codec_resume_cache(codec);
+	return 0;
+}
+#endif
+
 static struct hda_codec_ops ad198x_patch_ops = {
 	.build_controls = ad198x_build_controls,
 	.build_pcms = ad198x_build_pcms,
@@ -474,6 +537,11 @@ static struct hda_codec_ops ad198x_patch_ops = {
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	.check_power_status = ad198x_check_power_status,
 #endif
+#ifdef SND_HDA_NEEDS_RESUME
+	.suspend = ad198x_suspend,
+	.resume = ad198x_resume,
+#endif
+	.reboot_notify = ad198x_shutup,
 };
 
 
-- 
cgit v1.2.3


From c97259df3f2e163c72f4d0685c61fb2e026dc989 Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Sun, 27 Dec 2009 18:52:08 -0500
Subject: ALSA: hda: Refactor powerdown for Realtek HDA codecs

This patch converts the alc889 Aspire-specific powerdown to a generic
one. Like the previous effort, it currently only handles Front and PCM
but can be easily extended to cover other nids. The existing hook for
alc889 Aspire-specific remains enabled. Upon further testing, I've added
its use for ALC861_AUTO as well. Following patches will enable them for
other quirks.

Tested-by: Dr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 60 +++++++++++++++++++++++++++----------------
 1 file changed, 38 insertions(+), 22 deletions(-)

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index cd6d139b4fd..141ff446104 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -338,7 +338,7 @@ struct alc_spec {
 	void (*init_hook)(struct hda_codec *codec);
 	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-	void (*power_hook)(struct hda_codec *codec, int power);
+	void (*power_hook)(struct hda_codec *codec);
 #endif
 
 	/* for pin sensing */
@@ -391,7 +391,7 @@ struct alc_config_preset {
 	void (*init_hook)(struct hda_codec *);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	struct hda_amp_list *loopbacks;
-	void (*power_hook)(struct hda_codec *codec, int power);
+	void (*power_hook)(struct hda_codec *codec);
 #endif
 };
 
@@ -1835,16 +1835,6 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[2] = 0x1b;
 }
 
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static void alc889_power_eapd(struct hda_codec *codec, int power)
-{
-	snd_hda_codec_write(codec, 0x14, 0,
-			    AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
-	snd_hda_codec_write(codec, 0x15, 0,
-			    AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0);
-}
-#endif
-
 /*
  * ALC880 3-stack model
  *
@@ -3725,12 +3715,40 @@ static void alc_free(struct hda_codec *codec)
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
+static void alc_power_eapd(struct hda_codec *codec)
+{
+	/* We currently only handle front, HP */
+	switch (codec->vendor_id) {
+	case 0x10ec0260:
+		snd_hda_codec_write(codec, 0x0f, 0,
+				    AC_VERB_SET_EAPD_BTLENABLE, 0x00);
+		snd_hda_codec_write(codec, 0x10, 0,
+				    AC_VERB_SET_EAPD_BTLENABLE, 0x00);
+		break;
+	case 0x10ec0262:
+	case 0x10ec0267:
+	case 0x10ec0268:
+	case 0x10ec0269:
+	case 0x10ec0272:
+	case 0x10ec0660:
+	case 0x10ec0662:
+	case 0x10ec0663:
+	case 0x10ec0862:
+	case 0x10ec0889:
+		snd_hda_codec_write(codec, 0x14, 0,
+				    AC_VERB_SET_EAPD_BTLENABLE, 0x00);
+		snd_hda_codec_write(codec, 0x15, 0,
+				    AC_VERB_SET_EAPD_BTLENABLE, 0x00);
+		break;
+	}
+}
+
 static int alc_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	struct alc_spec *spec = codec->spec;
 	alc_shutup(codec);
 	if (spec && spec->power_hook)
-		spec->power_hook(codec, 0);
+		spec->power_hook(codec);
 	return 0;
 }
 #endif
@@ -3738,16 +3756,9 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state)
 #ifdef SND_HDA_NEEDS_RESUME
 static int alc_resume(struct hda_codec *codec)
 {
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-	struct alc_spec *spec = codec->spec;
-#endif
 	codec->patch_ops.init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-	if (spec && spec->power_hook)
-		spec->power_hook(codec, 1);
-#endif
 	return 0;
 }
 #endif
@@ -3767,6 +3778,7 @@ static struct hda_codec_ops alc_patch_ops = {
 	.suspend = alc_suspend,
 	.check_power_status = alc_check_power_status,
 #endif
+	.reboot_notify = alc_shutup,
 };
 
 
@@ -9547,7 +9559,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.setup = alc889_acer_aspire_8930g_setup,
 		.init_hook = alc_automute_amp,
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-		.power_hook = alc889_power_eapd,
+		.power_hook = alc_power_eapd,
 #endif
 	},
 	[ALC888_ACER_ASPIRE_7730G] = {
@@ -14984,8 +14996,12 @@ static int patch_alc861(struct hda_codec *codec)
 	spec->vmaster_nid = 0x03;
 
 	codec->patch_ops = alc_patch_ops;
-	if (board_config == ALC861_AUTO)
+	if (board_config == ALC861_AUTO) {
 		spec->init_hook = alc861_auto_init;
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+		spec->power_hook = alc_power_eapd;
+#endif
+	}
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
 		spec->loopback.amplist = alc861_loopbacks;
-- 
cgit v1.2.3


From cd9d95a55550555da8e587ead9cbba5f98a371a3 Mon Sep 17 00:00:00 2001
From: Ken Prox <kprox@users.sourceforge.net>
Date: Fri, 8 Jan 2010 09:01:47 +0100
Subject: ALSA: hda - conexant - Fixed microphone mixer for HP Compaq Presario
 F700

Added patch for Hewlett-Packard Company Device Subsystem id - 103c:30ea.

Signed-off-by: Ken Prox <kprox@users.sourceforge.net>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/hda/patch_conexant.c | 50 ++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 50 insertions(+)

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 1ab2958a290..b20c640f750 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -1720,6 +1720,22 @@ static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = {
 	{}
 };
 
+static struct snd_kcontrol_new cxt5051_f700_mixers[] = {
+	HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = cxt_eapd_info,
+		.get = cxt_eapd_get,
+		.put = cxt5051_hp_master_sw_put,
+		.private_value = 0x1a,
+	},
+
+	{}
+};
+
 static struct hda_verb cxt5051_init_verbs[] = {
 	/* Line in, Mic */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
@@ -1810,6 +1826,32 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
 	{ } /* end */
 };
 
+static struct hda_verb cxt5051_f700_init_verbs[] = {
+	/* Line in, Mic */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x03},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
+	/* SPK  */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP, Amp  */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* DAC1 */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Record selector: Int mic */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x1},
+	/* SPDIF route: PCM */
+	{0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* EAPD */
+	{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+	{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+	{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
+	{ } /* end */
+};
+
 /* initialize jack-sensing, too */
 static int cxt5051_init(struct hda_codec *codec)
 {
@@ -1829,6 +1871,7 @@ enum {
 	CXT5051_HP,	/* no docking */
 	CXT5051_HP_DV6736,	/* HP without mic switch */
 	CXT5051_LENOVO_X200,	/* Lenovo X200 laptop */
+	CXT5051_F700,       /* HP Compaq Presario F700 */
 	CXT5051_MODELS
 };
 
@@ -1837,6 +1880,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = {
 	[CXT5051_HP]		= "hp",
 	[CXT5051_HP_DV6736]	= "hp-dv6736",
 	[CXT5051_LENOVO_X200]	= "lenovo-x200",
+	[CXT5051_F700]          = "hp 700"
 };
 
 static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
@@ -1846,6 +1890,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
 		      CXT5051_LAPTOP),
 	SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200),
+	SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700),
 	{}
 };
 
@@ -1896,6 +1941,11 @@ static int patch_cxt5051(struct hda_codec *codec)
 	case CXT5051_LENOVO_X200:
 		spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs;
 		break;
+	case CXT5051_F700:
+		spec->init_verbs[0] = cxt5051_f700_init_verbs;
+		spec->mixers[0] = cxt5051_f700_mixers;
+		spec->no_auto_mic = 1;
+		break;
 	}
 
 	return 0;
-- 
cgit v1.2.3


From 75f8991d0e6969407d51501d5a0537f104075c99 Mon Sep 17 00:00:00 2001
From: Daniel Drake <dsd@laptop.org>
Date: Thu, 7 Jan 2010 13:46:25 +0100
Subject: ALSA: hda - Configure XO-1.5 microphones at capture time

The XO-1.5 has a microphone LED designed to indicate to the user when
something is being recorded.

This light is controlled by the microphone bias voltage and it is
currently coming on all the time.

This patch defers the microphone port configuration until when recording
is actually taking place, fixing the behaviour of the LED.

Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/hda/patch_conexant.c | 125 ++++++++++++++++++++++++++++-------------
 1 file changed, 85 insertions(+), 40 deletions(-)

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 01e46ba7269..3521f33d43c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -111,8 +111,12 @@ struct conexant_spec {
 
 	unsigned int dell_automute;
 	unsigned int port_d_mode;
-	unsigned char ext_mic_bias;
 	unsigned int dell_vostro;
+
+	unsigned int ext_mic_present;
+	unsigned int recording;
+	void (*capture_prepare)(struct hda_codec *codec);
+	void (*capture_cleanup)(struct hda_codec *codec);
 };
 
 static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -185,6 +189,8 @@ static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
 				      struct snd_pcm_substream *substream)
 {
 	struct conexant_spec *spec = codec->spec;
+	if (spec->capture_prepare)
+		spec->capture_prepare(codec);
 	snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
 				   stream_tag, 0, format);
 	return 0;
@@ -196,6 +202,8 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 {
 	struct conexant_spec *spec = codec->spec;
 	snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
+	if (spec->capture_cleanup)
+		spec->capture_cleanup(codec);
 	return 0;
 }
 
@@ -2016,53 +2024,53 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-/* toggle input of built-in and mic jack appropriately */
-static void cxt5066_automic(struct hda_codec *codec)
+/* OLPC defers mic widget control until when capture is started because the
+ * microphone LED comes on as soon as these settings are put in place. if we
+ * did this before recording, it would give the false indication that recording
+ * is happening when it is not. */
+static void cxt5066_olpc_select_mic(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
-	struct hda_verb ext_mic_present[] = {
-		/* enable external mic, port B */
-		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias},
-
-		/* switch to external mic input */
-		{0x17, AC_VERB_SET_CONNECT_SEL, 0},
+	if (!spec->recording)
+		return;
 
-		/* disable internal mic, port C */
-		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-		{}
-	};
-	static struct hda_verb ext_mic_absent[] = {
-		/* enable internal mic, port C */
-		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	/* external mic, port B */
+	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+		spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0);
 
-		/* switch to internal mic input */
-		{0x17, AC_VERB_SET_CONNECT_SEL, 1},
+	/* internal mic, port C */
+	snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+		spec->ext_mic_present ? 0 : PIN_VREF80);
+}
 
-		/* disable external mic, port B */
-		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
-		{}
-	};
+/* toggle input of built-in and mic jack appropriately */
+static void cxt5066_olpc_automic(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
 	unsigned int present;
 
-	present = snd_hda_jack_detect(codec, 0x1a);
-	if (present) {
+	present = snd_hda_codec_read(codec, 0x1a, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	if (present)
 		snd_printdd("CXT5066: external microphone detected\n");
-		snd_hda_sequence_write(codec, ext_mic_present);
-	} else {
+	else
 		snd_printdd("CXT5066: external microphone absent\n");
-		snd_hda_sequence_write(codec, ext_mic_absent);
-	}
+
+	snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL,
+		present ? 0 : 1);
+	spec->ext_mic_present = !!present;
+
+	cxt5066_olpc_select_mic(codec);
 }
 
 /* toggle input of built-in digital mic and mic jack appropriately */
 static void cxt5066_vostro_automic(struct hda_codec *codec)
 {
-	struct conexant_spec *spec = codec->spec;
 	unsigned int present;
 
 	struct hda_verb ext_mic_present[] = {
 		/* enable external mic, port B */
-		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias},
+		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 
 		/* switch to external mic input */
 		{0x17, AC_VERB_SET_CONNECT_SEL, 0},
@@ -2113,7 +2121,7 @@ static void cxt5066_hp_automute(struct hda_codec *codec)
 }
 
 /* unsolicited event for jack sensing */
-static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
+static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
 {
 	snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
 	switch (res >> 26) {
@@ -2121,7 +2129,7 @@ static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
 		cxt5066_hp_automute(codec);
 		break;
 	case CONEXANT_MIC_EVENT:
-		cxt5066_automic(codec);
+		cxt5066_olpc_automic(codec);
 		break;
 	}
 }
@@ -2197,6 +2205,31 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+static void cxt5066_olpc_capture_prepare(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	/* mark as recording and configure the microphone widget so that the
+	 * recording LED comes on. */
+	spec->recording = 1;
+	cxt5066_olpc_select_mic(codec);
+}
+
+static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	const struct hda_verb disable_mics[] = {
+		/* disable external mic, port B */
+		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+		/* disble internal mic, port C */
+		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+		{},
+	};
+
+	snd_hda_sequence_write(codec, disable_mics);
+	spec->recording = 0;
+}
+
 static struct hda_input_mux cxt5066_capture_source = {
 	.num_items = 4,
 	.items = {
@@ -2347,10 +2380,10 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
 	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
 
 	/* Port B: external microphone */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 
 	/* Port C: internal microphone */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 
 	/* Port D: unused */
 	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
@@ -2479,12 +2512,19 @@ static int cxt5066_init(struct hda_codec *codec)
 		cxt5066_hp_automute(codec);
 		if (spec->dell_vostro)
 			cxt5066_vostro_automic(codec);
-		else
-			cxt5066_automic(codec);
 	}
 	return 0;
 }
 
+static int cxt5066_olpc_init(struct hda_codec *codec)
+{
+	snd_printdd("CXT5066: init\n");
+	conexant_init(codec);
+	cxt5066_hp_automute(codec);
+	cxt5066_olpc_automic(codec);
+	return 0;
+}
+
 enum {
 	CXT5066_LAPTOP,			/* Laptops w/ EAPD support */
 	CXT5066_DELL_LAPTOP,	/* Dell Laptop */
@@ -2521,7 +2561,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 	codec->spec = spec;
 
 	codec->patch_ops = conexant_patch_ops;
-	codec->patch_ops.init = cxt5066_init;
+	codec->patch_ops.init = conexant_init;
 
 	spec->dell_automute = 0;
 	spec->multiout.max_channels = 2;
@@ -2534,7 +2574,6 @@ static int patch_cxt5066(struct hda_codec *codec)
 	spec->input_mux = &cxt5066_capture_source;
 
 	spec->port_d_mode = PIN_HP;
-	spec->ext_mic_bias = PIN_VREF80;
 
 	spec->num_init_verbs = 1;
 	spec->init_verbs[0] = cxt5066_init_verbs;
@@ -2561,20 +2600,26 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->dell_automute = 1;
 		break;
 	case CXT5066_OLPC_XO_1_5:
-		codec->patch_ops.unsol_event = cxt5066_unsol_event;
+		codec->patch_ops.init = cxt5066_olpc_init;
+		codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event;
 		spec->init_verbs[0] = cxt5066_init_verbs_olpc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		spec->port_d_mode = 0;
-		spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS;
 
 		/* no S/PDIF out */
 		spec->multiout.dig_out_nid = 0;
 
 		/* input source automatically selected */
 		spec->input_mux = NULL;
+
+		/* our capture hooks which allow us to turn on the microphone LED
+		 * at the right time */
+		spec->capture_prepare = cxt5066_olpc_capture_prepare;
+		spec->capture_cleanup = cxt5066_olpc_capture_cleanup;
 		break;
 	case CXT5066_DELL_VOSTO:
+		codec->patch_ops.init = cxt5066_init;
 		codec->patch_ops.unsol_event = cxt5066_vostro_event;
 		spec->init_verbs[0] = cxt5066_init_verbs_vostro;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
-- 
cgit v1.2.3


From c4cfe66c4c2d5a91b3734ffb4e2bad0badd5c874 Mon Sep 17 00:00:00 2001
From: Daniel Drake <dsd@laptop.org>
Date: Thu, 7 Jan 2010 13:47:04 +0100
Subject: ALSA: hda - support OLPC XO-1.5 DC input

The XO's audio hardware is wired up to allow DC sensors (e.g. light
sensors, thermistors, etc) to be plugged in through the microphone jack.

Add sound mixer controls to allow this mode to be enabled and tweaked.

Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/pci/hda/patch_conexant.c | 213 ++++++++++++++++++++++++++++++++++++-----
 1 file changed, 190 insertions(+), 23 deletions(-)

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 3521f33d43c..685015a5329 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -117,6 +117,16 @@ struct conexant_spec {
 	unsigned int recording;
 	void (*capture_prepare)(struct hda_codec *codec);
 	void (*capture_cleanup)(struct hda_codec *codec);
+
+	/* OLPC XO-1.5 supports DC input mode (e.g. for use with analog sensors)
+	 * through the microphone jack.
+	 * When the user enables this through a mixer switch, both internal and
+	 * external microphones are disabled. Gain is fixed at 0dB. In this mode,
+	 * we also allow the bias to be configured through a separate mixer
+	 * control. */
+	unsigned int dc_enable;
+	unsigned int dc_input_bias; /* offset into cxt5066_olpc_dc_bias */
+	unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */
 };
 
 static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -2024,6 +2034,26 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+static const struct hda_input_mux cxt5066_olpc_dc_bias = {
+	.num_items = 3,
+	.items = {
+		{ "Off", PIN_IN },
+		{ "50%", PIN_VREF50 },
+		{ "80%", PIN_VREF80 },
+	},
+};
+
+static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	/* Even though port F is the DC input, the bias is controlled on port B.
+	 * we also leave that port as an active input (but unselected) in DC mode
+	 * just in case that is necessary to make the bias setting take effect. */
+	return snd_hda_codec_write_cache(codec, 0x1a, 0,
+		AC_VERB_SET_PIN_WIDGET_CONTROL,
+		cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index);
+}
+
 /* OLPC defers mic widget control until when capture is started because the
  * microphone LED comes on as soon as these settings are put in place. if we
  * did this before recording, it would give the false indication that recording
@@ -2034,6 +2064,27 @@ static void cxt5066_olpc_select_mic(struct hda_codec *codec)
 	if (!spec->recording)
 		return;
 
+	if (spec->dc_enable) {
+		/* in DC mode we ignore presence detection and just use the jack
+		 * through our special DC port */
+		const struct hda_verb enable_dc_mode[] = {
+			/* disble internal mic, port C */
+			{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+			/* enable DC capture, port F */
+			{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+			{},
+		};
+
+		snd_hda_sequence_write(codec, enable_dc_mode);
+		/* port B input disabled (and bias set) through the following call */
+		cxt5066_set_olpc_dc_bias(codec);
+		return;
+	}
+
+	/* disable DC (port F) */
+	snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+
 	/* external mic, port B */
 	snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
 		spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0);
@@ -2049,6 +2100,9 @@ static void cxt5066_olpc_automic(struct hda_codec *codec)
 	struct conexant_spec *spec = codec->spec;
 	unsigned int present;
 
+	if (spec->dc_enable) /* don't do presence detection in DC mode */
+		return;
+
 	present = snd_hda_codec_read(codec, 0x1a, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 	if (present)
@@ -2123,13 +2177,16 @@ static void cxt5066_hp_automute(struct hda_codec *codec)
 /* unsolicited event for jack sensing */
 static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res)
 {
+	struct conexant_spec *spec = codec->spec;
 	snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
 	switch (res >> 26) {
 	case CONEXANT_HP_EVENT:
 		cxt5066_hp_automute(codec);
 		break;
 	case CONEXANT_MIC_EVENT:
-		cxt5066_olpc_automic(codec);
+		/* ignore mic events in DC mode; we're always using the jack */
+		if (!spec->dc_enable)
+			cxt5066_olpc_automic(codec);
 		break;
 	}
 }
@@ -2159,6 +2216,15 @@ static const struct hda_input_mux cxt5066_analog_mic_boost = {
 	},
 };
 
+static int cxt5066_set_mic_boost(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	return snd_hda_codec_write_cache(codec, 0x17, 0,
+		AC_VERB_SET_AMP_GAIN_MUTE,
+		AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT |
+			cxt5066_analog_mic_boost.items[spec->mic_boost].index);
+}
+
 static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol,
 					   struct snd_ctl_elem_info *uinfo)
 {
@@ -2169,15 +2235,8 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	int val;
-	hda_nid_t nid = kcontrol->private_value & 0xff;
-	int inout = (kcontrol->private_value & 0x100) ?
-		AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT;
-
-	val = snd_hda_codec_read(codec, nid, 0,
-		AC_VERB_GET_AMP_GAIN_MUTE, inout);
-
-	ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN;
+	struct conexant_spec *spec = codec->spec;
+	ucontrol->value.enumerated.item[0] = spec->mic_boost;
 	return 0;
 }
 
@@ -2185,23 +2244,101 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
 	const struct hda_input_mux *imux = &cxt5066_analog_mic_boost;
 	unsigned int idx;
-	hda_nid_t nid = kcontrol->private_value & 0xff;
-	int inout = (kcontrol->private_value & 0x100) ?
-		AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT;
+	idx = ucontrol->value.enumerated.item[0];
+	if (idx >= imux->num_items)
+		idx = imux->num_items - 1;
+
+	spec->mic_boost = idx;
+	if (!spec->dc_enable)
+		cxt5066_set_mic_boost(codec);
+	return 1;
+}
+
+static void cxt5066_enable_dc(struct hda_codec *codec)
+{
+	const struct hda_verb enable_dc_mode[] = {
+		/* disable gain */
+		{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+		/* switch to DC input */
+		{0x17, AC_VERB_SET_CONNECT_SEL, 3},
+		{}
+	};
+
+	/* configure as input source */
+	snd_hda_sequence_write(codec, enable_dc_mode);
+	cxt5066_olpc_select_mic(codec); /* also sets configured bias */
+}
+
+static void cxt5066_disable_dc(struct hda_codec *codec)
+{
+	/* reconfigure input source */
+	cxt5066_set_mic_boost(codec);
+	/* automic also selects the right mic if we're recording */
+	cxt5066_olpc_automic(codec);
+}
+
+static int cxt5066_olpc_dc_get(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+	ucontrol->value.integer.value[0] = spec->dc_enable;
+	return 0;
+}
 
-	if (!imux->num_items)
+static int cxt5066_olpc_dc_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+	int dc_enable = !!ucontrol->value.integer.value[0];
+
+	if (dc_enable == spec->dc_enable)
 		return 0;
+
+	spec->dc_enable = dc_enable;
+	if (dc_enable)
+		cxt5066_enable_dc(codec);
+	else
+		cxt5066_disable_dc(codec);
+
+	return 1;
+}
+
+static int cxt5066_olpc_dc_bias_enum_info(struct snd_kcontrol *kcontrol,
+					   struct snd_ctl_elem_info *uinfo)
+{
+	return snd_hda_input_mux_info(&cxt5066_olpc_dc_bias, uinfo);
+}
+
+static int cxt5066_olpc_dc_bias_enum_get(struct snd_kcontrol *kcontrol,
+					  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+	ucontrol->value.enumerated.item[0] = spec->dc_input_bias;
+	return 0;
+}
+
+static int cxt5066_olpc_dc_bias_enum_put(struct snd_kcontrol *kcontrol,
+					  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct conexant_spec *spec = codec->spec;
+	const struct hda_input_mux *imux = &cxt5066_analog_mic_boost;
+	unsigned int idx;
+
 	idx = ucontrol->value.enumerated.item[0];
 	if (idx >= imux->num_items)
 		idx = imux->num_items - 1;
 
-	snd_hda_codec_write_cache(codec, nid, 0,
-		AC_VERB_SET_AMP_GAIN_MUTE,
-		AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout |
-			imux->items[idx].index);
-
+	spec->dc_input_bias = idx;
+	if (spec->dc_enable)
+		cxt5066_set_olpc_dc_bias(codec);
 	return 1;
 }
 
@@ -2223,6 +2360,9 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec)
 
 		/* disble internal mic, port C */
 		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+		/* disable DC capture, port F */
+		{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 		{},
 	};
 
@@ -2282,6 +2422,24 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = {
 	{}
 };
 
+static struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "DC Mode Enable Switch",
+		.info = snd_ctl_boolean_mono_info,
+		.get = cxt5066_olpc_dc_get,
+		.put = cxt5066_olpc_dc_put,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "DC Input Bias Enum",
+		.info = cxt5066_olpc_dc_bias_enum_info,
+		.get = cxt5066_olpc_dc_bias_enum_get,
+		.put = cxt5066_olpc_dc_bias_enum_put,
+	},
+	{}
+};
+
 static struct snd_kcontrol_new cxt5066_mixers[] = {
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -2294,11 +2452,10 @@ static struct snd_kcontrol_new cxt5066_mixers[] = {
 
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Ext Mic Boost Capture Enum",
+		.name = "Analog Mic Boost Capture Enum",
 		.info = cxt5066_mic_boost_mux_enum_info,
 		.get = cxt5066_mic_boost_mux_enum_get,
 		.put = cxt5066_mic_boost_mux_enum_put,
-		.private_value = 0x17,
 	},
 
 	HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others),
@@ -2392,7 +2549,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = {
 	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 	{0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
 
-	/* Port F: unused */
+	/* Port F: external DC input through microphone port */
 	{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
 
 	/* Port G: internal speakers */
@@ -2513,15 +2670,22 @@ static int cxt5066_init(struct hda_codec *codec)
 		if (spec->dell_vostro)
 			cxt5066_vostro_automic(codec);
 	}
+	cxt5066_set_mic_boost(codec);
 	return 0;
 }
 
 static int cxt5066_olpc_init(struct hda_codec *codec)
 {
+	struct conexant_spec *spec = codec->spec;
 	snd_printdd("CXT5066: init\n");
 	conexant_init(codec);
 	cxt5066_hp_automute(codec);
-	cxt5066_olpc_automic(codec);
+	if (!spec->dc_enable) {
+		cxt5066_set_mic_boost(codec);
+		cxt5066_olpc_automic(codec);
+	} else {
+		cxt5066_enable_dc(codec);
+	}
 	return 0;
 }
 
@@ -2604,8 +2768,10 @@ static int patch_cxt5066(struct hda_codec *codec)
 		codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event;
 		spec->init_verbs[0] = cxt5066_init_verbs_olpc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
+		spec->mixers[spec->num_mixers++] = cxt5066_mixer_olpc_dc;
 		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
 		spec->port_d_mode = 0;
+		spec->mic_boost = 3; /* default 30dB gain */
 
 		/* no S/PDIF out */
 		spec->multiout.dig_out_nid = 0;
@@ -2627,6 +2793,7 @@ static int patch_cxt5066(struct hda_codec *codec)
 		spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers;
 		spec->port_d_mode = 0;
 		spec->dell_vostro = 1;
+		spec->mic_boost = 3; /* default 30dB gain */
 		snd_hda_attach_beep_device(codec, 0x13);
 
 		/* no S/PDIF out */
-- 
cgit v1.2.3


From ed69c6a8eef679f2783848ed624897a937a434ac Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 13 Jan 2010 08:12:31 +0100
Subject: ALSA: pcm_lib - fix wrong delta print for jiffies check

The previous jiffies delta was 0 in all cases. Use hw_ptr variable to
store and print original value.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
---
 sound/core/pcm_lib.c | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 0ee7e807c96..5417f7dce83 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -394,6 +394,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 								+ HZ/100);
 		/* move new_hw_ptr according jiffies not pos variable */
 		new_hw_ptr = old_hw_ptr;
+		hw_base = delta;
 		/* use loop to avoid checks for delta overflows */
 		/* the delta value is small or zero in most cases */
 		while (delta > 0) {
@@ -403,8 +404,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 			delta--;
 		}
 		/* align hw_base to buffer_size */
-		hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size);
-		delta = 0;
 		hw_ptr_error(substream,
 			     "hw_ptr skipping! %s"
 			     "(pos=%ld, delta=%ld, period=%ld, "
@@ -412,9 +411,12 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 			     in_interrupt ? "[Q] " : "",
 			     (long)pos, (long)hdelta,
 			     (long)runtime->period_size, jdelta,
-			     ((hdelta * HZ) / runtime->rate), delta,
+			     ((hdelta * HZ) / runtime->rate), hw_base,
 			     (unsigned long)old_hw_ptr,
 			     (unsigned long)new_hw_ptr);
+		/* reset values to proper state */
+		delta = 0;
+		hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size);
 	}
  no_jiffies_check:
 	if (delta > runtime->period_size + runtime->period_size / 2) {
-- 
cgit v1.2.3