Merge with /pub/scm/linux/kernel/git/torvalds/linux-2.6.git

83 files changed, 1506 insertions, 418 deletions

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index a2081803a82..d37346b12dc 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -216,14 +216,19 @@ config SND_CS46XX_NEW_DSP
 	  This works better than the old code, so say Y.
 
 config SND_CS5535AUDIO
-	tristate "CS5535 Audio"
+	tristate "CS5535/CS5536 Audio"
 	depends on SND && X86 && !X86_64
 	select SND_PCM
 	select SND_AC97_CODEC
 	help
 	  Say Y here to include support for audio on CS5535 chips. It is
 	  referred to as NS CS5535 IO or AMD CS5535 IO companion in
-	  various literature.
+	  various literature. This driver also supports the CS5536 audio
+	  device. However, for both chips, on certain boards, you may
+	  need to use ac97_quirk=hp_only if your board has physically 
+	  mapped headphone out to master output. If that works for you,
+	  send lspci -vvv output to the mailing list so that your board
+	  can be identified in the quirks list.
 
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-cs5535audio.
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index d05200741ac..0abf2808d59 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -253,6 +253,8 @@ void snd_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short va
 	ac97->bus->ops->write(ac97, reg, value);
 }
 
+EXPORT_SYMBOL(snd_ac97_write);
+
 /**
  * snd_ac97_read - read a value from the given register
  * 
@@ -281,6 +283,8 @@ static inline unsigned short snd_ac97_read_cache(struct snd_ac97 *ac97, unsigned
 	return ac97->regs[reg];
 }
 
+EXPORT_SYMBOL(snd_ac97_read);
+
 /**
  * snd_ac97_write_cache - write a value on the given register and update the cache
  * @ac97: the ac97 instance
@@ -302,6 +306,8 @@ void snd_ac97_write_cache(struct snd_ac97 *ac97, unsigned short reg, unsigned sh
 	mutex_unlock(&ac97->reg_mutex);
 }
 
+EXPORT_SYMBOL(snd_ac97_write_cache);
+
 /**
  * snd_ac97_update - update the value on the given register
  * @ac97: the ac97 instance
@@ -331,6 +337,8 @@ int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short va
 	return change;
 }
 
+EXPORT_SYMBOL(snd_ac97_update);
+
 /**
  * snd_ac97_update_bits - update the bits on the given register
  * @ac97: the ac97 instance
@@ -356,6 +364,8 @@ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned sho
 	return change;
 }
 
+EXPORT_SYMBOL(snd_ac97_update_bits);
+
 /* no lock version - see snd_ac97_updat_bits() */
 int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg,
 				unsigned short mask, unsigned short value)
@@ -563,7 +573,7 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1)
 };
 
 static const struct snd_kcontrol_new snd_ac97_controls_mic_boost =
-	AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0);
+	AC97_SINGLE("Mic Boost (+20dB) Switch", AC97_MIC, 6, 1, 0);
 
 
 static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"};
@@ -605,7 +615,7 @@ AC97_SINGLE("Simulated Stereo Enhancement", AC97_GENERAL_PURPOSE, 14, 1, 0),
 AC97_SINGLE("3D Control - Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
 AC97_SINGLE("Loudness (bass boost)", AC97_GENERAL_PURPOSE, 12, 1, 0),
 AC97_ENUM("Mono Output Select", std_enum[2]),
-AC97_ENUM("Mic Select", std_enum[3]),
+AC97_ENUM("Mic Select Capture Switch", std_enum[3]),
 AC97_SINGLE("ADC/DAC Loopback", AC97_GENERAL_PURPOSE, 7, 1, 0)
 };
 
@@ -1226,7 +1236,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
 	ac97->regs[AC97_CENTER_LFE_MASTER] = 0x8080;
 
 	/* build center controls */
-	if (snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) {
+	if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER)) 
+		&& !(ac97->flags & AC97_AD_MULTI)) {
 		if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_center[0], ac97))) < 0)
 			return err;
 		if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_center[1], ac97))) < 0)
@@ -1238,7 +1249,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
 	}
 
 	/* build LFE controls */
-	if (snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER+1)) {
+	if ((snd_ac97_try_volume_mix(ac97, AC97_CENTER_LFE_MASTER+1))
+		&& !(ac97->flags & AC97_AD_MULTI)) {
 		if ((err = snd_ctl_add(card, snd_ac97_cnew(&snd_ac97_controls_lfe[0], ac97))) < 0)
 			return err;
 		if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_lfe[1], ac97))) < 0)
@@ -1250,7 +1262,8 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
 	}
 
 	/* build surround controls */
-	if (snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER)) {
+	if ((snd_ac97_try_volume_mix(ac97, AC97_SURROUND_MASTER)) 
+		&& !(ac97->flags & AC97_AD_MULTI)) {
 		/* Surround Master (0x38) is with stereo mutes */
 		if ((err = snd_ac97_cmix_new_stereo(card, "Surround Playback", AC97_SURROUND_MASTER, 1, ac97)) < 0)
 			return err;
@@ -1335,9 +1348,11 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
 	}
 
 	/* build Aux controls */
-	if (snd_ac97_try_volume_mix(ac97, AC97_AUX)) {
-		if ((err = snd_ac97_cmix_new(card, "Aux Playback", AC97_AUX, ac97)) < 0)
-			return err;
+	if (!(ac97->flags & AC97_HAS_NO_AUX)) {
+		if (snd_ac97_try_volume_mix(ac97, AC97_AUX)) {
+			if ((err = snd_ac97_cmix_new(card, "Aux Playback", AC97_AUX, ac97)) < 0)
+				return err;
+		}
 	}
 
 	/* build PCM controls */
@@ -1682,6 +1697,7 @@ const char *snd_ac97_get_short_name(struct snd_ac97 *ac97)
 	return "unknown codec";
 }
 
+EXPORT_SYMBOL(snd_ac97_get_short_name);
 
 /* wait for a while until registers are accessible after RESET
  * return 0 if ok, negative not ready
@@ -1774,6 +1790,8 @@ int snd_ac97_bus(struct snd_card *card, int num, struct snd_ac97_bus_ops *ops,
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_ac97_bus);
+
 /* stop no dev release warning */
 static void ac97_device_release(struct device * dev)
 {
@@ -2117,6 +2135,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_ac97_mixer);
 
 /*
  * Power down the chip.
@@ -2166,6 +2185,8 @@ void snd_ac97_suspend(struct snd_ac97 *ac97)
 	snd_ac97_powerdown(ac97);
 }
 
+EXPORT_SYMBOL(snd_ac97_suspend);
+
 /*
  * restore ac97 status
  */
@@ -2267,6 +2288,8 @@ __reset_ready:
 		snd_ac97_restore_iec958(ac97);
 	}
 }
+
+EXPORT_SYMBOL(snd_ac97_resume);
 #endif
 
 
@@ -2590,29 +2613,7 @@ int snd_ac97_tune_hardware(struct snd_ac97 *ac97, struct ac97_quirk *quirk, cons
 	return 0;
 }
 
-
-/*
- *  Exported symbols
- */
-
-EXPORT_SYMBOL(snd_ac97_write);
-EXPORT_SYMBOL(snd_ac97_read);
-EXPORT_SYMBOL(snd_ac97_write_cache);
-EXPORT_SYMBOL(snd_ac97_update);
-EXPORT_SYMBOL(snd_ac97_update_bits);
-EXPORT_SYMBOL(snd_ac97_get_short_name);
-EXPORT_SYMBOL(snd_ac97_bus);
-EXPORT_SYMBOL(snd_ac97_mixer);
-EXPORT_SYMBOL(snd_ac97_pcm_assign);
-EXPORT_SYMBOL(snd_ac97_pcm_open);
-EXPORT_SYMBOL(snd_ac97_pcm_close);
-EXPORT_SYMBOL(snd_ac97_pcm_double_rate_rules);
 EXPORT_SYMBOL(snd_ac97_tune_hardware);
-EXPORT_SYMBOL(snd_ac97_set_rate);
-#ifdef CONFIG_PM
-EXPORT_SYMBOL(snd_ac97_resume);
-EXPORT_SYMBOL(snd_ac97_suspend);
-#endif
 
 /*
  *  INIT part
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 4d9cf37300f..7f197c78081 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -464,6 +464,10 @@ int patch_wolfson05(struct snd_ac97 * ac97)
 {
 	/* WM9705, WM9710 */
 	ac97->build_ops = &patch_wolfson_wm9705_ops;
+#ifdef CONFIG_TOUCHSCREEN_WM9705
+	/* WM9705 touchscreen uses AUX and VIDEO for touch */
+	ac97->flags |=3D AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX;
+#endif
 	return 0;
 }
 
@@ -1367,6 +1371,13 @@ static void ad18xx_resume(struct snd_ac97 *ac97)
 
 	snd_ac97_restore_iec958(ac97);
 }
+
+static void ad1888_resume(struct snd_ac97 *ac97)
+{
+	ad18xx_resume(ac97);
+	snd_ac97_write_cache(ac97, AC97_CODEC_CLASS_REV, 0x8080);
+}
+
 #endif
 
 int patch_ad1819(struct snd_ac97 * ac97)
@@ -1627,6 +1638,7 @@ static const struct snd_kcontrol_new snd_ac97_ad1981x_jack_sense[] = {
  * (SS vendor << 16 | device)
  */
 static unsigned int ad1981_jacks_blacklist[] = {
+	0x10140537, /* Thinkpad T41p */
 	0x10140554, /* Thinkpad T42p/R50p */
 	0 /* end */
 };
@@ -1839,7 +1851,7 @@ static struct snd_ac97_build_ops patch_ad1888_build_ops = {
 	.build_post_spdif = patch_ad198x_post_spdif,
 	.build_specific = patch_ad1888_specific,
 #ifdef CONFIG_PM
-	.resume = ad18xx_resume,
+	.resume = ad1888_resume,
 #endif
 	.update_jacks = ad1888_update_jacks,
 };
@@ -2048,7 +2060,10 @@ int patch_alc650(struct snd_ac97 * ac97)
 	/* Enable SPDIF-IN only on Rev.E and above */
 	val = snd_ac97_read(ac97, AC97_ALC650_CLOCK);
 	/* SPDIF IN with pin 47 */
-	if (ac97->spec.dev_flags)
+	if (ac97->spec.dev_flags &&
+	    /* ASUS A6KM requires EAPD */
+	    ! (ac97->subsystem_vendor == 0x1043 &&
+	       ac97->subsystem_device == 0x1103))
 		val |= 0x03; /* enable */
 	else
 		val &= ~0x03; /* disable */
diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c
index 512a3583b0c..f684aa2c006 100644
--- a/sound/pci/ac97/ac97_pcm.c
+++ b/sound/pci/ac97/ac97_pcm.c
@@ -317,6 +317,8 @@ int snd_ac97_set_rate(struct snd_ac97 *ac97, int reg, unsigned int rate)
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_ac97_set_rate);
+
 static unsigned short get_pslots(struct snd_ac97 *ac97, unsigned char *rate_table, unsigned short *spdif_slots)
 {
 	if (!ac97_is_audio(ac97))
@@ -550,6 +552,8 @@ int snd_ac97_pcm_assign(struct snd_ac97_bus *bus,
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_ac97_pcm_assign);
+
 /**
  * snd_ac97_pcm_open - opens the given AC97 pcm
  * @pcm: the ac97 pcm instance
@@ -633,6 +637,8 @@ int snd_ac97_pcm_open(struct ac97_pcm *pcm, unsigned int rate,
 	return err;
 }
 
+EXPORT_SYMBOL(snd_ac97_pcm_open);
+
 /**
  * snd_ac97_pcm_close - closes the given AC97 pcm
  * @pcm: the ac97 pcm instance
@@ -658,6 +664,8 @@ int snd_ac97_pcm_close(struct ac97_pcm *pcm)
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_ac97_pcm_close);
+
 static int double_rate_hw_constraint_rate(struct snd_pcm_hw_params *params,
 					  struct snd_pcm_hw_rule *rule)
 {
@@ -709,3 +717,5 @@ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime)
 				  SNDRV_PCM_HW_PARAM_RATE, -1);
 	return err;
 }
+
+EXPORT_SYMBOL(snd_ac97_pcm_double_rate_rules);
diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c
index 4d523df79cc..2118df50b9d 100644
--- a/sound/pci/ac97/ac97_proc.c
+++ b/sound/pci/ac97/ac97_proc.c
@@ -433,7 +433,7 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97)
 	prefix = ac97_is_audio(ac97) ? "ac97" : "mc97";
 	sprintf(name, "%s#%d-%d", prefix, ac97->addr, ac97->num);
 	if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) {
-		snd_info_set_text_ops(entry, ac97, 1024, snd_ac97_proc_read);
+		snd_info_set_text_ops(entry, ac97, snd_ac97_proc_read);
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
 			entry = NULL;
@@ -442,10 +442,9 @@ void snd_ac97_proc_init(struct snd_ac97 * ac97)
 	ac97->proc = entry;
 	sprintf(name, "%s#%d-%d+regs", prefix, ac97->addr, ac97->num);
 	if ((entry = snd_info_create_card_entry(ac97->bus->card, name, ac97->bus->proc)) != NULL) {
-		snd_info_set_text_ops(entry, ac97, 1024, snd_ac97_proc_regs_read);
+		snd_info_set_text_ops(entry, ac97, snd_ac97_proc_regs_read);
 #ifdef CONFIG_SND_DEBUG
 		entry->mode |= S_IWUSR;
-		entry->c.text.write_size = 1024;
 		entry->c.text.write = snd_ac97_proc_regs_write;
 #endif
 		if (snd_info_register(entry) < 0) {
diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c
index 0fb7b340731..94c26ec0588 100644
--- a/sound/pci/ac97/ak4531_codec.c
+++ b/sound/pci/ac97/ak4531_codec.c
@@ -453,7 +453,7 @@ static void snd_ak4531_proc_init(struct snd_card *card, struct snd_ak4531 *ak453
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(card, "ak4531", &entry))
-		snd_info_set_text_ops(entry, ak4531, 1024, snd_ak4531_proc_read);
+		snd_info_set_text_ops(entry, ak4531, snd_ak4531_proc_read);
 }
 #endif
 
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index eece1c7e55a..d42bf457036 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -753,7 +753,7 @@ snd_ad1889_proc_init(struct snd_ad1889 *chip)
 	struct snd_info_entry *entry;
 
 	if (!snd_card_proc_new(chip->card, chip->card->driver, &entry))
-		snd_info_set_text_ops(entry, chip, 1024, snd_ad1889_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_ad1889_proc_read);
 }
 
 static struct ac97_quirk ac97_quirks[] = {
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index e2dbc211890..5dfdbf6657f 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -49,7 +49,7 @@ MODULE_SUPPORTED_DEVICE("{{ALI,M5451,pci},{ALI,M5451}}");
 static int index = SNDRV_DEFAULT_IDX1;	/* Index */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
 static int pcm_channels = 32;
-static int spdif = 0;
+static int spdif;
 
 module_param(index, int, 0444);
 MODULE_PARM_DESC(index, "Index value for ALI M5451 PCI Audio.");
@@ -2173,7 +2173,7 @@ static void __devinit snd_ali_proc_init(struct snd_ali *codec)
 {
 	struct snd_info_entry *entry;
 	if(!snd_card_proc_new(codec->card, "ali5451", &entry))
-		snd_info_set_text_ops(entry, codec, 1024, snd_ali_proc_read);
+		snd_info_set_text_ops(entry, codec, snd_ali_proc_read);
 }
 
 static int __devinit snd_ali_resources(struct snd_ali *codec)
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 60423b1c678..a9f08066459 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -746,8 +746,8 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci,
 		card->shortname, chip->alt_port, chip->irq);
 
 	if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000,
-				        gcr+0x30, 1, pci->irq, 0,
-				        &chip->rmidi)) < 0) {
+				        gcr+0x30, MPU401_INFO_INTEGRATED,
+					pci->irq, 0, &chip->rmidi)) < 0) {
 		printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n", gcr+0x30);
 		goto out_err;
 	}
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index d0f759d86d3..f18a8c0e468 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1504,7 +1504,7 @@ static void __devinit snd_atiixp_proc_init(struct atiixp *chip)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(chip->card, "atiixp", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, snd_atiixp_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read);
 }
 #else /* !CONFIG_PROC_FS */
 #define snd_atiixp_proc_init(chip)
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 12a34c39caa..40739057076 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1177,7 +1177,7 @@ static void __devinit snd_atiixp_proc_init(struct atiixp_modem *chip)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(chip->card, "atiixp-modem", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, snd_atiixp_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_atiixp_proc_read);
 }
 #else
 #define snd_atiixp_proc_init(chip)
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index 126870ec063..8a3b118989b 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -261,6 +261,13 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return err;
 	}
 	snd_vortex_workaround(pci, pcifix[dev]);
+
+	// Card details needed in snd_vortex_midi
+	strcpy(card->driver, CARD_NAME_SHORT);
+	sprintf(card->shortname, "Aureal Vortex %s", CARD_NAME_SHORT);
+	sprintf(card->longname, "%s at 0x%lx irq %i",
+		card->shortname, chip->io, chip->irq);
+
 	// (4) Alloc components.
 	// ADB pcm.
 	if ((err = snd_vortex_new_pcm(chip, VORTEX_PCM_ADB, NR_ADB)) < 0) {
@@ -323,11 +330,6 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 #endif
 
 	// (5)
-	strcpy(card->driver, CARD_NAME_SHORT);
-	strcpy(card->shortname, CARD_NAME_SHORT);
-	sprintf(card->longname, "%s at 0x%lx irq %i",
-		card->shortname, chip->io, chip->irq);
-
 	if ((err = pci_read_config_word(pci, PCI_DEVICE_ID,
 				  &(chip->device))) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c
index 873f486b07b..c75d368ea08 100644
--- a/sound/pci/au88x0/au88x0_mpu401.c
+++ b/sound/pci/au88x0/au88x0_mpu401.c
@@ -47,7 +47,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex)
 	struct snd_rawmidi *rmidi;
 	int temp, mode;
 	struct snd_mpu401 *mpu;
-	int port;
+	unsigned long port;
 
 #ifdef VORTEX_MPU401_LEGACY
 	/* EnableHardCodedMPU401Port() */
@@ -70,9 +70,6 @@ static int __devinit snd_vortex_midi(vortex_t * vortex)
 	temp |= (MIDI_CLOCK_DIV << 8) | ((mode >> 24) & 0xff) << 4;
 	hwwrite(vortex->mmio, VORTEX_CTRL2, temp);
 	hwwrite(vortex->mmio, VORTEX_MIDI_CMD, MPU401_RESET);
-	/* Set some kind of mode */
-	if (mode)
-		hwwrite(vortex->mmio, VORTEX_MIDI_CMD, MPU401_ENTER_UART);
 
 	/* Check if anything is OK. */
 	temp = hwread(vortex->mmio, VORTEX_MIDI_DATA);
@@ -98,7 +95,8 @@ static int __devinit snd_vortex_midi(vortex_t * vortex)
 	port = (unsigned long)(vortex->mmio + VORTEX_MIDI_DATA);
 	if ((temp =
 	     snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port,
-				 1, 0, 0, &rmidi)) != 0) {
+				 MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO,
+				 0, 0, &rmidi)) != 0) {
 		hwwrite(vortex->mmio, VORTEX_CTRL,
 			(hwread(vortex->mmio, VORTEX_CTRL) &
 			 ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN);
@@ -107,6 +105,9 @@ static int __devinit snd_vortex_midi(vortex_t * vortex)
 	mpu = rmidi->private_data;
 	mpu->cport = (unsigned long)(vortex->mmio + VORTEX_MIDI_CMD);
 #endif
+	/* Overwrite MIDI name */
+	snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI %d", CARD_NAME_SHORT , vortex->card->number);
+
 	vortex->rmidi = rmidi;
 	return 0;
 }
diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c
index 4534e1882ad..b4151e208b7 100644
--- a/sound/pci/au88x0/au88x0_xtalk.c
+++ b/sound/pci/au88x0/au88x0_xtalk.c
@@ -66,31 +66,20 @@ static xtalk_gains_t const asXtalkGainsAllChan = {
 	0
 	    //0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff,0x7FFF,0x7FFF,0x7FFF,0x7FFF,0x7fff
 };
-static xtalk_gains_t const asXtalkGainsZeros = {
-	0, 0, 0, 0, 0, 0, 0, 0, 0, 0
-};
+static xtalk_gains_t const asXtalkGainsZeros;
 
-static xtalk_dline_t const alXtalkDlineZeros = {
-	0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	0, 0, 0,
-	0, 0, 0, 0, 0, 0, 0
-};
+static xtalk_dline_t const alXtalkDlineZeros;
 static xtalk_dline_t const alXtalkDlineTest = {
 	0xFC18, 0x03E8FFFF, 0x186A0, 0x7960FFFE, 1, 0xFFFFFFFF,
 	0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
 	0, 0, 0, 0
 };
 
-static xtalk_instate_t const asXtalkInStateZeros = { 0, 0, 0, 0 };
+static xtalk_instate_t const asXtalkInStateZeros;
 static xtalk_instate_t const asXtalkInStateTest =
     { 0xFF80, 0x0080, 0xFFFF, 0x0001 };
-static xtalk_state_t const asXtalkOutStateZeros = {
-	{0, 0, 0, 0},
-	{0, 0, 0, 0},
-	{0, 0, 0, 0},
-	{0, 0, 0, 0},
-	{0, 0, 0, 0}
-};
+static xtalk_state_t const asXtalkOutStateZeros;
+
 static short const sDiamondKLeftEq = 0x401d;
 static short const sDiamondKRightEq = 0x401d;
 static short const sDiamondKLeftXt = 0xF90E;
@@ -162,13 +151,7 @@ static xtalk_coefs_t const asXtalkNarrowCoefsRightXt = {
 	{0, 0, 0, 0, 0}
 };
 
-static xtalk_coefs_t const asXtalkCoefsZeros = {
-	{0, 0, 0, 0, 0},
-	{0, 0, 0, 0, 0},
-	{0, 0, 0, 0, 0},
-	{0, 0, 0, 0, 0},
-	{0, 0, 0, 0, 0}
-};
+static xtalk_coefs_t const asXtalkCoefsZeros;
 static xtalk_coefs_t const asXtalkCoefsPipe = {
 	{0, 0, 0x0FA0, 0, 0},
 	{0, 0, 0x0FA0, 0, 0},
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 52a36452426..6e62dafb66c 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -33,14 +33,21 @@
  *  in the first place >:-P}),
  *  I was forced to base this driver on reverse engineering
  *  (3 weeks' worth of evenings filled with driver work).
- *  (and no, I did NOT go the easy way: to pick up a PCI128 for 9 Euros)
+ *  (and no, I did NOT go the easy way: to pick up a SB PCI128 for 9 Euros)
  *
  *  The AZF3328 chip (note: AZF3328, *not* AZT3328, that's just the driver name
  *  for compatibility reasons) has the following features:
  *
  *  - builtin AC97 conformant codec (SNR over 80dB)
- *    (really AC97 compliant?? I really doubt it when looking
- *    at the mixer register layout)
+ *    Note that "conformant" != "compliant"!! this chip's mixer register layout
+ *    *differs* from the standard AC97 layout:
+ *    they chose to not implement the headphone register (which is not a
+ *    problem since it's merely optional), yet when doing this, they committed
+ *    the grave sin of letting other registers follow immediately instead of
+ *    keeping a headphone dummy register, thereby shifting the mixer register
+ *    addresses illegally. So far unfortunately it looks like the very flexible
+ *    ALSA AC97 support is still not enough to easily compensate for such a
+ *    grave layout violation despite all tweaks and quirks mechanisms it offers.
  *  - builtin genuine OPL3
  *  - full duplex 16bit playback/record at independent sampling rate
  *  - MPU401 (+ legacy address support) FIXME: how to enable legacy addr??
@@ -90,10 +97,15 @@
  * 
  * TODO
  *  - test MPU401 MIDI playback etc.
- *  - power management. See e.g. intel8x0 or cs4281.
- *    This would be nice since the chip runs a bit hot, and it's *required*
- *    anyway for proper ACPI power management.
+ *  - add some power micro-management (disable various units of the card
+ *    as long as they're unused). However this requires I/O ports which I
+ *    haven't figured out yet and which thus might not even exist...
+ *    The standard suspend/resume functionality could probably make use of
+ *    some improvement, too...
  *  - figure out what all unknown port bits are responsible for
+ *  - figure out some cleverly evil scheme to possibly make ALSA AC97 code
+ *    fully accept our quite incompatible ""AC97"" mixer and thus save some
+ *    code (but I'm not too optimistic that doing this is possible at all)
  */
 
 #include <sound/driver.h>
@@ -214,6 +226,16 @@ struct snd_azf3328 {
 
 	struct pci_dev *pci;
 	int irq;
+
+#ifdef CONFIG_PM
+	/* register value containers for power management
+	 * Note: not always full I/O range preserved (just like Win driver!) */
+	u16 saved_regs_codec [AZF_IO_SIZE_CODEC_PM / 2];
+	u16 saved_regs_io2   [AZF_IO_SIZE_IO2_PM / 2];
+	u16 saved_regs_mpu   [AZF_IO_SIZE_MPU_PM / 2];
+	u16 saved_regs_synth[AZF_IO_SIZE_SYNTH_PM / 2];
+	u16 saved_regs_mixer[AZF_IO_SIZE_MIXER_PM / 2];
+#endif
 };
 
 static const struct pci_device_id snd_azf3328_ids[] __devinitdata = {
@@ -317,10 +339,8 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg
 	else
 		dst_vol_left &= ~0x80;
 
-	do
-	{
-		if (!left_done)
-		{
+	do {
+		if (!left_done) {
 			if (curr_vol_left > dst_vol_left)
 				curr_vol_left--;
 			else
@@ -330,8 +350,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg
 			    left_done = 1;
 			outb(curr_vol_left, portbase + 1);
 		}
-		if (!right_done)
-		{
+		if (!right_done) {
 			if (curr_vol_right > dst_vol_right)
 				curr_vol_right--;
 			else
@@ -346,8 +365,7 @@ snd_azf3328_mixer_write_volume_gradually(const struct snd_azf3328 *chip, int reg
 		}
 		if (delay)
 			mdelay(delay);
-	}
-	while ((!left_done) || (!right_done));
+	} while ((!left_done) || (!right_done));
 	snd_azf3328_dbgcallleave();
 }
 
@@ -514,15 +532,18 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol,
 			    struct snd_ctl_elem_info *uinfo)
 {
 	static const char * const texts1[] = {
-		"ModemOut1", "ModemOut2"
+		"Mic1", "Mic2"
 	};
 	static const char * const texts2[] = {
-		"MonoSelectSource1", "MonoSelectSource2"
+		"Mix", "Mic"
 	};
 	static const char * const texts3[] = {
                 "Mic", "CD", "Video", "Aux",
 		"Line", "Mix", "Mix Mono", "Phone"
         };
+	static const char * const texts4[] = {
+		"pre 3D", "post 3D"
+        };
 	struct azf3328_mixer_reg reg;
 
 	snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
@@ -531,14 +552,19 @@ snd_azf3328_info_mixer_enum(struct snd_kcontrol *kcontrol,
         uinfo->value.enumerated.items = reg.enum_c;
         if (uinfo->value.enumerated.item > reg.enum_c - 1U)
                 uinfo->value.enumerated.item = reg.enum_c - 1U;
-	if (reg.reg == IDX_MIXER_ADVCTL2)
-	{
-		if (reg.lchan_shift == 8) /* modem out sel */
+	if (reg.reg == IDX_MIXER_ADVCTL2) {
+		switch(reg.lchan_shift) {
+		case 8: /* modem out sel */
 			strcpy(uinfo->value.enumerated.name, texts1[uinfo->value.enumerated.item]);
-		else /* mono sel source */
+			break;
+		case 9: /* mono sel source */
 			strcpy(uinfo->value.enumerated.name, texts2[uinfo->value.enumerated.item]);
-	}
-	else
+			break;
+		case 15: /* PCM Out Path */
+			strcpy(uinfo->value.enumerated.name, texts4[uinfo->value.enumerated.item]);
+			break;
+		}
+	} else
         	strcpy(uinfo->value.enumerated.name, texts3[uinfo->value.enumerated.item]
 );
         return 0;
@@ -554,12 +580,10 @@ snd_azf3328_get_mixer_enum(struct snd_kcontrol *kcontrol,
         
 	snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
 	val = snd_azf3328_mixer_inw(chip, reg.reg);
-	if (reg.reg == IDX_MIXER_REC_SELECT)
-	{
+	if (reg.reg == IDX_MIXER_REC_SELECT) {
         	ucontrol->value.enumerated.item[0] = (val >> 8) & (reg.enum_c - 1);
         	ucontrol->value.enumerated.item[1] = (val >> 0) & (reg.enum_c - 1);
-	}
-	else
+	} else
         	ucontrol->value.enumerated.item[0] = (val >> reg.lchan_shift) & (reg.enum_c - 1);
 
 	snd_azf3328_dbgmixer("get_enum: %02x is %04x -> %d|%d (shift %02d, enum_c %d)\n",
@@ -579,16 +603,13 @@ snd_azf3328_put_mixer_enum(struct snd_kcontrol *kcontrol,
 	snd_azf3328_mixer_reg_decode(&reg, kcontrol->private_value);
 	oreg = snd_azf3328_mixer_inw(chip, reg.reg);
 	val = oreg;
-	if (reg.reg == IDX_MIXER_REC_SELECT)
-	{
+	if (reg.reg == IDX_MIXER_REC_SELECT) {
         	if (ucontrol->value.enumerated.item[0] > reg.enum_c - 1U ||
             	ucontrol->value.enumerated.item[1] > reg.enum_c - 1U)
                 	return -EINVAL;
         	val = (ucontrol->value.enumerated.item[0] << 8) |
         	      (ucontrol->value.enumerated.item[1] << 0);
-	}
-	else
-	{
+	} else {
         	if (ucontrol->value.enumerated.item[0] > reg.enum_c - 1U)
                 	return -EINVAL;
 		val &= ~((reg.enum_c - 1) << reg.lchan_shift);
@@ -629,13 +650,14 @@ static const struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata
 	AZF3328_MIXER_VOL_MONO("Modem Playback Volume", IDX_MIXER_MODEMOUT, 0x1f, 1),
 	AZF3328_MIXER_SWITCH("Modem Capture Switch", IDX_MIXER_MODEMIN, 15, 1),
 	AZF3328_MIXER_VOL_MONO("Modem Capture Volume", IDX_MIXER_MODEMIN, 0x1f, 1),
-	AZF3328_MIXER_ENUM("Modem Out Select", IDX_MIXER_ADVCTL2, 2, 8),
-	AZF3328_MIXER_ENUM("Mono Select Source", IDX_MIXER_ADVCTL2, 2, 9),
+	AZF3328_MIXER_ENUM("Mic Select", IDX_MIXER_ADVCTL2, 2, 8),
+	AZF3328_MIXER_ENUM("Mono Output Select", IDX_MIXER_ADVCTL2, 2, 9),
+	AZF3328_MIXER_ENUM("PCM", IDX_MIXER_ADVCTL2, 2, 15), /* PCM Out Path, place in front since it controls *both* 3D and Bass/Treble! */
 	AZF3328_MIXER_VOL_SPECIAL("Tone Control - Treble", IDX_MIXER_BASSTREBLE, 0x07, 1, 0),
 	AZF3328_MIXER_VOL_SPECIAL("Tone Control - Bass", IDX_MIXER_BASSTREBLE, 0x07, 9, 0),
 	AZF3328_MIXER_SWITCH("3D Control - Switch", IDX_MIXER_ADVCTL2, 13, 0),
-	AZF3328_MIXER_VOL_SPECIAL("3D Control - Wide", IDX_MIXER_ADVCTL1, 0x07, 1, 0), /* "3D Width" */
-	AZF3328_MIXER_VOL_SPECIAL("3D Control - Space", IDX_MIXER_ADVCTL1, 0x03, 8, 0), /* "Hifi 3D" */
+	AZF3328_MIXER_VOL_SPECIAL("3D Control - Width", IDX_MIXER_ADVCTL1, 0x07, 1, 0), /* "3D Width" */
+	AZF3328_MIXER_VOL_SPECIAL("3D Control - Depth", IDX_MIXER_ADVCTL1, 0x03, 8, 0), /* "Hifi 3D" */
 #if MIXER_TESTING
 	AZF3328_MIXER_SWITCH("0", IDX_MIXER_ADVCTL2, 0, 0),
 	AZF3328_MIXER_SWITCH("1", IDX_MIXER_ADVCTL2, 1, 0),
@@ -813,22 +835,18 @@ snd_azf3328_setdmaa(struct snd_azf3328 *chip,
 	unsigned int is_running;
 
 	snd_azf3328_dbgcallenter();
-	if (do_recording)
-	{
+	if (do_recording) {
 		/* access capture registers, i.e. skip playback reg section */
 		portbase = chip->codec_port + 0x20;
 		is_running = chip->is_recording;
-	}
-	else
-	{
+	} else {
 		/* access the playback register section */
 		portbase = chip->codec_port + 0x00;
 		is_running = chip->is_playing;
 	}
 
 	/* AZF3328 uses a two buffer pointer DMA playback approach */
-	if (!is_running)
-	{
+	if (!is_running) {
 		unsigned long addr_area2;
 		unsigned long count_areas, count_tmp; /* width 32bit -- overflow!! */
 		count_areas = size/2;
@@ -961,6 +979,13 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 		chip->is_playing = 1;
 		snd_azf3328_dbgplay("STARTED PLAYBACK\n");
 		break;
+	case SNDRV_PCM_TRIGGER_RESUME:
+		snd_azf3328_dbgplay("RESUME PLAYBACK\n");
+		/* resume playback if we were active */
+		if (chip->is_playing)
+			snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+				snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) | DMA_RESUME);
+		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		snd_azf3328_dbgplay("STOP PLAYBACK\n");
 
@@ -988,6 +1013,12 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 		chip->is_playing = 0;
 		snd_azf3328_dbgplay("STOPPED PLAYBACK\n");
 		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		snd_azf3328_dbgplay("SUSPEND PLAYBACK\n");
+		/* make sure playback is stopped */
+		snd_azf3328_codec_outw(chip, IDX_IO_PLAY_FLAGS,
+			snd_azf3328_codec_inw(chip, IDX_IO_PLAY_FLAGS) & ~DMA_RESUME);
+		break;
         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n");
                 break;
@@ -995,6 +1026,7 @@ snd_azf3328_playback_trigger(struct snd_pcm_substream *substream, int cmd)
 		snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n");
                 break;
         default:
+		printk(KERN_ERR "FIXME: unknown trigger mode!\n");
                 return -EINVAL;
 	}
 	
@@ -1068,6 +1100,13 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 		chip->is_recording = 1;
 		snd_azf3328_dbgplay("STARTED CAPTURE\n");
 		break;
+	case SNDRV_PCM_TRIGGER_RESUME:
+		snd_azf3328_dbgplay("RESUME CAPTURE\n");
+		/* resume recording if we were active */
+		if (chip->is_recording)
+			snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
+				snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) | DMA_RESUME);
+		break;
         case SNDRV_PCM_TRIGGER_STOP:
 		snd_azf3328_dbgplay("STOP CAPTURE\n");
 
@@ -1088,6 +1127,12 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 		chip->is_recording = 0;
 		snd_azf3328_dbgplay("STOPPED CAPTURE\n");
 		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		snd_azf3328_dbgplay("SUSPEND CAPTURE\n");
+		/* make sure recording is stopped */
+		snd_azf3328_codec_outw(chip, IDX_IO_REC_FLAGS,
+			snd_azf3328_codec_inw(chip, IDX_IO_REC_FLAGS) & ~DMA_RESUME);
+		break;
         case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_PUSH NIY!\n");
                 break;
@@ -1095,6 +1140,7 @@ snd_azf3328_capture_trigger(struct snd_pcm_substream *substream, int cmd)
 		snd_printk(KERN_ERR "FIXME: SNDRV_PCM_TRIGGER_PAUSE_RELEASE NIY!\n");
                 break;
         default:
+		printk(KERN_ERR "FIXME: unknown trigger mode!\n");
                 return -EINVAL;
 	}
 	
@@ -1163,8 +1209,7 @@ snd_azf3328_interrupt(int irq, void *dev_id, struct pt_regs *regs)
 		snd_azf3328_codec_inw(chip, IDX_IO_PLAY_IRQTYPE),
 		status);
 		
-	if (status & IRQ_TIMER)
-	{
+	if (status & IRQ_TIMER) {
 		/* snd_azf3328_dbgplay("timer %ld\n", inl(chip->codec_port+IDX_IO_TIMER_VALUE) & TIMER_VALUE_MASK); */
 		if (chip->timer)
 			snd_timer_interrupt(chip->timer, chip->timer->sticks);
@@ -1174,50 +1219,43 @@ snd_azf3328_interrupt(int irq, void *dev_id, struct pt_regs *regs)
 		spin_unlock(&chip->reg_lock);
 		snd_azf3328_dbgplay("azt3328: timer IRQ\n");
 	}
-	if (status & IRQ_PLAYBACK)
-	{
+	if (status & IRQ_PLAYBACK) {
 		spin_lock(&chip->reg_lock);
 		which = snd_azf3328_codec_inb(chip, IDX_IO_PLAY_IRQTYPE);
 		/* ack all IRQ types immediately */
 		snd_azf3328_codec_outb(chip, IDX_IO_PLAY_IRQTYPE, which);
                	spin_unlock(&chip->reg_lock);
 
-		if (chip->pcm && chip->playback_substream)
-		{
+		if (chip->pcm && chip->playback_substream) {
 			snd_pcm_period_elapsed(chip->playback_substream);
 			snd_azf3328_dbgplay("PLAY period done (#%x), @ %x\n",
 				which,
 				inl(chip->codec_port+IDX_IO_PLAY_DMA_CURRPOS));
-		}
-		else
+		} else
 			snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n");
 		if (which & IRQ_PLAY_SOMETHING)
 			snd_azf3328_dbgplay("azt3328: unknown play IRQ type occurred, please report!\n");
 	}
-	if (status & IRQ_RECORDING)
-	{
+	if (status & IRQ_RECORDING) {
                 spin_lock(&chip->reg_lock);
 		which = snd_azf3328_codec_inb(chip, IDX_IO_REC_IRQTYPE);
 		/* ack all IRQ types immediately */
 		snd_azf3328_codec_outb(chip, IDX_IO_REC_IRQTYPE, which);
 		spin_unlock(&chip->reg_lock);
 
-		if (chip->pcm && chip->capture_substream)
-		{
+		if (chip->pcm && chip->capture_substream) {
 			snd_pcm_period_elapsed(chip->capture_substream);
 			snd_azf3328_dbgplay("REC  period done (#%x), @ %x\n",
 				which,
 				inl(chip->codec_port+IDX_IO_REC_DMA_CURRPOS));
-		}
-		else
+		} else
 			snd_azf3328_dbgplay("azt3328: ouch, irq handler problem!\n");
 		if (which & IRQ_REC_SOMETHING)
 			snd_azf3328_dbgplay("azt3328: unknown rec IRQ type occurred, please report!\n");
 	}
 	/* MPU401 has less critical IRQ requirements
 	 * than timer and playback/recording, right? */
-	if (status & IRQ_MPU401)
-	{
+	if (status & IRQ_MPU401) {
 		snd_mpu401_uart_interrupt(irq, chip->rmidi->private_data, regs);
 
 		/* hmm, do we have to ack the IRQ here somehow?
@@ -1511,8 +1549,7 @@ snd_azf3328_timer_start(struct snd_timer *timer)
 	snd_azf3328_dbgcallenter();
 	chip = snd_timer_chip(timer);
 	delay = ((timer->sticks * seqtimer_scaling) - 1) & TIMER_VALUE_MASK;
-	if (delay < 49)
-	{
+	if (delay < 49) {
 		/* uhoh, that's not good, since user-space won't know about
 		 * this timing tweak
 		 * (we need to do it to avoid a lockup, though) */
@@ -1766,9 +1803,11 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		goto out_err;
 	}
 
+	card->private_data = chip;
+
 	if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_MPU401,
-				        chip->mpu_port, 1, pci->irq, 0,
-				        &chip->rmidi)) < 0) {
+				        chip->mpu_port, MPU401_INFO_INTEGRATED,
+					pci->irq, 0, &chip->rmidi)) < 0) {
 		snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", chip->mpu_port);
 		goto out_err;
 	}
@@ -1791,6 +1830,8 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		}
 	}
 
+	opl3->private_data = chip;
+
 	sprintf(card->longname, "%s at 0x%lx, irq %i",
 		card->shortname, chip->codec_port, chip->irq);
 
@@ -1834,11 +1875,80 @@ snd_azf3328_remove(struct pci_dev *pci)
 	snd_azf3328_dbgcallleave();
 }
 
+#ifdef CONFIG_PM
+static int
+snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct snd_azf3328 *chip = card->private_data;
+	int reg;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	
+	snd_pcm_suspend_all(chip->pcm);
+
+	for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++)
+		chip->saved_regs_mixer[reg] = inw(chip->mixer_port + reg * 2);
+
+	/* make sure to disable master volume etc. to prevent looping sound */
+	snd_azf3328_mixer_set_mute(chip, IDX_MIXER_PLAY_MASTER, 1);
+	snd_azf3328_mixer_set_mute(chip, IDX_MIXER_WAVEOUT, 1);
+	
+	for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++)
+		chip->saved_regs_codec[reg] = inw(chip->codec_port + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++)
+		chip->saved_regs_io2[reg] = inw(chip->io2_port + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++)
+		chip->saved_regs_mpu[reg] = inw(chip->mpu_port + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++)
+		chip->saved_regs_synth[reg] = inw(chip->synth_port + reg * 2);
+
+	pci_set_power_state(pci, PCI_D3hot);
+	pci_disable_device(pci);
+	pci_save_state(pci);
+	return 0;
+}
+
+static int
+snd_azf3328_resume(struct pci_dev *pci)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct snd_azf3328 *chip = card->private_data;
+	int reg;
+
+	pci_restore_state(pci);
+	pci_enable_device(pci);
+	pci_set_power_state(pci, PCI_D0);
+	pci_set_master(pci);
+
+	for (reg = 0; reg < AZF_IO_SIZE_IO2_PM / 2; reg++)
+		outw(chip->saved_regs_io2[reg], chip->io2_port + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_MPU_PM / 2; reg++)
+		outw(chip->saved_regs_mpu[reg], chip->mpu_port + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_SYNTH_PM / 2; reg++)
+		outw(chip->saved_regs_synth[reg], chip->synth_port + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_MIXER_PM / 2; reg++)
+		outw(chip->saved_regs_mixer[reg], chip->mixer_port + reg * 2);
+	for (reg = 0; reg < AZF_IO_SIZE_CODEC_PM / 2; reg++)
+		outw(chip->saved_regs_codec[reg], chip->codec_port + reg * 2);
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	return 0;
+}
+#endif
+
+
+
+
 static struct pci_driver driver = {
 	.name = "AZF3328",
 	.id_table = snd_azf3328_ids,
 	.probe = snd_azf3328_probe,
 	.remove = __devexit_p(snd_azf3328_remove),
+#ifdef CONFIG_PM
+	.suspend = snd_azf3328_suspend,
+	.resume = snd_azf3328_resume,
+#endif
 };
 
 static int __init
diff --git a/sound/pci/azt3328.h b/sound/pci/azt3328.h
index f489bdaf6d4..b4f3e3cd006 100644
--- a/sound/pci/azt3328.h
+++ b/sound/pci/azt3328.h
@@ -5,6 +5,9 @@
 
 /*** main I/O area port indices ***/
 /* (only 0x70 of 0x80 bytes saved/restored by Windows driver) */
+#define AZF_IO_SIZE_CODEC	0x80
+#define AZF_IO_SIZE_CODEC_PM	0x70
+
 /* the driver initialisation suggests a layout of 4 main areas:
  * from 0x00 (playback), from 0x20 (recording) and from 0x40 (maybe MPU401??).
  * And another area from 0x60 to 0x6f (DirectX timer, IRQ management,
@@ -87,7 +90,7 @@
 #define IDX_IO_REC_DMA_CURROFS  0x34 /* PU:0x00000000 */
 #define IDX_IO_REC_SOUNDFORMAT  0x36 /* PU:0x0000 */
 
-/** hmm, what is this I/O area for? MPU401?? (after playback, recording, ???, timer) **/
+/** hmm, what is this I/O area for? MPU401?? or external DAC via I2S?? (after playback, recording, ???, timer) **/
 #define IDX_IO_SOMETHING_FLAGS	0x40 /* gets set to 0x34 just like port 0x0 and 0x20 on card init, PU:0x0000 */
 /* general */
 #define IDX_IO_42H		0x42 /* PU:0x0001 */
@@ -107,7 +110,8 @@
   #define IRQ_UNKNOWN2			0x0080 /* probably unused */
 #define IDX_IO_66H		0x66    /* writing 0xffff returns 0x0000 */
 #define IDX_IO_SOME_VALUE	0x68	/* this is set to e.g. 0x3ff or 0x300, and writable; maybe some buffer limit, but I couldn't find out more, PU:0x00ff */
-#define IDX_IO_6AH		0x6A	/* this WORD can be set to have bits 0x0028 activated; actually inhibits PCM playback!!! maybe power management?? */
+#define IDX_IO_6AH		0x6A	/* this WORD can be set to have bits 0x0028 activated (FIXME: correct??); actually inhibits PCM playback!!! maybe power management?? */
+  #define IO_6A_PAUSE_PLAYBACK		0x0200 /* bit 9; sure, this pauses playback, but what the heck is this really about?? */
 #define IDX_IO_6CH		0x6C
 #define IDX_IO_6EH		0x6E	/* writing 0xffff returns 0x83fe */
 /* further I/O indices not saved/restored, so probably not used */
@@ -115,15 +119,25 @@
 
 /*** I/O 2 area port indices ***/
 /* (only 0x06 of 0x08 bytes saved/restored by Windows driver) */ 
+#define AZF_IO_SIZE_IO2		0x08
+#define AZF_IO_SIZE_IO2_PM	0x06
+
 #define IDX_IO2_LEGACY_ADDR	0x04
   #define LEGACY_SOMETHING		0x01 /* OPL3?? */
   #define LEGACY_JOY			0x08
 
+#define AZF_IO_SIZE_MPU		0x04
+#define AZF_IO_SIZE_MPU_PM	0x04
+
+#define AZF_IO_SIZE_SYNTH	0x08
+#define AZF_IO_SIZE_SYNTH_PM	0x06
 
 /*** mixer I/O area port indices ***/
 /* (only 0x22 of 0x40 bytes saved/restored by Windows driver)
- * generally spoken: AC97 register index = AZF3328 mixer reg index + 2
- * (in other words: AZF3328 NOT fully AC97 compliant) */
+ * UNFORTUNATELY azf3328 is NOT truly AC97 compliant: see main file intro */
+#define AZF_IO_SIZE_MIXER	0x40
+#define AZF_IO_SIZE_MIXER_PM	0x22
+
   #define MIXER_VOLUME_RIGHT_MASK	0x001f
   #define MIXER_VOLUME_LEFT_MASK	0x1f00
   #define MIXER_MUTE_MASK		0x8000
@@ -156,14 +170,14 @@
 #define IDX_MIXER_ADVCTL1       0x1e
   /* unlisted bits are unmodifiable */
   #define MIXER_ADVCTL1_3DWIDTH_MASK	0x000e
-  #define MIXER_ADVCTL1_HIFI3D_MASK	0x0300
-#define IDX_MIXER_ADVCTL2       0x20 /* resembles AC97_GENERAL_PURPOSE reg! */
+  #define MIXER_ADVCTL1_HIFI3D_MASK	0x0300 /* yup, this is missing the high bit that official AC97 contains, plus it doesn't have linear bit value range behaviour but instead acts weirdly (possibly we're dealing with two *different* 3D settings here??) */
+#define IDX_MIXER_ADVCTL2       0x20 /* subset of AC97_GENERAL_PURPOSE reg! */
   /* unlisted bits are unmodifiable */
-  #define MIXER_ADVCTL2_BIT7		0x0080 /* WaveOut 3D Bypass? mutes WaveOut at LineOut */
-  #define MIXER_ADVCTL2_BIT8		0x0100 /* is this Modem Out Select? */
-  #define MIXER_ADVCTL2_BIT9		0x0200 /* Mono Select Source? */
-  #define MIXER_ADVCTL2_BIT13		0x2000 /* 3D enable? */
-  #define MIXER_ADVCTL2_BIT15		0x8000 /* unknown */
+  #define MIXER_ADVCTL2_LPBK		0x0080 /* Loopback mode -- Win driver: "WaveOut3DBypass"? mutes WaveOut at LineOut */
+  #define MIXER_ADVCTL2_MS		0x0100 /* Mic Select 0=Mic1, 1=Mic2 -- Win driver: "ModemOutSelect"?? */
+  #define MIXER_ADVCTL2_MIX		0x0200 /* Mono output select 0=Mix, 1=Mic; Win driver: "MonoSelectSource"?? */
+  #define MIXER_ADVCTL2_3D		0x2000 /* 3D Enhancement 1=on */
+  #define MIXER_ADVCTL2_POP		0x8000 /* Pcm Out Path, 0=pre 3D, 1=post 3D */
   
 #define IDX_MIXER_SOMETHING30H	0x30 /* used, but unknown??? */
 
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 9ee07d4aac1..c33642d8d9a 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -44,7 +44,7 @@ MODULE_SUPPORTED_DEVICE("{{Brooktree,Bt878},"
 static int index[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -2}; /* Exclude the first card */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
 static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card */
-static int digital_rate[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; /* digital input rate */
+static int digital_rate[SNDRV_CARDS];	/* digital input rate */
 static int load_all;	/* allow to load the non-whitelisted cards */
 
 module_param_array(index, int, NULL, 0444);
@@ -781,10 +781,12 @@ static struct pci_device_id snd_bt87x_ids[] __devinitdata = {
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, 32000),
 	/* Viewcast Osprey 200 */
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, 44100),
-	/* AVerMedia Studio No. 103, 203, ...? */
-	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000),
 	/* Leadtek Winfast tv 2000xp delux */
 	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, 32000),
+	/* Voodoo TV 200 */
+	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, 32000),
+	/* AVerMedia Studio No. 103, 203, ...? */
+	BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000),
 	{ }
 };
 MODULE_DEVICE_TABLE(pci, snd_bt87x_ids);
diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h
index c8131ea92ed..9cb66c59f52 100644
--- a/sound/pci/ca0106/ca0106.h
+++ b/sound/pci/ca0106/ca0106.h
@@ -537,9 +537,9 @@
 #endif
 
 #define ADC_MUX_MASK		0x0000000f	//Mask for ADC Mux
+#define ADC_MUX_PHONE		0x00000001	//Value to select TAD at ADC Mux (Not used)
 #define ADC_MUX_MIC		0x00000002	//Value to select Mic at ADC Mux
 #define ADC_MUX_LINEIN		0x00000004	//Value to select LineIn at ADC Mux
-#define ADC_MUX_PHONE		0x00000001	//Value to select TAD at ADC Mux (Not used)
 #define ADC_MUX_AUX		0x00000008	//Value to select Aux at ADC Mux
 
 #define SET_CHANNEL 0  /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
@@ -604,6 +604,8 @@ struct snd_ca0106 {
 	u32 spdif_bits[4];             /* s/pdif out setup */
 	int spdif_enable;
 	int capture_source;
+	int i2c_capture_source;
+	u8 i2c_capture_volume[4][2];
 	int capture_mic_line_in;
 
 	struct snd_dma_buffer buffer;
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index fd8bfebfbd5..59bf9bd0253 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -186,8 +186,8 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
 	 /* New Audigy SE. Has a different DAC. */
 	 /* SB0570:
 	  * CTRL:CA0106-DAT
-	  * ADC: WM8768GEDS
-	  * DAC: WM8775EDS
+	  * ADC: WM8775EDS
+	  * DAC: WM8768GEDS
 	  */
 	 { .serial = 0x100a1102,
 	   .name   = "Audigy SE [SB0570]",
@@ -195,9 +195,14 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
 	   .i2c_adc = 1,
 	   .spi_dac = 1 } ,
 	 /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */
+	 /* SB0438
+	  * CTRL:CA0106-DAT
+	  * ADC: WM8775SEDS
+	  * DAC: CS4382-KQZ
+	  */
 	 { .serial = 0x10091462,
 	   .name   = "MSI K8N Diamond MB [SB0438]",
-	   .gpio_type = 1,
+	   .gpio_type = 2,
 	   .i2c_adc = 1 } ,
 	 /* Shuttle XPC SD31P which has an onboard Creative Labs
 	  * Sound Blaster Live! 24-bit EAX
@@ -326,6 +331,7 @@ int snd_ca0106_spi_write(struct snd_ca0106 * emu,
 	return 0;
 }
 
+/* The ADC does not support i2c read, so only write is implemented */
 int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
 				u32 reg,
 				u32 value)
@@ -340,6 +346,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
 	}
 
 	tmp = reg << 25 | value << 16;
+	// snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value);
 	/* Not sure what this I2C channel controls. */
 	/* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */
 
@@ -348,8 +355,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
 
 	for (retry = 0; retry < 10; retry++) {
 		/* Send the data to i2c */
-		tmp = snd_ca0106_ptr_read(emu, I2C_A, 0);
-		tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK);
+		//tmp = snd_ca0106_ptr_read(emu, I2C_A, 0);
+		//tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK);
+		tmp = 0;
 		tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD);
 		snd_ca0106_ptr_write(emu, I2C_A, 0, tmp);
 
@@ -1181,7 +1189,7 @@ static unsigned int spi_dac_init[] = {
 	0x02ff,
 	0x0400,
 	0x0520,
-	0x0600,
+	0x0620, /* Set 24 bit. Was 0x0600 */
 	0x08ff,
 	0x0aff,
 	0x0cff,
@@ -1200,6 +1208,22 @@ static unsigned int spi_dac_init[] = {
 	0x1400,
 };
 
+static unsigned int i2c_adc_init[][2] = {
+	{ 0x17, 0x00 }, /* Reset */
+	{ 0x07, 0x00 }, /* Timeout */
+	{ 0x0b, 0x22 },  /* Interface control */
+	{ 0x0c, 0x22 },  /* Master mode control */
+	{ 0x0d, 0x08 },  /* Powerdown control */
+	{ 0x0e, 0xcf },  /* Attenuation Left  0x01 = -103dB, 0xff = 24dB */
+	{ 0x0f, 0xcf },  /* Attenuation Right 0.5dB steps */
+	{ 0x10, 0x7b },  /* ALC Control 1 */
+	{ 0x11, 0x00 },  /* ALC Control 2 */
+	{ 0x12, 0x32 },  /* ALC Control 3 */
+	{ 0x13, 0x00 },  /* Noise gate control */
+	{ 0x14, 0xa6 },  /* Limiter control */
+	{ 0x15, ADC_MUX_LINEIN },  /* ADC Mixer control */
+};
+
 static int __devinit snd_ca0106_create(struct snd_card *card,
 					 struct pci_dev *pci,
 					 struct snd_ca0106 **rchip)
@@ -1361,7 +1385,12 @@ static int __devinit snd_ca0106_create(struct snd_card *card,
         snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */
 	chip->capture_source = 3; /* Set CAPTURE_SOURCE */
 
-        if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */
+        if (chip->details->gpio_type == 2) { /* The SB0438 use GPIO differently. */
+		/* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
+		outl(0x0, chip->port+GPIO);
+		//outl(0x00f0e000, chip->port+GPIO); /* Analog */
+		outl(0x005f5301, chip->port+GPIO); /* Analog */
+	} else if (chip->details->gpio_type == 1) { /* The SB0410 and SB0413 use GPIO differently. */
 		/* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
 		outl(0x0, chip->port+GPIO);
 		//outl(0x00f0e000, chip->port+GPIO); /* Analog */
@@ -1379,7 +1408,19 @@ static int __devinit snd_ca0106_create(struct snd_card *card,
 	outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */
 
         if (chip->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */
-	        snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */
+		int size, n;
+
+		size = ARRAY_SIZE(i2c_adc_init);
+                //snd_printk("I2C:array size=0x%x\n", size);
+		for (n=0; n < size; n++) {
+			snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]);
+		}
+		for (n=0; n < 4; n++) {
+			chip->i2c_capture_volume[n][0]= 0xcf;
+			chip->i2c_capture_volume[n][1]= 0xcf;
+		}
+		chip->i2c_capture_source=2; /* Line in */
+	        //snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); /* Enable Line-in capture. MIC in currently untested. */
 	}
         if (chip->details->spi_dac == 1) { /* The SB0570 use SPI to control DAC. */
 		int size, n;
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index 06fe055674f..146eed70dce 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -171,6 +171,76 @@ static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol,
         return change;
 }
 
+static int snd_ca0106_i2c_capture_source_info(struct snd_kcontrol *kcontrol,
+					  struct snd_ctl_elem_info *uinfo)
+{
+	static char *texts[6] = {
+		"Phone", "Mic", "Line in", "Aux"
+	};
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 4;
+	if (uinfo->value.enumerated.item > 3)
+                uinfo->value.enumerated.item = 3;
+	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int snd_ca0106_i2c_capture_source_get(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.enumerated.item[0] = emu->i2c_capture_source;
+	return 0;
+}
+
+static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	unsigned int source_id;
+	unsigned int ngain, ogain;
+	int change = 0;
+	u32 source;
+	/* If the capture source has changed,
+	 * update the capture volume from the cached value
+	 * for the particular source.
+	 */
+	source_id = ucontrol->value.enumerated.item[0] ;
+	change = (emu->i2c_capture_source != source_id);
+	if (change) {
+		snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
+		ngain = emu->i2c_capture_volume[source_id][0]; /* Left */
+		ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */
+		if (ngain != ogain)
+			snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff));
+		ngain = emu->i2c_capture_volume[source_id][1]; /* Left */
+		ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Left */
+		if (ngain != ogain)
+			snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
+		source = 1 << source_id;
+		snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */
+		emu->i2c_capture_source = source_id;
+	}
+        return change;
+}
+
+static int snd_ca0106_capture_line_in_side_out_info(struct snd_kcontrol *kcontrol,
+					       struct snd_ctl_elem_info *uinfo)
+{
+	static char *texts[2] = { "Side out", "Line in" };
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 2;
+	if (uinfo->value.enumerated.item > 1)
+                uinfo->value.enumerated.item = 1;
+	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
 static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol,
 					       struct snd_ctl_elem_info *uinfo)
 {
@@ -207,16 +277,16 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
 	if (change) {
 		emu->capture_mic_line_in = val;
 		if (val) {
-			snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_PHONE); /* Mute input */
+			//snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
 			tmp = inl(emu->port+GPIO) & ~0x400;
 			tmp = tmp | 0x400;
 			outl(tmp, emu->port+GPIO);
-			snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC);
+			//snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC);
 		} else {
-			snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_PHONE); /* Mute input */
+			//snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */
 			tmp = inl(emu->port+GPIO) & ~0x400;
 			outl(tmp, emu->port+GPIO);
-			snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN);
+			//snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN);
 		}
 	}
         return change;
@@ -225,12 +295,22 @@ static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol,
 static struct snd_kcontrol_new snd_ca0106_capture_mic_line_in __devinitdata =
 {
 	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
-	.name =		"Mic/Line in Capture",
+	.name =		"Shared Mic/Line in Capture Switch",
 	.info =		snd_ca0106_capture_mic_line_in_info,
 	.get =		snd_ca0106_capture_mic_line_in_get,
 	.put =		snd_ca0106_capture_mic_line_in_put
 };
 
+static struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out __devinitdata =
+{
+	.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name =		"Shared Line in/Side out Capture Switch",
+	.info =		snd_ca0106_capture_line_in_side_out_info,
+	.get =		snd_ca0106_capture_mic_line_in_get,
+	.put =		snd_ca0106_capture_mic_line_in_put
+};
+
+
 static int snd_ca0106_spdif_info(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_info *uinfo)
 {
@@ -329,15 +409,81 @@ static int snd_ca0106_volume_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+static int snd_ca0106_i2c_volume_info(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = 2;
+        uinfo->value.integer.min = 0;
+        uinfo->value.integer.max = 255;
+        return 0;
+}
+
+static int snd_ca0106_i2c_volume_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+	int source_id;
+
+	source_id = kcontrol->private_value;
+
+        ucontrol->value.integer.value[0] = emu->i2c_capture_volume[source_id][0];
+        ucontrol->value.integer.value[1] = emu->i2c_capture_volume[source_id][1];
+        return 0;
+}
+
+static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol);
+        unsigned int ogain;
+        unsigned int ngain;
+	int source_id;
+	int change = 0;
+
+	source_id = kcontrol->private_value;
+	ogain = emu->i2c_capture_volume[source_id][0]; /* Left */
+	ngain = ucontrol->value.integer.value[0];
+	if (ngain > 0xff)
+		return 0;
+	if (ogain != ngain) {
+		if (emu->i2c_capture_source == source_id)
+			snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) );
+		emu->i2c_capture_volume[source_id][0] = ucontrol->value.integer.value[0];
+		change = 1;
+	}
+	ogain = emu->i2c_capture_volume[source_id][1]; /* Right */
+	ngain = ucontrol->value.integer.value[1];
+	if (ngain > 0xff)
+		return 0;
+	if (ogain != ngain) {
+		if (emu->i2c_capture_source == source_id)
+			snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff));
+		emu->i2c_capture_volume[source_id][1] = ucontrol->value.integer.value[1];
+		change = 1;
+	}
+
+	return change;
+}
+
 #define CA_VOLUME(xname,chid,reg) \
 {								\
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,	\
-	.info = snd_ca0106_volume_info,				\
-	.get =          snd_ca0106_volume_get,			\
-	.put =          snd_ca0106_volume_put,			\
+	.info =	 snd_ca0106_volume_info,			\
+	.get =   snd_ca0106_volume_get,				\
+	.put =   snd_ca0106_volume_put,				\
 	.private_value = ((chid) << 8) | (reg)			\
 }
 
+#define I2C_VOLUME(xname,chid) \
+{								\
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,	\
+	.info =  snd_ca0106_i2c_volume_info,			\
+	.get =   snd_ca0106_i2c_volume_get,			\
+	.put =   snd_ca0106_i2c_volume_put,			\
+	.private_value = chid					\
+}
+
 
 static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = {
 	CA_VOLUME("Analog Front Playback Volume",
@@ -361,6 +507,11 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = {
         CA_VOLUME("CAPTURE feedback Playback Volume",
 		  1, CAPTURE_CONTROL),
 
+        I2C_VOLUME("Phone Capture Volume", 0),
+        I2C_VOLUME("Mic Capture Volume", 1),
+        I2C_VOLUME("Line in Capture Volume", 2),
+        I2C_VOLUME("Aux Capture Volume", 3),
+
 	{
 		.access =	SNDRV_CTL_ELEM_ACCESS_READ,
 		.iface =        SNDRV_CTL_ELEM_IFACE_PCM,
@@ -378,12 +529,19 @@ static struct snd_kcontrol_new snd_ca0106_volume_ctls[] __devinitdata = {
 	},
 	{
 		.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name =		"Capture Source",
+		.name =		"Digital Capture Source",
 		.info =		snd_ca0106_capture_source_info,
 		.get =		snd_ca0106_capture_source_get,
 		.put =		snd_ca0106_capture_source_put
 	},
 	{
+		.iface =	SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name =		"Capture Source",
+		.info =		snd_ca0106_i2c_capture_source_info,
+		.get =		snd_ca0106_i2c_capture_source_get,
+		.put =		snd_ca0106_i2c_capture_source_put
+	},
+	{
 		.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
 		.name =         SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
 		.count =	4,
@@ -477,7 +635,10 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
 			return err;
 	}
 	if (emu->details->i2c_adc == 1) {
-		err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu));
+		if (emu->details->gpio_type == 1)
+			err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu));
+		else  /* gpio_type == 2 */
+			err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_line_in_side_out, emu));
 		if (err < 0)
 			return err;
 	}
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index 63757273bfb..75ca421eb3a 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -431,33 +431,30 @@ int __devinit snd_ca0106_proc_init(struct snd_ca0106 * emu)
 	struct snd_info_entry *entry;
 	
 	if(! snd_card_proc_new(emu->card, "iec958", &entry))
-		snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_iec958);
+		snd_info_set_text_ops(entry, emu, snd_ca0106_proc_iec958);
 	if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) {
-		snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read32);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32);
 		entry->c.text.write = snd_ca0106_proc_reg_write32;
 		entry->mode |= S_IWUSR;
 	}
 	if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry))
-		snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read16);
+		snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16);
 	if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry))
-		snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read8);
+		snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read8);
 	if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) {
-		snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read1);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1);
 		entry->c.text.write = snd_ca0106_proc_reg_write;
 		entry->mode |= S_IWUSR;
 //		entry->private_data = emu;
 	}
 	if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) {
-		snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_i2c_write);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_ca0106_proc_i2c_write);
 		entry->c.text.write = snd_ca0106_proc_i2c_write;
 		entry->mode |= S_IWUSR;
 //		entry->private_data = emu;
 	}
 	if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) 
-		snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read2);
+		snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2);
 	return 0;
 }
 
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index e5ce2dabd08..0938c158b5c 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -2121,7 +2121,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = {
 	CMIPCI_MIXER_VOL_MONO("Mic Capture Volume", CM_REG_MIXER2, CM_VADMIC_SHIFT, 7),
 	CMIPCI_SB_VOL_MONO("Phone Playback Volume", CM_REG_EXTENT_IND, 5, 7),
 	CMIPCI_DOUBLE("Phone Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 4, 4, 1, 0, 0),
-	CMIPCI_DOUBLE("PC Speaker Playnack Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0),
+	CMIPCI_DOUBLE("PC Speaker Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0),
 	CMIPCI_DOUBLE("Mic Boost Capture Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 0, 0, 1, 0, 0),
 };
 
@@ -2602,7 +2602,7 @@ static void __devinit snd_cmipci_proc_init(struct cmipci *cm)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(cm->card, "cmipci", &entry))
-		snd_info_set_text_ops(entry, cm, 1024, snd_cmipci_proc_read);
+		snd_info_set_text_ops(entry, cm, snd_cmipci_proc_read);
 }
 #else /* !CONFIG_PROC_FS */
 static inline void snd_cmipci_proc_init(struct cmipci *cm) {}
@@ -2932,7 +2932,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
 	}
 
 	integrated_midi = snd_cmipci_read_b(cm, CM_REG_MPU_PCI) != 0xff;
-	if (integrated_midi)
+	if (integrated_midi && mpu_port[dev] == 1)
 		iomidi = cm->iobase + CM_REG_MPU_PCI;
 	else {
 		iomidi = mpu_port[dev];
@@ -2981,7 +2981,9 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc
 
 	if (iomidi > 0) {
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
-					       iomidi, integrated_midi,
+					       iomidi,
+					       (integrated_midi ?
+						MPU401_INFO_INTEGRATED : 0),
 					       cm->irq, 0, &cm->rmidi)) < 0) {
 			printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi);
 		}
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index b3c94d83450..e77a4ce314b 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1184,7 +1184,7 @@ static void __devinit snd_cs4281_proc_init(struct cs4281 * chip)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(chip->card, "cs4281", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, snd_cs4281_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_cs4281_proc_read);
 	if (! snd_card_proc_new(chip->card, "cs4281_BA0", &entry)) {
 		entry->content = SNDRV_INFO_CONTENT_DATA;
 		entry->private_data = chip;
@@ -1379,6 +1379,13 @@ static int __devinit snd_cs4281_create(struct snd_card *card,
 	chip->ba0_addr = pci_resource_start(pci, 0);
 	chip->ba1_addr = pci_resource_start(pci, 1);
 
+	chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0));
+	chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1));
+	if (!chip->ba0 || !chip->ba1) {
+		snd_cs4281_free(chip);
+		return -ENOMEM;
+	}
+	
 	if (request_irq(pci->irq, snd_cs4281_interrupt, SA_INTERRUPT|SA_SHIRQ,
 			"CS4281", chip)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
@@ -1387,13 +1394,6 @@ static int __devinit snd_cs4281_create(struct snd_card *card,
 	}
 	chip->irq = pci->irq;
 
-	chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0));
-	chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1));
-	if (!chip->ba0 || !chip->ba1) {
-		snd_cs4281_free(chip);
-		return -ENOMEM;
-	}
-	
 	tmp = snd_cs4281_chip_init(chip);
 	if (tmp) {
 		snd_cs4281_free(chip);
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 848d772ae3c..772dc52bfeb 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -48,8 +48,8 @@ MODULE_SUPPORTED_DEVICE("{{Cirrus Logic,Sound Fusion (CS4280)},"
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
 static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card */
-static int external_amp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0};
-static int thinkpad[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0};
+static int external_amp[SNDRV_CARDS];
+static int thinkpad[SNDRV_CARDS];
 static int mmap_valid[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1};
 
 module_param_array(index, int, NULL, 0444);
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 69dbf542a6d..5c211443920 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -2877,14 +2877,15 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip)
 	if (chip->region.idx[0].resource)
 		snd_cs46xx_hw_stop(chip);
 
+	if (chip->irq >= 0)
+		free_irq(chip->irq, chip);
+
 	for (idx = 0; idx < 5; idx++) {
 		struct snd_cs46xx_region *region = &chip->region.idx[idx];
 		if (region->remap_addr)
 			iounmap(region->remap_addr);
 		release_and_free_resource(region->resource);
 	}
-	if (chip->irq >= 0)
-		free_irq(chip->irq, chip);
 
 	if (chip->active_ctrl)
 		chip->active_ctrl(chip, -chip->amplifier);
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
index f407d2a5ce3..5c9711c0265 100644
--- a/sound/pci/cs46xx/dsp_spos.c
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -767,7 +767,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip)
 	if ((entry = snd_info_create_card_entry(card, "dsp", card->proc_root)) != NULL) {
 		entry->content = SNDRV_INFO_CONTENT_TEXT;
 		entry->mode = S_IFDIR | S_IRUGO | S_IXUGO;
-		entry->c.text.read_size = 512;
       
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
@@ -784,7 +783,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip)
 		entry->content = SNDRV_INFO_CONTENT_TEXT;
 		entry->private_data = chip;
 		entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
-		entry->c.text.read_size = 512;
 		entry->c.text.read = cs46xx_dsp_proc_symbol_table_read;
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
@@ -797,7 +795,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip)
 		entry->content = SNDRV_INFO_CONTENT_TEXT;
 		entry->private_data = chip;
 		entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
-		entry->c.text.read_size = 512;
 		entry->c.text.read = cs46xx_dsp_proc_modules_read;
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
@@ -810,7 +807,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip)
 		entry->content = SNDRV_INFO_CONTENT_TEXT;
 		entry->private_data = chip;
 		entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
-		entry->c.text.read_size = 512;
 		entry->c.text.read = cs46xx_dsp_proc_parameter_dump_read;
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
@@ -823,7 +819,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip)
 		entry->content = SNDRV_INFO_CONTENT_TEXT;
 		entry->private_data = chip;
 		entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
-		entry->c.text.read_size = 512;
 		entry->c.text.read = cs46xx_dsp_proc_sample_dump_read;
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
@@ -836,7 +831,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip)
 		entry->content = SNDRV_INFO_CONTENT_TEXT;
 		entry->private_data = chip;
 		entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
-		entry->c.text.read_size = 512;
 		entry->c.text.read = cs46xx_dsp_proc_task_tree_read;
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
@@ -849,7 +843,6 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip)
 		entry->content = SNDRV_INFO_CONTENT_TEXT;
 		entry->private_data = chip;
 		entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
-		entry->c.text.read_size = 1024;
 		entry->c.text.read = cs46xx_dsp_proc_scb_read;
 		if (snd_info_register(entry) < 0) {
 			snd_info_free_entry(entry);
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 2c4ee45fe10..3844d18af19 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -267,7 +267,6 @@ void cs46xx_dsp_proc_register_scb_desc (struct snd_cs46xx *chip,
 			entry->private_data = scb_info;
 			entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
       
-			entry->c.text.read_size = 512;
 			entry->c.text.read = cs46xx_dsp_proc_scb_info_read;
       
 			if (snd_info_register(entry) < 0) {
diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile
index 08d8ee6547d..2911a8adc1f 100644
--- a/sound/pci/cs5535audio/Makefile
+++ b/sound/pci/cs5535audio/Makefile
@@ -4,5 +4,9 @@
 
 snd-cs5535audio-objs := cs5535audio.o cs5535audio_pcm.o
 
+ifdef CONFIG_PM
+snd-cs5535audio-objs += cs5535audio_pm.o
+endif
+
 # Toplevel Module Dependency
 obj-$(CONFIG_SND_CS5535AUDIO) += snd-cs5535audio.o
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 2c1213a35dc..91c18a11fe8 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -1,5 +1,5 @@
 /*
- * Driver for audio on multifunction CS5535 companion device
+ * Driver for audio on multifunction CS5535/6 companion device
  * Copyright (C) Jaya Kumar
  *
  * Based on Jaroslav Kysela and Takashi Iwai's examples.
@@ -40,16 +40,36 @@
 
 #define DRIVER_NAME "cs5535audio"
 
+static char *ac97_quirk;
+module_param(ac97_quirk, charp, 0444);
+MODULE_PARM_DESC(ac97_quirk, "AC'97 board specific workarounds.");
+
+static struct ac97_quirk ac97_quirks[] __devinitdata = {
+#if 0 /* Not yet confirmed if all 5536 boards are HP only */
+	{
+		.subvendor = PCI_VENDOR_ID_AMD, 
+		.subdevice = PCI_DEVICE_ID_AMD_CS5536_AUDIO, 
+		.name = "AMD RDK",     
+		.type = AC97_TUNE_HP_ONLY
+	},
+#endif
+	{}
+};
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
 static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
 
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " DRIVER_NAME);
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME);
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME);
+
 static struct pci_device_id snd_cs5535audio_ids[] __devinitdata = {
-	{ PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO,
-	  PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
-	{ PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO,
-	  PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
+	{ PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) },
+	{ PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) },
 	{}
 };
 
@@ -90,7 +110,8 @@ static unsigned short snd_cs5535audio_codec_read(struct cs5535audio *cs5535au,
 		udelay(1);
 	} while (--timeout);
 	if (!timeout)
-		snd_printk(KERN_ERR "Failure reading cs5535 codec\n");
+		snd_printk(KERN_ERR "Failure reading codec reg 0x%x,"
+					"Last value=0x%x\n", reg, val);
 
 	return (unsigned short) val;
 }
@@ -148,6 +169,8 @@ static int snd_cs5535audio_mixer(struct cs5535audio *cs5535au)
 		return err;
 	}
 
+	snd_ac97_tune_hardware(cs5535au->ac97, ac97_quirks, ac97_quirk);
+
 	return 0;
 }
 
@@ -347,6 +370,8 @@ static int __devinit snd_cs5535audio_probe(struct pci_dev *pci,
 	if ((err = snd_cs5535audio_create(card, pci, &cs5535au)) < 0)
 		goto probefail_out;
 
+	card->private_data = cs5535au;
+
 	if ((err = snd_cs5535audio_mixer(cs5535au)) < 0)
 		goto probefail_out;
 
@@ -383,6 +408,10 @@ static struct pci_driver driver = {
 	.id_table = snd_cs5535audio_ids,
 	.probe = snd_cs5535audio_probe,
 	.remove = __devexit_p(snd_cs5535audio_remove),
+#ifdef CONFIG_PM
+	.suspend = snd_cs5535audio_suspend,
+	.resume = snd_cs5535audio_resume,
+#endif
 };
 
 static int __init alsa_card_cs5535audio_init(void)
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index 5e55a1a1ed6..4fd1f31a6cf 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -74,6 +74,8 @@
 #define PRM_RDY_STS			0x00800000
 #define ACC_CODEC_CNTL_WR_CMD		(~0x80000000)
 #define ACC_CODEC_CNTL_RD_CMD		0x80000000
+#define ACC_CODEC_CNTL_LNK_SHUTDOWN	0x00040000
+#define ACC_CODEC_CNTL_LNK_WRM_RST	0x00020000
 #define PRD_JMP				0x2000
 #define PRD_EOP				0x4000
 #define PRD_EOT				0x8000
@@ -88,6 +90,7 @@ struct cs5535audio_dma_ops {
 	void (*disable_dma)(struct cs5535audio *cs5535au);
 	void (*pause_dma)(struct cs5535audio *cs5535au);
 	void (*setup_prd)(struct cs5535audio *cs5535au, u32 prd_addr);
+	u32 (*read_prd)(struct cs5535audio *cs5535au);
 	u32 (*read_dma_pntr)(struct cs5535audio *cs5535au);
 };
 
@@ -103,11 +106,14 @@ struct cs5535audio_dma {
 	struct snd_pcm_substream *substream;
 	unsigned int buf_addr, buf_bytes;
 	unsigned int period_bytes, periods;
+	int suspended;
+	u32 saved_prd;
 };
 
 struct cs5535audio {
 	struct snd_card *card;
 	struct snd_ac97 *ac97;
+	struct snd_pcm *pcm;
 	int irq;
 	struct pci_dev *pci;
 	unsigned long port;
@@ -117,6 +123,8 @@ struct cs5535audio {
 	struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS];
 };
 
+int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state);
+int snd_cs5535audio_resume(struct pci_dev *pci);
 int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535audio);
 
 #endif /* __SOUND_CS5535AUDIO_H */
diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c
index 60bb82b2ff4..f0a48693d68 100644
--- a/sound/pci/cs5535audio/cs5535audio_pcm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pcm.c
@@ -43,7 +43,8 @@ static struct snd_pcm_hardware snd_cs5535audio_playback =
 		 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
 		 		SNDRV_PCM_INFO_MMAP_VALID |
 		 		SNDRV_PCM_INFO_PAUSE |
-				SNDRV_PCM_INFO_SYNC_START
+				SNDRV_PCM_INFO_SYNC_START |
+				SNDRV_PCM_INFO_RESUME
 				),
 	.formats =		(
 				SNDRV_PCM_FMTBIT_S16_LE
@@ -193,6 +194,11 @@ static void cs5535audio_playback_setup_prd(struct cs5535audio *cs5535au,
 	cs_writel(cs5535au, ACC_BM0_PRD, prd_addr);
 }
 
+static u32 cs5535audio_playback_read_prd(struct cs5535audio *cs5535au)
+{
+	return cs_readl(cs5535au, ACC_BM0_PRD);
+}
+
 static u32 cs5535audio_playback_read_dma_pntr(struct cs5535audio *cs5535au)
 {
 	return cs_readl(cs5535au, ACC_BM0_PNTR);
@@ -219,6 +225,11 @@ static void cs5535audio_capture_setup_prd(struct cs5535audio *cs5535au,
 	cs_writel(cs5535au, ACC_BM1_PRD, prd_addr);
 }
 
+static u32 cs5535audio_capture_read_prd(struct cs5535audio *cs5535au)
+{
+	return cs_readl(cs5535au, ACC_BM1_PRD);
+}
+
 static u32 cs5535audio_capture_read_dma_pntr(struct cs5535audio *cs5535au)
 {
 	return cs_readl(cs5535au, ACC_BM1_PNTR);
@@ -285,9 +296,17 @@ static int snd_cs5535audio_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_START:
 		dma->ops->enable_dma(cs5535au);
 		break;
+	case SNDRV_PCM_TRIGGER_RESUME:
+		dma->ops->enable_dma(cs5535au);
+		dma->suspended = 0;
+		break;
 	case SNDRV_PCM_TRIGGER_STOP:
 		dma->ops->disable_dma(cs5535au);
 		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		dma->ops->disable_dma(cs5535au);
+		dma->suspended = 1;
+		break;
 	default:
 		snd_printk(KERN_ERR "unhandled trigger\n");
 		err = -EINVAL;
@@ -375,6 +394,7 @@ static struct cs5535audio_dma_ops snd_cs5535audio_playback_dma_ops = {
         .enable_dma = cs5535audio_playback_enable_dma,
         .disable_dma = cs5535audio_playback_disable_dma,
         .setup_prd = cs5535audio_playback_setup_prd,
+        .read_prd = cs5535audio_playback_read_prd,
         .pause_dma = cs5535audio_playback_pause_dma,
         .read_dma_pntr = cs5535audio_playback_read_dma_pntr,
 };
@@ -384,6 +404,7 @@ static struct cs5535audio_dma_ops snd_cs5535audio_capture_dma_ops = {
         .enable_dma = cs5535audio_capture_enable_dma,
         .disable_dma = cs5535audio_capture_disable_dma,
         .setup_prd = cs5535audio_capture_setup_prd,
+        .read_prd = cs5535audio_capture_read_prd,
         .pause_dma = cs5535audio_capture_pause_dma,
         .read_dma_pntr = cs5535audio_capture_read_dma_pntr,
 };
@@ -413,6 +434,7 @@ int __devinit snd_cs5535audio_pcm(struct cs5535audio *cs5535au)
 	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
 					snd_dma_pci_data(cs5535au->pci),
 					64*1024, 128*1024);
+	cs5535au->pcm = pcm;
 
 	return 0;
 }
diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c
new file mode 100644
index 00000000000..aad0e69db9c
--- /dev/null
+++ b/sound/pci/cs5535audio/cs5535audio_pm.c
@@ -0,0 +1,123 @@
+/*
+ * Power management for audio on multifunction CS5535 companion device
+ * Copyright (C) Jaya Kumar
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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/slab.h>
+#include <linux/pci.h>
+#include <linux/delay.h>
+#include <sound/driver.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/initval.h>
+#include <sound/asoundef.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include "cs5535audio.h"
+
+static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au)
+{
+	/* 
+	we depend on snd_ac97_suspend to tell the
+	AC97 codec to shutdown. the amd spec suggests
+	that the LNK_SHUTDOWN be done at the same time
+	that the codec power-down is issued. instead,
+	we do it just after rather than at the same 
+	time. excluding codec specific build_ops->suspend
+	ac97 powerdown hits:
+	0x8000 EAPD 
+	0x4000 Headphone amplifier 
+	0x0300 ADC & DAC 
+	0x0400 Analog Mixer powerdown (Vref on) 
+	I am not sure if this is the best that we can do.
+	The remainder to be investigated are:
+	- analog mixer (vref off) 0x0800
+	- AC-link powerdown 0x1000
+	- codec internal clock 0x2000
+	*/
+
+	/* set LNK_SHUTDOWN to shutdown AC link */
+	cs_writel(cs5535au, ACC_CODEC_CNTL, ACC_CODEC_CNTL_LNK_SHUTDOWN);
+
+}
+
+int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct cs5535audio *cs5535au = card->private_data;
+	int i;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) {
+		struct cs5535audio_dma *dma = &cs5535au->dmas[i];
+		if (dma && dma->substream && !dma->suspended) 
+			dma->saved_prd = dma->ops->read_prd(cs5535au);
+	}
+	snd_pcm_suspend_all(cs5535au->pcm);
+	snd_ac97_suspend(cs5535au->ac97);
+	/* save important regs, then disable aclink in hw */
+	snd_cs5535audio_stop_hardware(cs5535au);
+	pci_disable_device(pci);
+	pci_save_state(pci);
+
+	return 0;
+}
+
+int snd_cs5535audio_resume(struct pci_dev *pci)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct cs5535audio *cs5535au = card->private_data;
+	u32 tmp;
+	int timeout;
+	int i;
+
+	pci_restore_state(pci);
+	pci_enable_device(pci);
+	pci_set_master(pci);
+
+	/* set LNK_WRM_RST to reset AC link */
+	cs_writel(cs5535au, ACC_CODEC_CNTL, ACC_CODEC_CNTL_LNK_WRM_RST);
+
+	timeout = 50;
+	do {
+		tmp = cs_readl(cs5535au, ACC_CODEC_STATUS);
+		if (tmp & PRM_RDY_STS)
+			break;
+		udelay(1);
+	} while (--timeout);
+
+	if (!timeout)
+		snd_printk(KERN_ERR "Failure getting AC Link ready\n");
+
+	/* we depend on ac97 to perform the codec power up */
+	snd_ac97_resume(cs5535au->ac97);
+	/* set up rate regs, dma. actual initiation is done in trig */
+	for (i = 0; i < NUM_CS5535AUDIO_DMAS; i++) {
+		struct cs5535audio_dma *dma = &cs5535au->dmas[i];
+		if (dma && dma->substream && dma->suspended) {
+			dma->substream->ops->prepare(dma->substream);
+			dma->ops->setup_prd(cs5535au, dma->saved_prd);
+		}
+	}
+		
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+	return 0;
+}
+
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 42b11ba1d21..549673ea14a 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -46,13 +46,13 @@ MODULE_SUPPORTED_DEVICE("{{Creative Labs,SB Live!/PCI512/E-mu APS},"
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
 static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card */
-static int extin[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0};
-static int extout[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0};
+static int extin[SNDRV_CARDS];
+static int extout[SNDRV_CARDS];
 static int seq_ports[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4};
 static int max_synth_voices[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 64};
 static int max_buffer_size[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 128};
-static int enable_ir[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0};
-static uint subsystem[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0}; /* Force card subsystem model */
+static int enable_ir[SNDRV_CARDS];
+static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for the EMU10K1 soundcard.");
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 6bfa08436ef..42a358f989c 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -777,14 +777,6 @@ static int snd_emu10k1_dev_free(struct snd_device *device)
 
 static struct snd_emu_chip_details emu_chip_details[] = {
 	/* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/
-	/* Audigy4 SB0400 */
-	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102,
-	 .driver = "Audigy2", .name = "Audigy 4 [SB0400]", 
-	 .id = "Audigy2",
-	 .emu10k2_chip = 1,
-	 .ca0108_chip = 1,
-	 .spk71 = 1,
-	 .ac97_chip = 1} ,
 	/* Tested by James@superbug.co.uk 3rd July 2005 */
 	/* DSP: CA0108-IAT
 	 * DAC: CS4382-KQ
@@ -799,13 +791,59 @@ static struct snd_emu_chip_details emu_chip_details[] = {
 	 .ca0108_chip = 1,
 	 .spk71 = 1,
 	 .ac97_chip = 1} ,
+	/* Audigy4 (Not PRO) SB0610 */
+	/* Tested by James@superbug.co.uk 4th April 2006 */
+	/* A_IOCFG bits
+	 * Output
+	 * 0: ?
+	 * 1: ?
+	 * 2: ?
+	 * 3: 0 - Digital Out, 1 - Line in
+	 * 4: ?
+	 * 5: ?
+	 * 6: ?
+	 * 7: ?
+	 * Input
+	 * 8: ?
+	 * 9: ?
+	 * A: Green jack sense (Front)
+	 * B: ?
+	 * C: Black jack sense (Rear/Side Right)
+	 * D: Yellow jack sense (Center/LFE/Side Left)
+	 * E: ?
+	 * F: ?
+	 *
+	 * Digital Out/Line in switch using A_IOCFG bit 3 (0x08)
+	 * 0 - Digital Out
+	 * 1 - Line in
+	 */
+	/* Mic input not tested.
+	 * Analog CD input not tested
+	 * Digital Out not tested.
+	 * Line in working.
+	 * Audio output 5.1 working. Side outputs not working.
+	 */
+	/* DSP: CA10300-IAT LF
+	 * DAC: Cirrus Logic CS4382-KQZ
+	 * ADC: Philips 1361T
+	 * AC97: Sigmatel STAC9750
+	 * CA0151: None
+	 */
+	{.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102,
+	 .driver = "Audigy2", .name = "Audigy 4 [SB0610]", 
+	 .id = "Audigy2",
+	 .emu10k2_chip = 1,
+	 .ca0108_chip = 1,
+	 .spk71 = 1,
+	 .adc_1361t = 1,  /* 24 bit capture instead of 16bit */
+	 .ac97_chip = 1} ,
 	/* Audigy 2 ZS Notebook Cardbus card.*/
 	/* Tested by James@superbug.co.uk 22th December 2005 */
 	/* Audio output 7.1/Headphones working.
 	 * Digital output working. (AC3 not checked, only PCM)
 	 * Audio inputs not tested.
 	 */ 
-	/* DSP: Tiny2
+	/* DSP: Tina2
 	 * DAC: Wolfson WM8768/WM8568
 	 * ADC: Wolfson WM8775
 	 * AC97: None
@@ -1421,16 +1459,3 @@ void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu)
 	}
 }
 #endif
-
-/* memory.c */
-EXPORT_SYMBOL(snd_emu10k1_synth_alloc);
-EXPORT_SYMBOL(snd_emu10k1_synth_free);
-EXPORT_SYMBOL(snd_emu10k1_synth_bzero);
-EXPORT_SYMBOL(snd_emu10k1_synth_copy_from_user);
-EXPORT_SYMBOL(snd_emu10k1_memblk_map);
-/* voice.c */
-EXPORT_SYMBOL(snd_emu10k1_voice_alloc);
-EXPORT_SYMBOL(snd_emu10k1_voice_free);
-/* io.c */
-EXPORT_SYMBOL(snd_emu10k1_ptr_read);
-EXPORT_SYMBOL(snd_emu10k1_ptr_write);
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index d51290c1816..0fb27e4be07 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1055,8 +1055,7 @@ static int __devinit snd_emu10k1x_proc_init(struct emu10k1x * emu)
 	struct snd_info_entry *entry;
 	
 	if(! snd_card_proc_new(emu->card, "emu10k1x_regs", &entry)) {
-		snd_info_set_text_ops(entry, emu, 1024, snd_emu10k1x_proc_reg_read);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_emu10k1x_proc_reg_read);
 		entry->c.text.write = snd_emu10k1x_proc_reg_write;
 		entry->mode |= S_IWUSR;
 		entry->private_data = emu;
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index 2a9d12d1068..c31f3d0877f 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -777,6 +777,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
 	};
 	static char *audigy_remove_ctls[] = {
 		/* Master/PCM controls on ac97 of Audigy has no effect */
+		/* On the Audigy2 the AC97 playback is piped into
+		 * the Philips ADC for 24bit capture */
 		"PCM Playback Switch",
 		"PCM Playback Volume",
 		"Master Mono Playback Switch",
@@ -804,6 +806,47 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
 		"AMic Playback Volume", "Mic Playback Volume",
 		NULL
 	};
+	static char *audigy_remove_ctls_1361t_adc[] = {
+		/* On the Audigy2 the AC97 playback is piped into
+		 * the Philips ADC for 24bit capture */
+		"PCM Playback Switch",
+		"PCM Playback Volume",
+		"Master Mono Playback Switch",
+		"Master Mono Playback Volume",
+		"Capture Source",
+		"Capture Switch",
+		"Capture Volume",
+		"Mic Capture Volume",
+		"Headphone Playback Switch",
+		"Headphone Playback Volume",
+		"3D Control - Center",
+		"3D Control - Depth",
+		"3D Control - Switch",
+		"Line2 Playback Volume",
+		"Line2 Capture Volume",
+		NULL
+	};
+	static char *audigy_rename_ctls_1361t_adc[] = {
+		"Master Playback Switch", "Master Capture Switch",
+		"Master Playback Volume", "Master Capture Volume",
+		"Wave Master Playback Volume", "Master Playback Volume",
+		"PC Speaker Playback Switch", "PC Speaker Capture Switch",
+		"PC Speaker Playback Volume", "PC Speaker Capture Volume",
+		"Phone Playback Switch", "Phone Capture Switch",
+		"Phone Playback Volume", "Phone Capture Volume",
+		"Mic Playback Switch", "Mic Capture Switch",
+		"Mic Playback Volume", "Mic Capture Volume",
+		"Line Playback Switch", "Line Capture Switch",
+		"Line Playback Volume", "Line Capture Volume",
+		"CD Playback Switch", "CD Capture Switch",
+		"CD Playback Volume", "CD Capture Volume",
+		"Aux Playback Switch", "Aux Capture Switch",
+		"Aux Playback Volume", "Aux Capture Volume",
+		"Video Playback Switch", "Video Capture Switch",
+		"Video Playback Volume", "Video Capture Volume",
+
+		NULL
+	};
 
 	if (emu->card_capabilities->ac97_chip) {
 		struct snd_ac97_bus *pbus;
@@ -834,7 +877,10 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
 			snd_ac97_write_cache(emu->ac97, AC97_MASTER, 0x0000);
 			/* set capture source to mic */
 			snd_ac97_write_cache(emu->ac97, AC97_REC_SEL, 0x0000);
-			c = audigy_remove_ctls;
+			if (emu->card_capabilities->adc_1361t)
+				c = audigy_remove_ctls_1361t_adc;
+			else 
+				c = audigy_remove_ctls;
 		} else {
 			/*
 			 * Credits for cards based on STAC9758:
@@ -863,11 +909,15 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
 	}
 
 	if (emu->audigy)
-		c = audigy_rename_ctls;
+		if (emu->card_capabilities->adc_1361t)
+			c = audigy_rename_ctls_1361t_adc;
+		else
+			c = audigy_rename_ctls;
 	else
 		c = emu10k1_rename_ctls;
 	for (; *c; c += 2)
 		rename_ctl(card, c[0], c[1]);
+
 	if (emu->card_capabilities->subsystem == 0x20071102) {  /* Audigy 4 Pro */
 		rename_ctl(card, "Line2 Capture Volume", "Line1/Mic Capture Volume");
 		rename_ctl(card, "Analog Mix Capture Volume", "Line2 Capture Volume");
diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c
index 90f1c52703a..b939e03aaed 100644
--- a/sound/pci/emu10k1/emuproc.c
+++ b/sound/pci/emu10k1/emuproc.c
@@ -532,57 +532,51 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu)
 	struct snd_info_entry *entry;
 #ifdef CONFIG_SND_DEBUG
 	if (! snd_card_proc_new(emu->card, "io_regs", &entry)) {
-		snd_info_set_text_ops(entry, emu, 1024, snd_emu_proc_io_reg_read);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_emu_proc_io_reg_read);
 		entry->c.text.write = snd_emu_proc_io_reg_write;
 		entry->mode |= S_IWUSR;
 	}
 	if (! snd_card_proc_new(emu->card, "ptr_regs00a", &entry)) {
-		snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read00a);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00a);
 		entry->c.text.write = snd_emu_proc_ptr_reg_write00;
 		entry->mode |= S_IWUSR;
 	}
 	if (! snd_card_proc_new(emu->card, "ptr_regs00b", &entry)) {
-		snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read00b);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read00b);
 		entry->c.text.write = snd_emu_proc_ptr_reg_write00;
 		entry->mode |= S_IWUSR;
 	}
 	if (! snd_card_proc_new(emu->card, "ptr_regs20a", &entry)) {
-		snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20a);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20a);
 		entry->c.text.write = snd_emu_proc_ptr_reg_write20;
 		entry->mode |= S_IWUSR;
 	}
 	if (! snd_card_proc_new(emu->card, "ptr_regs20b", &entry)) {
-		snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20b);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20b);
 		entry->c.text.write = snd_emu_proc_ptr_reg_write20;
 		entry->mode |= S_IWUSR;
 	}
 	if (! snd_card_proc_new(emu->card, "ptr_regs20c", &entry)) {
-		snd_info_set_text_ops(entry, emu, 65536, snd_emu_proc_ptr_reg_read20c);
-		entry->c.text.write_size = 64;
+		snd_info_set_text_ops(entry, emu, snd_emu_proc_ptr_reg_read20c);
 		entry->c.text.write = snd_emu_proc_ptr_reg_write20;
 		entry->mode |= S_IWUSR;
 	}
 #endif
 	
 	if (! snd_card_proc_new(emu->card, "emu10k1", &entry))
-		snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_read);
+		snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_read);
 
 	if (emu->card_capabilities->emu10k2_chip) {
 		if (! snd_card_proc_new(emu->card, "spdif-in", &entry))
-			snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_spdif_read);
+			snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_spdif_read);
 	}
 	if (emu->card_capabilities->ca0151_chip) {
 		if (! snd_card_proc_new(emu->card, "capture-rates", &entry))
-			snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_rates_read);
+			snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_rates_read);
 	}
 
 	if (! snd_card_proc_new(emu->card, "voices", &entry))
-		snd_info_set_text_ops(entry, emu, 2048, snd_emu10k1_proc_voices_read);
+		snd_info_set_text_ops(entry, emu, snd_emu10k1_proc_voices_read);
 
 	if (! snd_card_proc_new(emu->card, "fx8010_gpr", &entry)) {
 		entry->content = SNDRV_INFO_CONTENT_DATA;
@@ -616,7 +610,6 @@ int __devinit snd_emu10k1_proc_init(struct snd_emu10k1 * emu)
 		entry->content = SNDRV_INFO_CONTENT_TEXT;
 		entry->private_data = emu;
 		entry->mode = S_IFREG | S_IRUGO /*| S_IWUSR*/;
-		entry->c.text.read_size = 128*1024;
 		entry->c.text.read = snd_emu10k1_proc_acode_read;
 	}
 	return 0;
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index ef5304df8c1..029e7856c43 100644
--- a/sound/pci/emu10k1/io.c
+++ b/sound/pci/emu10k1/io.c
@@ -62,6 +62,8 @@ unsigned int snd_emu10k1_ptr_read(struct snd_emu10k1 * emu, unsigned int reg, un
 	}
 }
 
+EXPORT_SYMBOL(snd_emu10k1_ptr_read);
+
 void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data)
 {
 	unsigned int regptr;
@@ -92,6 +94,8 @@ void snd_emu10k1_ptr_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned i
 	}
 }
 
+EXPORT_SYMBOL(snd_emu10k1_ptr_write);
+
 unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, 
 					  unsigned int reg, 
 					  unsigned int chn)
diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c
index e7ec98649f0..4fcaefe5a3c 100644
--- a/sound/pci/emu10k1/memory.c
+++ b/sound/pci/emu10k1/memory.c
@@ -287,6 +287,8 @@ int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *b
 	return err;
 }
 
+EXPORT_SYMBOL(snd_emu10k1_memblk_map);
+
 /*
  * page allocation for DMA
  */
@@ -387,6 +389,7 @@ snd_emu10k1_synth_alloc(struct snd_emu10k1 *hw, unsigned int size)
 	return (struct snd_util_memblk *)blk;
 }
 
+EXPORT_SYMBOL(snd_emu10k1_synth_alloc);
 
 /*
  * free a synth sample area
@@ -409,6 +412,7 @@ snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *memblk)
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_emu10k1_synth_free);
 
 /* check new allocation range */
 static void get_single_page_range(struct snd_util_memhdr *hdr,
@@ -540,6 +544,8 @@ int snd_emu10k1_synth_bzero(struct snd_emu10k1 *emu, struct snd_util_memblk *blk
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_emu10k1_synth_bzero);
+
 /*
  * copy_from_user(blk + offset, data, size)
  */
@@ -568,3 +574,5 @@ int snd_emu10k1_synth_copy_from_user(struct snd_emu10k1 *emu, struct snd_util_me
 	} while (offset < end_offset);
 	return 0;
 }
+
+EXPORT_SYMBOL(snd_emu10k1_synth_copy_from_user);
diff --git a/sound/pci/emu10k1/p17v.h b/sound/pci/emu10k1/p17v.h
new file mode 100644
index 00000000000..7ddb5be632c
--- /dev/null
+++ b/sound/pci/emu10k1/p17v.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
+ *  Driver p17v chips
+ *  Version: 0.01
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program 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
+ *
+ */
+
+/******************************************************************************/
+/* Audigy2Value Tina (P17V) pointer-offset register set,
+ * accessed through the PTR20 and DATA24 registers  */
+/******************************************************************************/
+
+/* 00 - 07: Not used */
+#define P17V_PLAYBACK_FIFO_PTR	0x08	/* Current playback fifo pointer
+					 * and number of sound samples in cache.
+					 */  
+/* 09 - 12: Not used */
+#define P17V_CAPTURE_FIFO_PTR	0x13	/* Current capture fifo pointer
+					 * and number of sound samples in cache.
+					 */  
+/* 14 - 17: Not used */
+#define P17V_PB_CHN_SEL		0x18	/* P17v playback channel select */
+#define P17V_SE_SLOT_SEL_L	0x19	/* Sound Engine slot select low */
+#define P17V_SE_SLOT_SEL_H	0x1a	/* Sound Engine slot select high */
+/* 1b - 1f: Not used */
+/* 20 - 2f: Not used */
+/* 30 - 3b: Not used */
+#define P17V_SPI		0x3c	/* SPI interface register */
+#define P17V_I2C_ADDR		0x3d	/* I2C Address */
+#define P17V_I2C_0		0x3e	/* I2C Data */
+#define P17V_I2C_1		0x3f	/* I2C Data */
+
+#define P17V_START_AUDIO	0x40	/* Start Audio bit */
+/* 41 - 47: Reserved */
+#define P17V_START_CAPTURE	0x48	/* Start Capture bit */
+#define P17V_CAPTURE_FIFO_BASE	0x49	/* Record FIFO base address */
+#define P17V_CAPTURE_FIFO_SIZE	0x4a	/* Record FIFO buffer size */
+#define P17V_CAPTURE_FIFO_INDEX	0x4b	/* Record FIFO capture index */
+#define P17V_CAPTURE_VOL_H	0x4c	/* P17v capture volume control */
+#define P17V_CAPTURE_VOL_L	0x4d	/* P17v capture volume control */
+/* 4e - 4f: Not used */
+/* 50 - 5f: Not used */
+#define P17V_SRCSel		0x60	/* SRC48 and SRCMulti sample rate select
+					 * and output select
+					 */
+#define P17V_MIXER_AC97_10K1_VOL_L	0x61	/* 10K to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_10K1_VOL_H	0x62	/* 10K to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_P17V_VOL_L	0x63	/* P17V to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_P17V_VOL_H	0x64	/* P17V to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_SRP_REC_VOL_L	0x65	/* SRP Record to Mixer_AC97 input volume control */
+#define P17V_MIXER_AC97_SRP_REC_VOL_H	0x66	/* SRP Record to Mixer_AC97 input volume control */
+/* 67 - 68: Reserved */
+#define P17V_MIXER_Spdif_10K1_VOL_L	0x69	/* 10K to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_10K1_VOL_H	0x6A	/* 10K to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_P17V_VOL_L	0x6B	/* P17V to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_P17V_VOL_H	0x6C	/* P17V to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_SRP_REC_VOL_L	0x6D	/* SRP Record to Mixer_Spdif input volume control */
+#define P17V_MIXER_Spdif_SRP_REC_VOL_H	0x6E	/* SRP Record to Mixer_Spdif input volume control */
+/* 6f - 70: Reserved */
+#define P17V_MIXER_I2S_10K1_VOL_L	0x71	/* 10K to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_10K1_VOL_H	0x72	/* 10K to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_P17V_VOL_L	0x73	/* P17V to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_P17V_VOL_H	0x74	/* P17V to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_SRP_REC_VOL_L	0x75	/* SRP Record to Mixer_I2S input volume control */
+#define P17V_MIXER_I2S_SRP_REC_VOL_H	0x76	/* SRP Record to Mixer_I2S input volume control */
+/* 77 - 78: Reserved */
+#define P17V_MIXER_AC97_ENABLE		0x79	/* Mixer AC97 input audio enable */
+#define P17V_MIXER_SPDIF_ENABLE		0x7A	/* Mixer SPDIF input audio enable */
+#define P17V_MIXER_I2S_ENABLE		0x7B	/* Mixer I2S input audio enable */
+#define P17V_AUDIO_OUT_ENABLE		0x7C	/* Audio out enable */
+#define P17V_MIXER_ATT			0x7D	/* SRP Mixer Attenuation Select */
+#define P17V_SRP_RECORD_SRR		0x7E	/* SRP Record channel source Select */
+#define P17V_SOFT_RESET_SRP_MIXER	0x7F	/* SRP and mixer soft reset */
+
+#define P17V_AC97_OUT_MASTER_VOL_L	0x80	/* AC97 Output master volume control */
+#define P17V_AC97_OUT_MASTER_VOL_H	0x81	/* AC97 Output master volume control */
+#define P17V_SPDIF_OUT_MASTER_VOL_L	0x82	/* SPDIF Output master volume control */
+#define P17V_SPDIF_OUT_MASTER_VOL_H	0x83	/* SPDIF Output master volume control */
+#define P17V_I2S_OUT_MASTER_VOL_L	0x84	/* I2S Output master volume control */
+#define P17V_I2S_OUT_MASTER_VOL_H	0x85	/* I2S Output master volume control */
+/* 86 - 87: Not used */
+#define P17V_I2S_CHANNEL_SWAP_PHASE_INVERSE	0x88	/* I2S out mono channel swap
+							 * and phase inverse */
+#define P17V_SPDIF_CHANNEL_SWAP_PHASE_INVERSE	0x89	/* SPDIF out mono channel swap
+							 * and phase inverse */
+/* 8A: Not used */
+#define P17V_SRP_P17V_ESR		0x8B	/* SRP_P17V estimated sample rate and rate lock */
+#define P17V_SRP_REC_ESR		0x8C	/* SRP_REC estimated sample rate and rate lock */
+#define P17V_SRP_BYPASS			0x8D	/* srps channel bypass and srps bypass */
+/* 8E - 92: Not used */
+#define P17V_I2S_SRC_SEL		0x93	/* I2SIN mode sel */
+
+
+
+
+
+
diff --git a/sound/pci/emu10k1/tina2.h b/sound/pci/emu10k1/tina2.h
index 5c43abf03e8..f2d8eb6c89e 100644
--- a/sound/pci/emu10k1/tina2.h
+++ b/sound/pci/emu10k1/tina2.h
@@ -1,11 +1,7 @@
 /*
  *  Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
- *  Driver p16v chips
- *  Version: 0.21
- *
- *
- *  This code was initally based on code from ALSA's emu10k1x.c which is:
- *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
+ *  Driver tina2 chips
+ *  Version: 0.1
  *
  *   This program is free software; you can redistribute it and/or modify
  *   it under the terms of the GNU General Public License as published by
diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c
index 56ffb7dc3ee..94eca82dd4f 100644
--- a/sound/pci/emu10k1/voice.c
+++ b/sound/pci/emu10k1/voice.c
@@ -139,6 +139,8 @@ int snd_emu10k1_voice_alloc(struct snd_emu10k1 *emu, int type, int number,
 	return result;
 }
 
+EXPORT_SYMBOL(snd_emu10k1_voice_alloc);
+
 int snd_emu10k1_voice_free(struct snd_emu10k1 *emu,
 			   struct snd_emu10k1_voice *pvoice)
 {
@@ -153,3 +155,5 @@ int snd_emu10k1_voice_free(struct snd_emu10k1 *emu,
 	spin_unlock_irqrestore(&emu->voice_lock, flags);
 	return 0;
 }
+
+EXPORT_SYMBOL(snd_emu10k1_voice_free);
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index ca9e34e88f6..9d46bbee2a4 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -1915,7 +1915,7 @@ static void __devinit snd_ensoniq_proc_init(struct ensoniq * ensoniq)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(ensoniq->card, "audiopci", &entry))
-		snd_info_set_text_ops(entry, ensoniq, 1024, snd_ensoniq_proc_read);
+		snd_info_set_text_ops(entry, ensoniq, snd_ensoniq_proc_read);
 }
 
 /*
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 6f9094ca4fb..ca6603fe0b1 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1756,7 +1756,8 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci,
 		}
 	}
 	if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
-				chip->mpu_port, 1, chip->irq, 0, &chip->rmidi) < 0) {
+				chip->mpu_port, MPU401_INFO_INTEGRATED,
+				chip->irq, 0, &chip->rmidi) < 0) {
 		printk(KERN_ERR "es1938: unable to initialize MPU-401\n");
 	} else {
 		// this line is vital for MIDI interrupt handling on ess-solo1
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 5ff4175c7b6..bfa0876e715 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -132,7 +132,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card *
 static int total_bufsize[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1024 };
 static int pcm_substreams_p[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 4 };
 static int pcm_substreams_c[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1 };
-static int clock[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0};
+static int clock[SNDRV_CARDS];
 static int use_pm[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
 static int enable_mpu[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
 #ifdef SUPPORT_JOYSTICK
@@ -2727,7 +2727,8 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci,
 	}
 	if (enable_mpu[dev]) {
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
-					       chip->io_port + ESM_MPU401_PORT, 1,
+					       chip->io_port + ESM_MPU401_PORT,
+					       MPU401_INFO_INTEGRATED,
 					       chip->irq, 0, &chip->rmidi)) < 0) {
 			printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n");
 		}
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index d72fc28c580..0afa573dd24 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -56,7 +56,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card *
  *    3 = MediaForte 64-PCR
  *  High 16-bits are video (radio) device number + 1
  */
-static int tea575x_tuner[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 };
+static int tea575x_tuner[SNDRV_CARDS];
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for the FM801 soundcard.");
@@ -1448,7 +1448,8 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
 		return err;
 	}
 	if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801,
-				       FM801_REG(chip, MPU401_DATA), 1,
+				       FM801_REG(chip, MPU401_DATA),
+				       MPU401_INFO_INTEGRATED,
 				       chip->irq, 0, &chip->rmidi)) < 0) {
 		snd_card_free(card);
 		return err;
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index ddfb5ff7fb8..dbacba6177d 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -1,5 +1,5 @@
 snd-hda-intel-objs := hda_intel.o
-snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o patch_sigmatel.o patch_si3054.o
+snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o patch_sigmatel.o patch_si3054.o patch_atihdmi.o
 ifdef CONFIG_PROC_FS
 snd-hda-codec-objs += hda_proc.o
 endif
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 5bee3b53647..8c2a8174ece 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -86,6 +86,8 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int dire
 	return res;
 }
 
+EXPORT_SYMBOL(snd_hda_codec_read);
+
 /**
  * snd_hda_codec_write - send a single command without waiting for response
  * @codec: the HDA codec
@@ -108,6 +110,8 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
 	return err;
 }
 
+EXPORT_SYMBOL(snd_hda_codec_write);
+
 /**
  * snd_hda_sequence_write - sequence writes
  * @codec: the HDA codec
@@ -122,6 +126,8 @@ void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq)
 		snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param);
 }
 
+EXPORT_SYMBOL(snd_hda_sequence_write);
+
 /**
  * snd_hda_get_sub_nodes - get the range of sub nodes
  * @codec: the HDA codec
@@ -140,6 +146,8 @@ int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *sta
 	return (int)(parm & 0x7fff);
 }
 
+EXPORT_SYMBOL(snd_hda_get_sub_nodes);
+
 /**
  * snd_hda_get_connections - get connection list
  * @codec: the HDA codec
@@ -256,6 +264,8 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_hda_queue_unsol_event);
+
 /*
  * process queueud unsolicited events
  */
@@ -384,6 +394,7 @@ int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_hda_bus_new);
 
 /*
  * find a matching codec preset
@@ -587,6 +598,8 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_hda_codec_new);
+
 /**
  * snd_hda_codec_setup_stream - set up the codec for streaming
  * @codec: the CODEC to set up
@@ -609,6 +622,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stre
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format);
 }
 
+EXPORT_SYMBOL(snd_hda_codec_setup_stream);
 
 /*
  * amp access functions
@@ -1294,6 +1308,7 @@ int snd_hda_build_controls(struct hda_bus *bus)
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_hda_build_controls);
 
 /*
  * stream formats
@@ -1382,6 +1397,8 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 	return val;
 }
 
+EXPORT_SYMBOL(snd_hda_calc_stream_format);
+
 /**
  * snd_hda_query_supported_pcm - query the supported PCM rates and formats
  * @codec: the HDA codec
@@ -1663,6 +1680,7 @@ int snd_hda_build_pcms(struct hda_bus *bus)
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_hda_build_pcms);
 
 /**
  * snd_hda_check_board_config - compare the current codec with the config table
@@ -2165,6 +2183,8 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state)
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_hda_suspend);
+
 /**
  * snd_hda_resume - resume the codecs
  * @bus: the HDA bus
@@ -2187,6 +2207,8 @@ int snd_hda_resume(struct hda_bus *bus)
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_hda_resume);
+
 /**
  * snd_hda_resume_ctls - resume controls in the new control list
  * @codec: the HDA codec
@@ -2247,25 +2269,6 @@ int snd_hda_resume_spdif_in(struct hda_codec *codec)
 #endif
 
 /*
- * symbols exported for controller modules
- */
-EXPORT_SYMBOL(snd_hda_codec_read);
-EXPORT_SYMBOL(snd_hda_codec_write);
-EXPORT_SYMBOL(snd_hda_sequence_write);
-EXPORT_SYMBOL(snd_hda_get_sub_nodes);
-EXPORT_SYMBOL(snd_hda_queue_unsol_event);
-EXPORT_SYMBOL(snd_hda_bus_new);
-EXPORT_SYMBOL(snd_hda_codec_new);
-EXPORT_SYMBOL(snd_hda_codec_setup_stream);
-EXPORT_SYMBOL(snd_hda_calc_stream_format);
-EXPORT_SYMBOL(snd_hda_build_pcms);
-EXPORT_SYMBOL(snd_hda_build_controls);
-#ifdef CONFIG_PM
-EXPORT_SYMBOL(snd_hda_suspend);
-EXPORT_SYMBOL(snd_hda_resume);
-#endif
-
-/*
  *  INIT part
  */
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index e821d65afa1..4070b5cd9b6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -82,6 +82,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
 			 "{Intel, ICH8},"
 			 "{ATI, SB450},"
 			 "{ATI, SB600},"
+			 "{ATI, RS600},"
 			 "{VIA, VT8251},"
 			 "{VIA, VT8237A},"
 			 "{SiS, SIS966},"
@@ -167,6 +168,12 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define ULI_PLAYBACK_INDEX	5
 #define ULI_NUM_PLAYBACK	6
 
+/* ATI HDMI has 1 playback and 0 capture */
+#define ATIHDMI_CAPTURE_INDEX	0
+#define ATIHDMI_NUM_CAPTURE	0
+#define ATIHDMI_PLAYBACK_INDEX	0
+#define ATIHDMI_NUM_PLAYBACK	1
+
 /* this number is statically defined for simplicity */
 #define MAX_AZX_DEV		16
 
@@ -331,6 +338,7 @@ struct azx {
 enum {
 	AZX_DRIVER_ICH,
 	AZX_DRIVER_ATI,
+	AZX_DRIVER_ATIHDMI,
 	AZX_DRIVER_VIA,
 	AZX_DRIVER_SIS,
 	AZX_DRIVER_ULI,
@@ -340,6 +348,7 @@ enum {
 static char *driver_short_names[] __devinitdata = {
 	[AZX_DRIVER_ICH] = "HDA Intel",
 	[AZX_DRIVER_ATI] = "HDA ATI SB",
+	[AZX_DRIVER_ATIHDMI] = "HDA ATI HDMI",
 	[AZX_DRIVER_VIA] = "HDA VIA VT82xx",
 	[AZX_DRIVER_SIS] = "HDA SIS966",
 	[AZX_DRIVER_ULI] = "HDA ULI M5461",
@@ -1393,10 +1402,10 @@ static int azx_free(struct azx *chip)
 		msleep(1);
 	}
 
-	if (chip->remap_addr)
-		iounmap(chip->remap_addr);
 	if (chip->irq >= 0)
 		free_irq(chip->irq, (void*)chip);
+	if (chip->remap_addr)
+		iounmap(chip->remap_addr);
 
 	if (chip->bdl.area)
 		snd_dma_free_pages(&chip->bdl);
@@ -1495,6 +1504,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 		chip->playback_index_offset = ULI_PLAYBACK_INDEX;
 		chip->capture_index_offset = ULI_CAPTURE_INDEX;
 		break;
+	case AZX_DRIVER_ATIHDMI:
+		chip->playback_streams = ATIHDMI_NUM_PLAYBACK;
+		chip->capture_streams = ATIHDMI_NUM_CAPTURE;
+		chip->playback_index_offset = ATIHDMI_PLAYBACK_INDEX;
+		chip->capture_index_offset = ATIHDMI_CAPTURE_INDEX;
+		break;
 	default:
 		chip->playback_streams = ICH6_NUM_PLAYBACK;
 		chip->capture_streams = ICH6_NUM_CAPTURE;
@@ -1621,6 +1636,7 @@ static struct pci_device_id azx_ids[] __devinitdata = {
 	{ 0x8086, 0x284b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ICH }, /* ICH8 */
 	{ 0x1002, 0x437b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB450 */
 	{ 0x1002, 0x4383, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATI }, /* ATI SB600 */
+	{ 0x1002, 0x793b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS600 HDMI */
 	{ 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */
 	{ 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */
 	{ 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */
diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h
index acaef3c811b..0b668793fac 100644
--- a/sound/pci/hda/hda_patch.h
+++ b/sound/pci/hda/hda_patch.h
@@ -12,6 +12,8 @@ extern struct hda_codec_preset snd_hda_preset_analog[];
 extern struct hda_codec_preset snd_hda_preset_sigmatel[];
 /* SiLabs 3054/3055 modem codecs */
 extern struct hda_codec_preset snd_hda_preset_si3054[];
+/* ATI HDMI codecs */
+extern struct hda_codec_preset snd_hda_preset_atihdmi[];
 
 static const struct hda_codec_preset *hda_preset_tables[] = {
 	snd_hda_preset_realtek,
@@ -19,5 +21,6 @@ static const struct hda_codec_preset *hda_preset_tables[] = {
 	snd_hda_preset_analog,
 	snd_hda_preset_sigmatel,
 	snd_hda_preset_si3054,
+	snd_hda_preset_atihdmi,
 	NULL
 };
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index ca514a6a587..c2f0fe85bf3 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -182,6 +182,10 @@ static void print_pin_caps(struct snd_info_buffer *buffer,
 		snd_iprintf(buffer, " OUT");
 	if (caps & AC_PINCAP_HP_DRV)
 		snd_iprintf(buffer, " HP");
+	if (caps & AC_PINCAP_EAPD)
+		snd_iprintf(buffer, " EAPD");
+	if (caps & AC_PINCAP_PRES_DETECT)
+		snd_iprintf(buffer, " Detect");
 	snd_iprintf(buffer, "\n");
 	caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
 	snd_iprintf(buffer, "  Pin Default 0x%08x: [%s] %s at %s %s\n", caps,
@@ -318,7 +322,7 @@ int snd_hda_codec_proc_new(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	snd_info_set_text_ops(entry, codec, 32 * 1024, print_codec_info);
+	snd_info_set_text_ops(entry, codec, print_codec_info);
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 40f000ba136..dd4e00a82b5 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -789,6 +789,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = {
 	{ .modelname = "3stack",	.config = AD1986A_3STACK },
 	{ .pci_subvendor = 0x10de, .pci_subdevice = 0xcb84,
 	  .config = AD1986A_3STACK }, /* ASUS A8N-VM CSM */
+	{ .pci_subvendor = 0x1043, .pci_subdevice = 0x81b3,
+	  .config = AD1986A_3STACK }, /* ASUS P5RD2-VM / P5GPL-X SE */
 	{ .modelname = "laptop",	.config = AD1986A_LAPTOP },
 	{ .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e,
 	  .config = AD1986A_LAPTOP }, /* FSC V2060 */
@@ -809,6 +811,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = {
 	  .config = AD1986A_LAPTOP_EAPD }, /* ASUS Z62F */
 	{ .pci_subvendor = 0x103c, .pci_subdevice = 0x30af,
 	  .config = AD1986A_LAPTOP_EAPD }, /* HP Compaq Presario B2800 */
+	{ .pci_subvendor = 0x17aa, .pci_subdevice = 0x2066,
+	  .config = AD1986A_LAPTOP_EAPD }, /* Lenovo 3000 N100-07684JU */
 	{}
 };
 
@@ -963,7 +967,7 @@ static struct snd_kcontrol_new ad1983_mixers[] = {
 	},
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route",
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
 		.info = ad1983_spdif_route_info,
 		.get = ad1983_spdif_route_get,
 		.put = ad1983_spdif_route_put,
@@ -1103,7 +1107,7 @@ static struct snd_kcontrol_new ad1981_mixers[] = {
 	/* identical with AD1983 */
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Route",
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
 		.info = ad1983_spdif_route_info,
 		.get = ad1983_spdif_route_get,
 		.put = ad1983_spdif_route_put,
@@ -1329,13 +1333,60 @@ static int ad1981_hp_init(struct hda_codec *codec)
 	return 0;
 }
 
+/* configuration for Lenovo Thinkpad T60 */
+static struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = ad198x_mux_enum_info,
+		.get = ad198x_mux_enum_get,
+		.put = ad198x_mux_enum_put,
+	},
+	/* identical with AD1983 */
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
+		.info = ad1983_spdif_route_info,
+		.get = ad1983_spdif_route_get,
+		.put = ad1983_spdif_route_put,
+	},
+	{ } /* end */
+};
+
+static struct hda_input_mux ad1981_thinkpad_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Mix", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
 /* models */
-enum { AD1981_BASIC, AD1981_HP };
+enum { AD1981_BASIC, AD1981_HP, AD1981_THINKPAD };
 
 static struct hda_board_config ad1981_cfg_tbl[] = {
 	{ .modelname = "hp", .config = AD1981_HP },
 	/* All HP models */
 	{ .pci_subvendor = 0x103c, .config = AD1981_HP },
+	{ .pci_subvendor = 0x30b0, .pci_subdevice = 0x103c,
+	  .config = AD1981_HP }, /* HP nx6320 (reversed SSID, H/W bug) */
+	{ .modelname = "thinkpad", .config = AD1981_THINKPAD },
+	/* Lenovo Thinkpad T60/X60/Z6xx */
+	{ .pci_subvendor = 0x17aa, .config = AD1981_THINKPAD },
+	{ .pci_subvendor = 0x1014, .pci_subdevice = 0x0597,
+	  .config = AD1981_THINKPAD }, /* Z60m/t */
 	{ .modelname = "basic", .config = AD1981_BASIC },
 	{}
 };
@@ -1381,6 +1432,10 @@ static int patch_ad1981(struct hda_codec *codec)
 		codec->patch_ops.init = ad1981_hp_init;
 		codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
 		break;
+	case AD1981_THINKPAD:
+		spec->mixers[0] = ad1981_thinkpad_mixers;
+		spec->input_mux = &ad1981_thinkpad_capture_source;
+		break;
 	}
 
 	return 0;
diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c
new file mode 100644
index 00000000000..a27440ffd1c
--- /dev/null
+++ b/sound/pci/hda/patch_atihdmi.c
@@ -0,0 +1,165 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for ATI HDMI codecs
+ *
+ * Copyright (c) 2006 ATI Technologies Inc.
+ *
+ *
+ *  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 <sound/driver.h>
+#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 atihdmi_spec {
+	struct hda_multi_out multiout;
+
+	struct hda_pcm pcm_rec;
+};
+
+static struct hda_verb atihdmi_basic_init[] = {
+	/* enable digital output on pin widget */
+	{ 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{} /* terminator */
+};
+
+/*
+ * Controls
+ */
+static int atihdmi_build_controls(struct hda_codec *codec)
+{
+	struct atihdmi_spec *spec = codec->spec;
+	int err;
+
+	err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+static int atihdmi_init(struct hda_codec *codec)
+{
+	snd_hda_sequence_write(codec, atihdmi_basic_init);
+	return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+ * resume
+ */
+static int atihdmi_resume(struct hda_codec *codec)
+{
+	atihdmi_init(codec);
+	snd_hda_resume_spdif_out(codec);
+
+	return 0;
+}
+#endif
+
+/*
+ * Digital out
+ */
+static int atihdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				     struct hda_codec *codec,
+				     struct snd_pcm_substream *substream)
+{
+	struct atihdmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int atihdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      struct snd_pcm_substream *substream)
+{
+	struct atihdmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static struct hda_pcm_stream atihdmi_pcm_digital_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0x2, /* NID to query formats and rates and setup streams */
+	.ops = {
+		.open = atihdmi_dig_playback_pcm_open,
+		.close = atihdmi_dig_playback_pcm_close
+	},
+};
+
+static int atihdmi_build_pcms(struct hda_codec *codec)
+{
+	struct atihdmi_spec *spec = codec->spec;
+	struct hda_pcm *info = &spec->pcm_rec;
+
+	codec->num_pcms = 1;
+	codec->pcm_info = info;
+
+	info->name = "ATI HDMI";
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback;
+
+	return 0;
+}
+
+static void atihdmi_free(struct hda_codec *codec)
+{
+	kfree(codec->spec);
+}
+
+static struct hda_codec_ops atihdmi_patch_ops = {
+	.build_controls = atihdmi_build_controls,
+	.build_pcms = atihdmi_build_pcms,
+	.init = atihdmi_init,
+	.free = atihdmi_free,
+#ifdef CONFIG_PM
+	.resume = atihdmi_resume,
+#endif
+};
+
+static int patch_atihdmi(struct hda_codec *codec)
+{
+	struct atihdmi_spec *spec;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	spec->multiout.num_dacs = 0;	  /* no analog */
+	spec->multiout.max_channels = 2;
+	spec->multiout.dig_out_nid = 0x2; /* NID for copying analog to digital,
+					   * seems to be unused in pure-digital
+					   * case. */
+
+	codec->patch_ops = atihdmi_patch_ops;
+
+	return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_atihdmi[] = {
+	{ .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi },
+	{} /* terminator */
+};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f0e9a9c9078..98b9f16c26f 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -2174,6 +2174,7 @@ static struct hda_board_config alc880_cfg_tbl[] = {
 
 	{ .modelname = "lg", .config = ALC880_LG },
 	{ .pci_subvendor = 0x1854, .pci_subdevice = 0x003b, .config = ALC880_LG },
+	{ .pci_subvendor = 0x1854, .pci_subdevice = 0x0068, .config = ALC880_LG },
 
 	{ .modelname = "lg-lw", .config = ALC880_LG_LW },
 	{ .pci_subvendor = 0x1854, .pci_subdevice = 0x0018, .config = ALC880_LG_LW },
@@ -3105,6 +3106,7 @@ static struct hda_verb alc260_init_verbs[] = {
 	{ }
 };
 
+#if 0 /* should be identical with alc260_init_verbs? */
 static struct hda_verb alc260_hp_init_verbs[] = {
 	/* Headphone and output */
 	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
@@ -3151,6 +3153,7 @@ static struct hda_verb alc260_hp_init_verbs[] = {
 	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
 	{ }
 };
+#endif
 
 static struct hda_verb alc260_hp_3013_init_verbs[] = {
 	/* Line out and output */
@@ -3822,12 +3825,16 @@ static struct hda_board_config alc260_cfg_tbl[] = {
 	{ .modelname = "basic", .config = ALC260_BASIC },
 	{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81bb,
 	  .config = ALC260_BASIC }, /* Sony VAIO */
+	{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81cc,
+	  .config = ALC260_BASIC }, /* Sony VAIO VGN-S3HP */
+	{ .pci_subvendor = 0x104d, .pci_subdevice = 0x81cd,
+	  .config = ALC260_BASIC }, /* Sony VAIO */
 	{ .pci_subvendor = 0x152d, .pci_subdevice = 0x0729,
 	  .config = ALC260_BASIC }, /* CTL Travel Master U553W */
 	{ .modelname = "hp", .config = ALC260_HP },
 	{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP },
 	{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP },
-	{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP },
+	{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3012, .config = ALC260_HP_3013 },
 	{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3013, .config = ALC260_HP_3013 },
 	{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3014, .config = ALC260_HP },
 	{ .pci_subvendor = 0x103c, .pci_subdevice = 0x3015, .config = ALC260_HP },
@@ -3862,7 +3869,7 @@ static struct alc_config_preset alc260_presets[] = {
 		.mixers = { alc260_base_output_mixer,
 			    alc260_input_mixer,
 			    alc260_capture_alt_mixer },
-		.init_verbs = { alc260_hp_init_verbs },
+		.init_verbs = { alc260_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
 		.num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids),
@@ -4094,21 +4101,6 @@ static struct snd_kcontrol_new alc882_base_mixer[] = {
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 3,
-		.info = alc882_mux_enum_info,
-		.get = alc882_mux_enum_get,
-		.put = alc882_mux_enum_put,
-	},
 	{ } /* end */
 };
 
@@ -4342,8 +4334,6 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.dig_out_nid = ALC882_DIGOUT_NID,
-		.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
-		.adc_nids = alc882_adc_nids,
 		.dig_in_nid = ALC882_DIGIN_NID,
 		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
 		.channel_mode = alc882_ch_modes,
@@ -4355,8 +4345,6 @@ static struct alc_config_preset alc882_presets[] = {
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.dig_out_nid = ALC882_DIGOUT_NID,
-		.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
-		.adc_nids = alc882_adc_nids,
 		.dig_in_nid = ALC882_DIGIN_NID,
 		.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
 		.channel_mode = alc882_sixstack_modes,
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8c440fb9860..36f199442fd 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -41,6 +41,7 @@
 #define STAC_REF		0
 #define STAC_D945GTP3		1
 #define STAC_D945GTP5		2
+#define STAC_MACMINI		3
 
 struct sigmatel_spec {
 	struct snd_kcontrol_new *mixers[4];
@@ -52,6 +53,7 @@ struct sigmatel_spec {
 	unsigned int mic_switch: 1;
 	unsigned int alt_switch: 1;
 	unsigned int hp_detect: 1;
+	unsigned int gpio_mute: 1;
 
 	/* playback */
 	struct hda_multi_out multiout;
@@ -293,6 +295,7 @@ static unsigned int *stac922x_brd_tbl[] = {
 	ref922x_pin_configs,
 	d945gtp3_pin_configs,
 	d945gtp5_pin_configs,
+	NULL,		/* STAC_MACMINI */
 };
 
 static struct hda_board_config stac922x_cfg_tbl[] = {
@@ -324,6 +327,9 @@ static struct hda_board_config stac922x_cfg_tbl[] = {
 	{ .pci_subvendor = PCI_VENDOR_ID_INTEL,
 	  .pci_subdevice = 0x0417,
 	  .config = STAC_D945GTP5 },	/* Intel D975XBK - 5 Stack */
+	{ .pci_subvendor = 0x8384,
+	  .pci_subdevice = 0x7680,
+	  .config = STAC_MACMINI },	/* Apple Mac Mini (early 2006) */
 	{} /* terminator */
 };
 
@@ -841,6 +847,19 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const
 		}
 	}
 
+	if (imux->num_items == 1) {
+		/*
+		 * Set the current input for the muxes.
+		 * The STAC9221 has two input muxes with identical source
+		 * NID lists.  Hopefully this won't get confused.
+		 */
+		for (i = 0; i < spec->num_muxes; i++) {
+			snd_hda_codec_write(codec, spec->mux_nids[i], 0,
+					    AC_VERB_SET_CONNECT_SEL,
+					    imux->items[0].index);
+		}
+	}
+
 	return 0;
 }
 
@@ -946,6 +965,45 @@ static int stac9200_parse_auto_config(struct hda_codec *codec)
 	return 1;
 }
 
+/*
+ * Early 2006 Intel Macintoshes with STAC9220X5 codecs seem to have a
+ * funky external mute control using GPIO pins.
+ */
+
+static void stac922x_gpio_mute(struct hda_codec *codec, int pin, int muted)
+{
+	unsigned int gpiostate, gpiomask, gpiodir;
+
+	gpiostate = snd_hda_codec_read(codec, codec->afg, 0,
+				       AC_VERB_GET_GPIO_DATA, 0);
+
+	if (!muted)
+		gpiostate |= (1 << pin);
+	else
+		gpiostate &= ~(1 << pin);
+
+	gpiomask = snd_hda_codec_read(codec, codec->afg, 0,
+				      AC_VERB_GET_GPIO_MASK, 0);
+	gpiomask |= (1 << pin);
+
+	gpiodir = snd_hda_codec_read(codec, codec->afg, 0,
+				     AC_VERB_GET_GPIO_DIRECTION, 0);
+	gpiodir |= (1 << pin);
+
+	/* AppleHDA seems to do this -- WTF is this verb?? */
+	snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0);
+
+	snd_hda_codec_write(codec, codec->afg, 0,
+			    AC_VERB_SET_GPIO_MASK, gpiomask);
+	snd_hda_codec_write(codec, codec->afg, 0,
+			    AC_VERB_SET_GPIO_DIRECTION, gpiodir);
+
+	msleep(1);
+
+	snd_hda_codec_write(codec, codec->afg, 0,
+			    AC_VERB_SET_GPIO_DATA, gpiostate);
+}
+
 static int stac92xx_init(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
@@ -982,6 +1040,11 @@ static int stac92xx_init(struct hda_codec *codec)
 		stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin,
 					 AC_PINCTL_IN_EN);
 
+	if (spec->gpio_mute) {
+		stac922x_gpio_mute(codec, 0, 0);
+		stac922x_gpio_mute(codec, 1, 0);
+	}
+
 	return 0;
 }
 
@@ -1132,7 +1195,7 @@ static int patch_stac922x(struct hda_codec *codec)
 	spec->board_config = snd_hda_check_board_config(codec, stac922x_cfg_tbl);
 	if (spec->board_config < 0)
                 snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, using BIOS defaults\n");
-	else {
+	else if (stac922x_brd_tbl[spec->board_config] != NULL) {
 		spec->num_pins = 10;
 		spec->pin_nids = stac922x_pin_nids;
 		spec->pin_configs = stac922x_brd_tbl[spec->board_config];
@@ -1154,6 +1217,9 @@ static int patch_stac922x(struct hda_codec *codec)
 		return err;
 	}
 
+	if (spec->board_config == STAC_MACMINI)
+		spec->gpio_mute = 1;
+
 	codec->patch_ops = stac92xx_patch_ops;
 
 	return 0;
@@ -1262,13 +1328,13 @@ static int vaio_master_sw_put(struct snd_kcontrol *kcontrol,
 	int change;
 
 	change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0,
-					  0x80, valp[0] & 0x80);
+					  0x80, (valp[0] ? 0 : 0x80));
 	change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0,
-					   0x80, valp[1] & 0x80);
+					   0x80, (valp[1] ? 0 : 0x80));
 	snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0,
-				 0x80, valp[0] & 0x80);
+				 0x80, (valp[0] ? 0 : 0x80));
 	snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0,
-				 0x80, valp[1] & 0x80);
+				 0x80, (valp[1] ? 0 : 0x80));
 	return change;
 }
 
@@ -1370,6 +1436,12 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = {
  	{ .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x },
  	{ .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x },
  	{ .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x },
+ 	{ .id = 0x83847618, .name = "STAC9227", .patch = patch_stac922x },
+ 	{ .id = 0x83847619, .name = "STAC9227", .patch = patch_stac922x },
+ 	{ .id = 0x83847616, .name = "STAC9228", .patch = patch_stac922x },
+ 	{ .id = 0x83847617, .name = "STAC9228", .patch = patch_stac922x },
+ 	{ .id = 0x83847614, .name = "STAC9229", .patch = patch_stac922x },
+ 	{ .id = 0x83847615, .name = "STAC9229", .patch = patch_stac922x },
  	{ .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x },
  	{ .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x },
  	{ .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x },
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 336dc489aee..ca74f5b85f4 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -1281,9 +1281,15 @@ static int aureon_set_headphone_amp(struct snd_ice1712 *ice, int enable)
 
 	tmp2 = tmp = snd_ice1712_gpio_read(ice);
 	if (enable)
-		tmp |= AUREON_HP_SEL;
+		if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT)
+			tmp |= AUREON_HP_SEL;
+		else
+			tmp |= PRODIGY_HP_SEL;
 	else
-		tmp &= ~ AUREON_HP_SEL;
+		if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT)
+			tmp &= ~ AUREON_HP_SEL;
+		else
+			tmp &= ~ PRODIGY_HP_SEL;
 	if (tmp != tmp2) {
 		snd_ice1712_gpio_write(ice, tmp);
 		return 1;
@@ -2079,16 +2085,16 @@ static unsigned char prodigy71_eeprom[] __devinitdata = {
 };
 
 static unsigned char prodigy71lt_eeprom[] __devinitdata = {
-	0x0b,	/* SYSCINF: clock 512, spdif-in/ADC, 4DACs */
+	0x4b,	/* SYSCINF: clock 512, spdif-in/ADC, 4DACs */
 	0x80,	/* ACLINK: I2S */
 	0xfc,	/* I2S: vol, 96k, 24bit, 192k */
-	0xc3,	/* SPDUF: out-en, out-int */
-	0x00,	/* GPIO_DIR */
-	0x07,	/* GPIO_DIR1 */
-	0x00,	/* GPIO_DIR2 */
-	0xff,	/* GPIO_MASK */
-	0xf8,	/* GPIO_MASK1 */
-	0xff,	/* GPIO_MASK2 */
+	0xc3,	/* SPDIF: out-en, out-int, spdif-in */
+	0xff,	/* GPIO_DIR */
+	0xff,	/* GPIO_DIR1 */
+	0x5f,	/* GPIO_DIR2 */
+	0x00,	/* GPIO_MASK */
+	0x00,	/* GPIO_MASK1 */
+	0x00,	/* GPIO_MASK2 */
 	0x00,	/* GPIO_STATE */
 	0x00,	/* GPIO_STATE1 */
 	0x00,	/* GPIO_STATE2 */
diff --git a/sound/pci/ice1712/aureon.h b/sound/pci/ice1712/aureon.h
index 98a6752280f..3b7bea656c5 100644
--- a/sound/pci/ice1712/aureon.h
+++ b/sound/pci/ice1712/aureon.h
@@ -58,5 +58,6 @@ extern struct snd_ice1712_card_info  snd_vt1724_aureon_cards[];
 #define PRODIGY_WM_CS		(1 << 8)
 #define PRODIGY_SPI_MOSI	(1 << 10)
 #define PRODIGY_SPI_CLK		(1 << 9)
+#define PRODIGY_HP_SEL		(1 << 5)
 
 #endif /* __SOUND_AUREON_H */
diff --git a/sound/pci/ice1712/ews.c b/sound/pci/ice1712/ews.c
index 2c529e74138..b135389fec6 100644
--- a/sound/pci/ice1712/ews.c
+++ b/sound/pci/ice1712/ews.c
@@ -1031,6 +1031,9 @@ struct snd_ice1712_card_info snd_ice1712_ews_cards[] __devinitdata = {
 		.model = "dmx6fire",
 		.chip_init = snd_ice1712_ews_init,
 		.build_controls = snd_ice1712_ews_add_controls,
+		.mpu401_1_name = "MIDI-Front DMX6fire",
+		.mpu401_2_name = "Wavetable DMX6fire",
+		.mpu401_2_info_flags = MPU401_INFO_OUTPUT,
 	},
 	{ } /* terminator */
 };
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index c56793b381e..845907159b7 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -61,7 +61,6 @@
 #include <sound/core.h>
 #include <sound/cs8427.h>
 #include <sound/info.h>
-#include <sound/mpu401.h>
 #include <sound/initval.h>
 
 #include <sound/asoundef.h>
@@ -1596,7 +1595,7 @@ static void __devinit snd_ice1712_proc_init(struct snd_ice1712 * ice)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(ice->card, "ice1712", &entry))
-		snd_info_set_text_ops(entry, ice, 1024, snd_ice1712_proc_read);
+		snd_info_set_text_ops(entry, ice, snd_ice1712_proc_read);
 }
 
 /*
@@ -2398,13 +2397,14 @@ static int __devinit snd_ice1712_chip_init(struct snd_ice1712 *ice)
 	udelay(200);
 	outb(ICE1712_NATIVE, ICEREG(ice, CONTROL));
 	udelay(200);
-	if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DMX6FIRE && !ice->dxr_enable) {
-                /* Limit active ADCs and DACs to 6;  */
-                /* Note: DXR extension not supported */
-		pci_write_config_byte(ice->pci, 0x60, 0x2a);
-	} else {
-		pci_write_config_byte(ice->pci, 0x60, ice->eeprom.data[ICE_EEP1_CODEC]);
-	}
+	if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DMX6FIRE &&
+	    !ice->dxr_enable)
+		/*  Set eeprom value to limit active ADCs and DACs to 6;
+		 *  Also disable AC97 as no hardware in standard 6fire card/box
+		 *  Note: DXR extensions are not currently supported
+		 */
+		ice->eeprom.data[ICE_EEP1_CODEC] = 0x3a;
+	pci_write_config_byte(ice->pci, 0x60, ice->eeprom.data[ICE_EEP1_CODEC]);
 	pci_write_config_byte(ice->pci, 0x61, ice->eeprom.data[ICE_EEP1_ACLINK]);
 	pci_write_config_byte(ice->pci, 0x62, ice->eeprom.data[ICE_EEP1_I2SID]);
 	pci_write_config_byte(ice->pci, 0x63, ice->eeprom.data[ICE_EEP1_SPDIF]);
@@ -2737,21 +2737,38 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 
 	if (! c->no_mpu401) {
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
-					       ICEREG(ice, MPU1_CTRL), 1,
+					       ICEREG(ice, MPU1_CTRL),
+					       (c->mpu401_1_info_flags |
+						MPU401_INFO_INTEGRATED),
 					       ice->irq, 0,
 					       &ice->rmidi[0])) < 0) {
 			snd_card_free(card);
 			return err;
 		}
-
-		if (ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401)
+		if (c->mpu401_1_name)
+			/*  Prefered name available in card_info */
+			snprintf(ice->rmidi[0]->name,
+				 sizeof(ice->rmidi[0]->name),
+				 "%s %d", c->mpu401_1_name, card->number);
+
+		if (ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) {
+			/*  2nd port used  */
 			if ((err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712,
-						       ICEREG(ice, MPU2_CTRL), 1,
+						       ICEREG(ice, MPU2_CTRL),
+						       (c->mpu401_2_info_flags |
+							MPU401_INFO_INTEGRATED),
 						       ice->irq, 0,
 						       &ice->rmidi[1])) < 0) {
 				snd_card_free(card);
 				return err;
 			}
+			if (c->mpu401_2_name)
+				/*  Prefered name available in card_info */
+				snprintf(ice->rmidi[1]->name,
+					 sizeof(ice->rmidi[1]->name),
+					 "%s %d", c->mpu401_2_name,
+					 card->number);
+		}
 	}
 
 	snd_ice1712_set_input_clock_source(ice, 0);
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 053f8e56fd6..ce27eac40d4 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -29,6 +29,7 @@
 #include <sound/ak4xxx-adda.h>
 #include <sound/ak4114.h>
 #include <sound/pcm.h>
+#include <sound/mpu401.h>
 
 
 /*
@@ -495,6 +496,10 @@ struct snd_ice1712_card_info {
 	int (*chip_init)(struct snd_ice1712 *);
 	int (*build_controls)(struct snd_ice1712 *);
 	unsigned int no_mpu401: 1;
+	unsigned int mpu401_1_info_flags;
+	unsigned int mpu401_2_info_flags;
+	const char *mpu401_1_name;
+	const char *mpu401_2_name;
 	unsigned int eeprom_size;
 	unsigned char *eeprom_data;
 };
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index b1c007e022d..34a58c629f4 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -1293,7 +1293,7 @@ static void __devinit snd_vt1724_proc_init(struct snd_ice1712 * ice)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(ice->card, "ice1724", &entry))
-		snd_info_set_text_ops(entry, ice, 1024, snd_vt1724_proc_read);
+		snd_info_set_text_ops(entry, ice, snd_vt1724_proc_read);
 }
 
 /*
@@ -2388,7 +2388,8 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
 	if (! c->no_mpu401) {
 		if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) {
 			if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712,
-						       ICEREG1724(ice, MPU_CTRL), 1,
+						       ICEREG1724(ice, MPU_CTRL),
+						       MPU401_INFO_INTEGRATED,
 						       ice->irq, 0,
 						       &ice->rmidi[0])) < 0) {
 				snd_card_free(card);
diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c
index d23fb3fc213..0efcad9260a 100644
--- a/sound/pci/ice1712/pontis.c
+++ b/sound/pci/ice1712/pontis.c
@@ -680,9 +680,8 @@ static void wm_proc_init(struct snd_ice1712 *ice)
 {
 	struct snd_info_entry *entry;
 	if (! snd_card_proc_new(ice->card, "wm_codec", &entry)) {
-		snd_info_set_text_ops(entry, ice, 1024, wm_proc_regs_read);
+		snd_info_set_text_ops(entry, ice, wm_proc_regs_read);
 		entry->mode |= S_IWUSR;
-		entry->c.text.write_size = 1024;
 		entry->c.text.write = wm_proc_regs_write;
 	}
 }
@@ -705,9 +704,8 @@ static void cs_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buff
 static void cs_proc_init(struct snd_ice1712 *ice)
 {
 	struct snd_info_entry *entry;
-	if (! snd_card_proc_new(ice->card, "cs_codec", &entry)) {
-		snd_info_set_text_ops(entry, ice, 1024, cs_proc_regs_read);
-	}
+	if (! snd_card_proc_new(ice->card, "cs_codec", &entry))
+		snd_info_set_text_ops(entry, ice, cs_proc_regs_read);
 }
 
 
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 0df7602568e..edc14475ef8 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -66,7 +66,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH},"
 
 static int index = SNDRV_DEFAULT_IDX1;	/* Index 0-MAX */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
-static int ac97_clock = 0;
+static int ac97_clock;
 static char *ac97_quirk;
 static int buggy_semaphore;
 static int buggy_irq = -1; /* auto-check */
@@ -1807,6 +1807,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 	},
 	{
 		.subvendor = 0x1028,
+		.subdevice = 0x014e,
+		.name = "Dell D800", /* STAC9750/51 */
+		.type = AC97_TUNE_HP_ONLY
+	},
+	{
+		.subvendor = 0x1028,
 		.subdevice = 0x0163,
 		.name = "Dell Unknown",	/* STAC9750/51 */
 		.type = AC97_TUNE_HP_ONLY
@@ -2645,7 +2651,7 @@ static void __devinit snd_intel8x0_proc_init(struct intel8x0 * chip)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(chip->card, "intel8x0", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, snd_intel8x0_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_intel8x0_proc_read);
 }
 #else
 #define snd_intel8x0_proc_init(x)
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 720635f0cb8..24703d75b65 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -59,7 +59,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel,82801AA-ICH},"
 
 static int index = -2; /* Exclude the first card */
 static char *id = SNDRV_DEFAULT_STR1;	/* ID for this card */
-static int ac97_clock = 0;
+static int ac97_clock;
 
 module_param(index, int, 0444);
 MODULE_PARM_DESC(index, "Index value for Intel i8x0 modemcard.");
@@ -1092,7 +1092,7 @@ static void __devinit snd_intel8x0m_proc_init(struct intel8x0m * chip)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(chip->card, "intel8x0m", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, snd_intel8x0m_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_intel8x0m_proc_read);
 }
 #else /* !CONFIG_PROC_FS */
 #define snd_intel8x0m_proc_init(chip)
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index e39fad1a420..6e97932de34 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2085,7 +2085,7 @@ static void __devinit snd_korg1212_proc_init(struct snd_korg1212 *korg1212)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(korg1212->card, "korg1212", &entry))
-		snd_info_set_text_ops(entry, korg1212, 1024, snd_korg1212_proc_read);
+		snd_info_set_text_ops(entry, korg1212, snd_korg1212_proc_read);
 }
 
 static int
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 1928e06b6d8..1c344fbd964 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2861,7 +2861,8 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 #if 0 /* TODO: not supported yet */
 	/* TODO enable MIDI IRQ and I/O */
 	err = snd_mpu401_uart_new(chip->card, 0, MPU401_HW_MPU401,
-				  chip->iobase + MPU401_DATA_PORT, 1,
+				  chip->iobase + MPU401_DATA_PORT,
+				  MPU401_INFO_INTEGRATED,
 				  chip->irq, 0, &chip->rmidi);
 	if (err < 0)
 		printk(KERN_WARNING "maestro3: no MIDI support.\n");
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index 09cc0786495..366c4a7e65c 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -1244,7 +1244,6 @@ static void __devinit snd_mixart_proc_init(struct snd_mixart *chip)
 	/* text interface to read perf and temp meters */
 	if (! snd_card_proc_new(chip->card, "board_info", &entry)) {
 		entry->private_data = chip;
-		entry->c.text.read_size = 1024;
 		entry->c.text.read = snd_mixart_proc_read;
 	}
 
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index dafa2235aba..8198884b51e 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1150,9 +1150,9 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(chip->card, "info", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, pcxhr_proc_info);
+		snd_info_set_text_ops(entry, chip, pcxhr_proc_info);
 	if (! snd_card_proc_new(chip->card, "sync", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, pcxhr_proc_sync);
+		snd_info_set_text_ops(entry, chip, pcxhr_proc_sync);
 }
 /* end of proc interface */
 
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index d8cc985d724..5618ec9740b 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1836,11 +1836,11 @@ static int snd_riptide_free(struct snd_riptide *chip)
 		UNSET_GRESET(cif->hwport);
 		kfree(chip->cif);
 	}
+	if (chip->irq >= 0)
+		free_irq(chip->irq, chip);
 	if (chip->fw_entry)
 		release_firmware(chip->fw_entry);
 	release_and_free_resource(chip->res_port);
-	if (chip->irq >= 0)
-		free_irq(chip->irq, chip);
 	kfree(chip);
 	return 0;
 }
@@ -1992,7 +1992,7 @@ static void __devinit snd_riptide_proc_init(struct snd_riptide *chip)
 	struct snd_info_entry *entry;
 
 	if (!snd_card_proc_new(chip->card, "riptide", &entry))
-		snd_info_set_text_ops(entry, chip, 4096, snd_riptide_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_riptide_proc_read);
 }
 
 static int __devinit snd_riptide_mixer(struct snd_riptide *chip)
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index 55b1d4838d9..2cb9fe98db2 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -1368,18 +1368,18 @@ static int __devinit snd_rme32_create(struct rme32 * rme32)
 		return err;
 	rme32->port = pci_resource_start(rme32->pci, 0);
 
-	if (request_irq(pci->irq, snd_rme32_interrupt, SA_INTERRUPT | SA_SHIRQ, "RME32", (void *) rme32)) {
-		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
-		return -EBUSY;
-	}
-	rme32->irq = pci->irq;
-
 	if ((rme32->iobase = ioremap_nocache(rme32->port, RME32_IO_SIZE)) == 0) {
 		snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n",
 			   rme32->port, rme32->port + RME32_IO_SIZE - 1);
 		return -ENOMEM;
 	}
 
+	if (request_irq(pci->irq, snd_rme32_interrupt, SA_INTERRUPT | SA_SHIRQ, "RME32", (void *) rme32)) {
+		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
+		return -EBUSY;
+	}
+	rme32->irq = pci->irq;
+
 	/* read the card's revision number */
 	pci_read_config_byte(pci, 8, &rme32->rev);
 
@@ -1578,7 +1578,7 @@ static void __devinit snd_rme32_proc_init(struct rme32 * rme32)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(rme32->card, "rme32", &entry))
-		snd_info_set_text_ops(entry, rme32, 1024, snd_rme32_proc_read);
+		snd_info_set_text_ops(entry, rme32, snd_rme32_proc_read);
 }
 
 /*
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 3c1bc533d51..991cb18c14f 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -1151,6 +1151,25 @@ static struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = {
 	.mask = 0
 };
 
+static void
+rme96_set_buffer_size_constraint(struct rme96 *rme96,
+				 struct snd_pcm_runtime *runtime)
+{
+	unsigned int size;
+
+	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+				     RME96_BUFFER_SIZE, RME96_BUFFER_SIZE);
+	if ((size = rme96->playback_periodsize) != 0 ||
+	    (size = rme96->capture_periodsize) != 0)
+		snd_pcm_hw_constraint_minmax(runtime,
+					     SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+					     size, size);
+	else
+		snd_pcm_hw_constraint_list(runtime, 0,
+					   SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+					   &hw_constraints_period_bytes);
+}
+
 static int
 snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
 {
@@ -1180,8 +1199,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
                 runtime->hw.rate_min = rate;
                 runtime->hw.rate_max = rate;
 	}        
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE);
-	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes);
+	rme96_set_buffer_size_constraint(rme96, runtime);
 
 	rme96->wcreg_spdif_stream = rme96->wcreg_spdif;
 	rme96->spdif_ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
@@ -1219,9 +1237,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream)
 	rme96->capture_substream = substream;
 	spin_unlock_irq(&rme96->lock);
 	
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE);
-	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes);
-
+	rme96_set_buffer_size_constraint(rme96, runtime);
 	return 0;
 }
 
@@ -1254,8 +1270,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream)
                 runtime->hw.rate_min = rate;
                 runtime->hw.rate_max = rate;
 	}        
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE);
-	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes);
+	rme96_set_buffer_size_constraint(rme96, runtime);
 	return 0;
 }
 
@@ -1291,8 +1306,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream)
 	rme96->capture_substream = substream;
 	spin_unlock_irq(&rme96->lock);
 
-	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, RME96_BUFFER_SIZE, RME96_BUFFER_SIZE);
-	snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_bytes);
+	rme96_set_buffer_size_constraint(rme96, runtime);
 	return 0;
 }
 
@@ -1569,17 +1583,17 @@ snd_rme96_create(struct rme96 *rme96)
 		return err;
 	rme96->port = pci_resource_start(rme96->pci, 0);
 
+	if ((rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE)) == 0) {
+		snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme96->port, rme96->port + RME96_IO_SIZE - 1);
+		return -ENOMEM;
+	}
+
 	if (request_irq(pci->irq, snd_rme96_interrupt, SA_INTERRUPT|SA_SHIRQ, "RME96", (void *)rme96)) {
 		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		return -EBUSY;
 	}
 	rme96->irq = pci->irq;
 
-	if ((rme96->iobase = ioremap_nocache(rme96->port, RME96_IO_SIZE)) == 0) {
-		snd_printk(KERN_ERR "unable to remap memory region 0x%lx-0x%lx\n", rme96->port, rme96->port + RME96_IO_SIZE - 1);
-		return -ENOMEM;
-	}
-
 	/* read the card's revision number */
 	pci_read_config_byte(pci, 8, &rme96->rev);	
 	
@@ -1805,7 +1819,7 @@ snd_rme96_proc_init(struct rme96 *rme96)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(rme96->card, "rme96", &entry))
-		snd_info_set_text_ops(entry, rme96, 1024, snd_rme96_proc_read);
+		snd_info_set_text_ops(entry, rme96, snd_rme96_proc_read);
 }
 
 /*
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 61f82f0d5cc..eaf3c22449a 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -389,7 +389,7 @@ MODULE_SUPPORTED_DEVICE("{{RME Hammerfall-DSP},"
 
 /* use hotplug firmeare loader? */
 #if defined(CONFIG_FW_LOADER) || defined(CONFIG_FW_LOADER_MODULE)
-#ifndef HDSP_USE_HWDEP_LOADER
+#if !defined(HDSP_USE_HWDEP_LOADER) && !defined(CONFIG_SND_HDSP)
 #define HDSP_FW_LOADER
 #endif
 #endif
@@ -3169,9 +3169,10 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 	char *clock_source;
 	int x;
 
-	if (hdsp_check_for_iobox (hdsp))
+	if (hdsp_check_for_iobox (hdsp)) {
 		snd_iprintf(buffer, "No I/O box connected.\nPlease connect one and upload firmware.\n");
 		return;
+        }
 
 	if (hdsp_check_for_firmware(hdsp, 0)) {
 		if (hdsp->state & HDSP_FirmwareCached) {
@@ -3470,7 +3471,7 @@ static void __devinit snd_hdsp_proc_init(struct hdsp *hdsp)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(hdsp->card, "hdsp", &entry))
-		snd_info_set_text_ops(entry, hdsp, 1024, snd_hdsp_proc_read);
+		snd_info_set_text_ops(entry, hdsp, snd_hdsp_proc_read);
 }
 
 static void snd_hdsp_free_buffers(struct hdsp *hdsp)
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 722b9e6ce54..bba1615504d 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -2489,7 +2489,7 @@ static void __devinit snd_hdspm_proc_init(struct hdspm * hdspm)
 	struct snd_info_entry *entry;
 
 	if (!snd_card_proc_new(hdspm->card, "hdspm", &entry))
-		snd_info_set_text_ops(entry, hdspm, 1024,
+		snd_info_set_text_ops(entry, hdspm,
 				      snd_hdspm_proc_read);
 }
 
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index 75d6406303d..3b945e8c1b1 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -41,7 +41,7 @@
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
 static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card */
-static int precise_ptr[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 0 }; /* Enable precise pointer */
+static int precise_ptr[SNDRV_CARDS];			/* Enable precise pointer */
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for RME Digi9652 (Hammerfall) soundcard.");
@@ -1787,7 +1787,7 @@ static void __devinit snd_rme9652_proc_init(struct snd_rme9652 *rme9652)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(rme9652->card, "rme9652", &entry))
-		snd_info_set_text_ops(entry, rme9652, 1024, snd_rme9652_proc_read);
+		snd_info_set_text_ops(entry, rme9652, snd_rme9652_proc_read);
 }
 
 static void snd_rme9652_free_buffers(struct snd_rme9652 *rme9652)
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index 91f8bf3ae9f..dcf40294834 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -54,8 +54,8 @@ MODULE_SUPPORTED_DEVICE("{{S3,SonicVibes PCI}}");
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
 static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card */
-static int reverb[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0};
-static int mge[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 0};
+static int reverb[SNDRV_CARDS];
+static int mge[SNDRV_CARDS];
 static unsigned int dmaio = 0x7a00;	/* DDMA i/o address */
 
 module_param_array(index, int, NULL, 0444);
@@ -1144,7 +1144,7 @@ static void __devinit snd_sonicvibes_proc_init(struct sonicvibes * sonic)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(sonic->card, "sonicvibes", &entry))
-		snd_info_set_text_ops(entry, sonic, 1024, snd_sonicvibes_proc_read);
+		snd_info_set_text_ops(entry, sonic, snd_sonicvibes_proc_read);
 }
 
 /*
@@ -1456,7 +1456,7 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci,
 		return err;
 	}
 	if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES,
-				       sonic->midi_port, 1,
+				       sonic->midi_port, MPU401_INFO_INTEGRATED,
 				       sonic->irq, 0,
 				       &midi_uart)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index 9624a5f2b87..5629b7eba96 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -148,7 +148,8 @@ static int __devinit snd_trident_probe(struct pci_dev *pci,
 	}
 	if (trident->device != TRIDENT_DEVICE_ID_SI7018 &&
 	    (err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE,
-				       trident->midi_port, 1,
+				       trident->midi_port,
+				       MPU401_INFO_INTEGRATED,
 				       trident->irq, 0, &trident->rmidi)) < 0) {
 		snd_card_free(card);
 		return err;
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index 52178b8ad49..d99ed723775 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -306,6 +306,8 @@ void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice)
 	outl(mask, TRID_REG(trident, reg));
 }
 
+EXPORT_SYMBOL(snd_trident_start_voice);
+
 /*---------------------------------------------------------------------------
    void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice)
 
@@ -328,6 +330,8 @@ void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice)
 	outl(mask, TRID_REG(trident, reg));
 }
 
+EXPORT_SYMBOL(snd_trident_stop_voice);
+
 /*---------------------------------------------------------------------------
     int snd_trident_allocate_pcm_channel(struct snd_trident *trident)
   
@@ -502,6 +506,8 @@ void snd_trident_write_voice_regs(struct snd_trident * trident,
 #endif
 }
 
+EXPORT_SYMBOL(snd_trident_write_voice_regs);
+
 /*---------------------------------------------------------------------------
    snd_trident_write_cso_reg
   
@@ -3332,7 +3338,7 @@ static void __devinit snd_trident_proc_init(struct snd_trident * trident)
 	if (trident->device == TRIDENT_DEVICE_ID_SI7018)
 		s = "sis7018";
 	if (! snd_card_proc_new(trident->card, s, &entry))
-		snd_info_set_text_ops(entry, trident, 1024, snd_trident_proc_read);
+		snd_info_set_text_ops(entry, trident, snd_trident_proc_read);
 }
 
 static int snd_trident_dev_free(struct snd_device *device)
@@ -3884,6 +3890,8 @@ struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident,
 	return NULL;
 }
 
+EXPORT_SYMBOL(snd_trident_alloc_voice);
+
 void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voice *voice)
 {
 	unsigned long flags;
@@ -3912,6 +3920,8 @@ void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voi
 		private_free(voice);
 }
 
+EXPORT_SYMBOL(snd_trident_free_voice);
+
 static void snd_trident_clear_voices(struct snd_trident * trident, unsigned short v_min, unsigned short v_max)
 {
 	unsigned int i, val, mask[2] = { 0, 0 };
@@ -3993,13 +4003,3 @@ int snd_trident_resume(struct pci_dev *pci)
 	return 0;
 }
 #endif /* CONFIG_PM */
-
-EXPORT_SYMBOL(snd_trident_alloc_voice);
-EXPORT_SYMBOL(snd_trident_free_voice);
-EXPORT_SYMBOL(snd_trident_start_voice);
-EXPORT_SYMBOL(snd_trident_stop_voice);
-EXPORT_SYMBOL(snd_trident_write_voice_regs);
-/* trident_memory.c symbols */
-EXPORT_SYMBOL(snd_trident_synth_alloc);
-EXPORT_SYMBOL(snd_trident_synth_free);
-EXPORT_SYMBOL(snd_trident_synth_copy_from_user);
diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c
index 46c6982c9e8..aff3f874131 100644
--- a/sound/pci/trident/trident_memory.c
+++ b/sound/pci/trident/trident_memory.c
@@ -349,6 +349,7 @@ snd_trident_synth_alloc(struct snd_trident *hw, unsigned int size)
 	return blk;
 }
 
+EXPORT_SYMBOL(snd_trident_synth_alloc);
 
 /*
  * free a synth sample area
@@ -365,6 +366,7 @@ snd_trident_synth_free(struct snd_trident *hw, struct snd_util_memblk *blk)
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_trident_synth_free);
 
 /*
  * reset TLB entry and free kernel page
@@ -486,3 +488,4 @@ int snd_trident_synth_copy_from_user(struct snd_trident *trident,
 	return 0;
 }
 
+EXPORT_SYMBOL(snd_trident_synth_copy_from_user);
diff --git a/sound/pci/trident/trident_synth.c b/sound/pci/trident/trident_synth.c
index cc7af8bc55a..9b7dee84743 100644
--- a/sound/pci/trident/trident_synth.c
+++ b/sound/pci/trident/trident_synth.c
@@ -914,7 +914,9 @@ static int snd_trident_synth_create_port(struct snd_trident * trident, int idx)
 						   &callbacks,
 						   SNDRV_SEQ_PORT_CAP_WRITE | SNDRV_SEQ_PORT_CAP_SUBS_WRITE,
 						   SNDRV_SEQ_PORT_TYPE_DIRECT_SAMPLE |
-						   SNDRV_SEQ_PORT_TYPE_SYNTH,
+						   SNDRV_SEQ_PORT_TYPE_SYNTH |
+						   SNDRV_SEQ_PORT_TYPE_HARDWARE |
+						   SNDRV_SEQ_PORT_TYPE_SYNTHESIZER,
 						   16, 0,
 						   name);
 	if (p->chset->port < 0) {
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 39daf62d2ba..2527bbd958c 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -1775,6 +1775,12 @@ static struct ac97_quirk ac97_quirks[] = {
 		.name = "Targa Traveller 811",
 		.type = AC97_TUNE_HP_ONLY,
 	},
+	{
+		.subvendor = 0x161f,
+		.subdevice = 0x2032,
+		.name = "m680x",
+		.type = AC97_TUNE_HP_ONLY, /* http://launchpad.net/bugs/38546 */
+	},
 	{ } /* terminator */
 };
 
@@ -1973,7 +1979,7 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip)
 	pci_write_config_byte(chip->pci, VIA_PNP_CONTROL, legacy_cfg);
 	if (chip->mpu_res) {
 		if (snd_mpu401_uart_new(chip->card, 0, MPU401_HW_VIA686A,
-					mpu_port, 1,
+					mpu_port, MPU401_INFO_INTEGRATED,
 					chip->irq, 0, &chip->rmidi) < 0) {
 			printk(KERN_WARNING "unable to initialize MPU-401"
 			       " at 0x%lx, skipping\n", mpu_port);
@@ -2015,7 +2021,7 @@ static void __devinit snd_via82xx_proc_init(struct via82xx *chip)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(chip->card, "via82xx", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, snd_via82xx_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read);
 }
 
 /*
@@ -2365,7 +2371,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci, int revision)
 		{ .subvendor = 0x1462, .subdevice = 0x0470, .action = VIA_DXS_SRC }, /* MSI KT880 Delta-FSR */
 		{ .subvendor = 0x1462, .subdevice = 0x3800, .action = VIA_DXS_ENABLE }, /* MSI KT266 */
 		{ .subvendor = 0x1462, .subdevice = 0x5901, .action = VIA_DXS_NO_VRA }, /* MSI KT6 Delta-SR */
-		{ .subvendor = 0x1462, .subdevice = 0x7023, .action = VIA_DXS_NO_VRA }, /* MSI K8T Neo2-FI */
+		{ .subvendor = 0x1462, .subdevice = 0x7023, .action = VIA_DXS_SRC }, /* MSI K8T Neo2-FI */
 		{ .subvendor = 0x1462, .subdevice = 0x7120, .action = VIA_DXS_ENABLE }, /* MSI KT4V */
 		{ .subvendor = 0x1462, .subdevice = 0x7142, .action = VIA_DXS_ENABLE }, /* MSI K8MM-V */
 		{ .subvendor = 0x1462, .subdevice = 0xb012, .action = VIA_DXS_SRC }, /* P4M800/VIA8237R */
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index ef97e50cd6c..577a2b03759 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -929,7 +929,7 @@ static void __devinit snd_via82xx_proc_init(struct via82xx_modem *chip)
 	struct snd_info_entry *entry;
 
 	if (! snd_card_proc_new(chip->card, "via82xx", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, snd_via82xx_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_via82xx_proc_read);
 }
 
 /*
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 65ebf5f1933..26aa775b7b6 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -308,7 +308,8 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
 	}
 	if (chip->mpu_res) {
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI,
-					       mpu_port[dev], 1,
+					       mpu_port[dev],
+					       MPU401_INFO_INTEGRATED,
 					       pci->irq, 0, &chip->rawmidi)) < 0) {
 			printk(KERN_WARNING "ymfpci: cannot initialize MPU401 at 0x%lx, skipping...\n", mpu_port[dev]);
 			legacy_ctrl &= ~YMFPCI_LEGACY_MIEN; /* disable MPU401 irq */
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 8ac5ab50b5c..f894752523b 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -1919,7 +1919,7 @@ static int __devinit snd_ymfpci_proc_init(struct snd_card *card, struct snd_ymfp
 	struct snd_info_entry *entry;
 	
 	if (! snd_card_proc_new(card, "ymfpci", &entry))
-		snd_info_set_text_ops(entry, chip, 1024, snd_ymfpci_proc_read);
+		snd_info_set_text_ops(entry, chip, snd_ymfpci_proc_read);
 	return 0;
 }