Merge commit 'v2.6.33-rc2' into for-2.6.33

65 files changed, 9443 insertions, 2371 deletions

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 75c602b5b13..351654cf7b0 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -570,6 +570,7 @@ config SND_ICE1712
 	tristate "ICEnsemble ICE1712 (Envy24)"
 	select SND_MPU401_UART
 	select SND_AC97_CODEC
+	select BITREVERSE
 	help
 	  Say Y here to include support for soundcards based on the
 	  ICE1712 (Envy24) chip.
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 78288dbfc17..c1192062300 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -603,8 +603,8 @@ AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1)
 };
 
 static const struct snd_kcontrol_new snd_ac97_controls_pc_beep[2] = {
-AC97_SINGLE("PC Speaker Playback Switch", AC97_PC_BEEP, 15, 1, 1),
-AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1)
+AC97_SINGLE("Beep Playback Switch", AC97_PC_BEEP, 15, 1, 1),
+AC97_SINGLE("Beep Playback Volume", AC97_PC_BEEP, 1, 15, 1)
 };
 
 static const struct snd_kcontrol_new snd_ac97_controls_mic_boost =
@@ -1393,7 +1393,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97)
 		}
 	}
 	
-	/* build PC Speaker controls */
+	/* build Beep controls */
 	if (!(ac97->flags & AC97_HAS_NO_PC_BEEP) && 
 		((ac97->flags & AC97_HAS_PC_BEEP) ||
 	    snd_ac97_try_volume_mix(ac97, AC97_PC_BEEP))) {
@@ -2122,7 +2122,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template,
 		}
 		/* nothing should be in powerdown mode */
 		snd_ac97_write_cache(ac97, AC97_GENERAL_PURPOSE, 0);
-		end_time = jiffies + msecs_to_jiffies(120);
+		end_time = jiffies + msecs_to_jiffies(5000);
 		do {
 			if ((snd_ac97_read(ac97, AC97_POWERDOWN) & 0x0f) == 0x0f)
 				goto __ready_ok;
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 7337abdbe4e..139cf3b2b9d 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -800,12 +800,12 @@ AC97_SINGLE("Mono Switch", AC97_MASTER_TONE, 7, 1, 1),
 AC97_SINGLE("Mono ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
 AC97_SINGLE("Mono Volume", AC97_MASTER_TONE, 0, 31, 1),
 
-AC97_SINGLE("PC Beep to Headphone Switch", AC97_AUX, 15, 1, 1),
-AC97_SINGLE("PC Beep to Headphone Volume", AC97_AUX, 12, 7, 1),
-AC97_SINGLE("PC Beep to Master Switch", AC97_AUX, 11, 1, 1),
-AC97_SINGLE("PC Beep to Master Volume", AC97_AUX, 8, 7, 1),
-AC97_SINGLE("PC Beep to Mono Switch", AC97_AUX, 7, 1, 1),
-AC97_SINGLE("PC Beep to Mono Volume", AC97_AUX, 4, 7, 1),
+AC97_SINGLE("Beep to Headphone Switch", AC97_AUX, 15, 1, 1),
+AC97_SINGLE("Beep to Headphone Volume", AC97_AUX, 12, 7, 1),
+AC97_SINGLE("Beep to Master Switch", AC97_AUX, 11, 1, 1),
+AC97_SINGLE("Beep to Master Volume", AC97_AUX, 8, 7, 1),
+AC97_SINGLE("Beep to Mono Switch", AC97_AUX, 7, 1, 1),
+AC97_SINGLE("Beep to Mono Volume", AC97_AUX, 4, 7, 1),
 
 AC97_SINGLE("Voice to Headphone Switch", AC97_PCM, 15, 1, 1),
 AC97_SINGLE("Voice to Headphone Volume", AC97_PCM, 12, 7, 1),
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 8451a0169f3..69867ace786 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -830,8 +830,8 @@ static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = {
 	AZF3328_MIXER_SWITCH("Mic Boost (+20dB)", IDX_MIXER_MIC, 6, 0),
 	AZF3328_MIXER_SWITCH("Line Playback Switch", IDX_MIXER_LINEIN, 15, 1),
 	AZF3328_MIXER_VOL_STEREO("Line Playback Volume", IDX_MIXER_LINEIN, 0x1f, 1),
-	AZF3328_MIXER_SWITCH("PC Speaker Playback Switch", IDX_MIXER_PCBEEP, 15, 1),
-	AZF3328_MIXER_VOL_SPECIAL("PC Speaker Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1),
+	AZF3328_MIXER_SWITCH("Beep Playback Switch", IDX_MIXER_PCBEEP, 15, 1),
+	AZF3328_MIXER_VOL_SPECIAL("Beep Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1),
 	AZF3328_MIXER_SWITCH("Video Playback Switch", IDX_MIXER_VIDEO, 15, 1),
 	AZF3328_MIXER_VOL_STEREO("Video Playback Volume", IDX_MIXER_VIDEO, 0x1f, 1),
 	AZF3328_MIXER_SWITCH("Aux Playback Switch", IDX_MIXER_AUX, 15, 1),
diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c
index c8c6f437f5b..8f443a9d61e 100644
--- a/sound/pci/ca0106/ca0106_mixer.c
+++ b/sound/pci/ca0106/ca0106_mixer.c
@@ -792,8 +792,8 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu)
 		"Phone Playback Volume",
 		"Video Playback Switch",
 		"Video Playback Volume",
-		"PC Speaker Playback Switch",
-		"PC Speaker Playback Volume",
+		"Beep Playback Switch",
+		"Beep Playback Volume",
 		"Mono Output Select",
 		"Capture Source",
 		"Capture Switch",
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index c62b7d10ec6..0470461cc03 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -233,7 +233,7 @@ static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 val
 			snd_iprintf(buffer, "user-defined\n");
 			break;
 		default:
-			snd_iprintf(buffer, "unkown\n");
+			snd_iprintf(buffer, "unknown\n");
 			break;
 		}
 		snd_iprintf(buffer, "Sample Bits: ");
@@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry,
         while (!snd_info_get_line(buffer, line, sizeof(line))) {
                 if (sscanf(line, "%x %x", &reg, &val) != 2)
                         continue;
-                if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) {
+		if (reg < 0x40 && val <= 0xffffffff) {
 			spin_lock_irqsave(&emu->emu_lock, flags);
 			outl(val, emu->port + (reg & 0xfffffffc));
 			spin_unlock_irqrestore(&emu->emu_lock, flags);
@@ -405,7 +405,7 @@ static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry,
         while (!snd_info_get_line(buffer, line, sizeof(line))) {
                 if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
                         continue;
-                if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) )
+		if (reg < 0x80 && val <= 0xffffffff && channel_id <= 3)
                         snd_ca0106_ptr_write(emu, reg, channel_id, val);
         }
 }
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index ddcd4a9fd7e..a312bae08f5 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -2302,7 +2302,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = {
 	CMIPCI_SB_VOL_MONO("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31),
 	CMIPCI_SB_SW_MONO("Mic Playback Switch", 0),
 	CMIPCI_DOUBLE("Mic Capture Switch", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0, 1, 0, 0),
-	CMIPCI_SB_VOL_MONO("PC Speaker Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
+	CMIPCI_SB_VOL_MONO("Beep Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
 	CMIPCI_MIXER_VOL_STEREO("Aux Playback Volume", CM_REG_AUX_VOL, 4, 0, 15),
 	CMIPCI_MIXER_SW_STEREO("Aux Playback Switch", CM_REG_MIXER2, CM_VAUXLM_SHIFT, CM_VAUXRM_SHIFT, 0),
 	CMIPCI_MIXER_SW_STEREO("Aux Capture Switch", CM_REG_MIXER2, CM_RAUXLEN_SHIFT, CM_RAUXREN_SHIFT, 0),
@@ -2310,7 +2310,7 @@ 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 Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0),
+	CMIPCI_DOUBLE("Beep 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),
 };
 
diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp
index 09d24c76f03..a65e1193c89 100644
--- a/sound/pci/cs46xx/imgs/cwcdma.asp
+++ b/sound/pci/cs46xx/imgs/cwcdma.asp
@@ -26,10 +26,11 @@
 //
 //
 // The purpose of this code is very simple: make it possible to tranfser
-// the samples 'as they are' with no alteration from a PCMreader SCB (DMA from host)
-// to any other SCB. This is useful for AC3 throug SPDIF. SRC (source rate converters) 
-// task always alters the samples in some how, however it's from 48khz -> 48khz. The
-// alterations are not audible, but AC3 wont work. 
+// the samples 'as they are' with no alteration from a PCMreader
+// SCB (DMA from host) to any other SCB. This is useful for AC3 through SPDIF.
+// SRC (source rate converters) task always alters the samples in somehow,
+// however it's from 48khz -> 48khz.
+// The alterations are not audible, but AC3 wont work. 
 //
 //        ...
 //         |
diff --git a/sound/pci/cs5535audio/Makefile b/sound/pci/cs5535audio/Makefile
index fda7a94c992..ccc642269b9 100644
--- a/sound/pci/cs5535audio/Makefile
+++ b/sound/pci/cs5535audio/Makefile
@@ -4,9 +4,7 @@
 
 snd-cs5535audio-y := cs5535audio.o cs5535audio_pcm.o
 snd-cs5535audio-$(CONFIG_PM) += cs5535audio_pm.o
-ifdef CONFIG_MGEODE_LX
 snd-cs5535audio-$(CONFIG_OLPC) += cs5535audio_olpc.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 05f56e04849..91e7faf69bb 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -389,6 +389,7 @@ probefail_out:
 
 static void __devexit snd_cs5535audio_remove(struct pci_dev *pci)
 {
+	olpc_quirks_cleanup();
 	snd_card_free(pci_get_drvdata(pci));
 	pci_set_drvdata(pci, NULL);
 }
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index 7a298ac662e..51966d782a3 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -99,10 +99,11 @@ int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state);
 int snd_cs5535audio_resume(struct pci_dev *pci);
 #endif
 
-#if defined(CONFIG_OLPC) && defined(CONFIG_MGEODE_LX)
+#ifdef CONFIG_OLPC
 void __devinit olpc_prequirks(struct snd_card *card,
 		struct snd_ac97_template *ac97);
 int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97);
+void __devexit olpc_quirks_cleanup(void);
 void olpc_analog_input(struct snd_ac97 *ac97, int on);
 void olpc_mic_bias(struct snd_ac97 *ac97, int on);
 
@@ -128,6 +129,7 @@ static inline int olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
 {
 	return 0;
 }
+static inline void olpc_quirks_cleanup(void) { }
 static inline void olpc_analog_input(struct snd_ac97 *ac97, int on) { }
 static inline void olpc_mic_bias(struct snd_ac97 *ac97, int on) { }
 static inline void olpc_capture_open(struct snd_ac97 *ac97) { }
diff --git a/sound/pci/cs5535audio/cs5535audio_olpc.c b/sound/pci/cs5535audio/cs5535audio_olpc.c
index 5c6814335cd..50da49be9ae 100644
--- a/sound/pci/cs5535audio/cs5535audio_olpc.c
+++ b/sound/pci/cs5535audio/cs5535audio_olpc.c
@@ -13,10 +13,13 @@
 #include <sound/info.h>
 #include <sound/control.h>
 #include <sound/ac97_codec.h>
+#include <linux/gpio.h>
 
 #include <asm/olpc.h>
 #include "cs5535audio.h"
 
+#define DRV_NAME "cs5535audio-olpc"
+
 /*
  * OLPC has an additional feature on top of the regular AD1888 codec features.
  * It has an Analog Input mode that is switched into (after disabling the
@@ -38,10 +41,7 @@ void olpc_analog_input(struct snd_ac97 *ac97, int on)
 	}
 
 	/* set Analog Input through GPIO */
-	if (on)
-		geode_gpio_set(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
-	else
-		geode_gpio_clear(OLPC_GPIO_MIC_AC, GPIO_OUTPUT_VAL);
+	gpio_set_value(OLPC_GPIO_MIC_AC, on);
 }
 
 /*
@@ -73,8 +73,7 @@ static int olpc_dc_info(struct snd_kcontrol *kctl,
 
 static int olpc_dc_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *v)
 {
-	v->value.integer.value[0] = geode_gpio_isset(OLPC_GPIO_MIC_AC,
-			GPIO_OUTPUT_VAL);
+	v->value.integer.value[0] = gpio_get_value(OLPC_GPIO_MIC_AC);
 	return 0;
 }
 
@@ -153,6 +152,12 @@ int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
 	if (!machine_is_olpc())
 		return 0;
 
+	if (gpio_request(OLPC_GPIO_MIC_AC, DRV_NAME)) {
+		printk(KERN_ERR DRV_NAME ": unable to allocate MIC GPIO\n");
+		return -EIO;
+	}
+	gpio_direction_output(OLPC_GPIO_MIC_AC, 0);
+
 	/* drop the original AD1888 HPF control */
 	memset(&elem, 0, sizeof(elem));
 	elem.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
@@ -169,11 +174,18 @@ int __devinit olpc_quirks(struct snd_card *card, struct snd_ac97 *ac97)
 	for (i = 0; i < ARRAY_SIZE(olpc_cs5535audio_ctls); i++) {
 		err = snd_ctl_add(card, snd_ctl_new1(&olpc_cs5535audio_ctls[i],
 				ac97->private_data));
-		if (err < 0)
+		if (err < 0) {
+			gpio_free(OLPC_GPIO_MIC_AC);
 			return err;
+		}
 	}
 
 	/* turn off the mic by default */
 	olpc_mic_bias(ac97, 0);
 	return 0;
 }
+
+void __devexit olpc_quirks_cleanup(void)
+{
+	gpio_free(OLPC_GPIO_MIC_AC);
+}
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index 75454648d50..cb65bd0dd35 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -240,7 +240,7 @@ static int select_rom(unsigned int pitch)
 	} else if (pitch == 0x02000000) {
 		/* pitch == 2 */
 		return 3;
-	} else if (pitch >= 0x0 && pitch <= 0x08000000) {
+	} else if (pitch <= 0x08000000) {
 		/* 0 <= pitch <= 8 */
 		return 0;
 	} else {
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 36e08bd2b3c..1d369ff7380 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -184,7 +184,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard.");
  * The hardware has 3 channels for playback and 1 for capture.
  *  - channel 0 is the front channel
  *  - channel 1 is the rear channel
- *  - channel 2 is the center/lfe chanel
+ *  - channel 2 is the center/lfe channel
  * Volume is controlled by the AC97 for the front and rear channels by
  * the PCM Playback Volume, Sigmatel Surround Playback Volume and 
  * Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects
@@ -1040,8 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry,
 		if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
 			continue;
 
-		if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) 
-		    && (channel_id >= 0) && (channel_id <= 2) )
+		if (reg < 0x49 && val <= 0xffffffff && channel_id <= 2)
 			snd_emu10k1x_ptr_write(emu, reg, channel_id, val);
 	}
 }
diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c
index b0fb6c917c3..05afe06e353 100644
--- a/sound/pci/emu10k1/emumixer.c
+++ b/sound/pci/emu10k1/emumixer.c
@@ -1818,8 +1818,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu,
 		"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",
+		"Beep Playback Switch", "Beep Capture Switch",
+		"Beep Playback Volume", "Beep Capture Volume",
 		"Phone Playback Switch", "Phone Capture Switch",
 		"Phone Playback Volume", "Phone Capture Volume",
 		"Mic Playback Switch", "Mic Capture Switch",
diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c
index 216f9748aff..baa7cd508cd 100644
--- a/sound/pci/emu10k1/emuproc.c
+++ b/sound/pci/emu10k1/emuproc.c
@@ -451,7 +451,7 @@ static void snd_emu_proc_io_reg_write(struct snd_info_entry *entry,
 	while (!snd_info_get_line(buffer, line, sizeof(line))) {
 		if (sscanf(line, "%x %x", &reg, &val) != 2)
 			continue;
-		if ((reg < 0x40) && (reg >= 0) && (val <= 0xffffffff) ) {
+		if (reg < 0x40 && val <= 0xffffffff) {
 			spin_lock_irqsave(&emu->emu_lock, flags);
 			outl(val, emu->port + (reg & 0xfffffffc));
 			spin_unlock_irqrestore(&emu->emu_lock, flags);
@@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry,
 	while (!snd_info_get_line(buffer, line, sizeof(line))) {
 		if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
 			continue;
-		if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) )
+		if (reg < 0xa0 && val <= 0xffffffff && channel_id <= 3)
 			snd_ptr_write(emu, iobase, reg, channel_id, val);
 	}
 }
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index c1a5aa15af8..5ef7080e14d 100644
--- a/sound/pci/emu10k1/io.c
+++ b/sound/pci/emu10k1/io.c
@@ -256,7 +256,7 @@ int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value)
 	if (reg > 0x3f)
 		return 1;
 	reg += 0x40; /* 0x40 upwards are registers. */
-	if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */
+	if (value > 0x3f) /* 0 to 0x3f are values */
 		return 1;
 	spin_lock_irqsave(&emu->emu_lock, flags);
 	outl(reg, emu->port + A_IOCFG);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 820318ee62c..fb83e1ffa5c 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1387,7 +1387,7 @@ ES1938_DOUBLE_TLV("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0,
 		  db_scale_line),
 ES1938_DOUBLE_TLV("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0,
 		  db_scale_capture),
-ES1938_SINGLE("PC Speaker Volume", 0, 0x3c, 0, 7, 0),
+ES1938_SINGLE("Beep Volume", 0, 0x3c, 0, 7, 0),
 ES1938_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0),
 ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1),
 {
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 60cdb9e0b68..83508b3964f 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -55,7 +55,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;	/* Enable this card *
  *    1 = MediaForte 256-PCS
  *    2 = MediaForte 256-PCPR
  *    3 = MediaForte 64-PCR
- *   16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card
+ *   16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card
  *  High 16-bits are video (radio) device number + 1
  */
 static int tea575x_tuner[SNDRV_CARDS];
@@ -67,7 +67,10 @@ MODULE_PARM_DESC(id, "ID string for the FM801 soundcard.");
 module_param_array(enable, bool, NULL, 0444);
 MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
 module_param_array(tea575x_tuner, int, NULL, 0444);
-MODULE_PARM_DESC(tea575x_tuner, "Enable TEA575x tuner.");
+MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only).");
+
+#define TUNER_ONLY		(1<<4)
+#define TUNER_TYPE_MASK		(~TUNER_ONLY & 0xFFFF)
 
 /*
  *  Direct registers
@@ -160,7 +163,7 @@ struct fm801 {
 	unsigned int multichannel: 1,	/* multichannel support */
 		     secon