Merge remote branch 'alsa/fixes' into fix/hda

15 files changed, 96 insertions, 55 deletions

diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 75c602b..351654c 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 20cb60a..c119206 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -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/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index 15523e6..0470461 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: ");
diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp
index 09d24c7..a65e119 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/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 6b8ae7b..1d369ff 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
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 60cdb9e..83508b3 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 */
 		     secondary: 1;	/* secondary codec */
 	unsigned char secondary_addr;	/* address of the secondary codec */
-	unsigned int tea575x_tuner;	/* tuner flags */
+	unsigned int tea575x_tuner;	/* tuner access method & flags */
 
 	unsigned short ply_ctrl; /* playback control */
 	unsigned short cap_ctrl; /* capture control */
@@ -1287,7 +1290,7 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume)
 {
 	unsigned short cmdw;
 
-	if (chip->tea575x_tuner & 0x0010)
+	if (chip->tea575x_tuner & TUNER_ONLY)
 		goto __ac97_ok;
 
 	/* codec cold reset + AC'97 warm reset */
@@ -1296,11 +1299,13 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume)
 	udelay(100);
 	outw(0, FM801_REG(chip, CODEC_CTRL));
 
-	if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) {
-		snd_printk(KERN_ERR "Primary AC'97 codec not found\n");
-		if (! resume)
-			return -EIO;
-	}
+	if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0)
+		if (!resume) {
+			snd_printk(KERN_INFO "Primary AC'97 codec not found, "
+					    "assume SF64-PCR (tuner-only)\n");
+			chip->tea575x_tuner = 3 | TUNER_ONLY;
+			goto __ac97_ok;
+		}
 
 	if (chip->multichannel) {
 		if (chip->secondary_addr) {
@@ -1414,7 +1419,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 		return err;
 	}
 	chip->port = pci_resource_start(pci, 0);
-	if ((tea575x_tuner & 0x0010) == 0) {
+	if ((tea575x_tuner & TUNER_ONLY) == 0) {
 		if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED,
 				"FM801", chip)) {
 			snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq);
@@ -1429,6 +1434,14 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 		chip->multichannel = 1;
 
 	snd_fm801_chip_init(chip, 0);
+	/* init might set tuner access method */
+	tea575x_tuner = chip->tea575x_tuner;
+
+	if (chip->irq >= 0 && (tea575x_tuner & TUNER_ONLY)) {
+		pci_clear_master(pci);
+		free_irq(chip->irq, chip);
+		chip->irq = -1;
+	}
 
 	if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
 		snd_fm801_free(chip);
@@ -1438,12 +1451,13 @@ static int __devinit snd_fm801_create(struct snd_card *card,
 	snd_card_set_dev(card, &pci->dev);
 
 #ifdef TEA575X_RADIO
-	if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) {
+	if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 &&
+	    (tea575x_tuner & TUNER_TYPE_MASK) < 4) {
 		chip->tea.dev_nr = tea575x_tuner >> 16;
 		chip->tea.card = card;
 		chip->tea.freq_fixup = 10700;
 		chip->tea.private_data = chip;
-		chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1];
+		chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & TUNER_TYPE_MASK) - 1];
 		snd_tea575x_init(&chip->tea);
 	}
 #endif
@@ -1483,7 +1497,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
 	sprintf(card->longname, "%s at 0x%lx, irq %i",
 		card->shortname, chip->port, chip->irq);
 
-	if (tea575x_tuner[dev] & 0x0010)
+	if (chip->tea575x_tuner & TUNER_ONLY)
 		goto __fm801_tuner_only;
 
 	if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) {
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 2439e84..4b200da 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -938,7 +938,7 @@ static void init_input(struct hda_codec *codec)
 		coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */
 	if (is_active_pin(codec, CS_DMIC1_PIN_NID))
 		coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 
-				 * No effect if SPDIF_OUT2 is slected in 
+				 * No effect if SPDIF_OUT2 is selected in 
 				 * IDX_SPDIF_CTL.
 				  */
 	cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 85c81fe..a45c116 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -66,7 +66,7 @@ struct cmi_spec {
 
 	struct hda_pcm pcm_rec[2];	/* PCM information */
 
-	/* pin deafault configuration */
+	/* pin default configuration */
 	hda_nid_t pin_nid[NUM_PINS];
 	unsigned int def_conf[NUM_PINS];
 	unsigned int pin_def_confs;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b3abe9c..0877bae 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6673,7 +6673,7 @@ static struct hda_input_mux alc889A_mb31_capture_source = {
 		/* Front Mic (0x01) unused */
 		{ "Line", 0x2 },
 		/* Line 2 (0x03) unused */
-		/* CD (0x04) unsused? */
+		/* CD (0x04) unused? */
 	},
 };
 
@@ -9287,8 +9287,6 @@ static struct alc_config_preset alc882_presets[] = {
 		.dac_nids = alc883_dac_nids,
 		.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
 		.adc_nids = alc889_adc_nids,
-		.capsrc_nids = alc889_capsrc_nids,
-		.capsrc_nids = alc889_capsrc_nids,
 		.dig_out_nid = ALC883_DIGOUT_NID,
 		.dig_in_nid = ALC883_DIGIN_NID,
 		.slave_dig_outs = alc883_slave_dig_outs,
diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c
index 110d16e..765d7bd 100644
--- a/sound/pci/ice1712/aureon.c
+++ b/sound/pci/ice1712/aureon.c
@@ -689,32 +689,14 @@ static int aureon_ac97_mmute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	return change;
 }
 
-static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -12700, 100, 1);
+static const DECLARE_TLV_DB_SCALE(db_scale_wm_dac, -10000, 100, 1);
 static const DECLARE_TLV_DB_SCALE(db_scale_wm_pcm, -6400, 50, 1);
 static const DECLARE_TLV_DB_SCALE(db_scale_wm_adc, -1200, 100, 0);
 static const DECLARE_TLV_DB_SCALE(db_scale_ac97_master, -4650, 150, 0);
 static const DECLARE_TLV_DB_SCALE(db_scale_ac97_gain, -3450, 150, 0);
 
-/*
- * Logarithmic volume values for WM8770
- * Computed as 20 * Log10(255 / x)
- */
-static const unsigned char wm_vol[256] = {
-	127, 48, 42, 39, 36, 34, 33, 31, 30, 29, 28, 27, 27, 26, 25, 25, 24, 24, 23,
-	23, 22, 22, 21, 21, 21, 20, 20, 20, 19, 19, 19, 18, 18, 18, 18, 17, 17, 17,
-	17, 16, 16, 16, 16, 15, 15, 15, 15, 15, 15, 14, 14, 14, 14, 14, 13, 13, 13,
-	13, 13, 13, 13, 12, 12, 12, 12, 12, 12, 12, 11, 11, 11, 11, 11, 11, 11, 11,
-	11, 10, 10, 10, 10, 10, 10, 10, 10, 10, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 8, 8,
-	8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 6, 6, 6,
-	6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 6, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
-	5, 5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 3, 3, 3, 3, 3,
-	3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
-	2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
-	1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	0, 0
-};
-
-#define WM_VOL_MAX	(sizeof(wm_vol) - 1)
+#define WM_VOL_MAX	100
+#define WM_VOL_CNT	101	/* 0dB .. -100dB */
 #define WM_VOL_MUTE	0x8000
 
 static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned short vol, unsigned short master)
@@ -724,7 +706,8 @@ static void wm_set_vol(struct snd_ice1712 *ice, unsigned int index, unsigned sho
 	if ((master & WM_VOL_MUTE) || (vol & WM_VOL_MUTE))
 		nvol = 0;
 	else
-		nvol = 127 - wm_vol[(((vol & ~WM_VOL_MUTE) * (master & ~WM_VOL_MUTE)) / 127) & WM_VOL_MAX];
+		nvol = ((vol % WM_VOL_CNT) * (master % WM_VOL_CNT)) /
+								WM_VOL_MAX;
 
 	wm_put(ice, index, nvol);
 	wm_put_nocache(ice, index, 0x180 | nvol);
@@ -820,7 +803,7 @@ static int wm_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 	uinfo->count = voices;
 	uinfo->value.integer.min = 0;		/* mute (-101dB) */
-	uinfo->value.integer.max = 0x7F;	/* 0dB */
+	uinfo->value.integer.max = WM_VOL_MAX;	/* 0dB */
 	return 0;
 }
 
@@ -850,7 +833,7 @@ static int wm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *
 	snd_ice1712_save_gpio_status(ice);
 	for (i = 0; i < voices; i++) {
 		unsigned int vol = ucontrol->value.integer.value[i];
-		if (vol > 0x7f)
+		if (vol > WM_VOL_MAX)
 			continue;
 		vol |= spec->vol[ofs+i];
 		if (vol != spec->vol[ofs+i]) {
diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h
index 4615bca..0da778a 100644
--- a/sound/pci/ice1712/ice1712.h
+++ b/sound/pci/ice1712/ice1712.h
@@ -387,8 +387,8 @@ struct snd_ice1712 {
 #ifdef CONFIG_PM
 	int (*pm_suspend)(struct snd_ice1712 *);
 	int (*pm_resume)(struct snd_ice1712 *);
-	int pm_suspend_enabled:1;
-	int pm_saved_is_spdif_master:1;
+	unsigned int pm_suspend_enabled:1;
+	unsigned int pm_saved_is_spdif_master:1;
 	unsigned int pm_saved_spdif_ctrl;
 	unsigned char pm_saved_spdif_cfg;
 	unsigned int pm_saved_route;
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index 4bed963..98bc3b7 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -380,7 +380,7 @@ static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = {
 	 * inputs) are fed from Xilinx.
 	 *
 	 * I even checked traces on board and coded a support in driver for
-	 * an alternative possiblity - the unused I2S ICE output channels
+	 * an alternative possibility - the unused I2S ICE output channels
 	 * switched to HW-IN/SPDIF-IN and providing the monitoring signal to
 	 * the DAC - to no avail. The I2S outputs seem to be unconnected.
 	 *
@@ -483,6 +483,31 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice)
 }
 
 /*
+ * suspend/resume
+ * */
+
+#ifdef CONFIG_PM
+static int juli_resume(struct snd_ice1712 *ice)
+{
+	struct snd_akm4xxx *ak = ice->akm;
+	struct juli_spec *spec = ice->spec;
+	/* akm4358 un-reset, un-mute */
+	snd_akm4xxx_reset(ak, 0);
+	/* reinit ak4114 */
+	snd_ak4114_reinit(spec->ak4114);
+	return 0;
+}
+
+static int juli_suspend(struct snd_ice1712 *ice)
+{
+	struct snd_akm4xxx *ak = ice->akm;
+	/* akm4358 reset and soft-mute */
+	snd_akm4xxx_reset(ak, 1);
+	return 0;
+}
+#endif
+
+/*
  * initialize the chip
  */
 
@@ -626,6 +651,13 @@ static int __devinit juli_init(struct snd_ice1712 *ice)
 	ice->set_spdif_clock = juli_set_spdif_clock;
 
 	ice->spdif.ops.open = juli_spdif_in_open;
+
+#ifdef CONFIG_PM
+	ice->pm_resume = juli_resume;
+	ice->pm_suspend = juli_suspend;
+	ice->pm_suspend_enabled = 1;
+#endif
+
 	return 0;
 }
 
diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c
index c75515f..6a9fee3 100644
--- a/sound/pci/ice1712/prodigy_hifi.c
+++ b/sound/pci/ice1712/prodigy_hifi.c
@@ -1100,7 +1100,7 @@ static void ak4396_init(struct snd_ice1712 *ice)
 }
 
 #ifdef CONFIG_PM
-static int __devinit prodigy_hd2_resume(struct snd_ice1712 *ice)
+static int prodigy_hd2_resume(struct snd_ice1712 *ice)
 {
 	/* initialize ak4396 codec and restore previous mixer volumes */
 	struct prodigy_hifi_spec *spec = ice->spec;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 754867e..b990143 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -1950,6 +1950,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 	},
 	{
 		.subvendor = 0x104d,
+		.subdevice = 0x8144,
+		.name = "Sony",
+		.type = AC97_TUNE_INV_EAPD
+	},
+	{
+		.subvendor = 0x104d,
 		.subdevice = 0x8197,
 		.name = "Sony S1XP",
 		.type = AC97_TUNE_INV_EAPD
@@ -2057,6 +2063,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
 		.type = AC97_TUNE_HP_ONLY
 	},
 	{
+		.subvendor = 0x161f,
+		.subdevice = 0x203a,
+		.name = "Gateway 4525GZ",		/* AD1981B */
+		.type = AC97_TUNE_INV_EAPD
+	},
+	{
 		.subvendor = 0x1734,
 		.subdevice = 0x0088,
 		.name = "Fujitsu-Siemens D1522",	/* AD1981 */
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 0dce331..a1b10d1 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -3017,7 +3017,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
 		insel = "Coaxial";
 		break;
 	default:
-		insel = "Unkown";
+		insel = "Unknown";
 	}
 
 	switch (hdspm->control_register & HDSPM_SyncRefMask) {
@@ -3028,7 +3028,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
 		syncref = "MADI";
 		break;
 	default:
-		syncref = "Unkown";
+		syncref = "Unknown";
 	}
 	snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel,
 		    syncref);