From edf8e4565c44bffbb4d09e8984df941d0ae9e6e8 Mon Sep 17 00:00:00 2001
From: Mikael Magnusson <mikma@users.sourceforge.net>
Date: Tue, 13 Sep 2005 11:32:58 +0200
Subject: [ALSA] emu10k1: Front channels via fxbus 8 and 9

Modules: EMU10K1/EMU10K2 driver

Adds left and right front channel outputs using fxbus 8 and 9 and 'Front'
playback and capture volume controls.

Signed-off-by: Mikael Magnusson <mikma@users.sourceforge.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/emu10k1/emufx.c | 23 ++++++++++++++++++++---
 1 file changed, 20 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 646b5d972e6..20db3ac6cd6 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -964,8 +964,8 @@ static int snd_emu10k1_ipcm_peek(emu10k1_t *emu, emu10k1_fx8010_pcm_t *ipcm)
 	return err;
 }
 
-#define SND_EMU10K1_GPR_CONTROLS	41
-#define SND_EMU10K1_INPUTS		10
+#define SND_EMU10K1_GPR_CONTROLS	44
+#define SND_EMU10K1_INPUTS		12
 #define SND_EMU10K1_PLAYBACK_CHANNELS	8
 #define SND_EMU10K1_CAPTURE_CHANNELS	4
 
@@ -1527,7 +1527,7 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu)
 
 	strcpy(icode->name, "SB Live! FX8010 code for ALSA v1.2 by Jaroslav Kysela");
 	ptr = 0; i = 0;
-	/* we have 10 inputs */
+	/* we have 12 inputs */
 	playback = SND_EMU10K1_INPUTS;
 	/* we have 6 playback channels and tone control doubles */
 	capture = playback + (SND_EMU10K1_PLAYBACK_CHANNELS * 2);
@@ -1551,6 +1551,8 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu)
 	OP(icode, &ptr, iMACINT0, GPR(7), C_00000000, FXBUS(FXBUS_PCM_LFE), C_00000004);
 	OP(icode, &ptr, iMACINT0, GPR(8), C_00000000, C_00000000, C_00000000);	/* S/PDIF left */
 	OP(icode, &ptr, iMACINT0, GPR(9), C_00000000, C_00000000, C_00000000);	/* S/PDIF right */
+	OP(icode, &ptr, iMACINT0, GPR(10), C_00000000, FXBUS(FXBUS_PCM_LEFT_FRONT), C_00000004);
+	OP(icode, &ptr, iMACINT0, GPR(11), C_00000000, FXBUS(FXBUS_PCM_RIGHT_FRONT), C_00000004);
 
 	/* Raw S/PDIF PCM */
 	ipcm->substream = 0;
@@ -1697,6 +1699,21 @@ static int __devinit _snd_emu10k1_init_efx(emu10k1_t *emu)
 	VOLUME_ADD(icode, &ptr, playback + 5, 7, gpr);
 	snd_emu10k1_init_mono_control(controls + i++, "LFE Digital Playback Volume", gpr++, 100);
 
+	/* Front Playback Volume */
+	for (z = 0; z < 2; z++)
+		VOLUME_ADD(icode, &ptr, playback + z, 10 + z, gpr + z);
+	snd_emu10k1_init_stereo_control(controls + i++, "Front Playback Volume", gpr, 100);
+	gpr += 2;
+
+	/* Front Capture Volume + Switch */
+	for (z = 0; z < 2; z++) {
+		SWITCH(icode, &ptr, tmp + 0, 10 + z, gpr + 2);
+		VOLUME_ADD(icode, &ptr, capture + z, tmp + 0, gpr + z);
+	}
+	snd_emu10k1_init_stereo_control(controls + i++, "Front Capture Volume", gpr, 0);
+	snd_emu10k1_init_mono_onoff_control(controls + i++, "Front Capture Switch", gpr + 2, 0);
+	gpr += 3;
+
 	/*
 	 *  Process inputs
 	 */
-- 
cgit v1.2.3-18-g5258


From 4525c9f31f4606881ed49d822b44c2c5d6080ef4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Sep 2005 11:47:07 +0200
Subject: [ALSA] Fix surround control of CMI9761

Modules: AC97 Codec

- Fixed surround controls of CMI9761 (model 83) in update_jacks callback.
- Clean up ad1888 and ad1985 update_jacks callbacks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_patch.c | 77 +++++++++++++++++++++++++++++----------------
 1 file changed, 50 insertions(+), 27 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 0238cc65d32..ffc8d652107 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -163,14 +163,24 @@ static int ac97_channel_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t
 		.private_value = 1, \
 	}
 
+static inline int is_surround_on(ac97_t *ac97)
+{
+	return ac97->channel_mode >= 1;
+}
+
+static inline int is_clfe_on(ac97_t *ac97)
+{
+	return a97->channel_mode >= 2;
+}
+
 static inline int is_shared_linein(ac97_t *ac97)
 {
-	return ! ac97->indep_surround && ac97->channel_mode >= 1;
+	return ! ac97->indep_surround && is_surround_on(ac97);
 }
 
 static inline int is_shared_micin(ac97_t *ac97)
 {
-	return ! ac97->indep_surround && ac97->channel_mode >= 2;
+	return ! ac97->indep_surround && is_clfe_on(ac97);
 }
 
 
@@ -1753,12 +1763,13 @@ static int snd_ac97_ad1888_downmix_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_va
 
 static void ad1888_update_jacks(ac97_t *ac97)
 {
+	unsigned short val = 0;
+	if (! is_shared_linein(ac97))
+		val |= (1 << 12);
+	if (! is_shared_micin(ac97))
+		val |= (1 << 11);
 	/* shared Line-In */
-	snd_ac97_update_bits(ac97, AC97_AD_MISC, 1 << 12,
-			     is_shared_linein(ac97) ? 0 : 1 << 12);
-	/* shared Mic */
-	snd_ac97_update_bits(ac97, AC97_AD_MISC, 1 << 11,
-			     is_shared_micin(ac97) ? 0 : 1 << 11);
+	snd_ac97_update_bits(ac97, AC97_AD_MISC, (1 << 11) | (1 << 12), val);
 }
 
 static const snd_kcontrol_new_t snd_ac97_ad1888_controls[] = {
@@ -1852,12 +1863,7 @@ static const snd_kcontrol_new_t snd_ac97_ad1985_controls[] = {
 
 static void ad1985_update_jacks(ac97_t *ac97)
 {
-	/* shared Line-In */
-	snd_ac97_update_bits(ac97, AC97_AD_MISC, 1 << 12,
-			     is_shared_linein(ac97) ? 0 : 1 << 12);
-	/* shared Mic */
-	snd_ac97_update_bits(ac97, AC97_AD_MISC, 1 << 11,
-			     is_shared_micin(ac97) ? 0 : 1 << 11);
+	ad1888_update_jacks(ac97);
 	snd_ac97_update_bits(ac97, AC97_AD_SERIAL_CFG, 1 << 9,
 			     is_shared_micin(ac97) ? 0 : 1 << 9);
 }
@@ -2442,21 +2448,37 @@ int patch_cm9739(ac97_t * ac97)
 
 static void cm9761_update_jacks(ac97_t *ac97)
 {
-	unsigned short surr_vals[2][2] = {
-		{ 0x0008, 0x0400 }, /* off, on */
-		{ 0x0000, 0x0408 }, /* off, on (9761-82 rev.B) */
+	/* FIXME: check the bits for each model
+	 *        model 83 is confirmed to work
+	 */
+	static unsigned short surr_on[3][2] = {
+		{ 0x0008, 0x0000 }, /* 9761-78 & 82 */
+		{ 0x0000, 0x0008 }, /* 9761-82 rev.B */
+		{ 0x0000, 0x0008 }, /* 9761-83 */
+	};
+	static unsigned short clfe_on[3][2] = {
+		{ 0x0000, 0x1000 }, /* 9761-78 & 82 */
+		{ 0x1000, 0x0000 }, /* 9761-82 rev.B */
+		{ 0x0000, 0x1000 }, /* 9761-83 */
+	};
+	static unsigned short surr_shared[3][2] = {
+		{ 0x0000, 0x0400 }, /* 9761-78 & 82 */
+		{ 0x0000, 0x0400 }, /* 9761-82 rev.B */
+		{ 0x0000, 0x0400 }, /* 9761-83 */
 	};
-	unsigned short clfe_vals[2][2] = {
-		{ 0x2000, 0x1880 }, /* off, on */
-		{ 0x1000, 0x2880 }, /* off, on (9761-82 rev.B) */
+	static unsigned short clfe_shared[3][2] = {
+		{ 0x2000, 0x0880 }, /* 9761-78 & 82 */
+		{ 0x0000, 0x2880 }, /* 9761-82 rev.B */
+		{ 0x2000, 0x0800 }, /* 9761-83 */
 	};
+	unsigned short val = 0;
 
-	/* shared Line-In */
-	snd_ac97_update_bits(ac97, AC97_CM9761_MULTI_CHAN, 0x0408,
-			     surr_vals[ac97->spec.dev_flags][is_shared_linein(ac97)]);
-	/* shared Mic */
-	snd_ac97_update_bits(ac97, AC97_CM9761_MULTI_CHAN, 0x3880,
-			     clfe_vals[ac97->spec.dev_flags][is_shared_micin(ac97)]);
+	val |= surr_on[ac97->spec.dev_flags][is_surround_on(ac97)];
+	val |= clfe_on[ac97->spec.dev_flags][is_clfe_on(ac97)];
+	val |= surr_shared[ac97->spec.dev_flags][is_shared_linein(ac97)];
+	val |= clfe_shared[ac97->spec.dev_flags][is_shared_micin(ac97)];
+
+	snd_ac97_update_bits(ac97, AC97_CM9761_MULTI_CHAN, 0x3c88, val);
 }
 
 static const snd_kcontrol_new_t snd_ac97_cm9761_controls[] = {
@@ -2551,7 +2573,7 @@ int patch_cm9761(ac97_t *ac97)
 	snd_ac97_write_cache(ac97, AC97_MASTER, 0x8808);
 	snd_ac97_write_cache(ac97, AC97_PCM, 0x8808);
 
-	ac97->spec.dev_flags = 0; /* 1 = model 82 revision B */
+	ac97->spec.dev_flags = 0; /* 1 = model 82 revision B, 2 = model 83 */
 	if (ac97->id == AC97_ID_CM9761_82) {
 		unsigned short tmp;
 		/* check page 1, reg 0x60 */
@@ -2560,7 +2582,8 @@ int patch_cm9761(ac97_t *ac97)
 		tmp = snd_ac97_read(ac97, 0x60);
 		ac97->spec.dev_flags = tmp & 1; /* revision B? */
 		snd_ac97_write_cache(ac97, AC97_INT_PAGING, val);
-	}
+	} else if (ac97->id == AC97_ID_CM9761_83)
+		ac97->spec.dev_flags = 2;
 
 	ac97->build_ops = &patch_cm9761_ops;
 
-- 
cgit v1.2.3-18-g5258


From eb9b4142b3af98d808012081c4dd6d36e2964d43 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 13 Sep 2005 18:39:21 +0200
Subject: [ALSA] Fix a typo

Modules: AC97 Codec

Fix a typo in the last patch.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_patch.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index ffc8d652107..b24beb32961 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -170,7 +170,7 @@ static inline int is_surround_on(ac97_t *ac97)
 
 static inline int is_clfe_on(ac97_t *ac97)
 {
-	return a97->channel_mode >= 2;
+	return ac97->channel_mode >= 2;
 }
 
 static inline int is_shared_linein(ac97_t *ac97)
-- 
cgit v1.2.3-18-g5258


From da3fca21996414a263d36804d9afb2e701abbfef Mon Sep 17 00:00:00 2001
From: Vinod G <vinodg@nvidia.com>
Date: Tue, 13 Sep 2005 18:49:12 +0200
Subject: [ALSA] hda-intel - Add NVidia support

Modules: HDA Intel driver

This patch is to make the Intel HDA code work for NVIDIA azalia controller.

Modified by Takashi Iwai <tiwai@suse.de>

Signed-off-by: Vinod G. <vinodg@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 33 ++++++++++++++++++++++++---------
 1 file changed, 24 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 6fe696e53ea..2c5d4114771 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -223,6 +223,9 @@ enum {
 #define ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR   0x42
 #define ATI_SB450_HDAUDIO_ENABLE_SNOOP      0x02
 
+/* Defines for Nvidia HDA support */
+#define NVIDIA_HDA_TRANSREG_ADDR      0x4e
+#define NVIDIA_HDA_ENABLE_COHBITS     0x0f
 
 /*
  * Use CORB/RIRB for communication from/to codecs.
@@ -328,6 +331,7 @@ enum {
 	AZX_DRIVER_VIA,
 	AZX_DRIVER_SIS,
 	AZX_DRIVER_ULI,
+	AZX_DRIVER_NVIDIA,
 };
 
 static char *driver_short_names[] __devinitdata = {
@@ -335,7 +339,8 @@ static char *driver_short_names[] __devinitdata = {
 	[AZX_DRIVER_ATI] = "HDA ATI SB",
 	[AZX_DRIVER_VIA] = "HDA VIA VT82xx",
 	[AZX_DRIVER_SIS] = "HDA SIS966",
-	[AZX_DRIVER_ULI] = "HDA ULI M5461"
+	[AZX_DRIVER_ULI] = "HDA ULI M5461",
+	[AZX_DRIVER_NVIDIA] = "HDA NVidia",
 };
 
 /*
@@ -710,14 +715,14 @@ static void azx_stream_stop(azx_t *chip, azx_dev_t *azx_dev)
  */
 static void azx_init_chip(azx_t *chip)
 {
-	unsigned char tcsel_reg, ati_misc_cntl2;
+	unsigned char reg;
 
 	/* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
 	 * TCSEL == Traffic Class Select Register, which sets PCI express QOS
 	 * Ensuring these bits are 0 clears playback static on some HD Audio codecs
 	 */
-	pci_read_config_byte (chip->pci, ICH6_PCIREG_TCSEL, &tcsel_reg);
-	pci_write_config_byte(chip->pci, ICH6_PCIREG_TCSEL, tcsel_reg & 0xf8);
+	pci_read_config_byte (chip->pci, ICH6_PCIREG_TCSEL, &reg);
+	pci_write_config_byte(chip->pci, ICH6_PCIREG_TCSEL, reg & 0xf8);
 
 	/* reset controller */
 	azx_reset(chip);
@@ -733,13 +738,21 @@ static void azx_init_chip(azx_t *chip)
 	azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
 	azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr));
 
-	/* For ATI SB450 azalia HD audio, we need to enable snoop */
-	if (chip->driver_type == AZX_DRIVER_ATI) {
+	switch (chip->driver_type) {
+	case AZX_DRIVER_ATI:
+		/* For ATI SB450 azalia HD audio, we need to enable snoop */
 		pci_read_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 
-				     &ati_misc_cntl2);
+				     &reg);
 		pci_write_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR, 
-				      (ati_misc_cntl2 & 0xf8) | ATI_SB450_HDAUDIO_ENABLE_SNOOP);
-	}
+				      (reg & 0xf8) | ATI_SB450_HDAUDIO_ENABLE_SNOOP);
+		break;
+	case AZX_DRIVER_NVIDIA:
+		/* For NVIDIA HDA, enable snoop */
+		pci_read_config_byte(chip->pci,NVIDIA_HDA_TRANSREG_ADDR, &reg);
+		pci_write_config_byte(chip->pci,NVIDIA_HDA_TRANSREG_ADDR,
+				      (reg & 0xf0) | NVIDIA_HDA_ENABLE_COHBITS);
+		break;
+        }
 }
 
 
@@ -1601,6 +1614,8 @@ static struct pci_device_id azx_ids[] = {
 	{ 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 */
+	{ 0x10de, 0x026c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA 026c */
+	{ 0x10de, 0x0371, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA 0371 */
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, azx_ids);
-- 
cgit v1.2.3-18-g5258


From 5747e54042c710272cefed74cc457531a01768c9 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Wed, 14 Sep 2005 08:33:46 +0200
Subject: [ALSA] cmipci: use FM/MIDI ports in PCI port space

Modules: CMIPCI driver

If possible, use ports in the card's PCI port address range instead of
the legacy ports.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/pci/cmipci.c | 118 ++++++++++++++++++++++++++++++-----------------------
 1 file changed, 68 insertions(+), 50 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 1eb3315d136..316afb78488 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -446,9 +446,6 @@ struct snd_stru_cmipci {
 	snd_kcontrol_t *mixer_res_ctl[CM_SAVED_MIXERS];
 	int mixer_res_status[CM_SAVED_MIXERS];
 
-	opl3_t *opl3;
-	snd_hwdep_t *opl3hwdep;
-
 	cmipci_pcm_t channel[2];	/* ch0 - DAC, ch1 - ADC or 2nd DAC */
 
 	/* external MIDI */
@@ -2753,6 +2750,51 @@ static int snd_cmipci_dev_free(snd_device_t *device)
 	return snd_cmipci_free(cm);
 }
 
+static int __devinit snd_cmipci_create_fm(cmipci_t *cm, long fm_port)
+{
+	long iosynth;
+	unsigned int val;
+	opl3_t *opl3;
+	int err;
+
+	/* first try FM regs in PCI port range */
+	iosynth = cm->iobase + CM_REG_FM_PCI;
+	err = snd_opl3_create(cm->card, iosynth, iosynth + 2,
+			      OPL3_HW_OPL3, 1, &opl3);
+	if (err < 0) {
+		/* then try legacy ports */
+		val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL) & ~CM_FMSEL_MASK;
+		iosynth = fm_port;
+		switch (iosynth) {
+		case 0x3E8: val |= CM_FMSEL_3E8; break;
+		case 0x3E0: val |= CM_FMSEL_3E0; break;
+		case 0x3C8: val |= CM_FMSEL_3C8; break;
+		case 0x388: val |= CM_FMSEL_388; break;
+		default:
+			    return 0;
+		}
+		snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val);
+		/* enable FM */
+		snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_FM_EN);
+
+		if (snd_opl3_create(cm->card, iosynth, iosynth + 2,
+				    OPL3_HW_OPL3, 0, &opl3) < 0) {
+			printk(KERN_ERR "cmipci: no OPL device at %#lx, "
+			       "skipping...\n", iosynth);
+			/* disable FM */
+			snd_cmipci_write(cm, CM_REG_LEGACY_CTRL,
+					 val & ~CM_FMSEL_MASK);
+			snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_FM_EN);
+			return 0;
+		}
+	}
+	if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+		printk(KERN_ERR "cmipci: cannot create OPL3 hwdep\n");
+		return err;
+	}
+	return 0;
+}
+
 static int __devinit snd_cmipci_create(snd_card_t *card, struct pci_dev *pci,
 				       int dev, cmipci_t **rcmipci)
 {
@@ -2762,8 +2804,8 @@ static int __devinit snd_cmipci_create(snd_card_t *card, struct pci_dev *pci,
 		.dev_free =	snd_cmipci_dev_free,
 	};
 	unsigned int val = 0;
-	long iomidi = mpu_port[dev];
-	long iosynth = fm_port[dev];
+	long iomidi;
+	int integrated_midi;
 	int pcm_index, pcm_spdif_index;
 	static struct pci_device_id intel_82437vx[] = {
 		{ PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) },
@@ -2867,52 +2909,28 @@ static int __devinit snd_cmipci_create(snd_card_t *card, struct pci_dev *pci,
 		return err;
 	}
 
-	/* set MPU address */
-	switch (iomidi) {
-	case 0x320: val = CM_VMPU_320; break;
-	case 0x310: val = CM_VMPU_310; break;
-	case 0x300: val = CM_VMPU_300; break;
-	case 0x330: val = CM_VMPU_330; break;
-	default:
-		iomidi = 0; break;
-	}
-	if (iomidi > 0) {
-		snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val);
-		/* enable UART */
-		snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_UART_EN);
-	}
-
-	/* set FM address */
-	val = snd_cmipci_read(cm, CM_REG_LEGACY_CTRL) & ~CM_FMSEL_MASK;
-	switch (iosynth) {
-	case 0x3E8: val |= CM_FMSEL_3E8; break;
-	case 0x3E0: val |= CM_FMSEL_3E0; break;
-	case 0x3C8: val |= CM_FMSEL_3C8; break;
-	case 0x388: val |= CM_FMSEL_388; break;
-	default:
-		iosynth = 0; break;
-	}
-	if (iosynth > 0) {
-		snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val);
-		/* enable FM */
-		snd_cmipci_set_bit(cm, CM_REG_MISC_CTRL, CM_FM_EN);
-
-		if (snd_opl3_create(card, iosynth, iosynth + 2,
-				    OPL3_HW_OPL3, 0, &cm->opl3) < 0) {
-			printk(KERN_ERR "cmipci: no OPL device at 0x%lx, skipping...\n", iosynth);
-			iosynth = 0;
-		} else {
-			if ((err = snd_opl3_hwdep_new(cm->opl3, 0, 1, &cm->opl3hwdep)) < 0) {
-				printk(KERN_ERR "cmipci: cannot create OPL3 hwdep\n");
-				return err;
-			}
+	integrated_midi = snd_cmipci_read_b(cm, CM_REG_MPU_PCI) != 0xff;
+	if (integrated_midi)
+		iomidi = cm->iobase + CM_REG_MPU_PCI;
+	else {
+		iomidi = mpu_port[dev];
+		switch (iomidi) {
+		case 0x320: val = CM_VMPU_320; break;
+		case 0x310: val = CM_VMPU_310; break;
+		case 0x300: val = CM_VMPU_300; break;
+		case 0x330: val = CM_VMPU_330; break;
+		default:
+			    iomidi = 0; break;
+		}
+		if (iomidi > 0) {
+			snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val);
+			/* enable UART */
+			snd_cmipci_set_bit(cm, CM_REG_FUNCTRL1, CM_UART_EN);
 		}
 	}
-	if (! iosynth) {
-		/* disable FM */
-		snd_cmipci_write(cm, CM_REG_LEGACY_CTRL, val & ~CM_FMSEL_MASK);
-		snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_FM_EN);
-	}
+
+	if ((err = snd_cmipci_create_fm(cm, fm_port[dev])) < 0)
+		return err;
 
 	/* reset mixer */
 	snd_cmipci_mixer_write(cm, 0, 0);
@@ -2941,7 +2959,7 @@ static int __devinit snd_cmipci_create(snd_card_t *card, struct pci_dev *pci,
 
 	if (iomidi > 0) {
 		if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI,
-					       iomidi, 0,
+					       iomidi, integrated_midi,
 					       cm->irq, 0, &cm->rmidi)) < 0) {
 			printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi);
 		}
-- 
cgit v1.2.3-18-g5258


From d1bda0455478a9e2a13381044e9bb80a17ec92d2 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Wed, 14 Sep 2005 08:36:03 +0200
Subject: [ALSA] usb-audio: simplify MIDI quirk handling

Modules: USB generic driver

Simplify the handling of MIDI quirks by treating an interface without
quirks as a QUIRK_MIDI_STANDARD_INTERFACE.

This also fixes the bug where a MIDI_STANDARD quirk would not be
recognized.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/usb/usbaudio.c | 24 +++++-----------
 sound/usb/usbmidi.c  | 77 ++++++++++++++++++++++++++--------------------------
 2 files changed, 45 insertions(+), 56 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 2ead878bcb8..ea1b1f87c1b 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -2755,9 +2755,9 @@ static int create_fixed_stream_quirk(snd_usb_audio_t *chip,
 /*
  * create a stream for an interface with proper descriptors
  */
-static int create_standard_interface_quirk(snd_usb_audio_t *chip,
-					   struct usb_interface *iface,
-					   const snd_usb_audio_quirk_t *quirk)
+static int create_standard_audio_quirk(snd_usb_audio_t *chip,
+				       struct usb_interface *iface,
+				       const snd_usb_audio_quirk_t *quirk)
 {
 	struct usb_host_interface *alts;
 	struct usb_interface_descriptor *altsd;
@@ -2765,24 +2765,14 @@ static int create_standard_interface_quirk(snd_usb_audio_t *chip,
 
 	alts = &iface->altsetting[0];
 	altsd = get_iface_desc(alts);
-	switch (quirk->type) {
-	case QUIRK_AUDIO_STANDARD_INTERFACE:
-		err = parse_audio_endpoints(chip, altsd->bInterfaceNumber);
-		if (!err)
-			usb_set_interface(chip->dev, altsd->bInterfaceNumber, 0); /* reset the current interface */
-		break;
-	case QUIRK_MIDI_STANDARD_INTERFACE:
-		err = snd_usb_create_midi_interface(chip, iface, NULL);
-		break;
-	default:
-		snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type);
-		return -ENXIO;
-	}
+	err = parse_audio_endpoints(chip, altsd->bInterfaceNumber);
 	if (err < 0) {
 		snd_printk(KERN_ERR "cannot setup if %d: error %d\n",
 			   altsd->bInterfaceNumber, err);
 		return err;
 	}
+	/* reset the current interface */
+	usb_set_interface(chip->dev, altsd->bInterfaceNumber, 0);
 	return 0;
 }
 
@@ -3044,7 +3034,7 @@ static int snd_usb_create_quirk(snd_usb_audio_t *chip,
 		[QUIRK_MIDI_RAW] = snd_usb_create_midi_interface,
 		[QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface,
 		[QUIRK_MIDI_MIDITECH] = snd_usb_create_midi_interface,
-		[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_interface_quirk,
+		[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
 		[QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
 		[QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk,
 		[QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk,
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index f1a2e2c2e02..32be9f99c02 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -1550,46 +1550,45 @@ int snd_usb_create_midi_interface(snd_usb_audio_t* chip,
 
 	/* detect the endpoint(s) to use */
 	memset(endpoints, 0, sizeof(endpoints));
-	if (!quirk) {
+	switch (quirk ? quirk->type : QUIRK_MIDI_STANDARD_INTERFACE) {
+	case QUIRK_MIDI_STANDARD_INTERFACE:
 		err = snd_usbmidi_get_ms_info(umidi, endpoints);
-	} else {
-		switch (quirk->type) {
-		case QUIRK_MIDI_FIXED_ENDPOINT:
-			memcpy(&endpoints[0], quirk->data,
-			       sizeof(snd_usb_midi_endpoint_info_t));
-			err = snd_usbmidi_detect_endpoints(umidi, &endpoints[0], 1);
-			break;
-		case QUIRK_MIDI_YAMAHA:
-			err = snd_usbmidi_detect_yamaha(umidi, &endpoints[0]);
-			break;
-		case QUIRK_MIDI_MIDIMAN:
-			umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops;
-			memcpy(&endpoints[0], quirk->data,
-			       sizeof(snd_usb_midi_endpoint_info_t));
-			err = 0;
-			break;
-		case QUIRK_MIDI_NOVATION:
-			umidi->usb_protocol_ops = &snd_usbmidi_novation_ops;
-			err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
-			break;
-		case QUIRK_MIDI_RAW:
-			umidi->usb_protocol_ops = &snd_usbmidi_raw_ops;
-			err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
-			break;
-		case QUIRK_MIDI_EMAGIC:
-			umidi->usb_protocol_ops = &snd_usbmidi_emagic_ops;
-			memcpy(&endpoints[0], quirk->data,
-			       sizeof(snd_usb_midi_endpoint_info_t));
-			err = snd_usbmidi_detect_endpoints(umidi, &endpoints[0], 1);
-			break;
-		case QUIRK_MIDI_MIDITECH:
-			err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
-			break;
-		default:
-			snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type);
-			err = -ENXIO;
-			break;
-		}
+		break;
+	case QUIRK_MIDI_FIXED_ENDPOINT:
+		memcpy(&endpoints[0], quirk->data,
+		       sizeof(snd_usb_midi_endpoint_info_t));
+		err = snd_usbmidi_detect_endpoints(umidi, &endpoints[0], 1);
+		break;
+	case QUIRK_MIDI_YAMAHA:
+		err = snd_usbmidi_detect_yamaha(umidi, &endpoints[0]);
+		break;
+	case QUIRK_MIDI_MIDIMAN:
+		umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops;
+		memcpy(&endpoints[0], quirk->data,
+		       sizeof(snd_usb_midi_endpoint_info_t));
+		err = 0;
+		break;
+	case QUIRK_MIDI_NOVATION:
+		umidi->usb_protocol_ops = &snd_usbmidi_novation_ops;
+		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+		break;
+	case QUIRK_MIDI_RAW:
+		umidi->usb_protocol_ops = &snd_usbmidi_raw_ops;
+		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+		break;
+	case QUIRK_MIDI_EMAGIC:
+		umidi->usb_protocol_ops = &snd_usbmidi_emagic_ops;
+		memcpy(&endpoints[0], quirk->data,
+		       sizeof(snd_usb_midi_endpoint_info_t));
+		err = snd_usbmidi_detect_endpoints(umidi, &endpoints[0], 1);
+		break;
+	case QUIRK_MIDI_MIDITECH:
+		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+		break;
+	default:
+		snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type);
+		err = -ENXIO;
+		break;
 	}
 	if (err < 0) {
 		kfree(umidi);
-- 
cgit v1.2.3-18-g5258


From dfedc5f47cfd672a57923506c71b2042b33ae306 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Fri, 16 Sep 2005 16:56:01 +0200
Subject: [ALSA] usb-audio: ignore Hercules DJ Console mixer errors

Modules: USB generic driver

Add a quirk entry for the Hercules DJ Console to ignore timeouts on
some mixer control transfers.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/usb/usbmixer_maps.c | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'sound')

diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c
index c1264434e50..9a8534b9eff 100644
--- a/sound/usb/usbmixer_maps.c
+++ b/sound/usb/usbmixer_maps.c
@@ -247,6 +247,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.id = USB_ID(0x06f8, 0xd002),
 		.ignore_ctl_error = 1,
 	},
+	{
+		/* Hercules DJ Console */
+		.id = USB_ID(0x06f8, 0xd002),
+		.ignore_ctl_error = 1,
+	},
 	{
 		.id = USB_ID(0x08bb, 0x2702),
 		.map = linex_map,
-- 
cgit v1.2.3-18-g5258


From a9430dd8fc232cfddcfaedde1a6a915e241366a8 Mon Sep 17 00:00:00 2001
From: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Date: Fri, 16 Sep 2005 19:12:48 +0200
Subject: [ALSA] hda-codec - A new model for Fujitsu S7020

Modules: HDA Codec driver

Added a new model 'fujitsu' to ALC260 config for Fujitsu S7020.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 160 +++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 157 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7327deb6df9..48356ab0457 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -57,6 +57,7 @@ enum {
 enum {
 	ALC260_BASIC,
 	ALC260_HP,
+	ALC260_FUJITSU_S702x,
 	ALC260_MODEL_LAST /* last tag */
 };
 
@@ -72,6 +73,7 @@ enum {
 #define PIN_VREF50	0x21
 #define PIN_OUT		0x40
 #define PIN_HP		0xc0
+#define PIN_HP_AMP	0x80
 
 struct alc_spec {
 	/* codec parameterization */
@@ -284,6 +286,54 @@ static int alc_bind_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *u
 
 #define ALC_BIND_MUTE(xname,nid,indices,dir) ALC_BIND_MUTE_MONO(xname,nid,3,indices,dir)
 
+/*
+ * Control of pin widget settings via the mixer.  Only boolean settings are
+ * supported, so VrefEn can't be controlled using these functions as they
+ * stand.
+ */
+static int alc_pinctl_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+
+static int alc_pinctl_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	long mask = (kcontrol->private_value >> 16) & 0xff;
+	long *valp = ucontrol->value.integer.value;
+
+	*valp = 0;
+	if (snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00) & mask)
+		*valp = 1;
+	return 0;
+}
+
+static int alc_pinctl_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value & 0xffff;
+	long mask = (kcontrol->private_value >> 16) & 0xff;
+	long *valp = ucontrol->value.integer.value;
+	unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00);
+	int change = ((pinctl & mask)!=0) != *valp;
+
+	if (change)
+		snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL,
+			*valp?(pinctl|mask):(pinctl&~mask));
+	return change;
+}
+
+#define ALC_PINCTL_SWITCH(xname, nid, mask) \
+	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0,  \
+	  .info = alc_pinctl_switch_info, \
+	  .get = alc_pinctl_switch_get, \
+	  .put = alc_pinctl_switch_put, \
+	  .private_value = (nid) | (mask<<16) }
 
 /*
  * ALC880 3-stack model
@@ -2205,6 +2255,17 @@ static struct hda_input_mux alc260_capture_source = {
 	},
 };
 
+/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack
+ * and the internal CD lines.
+ */
+static struct hda_input_mux alc260_fujitsu_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic/Line", 0x0 },
+		{ "CD", 0x4 },
+	},
+};
+
 /*
  * This is just place-holder, so there's something for alc_build_pcms to look
  * at when it calculates the maximum number of channels. ALC260 has no mixer
@@ -2271,6 +2332,30 @@ static snd_kcontrol_new_t alc260_hp_mixer[] = {
 	{ } /* end */
 };
 
+static snd_kcontrol_new_t alc260_fujitsu_mixer[] = {
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+	ALC_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
+	ALC_PINCTL_SWITCH("Headphone Amp Switch", 0x14, PIN_HP_AMP),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
+	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
+	HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+	ALC_BIND_MUTE("Internal Speaker Playback Switch", 0x09, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
+	},
+	{ } /* end */
+};
+
 static struct hda_verb alc260_init_verbs[] = {
 	/* Line In pin widget for input */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
@@ -2332,6 +2417,60 @@ static struct hda_verb alc260_init_verbs[] = {
 	{ }
 };
 
+/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
+ * laptops.
+ */
+static struct hda_verb alc260_fujitsu_init_verbs[] = {
+	/* Disable all GPIOs */
+	{0x01, AC_VERB_SET_GPIO_MASK, 0},
+	/* Internal speaker is connected to headphone pin */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Headphone/Line-out jack connects to Line1 pin; make it an output */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+        /* Mic/Line-in jack is connected to mic1 pin, so make it an input */
+        {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+        /* Ensure all other unused pins are disabled and muted.
+	 * Note: trying to set widget 0x15 to anything blocks all audio
+	 * output for some reason, so just leave that at the default.
+	 */
+        {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+        {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+        {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+        {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        /* Disable digital (SPDIF) pins */
+        {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+        {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+        /* Start with mixer outputs muted */
+        {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+        /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
+        {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        /* Unmute Line1 pin widget amp left and right (no equiv mixer ctrl) */
+        {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Unmute pin widget used for Line-in (no equiv mixer ctrl) */
+        {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+        /* Mute capture amp left and right */
+        {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        /* Set ADC connection select to line in (on mic1 pin) */
+        {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+        /* Mute all inputs to mixer widget (even unconnected ones) */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+};
+
 static struct hda_pcm_stream alc260_pcm_analog_playback = {
 	.substreams = 1,
 	.channels_min = 2,
@@ -2347,6 +2486,8 @@ static struct hda_pcm_stream alc260_pcm_analog_capture = {
 static struct hda_board_config alc260_cfg_tbl[] = {
 	{ .modelname = "hp", .config = ALC260_HP },
 	{ .pci_subvendor = 0x103c, .config = ALC260_HP },
+	{ .modelname = "fujitsu", .config = ALC260_FUJITSU_S702x },
+	{ .pci_subvendor = 0x10cf, .pci_subdevice = 0x1326, .config = ALC260_FUJITSU_S702x },
 	{}
 };
 
@@ -2373,14 +2514,23 @@ static int patch_alc260(struct hda_codec *codec)
 		spec->mixers[spec->num_mixers] = alc260_hp_mixer;
 		spec->num_mixers++;
 		break;
+	case ALC260_FUJITSU_S702x:
+		spec->mixers[spec->num_mixers] = alc260_fujitsu_mixer;
+		spec->num_mixers++;
+		break;
 	default:
 		spec->mixers[spec->num_mixers] = alc260_base_mixer;
 		spec->num_mixers++;
 		break;
 	}
 
-	spec->init_verbs[0] = alc260_init_verbs;
-	spec->num_init_verbs = 1;
+	if (board_config != ALC260_FUJITSU_S702x) {
+		spec->init_verbs[0] = alc260_init_verbs;
+		spec->num_init_verbs = 1;
+	} else {
+		spec->init_verbs[0] = alc260_fujitsu_init_verbs;
+		spec->num_init_verbs = 1;
+	}
 
 	spec->channel_mode = alc260_modes;
 	spec->num_channel_mode = ARRAY_SIZE(alc260_modes);
@@ -2393,7 +2543,11 @@ static int patch_alc260(struct hda_codec *codec)
 	spec->multiout.num_dacs = ARRAY_SIZE(alc260_dac_nids);
 	spec->multiout.dac_nids = alc260_dac_nids;
 
-	spec->input_mux = &alc260_capture_source;
+	if (board_config != ALC260_FUJITSU_S702x) {
+		spec->input_mux = &alc260_capture_source;
+	} else {
+		spec->input_mux = &alc260_fujitsu_capture_source;
+	}
 	switch (board_config) {
 	case ALC260_HP:
 		spec->num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids);
-- 
cgit v1.2.3-18-g5258


From f2b31737e509a70952421e7cca488a2001a3b087 Mon Sep 17 00:00:00 2001
From: Sasha Khapyorsky <sashak@smlink.com>
Date: Fri, 16 Sep 2005 19:22:44 +0200
Subject: [ALSA] au88x0: codec access procs for multiple AC97 codecs

Modules: au88x0 driver

This patch extends au88x0 AC97 codec access procedures to handle multiple
codecs properly.

Signed-off-by: Sasha Khapyorsky <sashak@smlink.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/au88x0/au8810.h      | 5 -----
 sound/pci/au88x0/au8820.h      | 5 -----
 sound/pci/au88x0/au8830.h      | 5 -----
 sound/pci/au88x0/au88x0.h      | 8 ++++++++
 sound/pci/au88x0/au88x0_core.c | 6 ++++--
 5 files changed, 12 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/au88x0/au8810.h b/sound/pci/au88x0/au8810.h
index 3837d2ba5e6..5d69c31fe3f 100644
--- a/sound/pci/au88x0/au8810.h
+++ b/sound/pci/au88x0/au8810.h
@@ -178,11 +178,6 @@
 #define		EN_SPDIF	0x000c0000
 
 #define VORTEX_CODEC_CHN 	0x29080
-#define VORTEX_CODEC_WRITE	0x00800000
-#define VORTEX_CODEC_ADDSHIFT 	16
-#define VORTEX_CODEC_ADDMASK	0x7f0000	/* 0x000f0000 */
-#define VORTEX_CODEC_DATSHIFT	0
-#define VORTEX_CODEC_DATMASK	0xffff
 #define VORTEX_CODEC_IO		0x29188
 
 /* SPDIF */
diff --git a/sound/pci/au88x0/au8820.h b/sound/pci/au88x0/au8820.h
index be8022e7871..abbe85e4f7a 100644
--- a/sound/pci/au88x0/au8820.h
+++ b/sound/pci/au88x0/au8820.h
@@ -162,11 +162,6 @@
 #define		EN_SPORT	0x00030000
 #define		EN_SPDIF	0x000c0000
 #define VORTEX_CODEC_CHN 0x11880
-#define VORTEX_CODEC_WRITE 0x00800000
-#define VORTEX_CODEC_ADDSHIFT 16
-#define VORTEX_CODEC_ADDMASK 0x7f0000	/* 0x000f0000 */
-#define VORTEX_CODEC_DATSHIFT 0
-#define VORTEX_CODEC_DATMASK 0xffff
 #define VORTEX_CODEC_IO 0x11988
 
 #define VORTEX_SPDIF_FLAGS		0x1005c	/* FIXME */
diff --git a/sound/pci/au88x0/au8830.h b/sound/pci/au88x0/au8830.h
index aa77826b5e5..04ece1b1c21 100644
--- a/sound/pci/au88x0/au8830.h
+++ b/sound/pci/au88x0/au8830.h
@@ -194,11 +194,6 @@
 
 #define VORTEX_CODEC_CTRL 0x29184
 #define VORTEX_CODEC_IO 0x29188
-#define 	VORTEX_CODEC_WRITE 0x00800000
-#define 	VORTEX_CODEC_ADDSHIFT 16
-#define 	VORTEX_CODEC_ADDMASK 0x7f0000	/* 0x000f0000 */
-#define 	VORTEX_CODEC_DATSHIFT 0
-#define 	VORTEX_CODEC_DATMASK 0xffff
 
 #define VORTEX_CODEC_SPORTCTRL 0x2918c
 
diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h
index ee1ede1979f..b1197cfab3f 100644
--- a/sound/pci/au88x0/au88x0.h
+++ b/sound/pci/au88x0/au88x0.h
@@ -79,6 +79,14 @@
 #define VORTEX_RESOURCE_A3D	0x00000004
 #define VORTEX_RESOURCE_LAST	0x00000005
 
+/* codec io: VORTEX_CODEC_IO bits */
+#define VORTEX_CODEC_ID_SHIFT	24
+#define VORTEX_CODEC_WRITE	0x00800000
+#define VORTEX_CODEC_ADDSHIFT 	16
+#define VORTEX_CODEC_ADDMASK	0x7f0000
+#define VORTEX_CODEC_DATSHIFT	0
+#define VORTEX_CODEC_DATMASK	0xffff
+
 /* Check for SDAC bit in "Extended audio ID" AC97 register */
 //#define VORTEX_IS_QUAD(x) (((x)->codec == NULL) ?  0 : ((x)->codec->ext_id&0x80))
 #define VORTEX_IS_QUAD(x) ((x)->isquad)
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index f0eda4bbbb3..91ac4f3a2aa 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -2532,7 +2532,8 @@ vortex_codec_write(ac97_t * codec, unsigned short addr, unsigned short data)
 	hwwrite(card->mmio, VORTEX_CODEC_IO,
 		((addr << VORTEX_CODEC_ADDSHIFT) & VORTEX_CODEC_ADDMASK) |
 		((data << VORTEX_CODEC_DATSHIFT) & VORTEX_CODEC_DATMASK) |
-		VORTEX_CODEC_WRITE);
+		VORTEX_CODEC_WRITE |
+		(codec->num << VORTEX_CODEC_ID_SHIFT) );
 
 	/* Flush Caches. */
 	hwread(card->mmio, VORTEX_CODEC_IO);
@@ -2554,7 +2555,8 @@ static unsigned short vortex_codec_read(ac97_t * codec, unsigned short addr)
 		}
 	}
 	/* set up read address */
-	read_addr = ((addr << VORTEX_CODEC_ADDSHIFT) & VORTEX_CODEC_ADDMASK);
+	read_addr = ((addr << VORTEX_CODEC_ADDSHIFT) & VORTEX_CODEC_ADDMASK) |
+		(codec->num << VORTEX_CODEC_ID_SHIFT) ;
 	hwwrite(card->mmio, VORTEX_CODEC_IO, read_addr);
 
 	/* wait for address */
-- 
cgit v1.2.3-18-g5258


From d879f0ccf55a9ac5752b24dc210a446746e3c6da Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Wed, 21 Sep 2005 16:33:49 +0200
Subject: [ALSA] usb-audio: add quirk comments

Modules: USB generic driver

Add more comments about other device modes and unsupported devices to
the Roland part of the quirks table.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/usb/usbquirks.h | 40 +++++++++++++++++++++++++++++++++++++++-
 1 file changed, 39 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 948759da656..ba506c3871f 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -294,6 +294,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* a later revision uses ID 0x0099 */
 	USB_DEVICE(0x0582, 0x0005),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -384,6 +385,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* a later revision uses ID 0x009d */
 	USB_DEVICE(0x0582, 0x0009),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -532,6 +534,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0013 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0012),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -545,6 +548,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0015 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0014),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -558,6 +562,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0017 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0016),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -588,6 +593,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x001c when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x001b),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -618,6 +624,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x001e when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x001d),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -631,6 +638,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0024 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0023),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -675,6 +683,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0028 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0027),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -688,6 +697,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x002a when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0029),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -732,6 +742,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x002e when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x002d),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -745,6 +756,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0030 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x002f),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -758,6 +770,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0034 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0033),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -770,7 +783,12 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+	/* TODO: add Roland M-1000 support */
 {
+	/*
+	 * Has ID 0x0038 when not in "Advanced Driver" mode;
+	 * later revisions use IDs 0x0054 and 0x00a2.
+	 */
 	USB_DEVICE(0x0582, 0x0037),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -815,6 +833,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0041 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0040),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -828,6 +847,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0043 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0042),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -871,6 +891,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x004a when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0048),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -883,7 +904,9 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+	/* TODO: add Edirol M-100FX support */
 {
+	/* has ID 0x004f when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x004d),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -931,7 +954,9 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		.type = QUIRK_MIDI_STANDARD_INTERFACE
 	}
 },
+	/* TODO: add Roland EXR support */
 {
+	/* has ID 0x0067 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0065),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "EDIROL",
@@ -945,6 +970,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x006b when not in "Advanced Driver" mode */
 	USB_DEVICE_VENDOR_SPEC(0x0582, 0x006a),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -958,6 +984,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x006e when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x006d),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
@@ -1002,6 +1029,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x0076 when not in "Advanced Driver" mode */
 	USB_DEVICE(0x0582, 0x0075),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "BOSS",
@@ -1015,10 +1043,11 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
+	/* has ID 0x007b when not in "Advanced Driver" mode */
 	USB_DEVICE_VENDOR_SPEC(0x0582, 0x007a),
 	.driver_info = (unsigned long) & (const snd_usb_audio_quirk_t) {
 		.vendor_name = "Roland",
-		/* RD-700SX, RD-300SX */
+		/* "RD" or "RD-700SX"? */
 		.ifnum = 0,
 		.type = QUIRK_MIDI_FIXED_ENDPOINT,
 		.data = & (const snd_usb_midi_endpoint_info_t) {
@@ -1048,6 +1077,15 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 		}
 	}
 },
+	/* TODO: add Edirol UA-101 support */
+	/* TODO: add Roland G-70 support */
+	/* TODO: add Roland V-SYNTH XT support */
+	/* TODO: add BOSS GT-PRO support */
+	/* TODO: add Edirol PC-50 support */
+	/* TODO: add Edirol PC-80 support */
+	/* TODO: add Edirol UA-1EX support */
+	/* TODO: add Edirol UM-3 support */
+	/* TODO: add Edirol MD-P1 support */
 
 /* Midiman/M-Audio devices */
 {
-- 
cgit v1.2.3-18-g5258


From 3cfc1eb181b49dfbfeaf20ebfeb5cd2af4c4c4b5 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Mon, 26 Sep 2005 10:01:12 +0200
Subject: [ALSA] usb-audio: remove old compatibility wrappers (1/2)

Modules: USB generic driver

Move the usb_pipe_needs_resubmit() compatibility wrapper out of the
kernel tree.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/usb/usbaudio.h | 4 ----
 sound/usb/usbmidi.c  | 6 ++----
 2 files changed, 2 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index ad9eab211d8..b58dc0ca7d3 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -249,10 +249,6 @@ void snd_usbmidi_disconnect(struct list_head *p);
 #define get_cfg_desc(cfg)	(&(cfg)->desc)
 #endif
 
-#ifndef usb_pipe_needs_resubmit
-#define usb_pipe_needs_resubmit(pipe) 1
-#endif
-
 #ifndef snd_usb_complete_callback
 #define snd_usb_complete_callback(x) (x)
 #endif
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 32be9f99c02..369319d672b 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -246,10 +246,8 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb, struct pt_regs *regs)
 		}
 	}
 
-	if (usb_pipe_needs_resubmit(urb->pipe)) {
-		urb->dev = ep->umidi->chip->dev;
-		snd_usbmidi_submit_urb(urb, GFP_ATOMIC);
-	}
+	urb->dev = ep->umidi->chip->dev;
+	snd_usbmidi_submit_urb(urb, GFP_ATOMIC);
 }
 
 static void snd_usbmidi_out_urb_complete(struct urb* urb, struct pt_regs *regs)
-- 
cgit v1.2.3-18-g5258


From 3527a008cbc51dd23a8bf4927e3ab29423b0d05b Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Mon, 26 Sep 2005 10:03:09 +0200
Subject: [ALSA] usb-audio: remove old compatibility wrappers (2/2)

Modules: USB generic driver

Move the usb_complete_callback() compatibility wrapper out of the
kernel tree.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/usb/usbaudio.c |  4 ++--
 sound/usb/usbaudio.h |  4 ----
 sound/usb/usbmidi.c  | 14 ++++++--------
 3 files changed, 8 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index ea1b1f87c1b..5429b169877 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -1044,7 +1044,7 @@ static int init_substream_urbs(snd_usb_substream_t *subs, unsigned int period_by
 		u->urb->transfer_flags = URB_ISO_ASAP | URB_NO_TRANSFER_DMA_MAP;
 		u->urb->interval = 1 << subs->datainterval;
 		u->urb->context = u;
-		u->urb->complete = snd_usb_complete_callback(snd_complete_urb);
+		u->urb->complete = snd_complete_urb;
 	}
 
 	if (subs->syncpipe) {
@@ -1070,7 +1070,7 @@ static int init_substream_urbs(snd_usb_substream_t *subs, unsigned int period_by
 			u->urb->number_of_packets = 1;
 			u->urb->interval = 1 << subs->syncinterval;
 			u->urb->context = u;
-			u->urb->complete = snd_usb_complete_callback(snd_complete_sync_urb);
+			u->urb->complete = snd_complete_sync_urb;
 		}
 	}
 	return 0;
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index b58dc0ca7d3..b5802022bb1 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -249,10 +249,6 @@ void snd_usbmidi_disconnect(struct list_head *p);
 #define get_cfg_desc(cfg)	(&(cfg)->desc)
 #endif
 
-#ifndef snd_usb_complete_callback
-#define snd_usb_complete_callback(x) (x)
-#endif
-
 #ifndef snd_usb_get_speed
 #define snd_usb_get_speed(dev) ((dev)->speed)
 #endif
diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c
index 369319d672b..2885b8974da 100644
--- a/sound/usb/usbmidi.c
+++ b/sound/usb/usbmidi.c
@@ -861,13 +861,12 @@ static int snd_usbmidi_in_endpoint_create(snd_usb_midi_t* umidi,
 		return -ENOMEM;
 	}
 	if (ep_info->in_interval)
-		usb_fill_int_urb(ep->urb, umidi->chip->dev, pipe, buffer, length,
-				 snd_usb_complete_callback(snd_usbmidi_in_urb_complete),
-				 ep, ep_info->in_interval);
+		usb_fill_int_urb(ep->urb, umidi->chip->dev, pipe, buffer,
+				 length, snd_usbmidi_in_urb_complete, ep,
+				 ep_info->in_interval);
 	else
-		usb_fill_bulk_urb(ep->urb, umidi->chip->dev, pipe, buffer, length,
-				  snd_usb_complete_callback(snd_usbmidi_in_urb_complete),
-				  ep);
+		usb_fill_bulk_urb(ep->urb, umidi->chip->dev, pipe, buffer,
+				  length, snd_usbmidi_in_urb_complete, ep);
 	ep->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
 
 	rep->in = ep;
@@ -931,8 +930,7 @@ static int snd_usbmidi_out_endpoint_create(snd_usb_midi_t* umidi,
 		return -ENOMEM;
 	}
 	usb_fill_bulk_urb(ep->urb, umidi->chip->dev, pipe, buffer,
-			  ep->max_transfer,
-			  snd_usb_complete_callback(snd_usbmidi_out_urb_complete), ep);
+			  ep->max_transfer, snd_usbmidi_out_urb_complete, ep);
 	ep->urb->transfer_flags = URB_NO_TRANSFER_DMA_MAP;
 
 	spin_lock_init(&ep->buffer_lock);
-- 
cgit v1.2.3-18-g5258


From adf25df1be2e3843f786a2562202c7897bbd149d Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Tue, 27 Sep 2005 15:56:28 +0200
Subject: [ALSA] rtctimer: remove superfluous rtc_inc variable

Modules: RTC timer driver

The rtc_inc variable is never used outside the interrupt handler, and
is always one where it matters, so we can just remove it.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/core/rtctimer.c | 9 +--------
 1 file changed, 1 insertion(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c
index bd5d584d284..85627dbe6a5 100644
--- a/sound/core/rtctimer.c
+++ b/sound/core/rtctimer.c
@@ -60,7 +60,6 @@ static struct _snd_timer_hardware rtc_hw = {
 
 static int rtctimer_freq = RTC_FREQ;		/* frequency */
 static snd_timer_t *rtctimer;
-static atomic_t rtc_inc = ATOMIC_INIT(0);
 static rtc_task_t rtc_task;
 
 
@@ -94,7 +93,6 @@ rtctimer_start(snd_timer_t *timer)
 	snd_assert(rtc != NULL, return -EINVAL);
 	rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq);
 	rtc_control(rtc, RTC_PIE_ON, 0);
-	atomic_set(&rtc_inc, 0);
 	return 0;
 }
 
@@ -112,12 +110,7 @@ rtctimer_stop(snd_timer_t *timer)
  */
 static void rtctimer_interrupt(void *private_data)
 {
-	int ticks;
-
-	atomic_inc(&rtc_inc);
-	ticks = atomic_read(&rtc_inc);
-	snd_timer_interrupt((snd_timer_t*)private_data, ticks);
-	atomic_sub(ticks, &rtc_inc);
+	snd_timer_interrupt(private_data, 1);
 }
 
 
-- 
cgit v1.2.3-18-g5258


From d9ad1bdd6d72606a59cdc07e571fbe695e32f271 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Tue, 27 Sep 2005 15:57:24 +0200
Subject: [ALSA] rtctimer: optimize module parameter validation

Modules: RTC timer driver

The check whether rtctimer_freq is a power of two can be done easier
with a simple bit operation.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/core/rtctimer.c | 11 +++--------
 1 file changed, 3 insertions(+), 8 deletions(-)

(limited to 'sound')

diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c
index 85627dbe6a5..8762ff8938c 100644
--- a/sound/core/rtctimer.c
+++ b/sound/core/rtctimer.c
@@ -119,16 +119,11 @@ static void rtctimer_interrupt(void *private_data)
  */
 static int __init rtctimer_init(void)
 {
-	int order, err;
+	int err;
 	snd_timer_t *timer;
 
-	if (rtctimer_freq < 2 || rtctimer_freq > 8192) {
-		snd_printk(KERN_ERR "rtctimer: invalid frequency %d\n", rtctimer_freq);
-		return -EINVAL;
-	}
-	for (order = 1; rtctimer_freq > order; order <<= 1)
-		;
-	if (rtctimer_freq != order) {
+	if (rtctimer_freq < 2 || rtctimer_freq > 8192 ||
+	    (rtctimer_freq & (rtctimer_freq - 1)) != 0) {
 		snd_printk(KERN_ERR "rtctimer: invalid frequency %d\n", rtctimer_freq);
 		return -EINVAL;
 	}
-- 
cgit v1.2.3-18-g5258


From bf3b644039d6a99076f5a87c714d94dfea3a6814 Mon Sep 17 00:00:00 2001
From: Clemens Ladisch <clemens@ladisch.de>
Date: Wed, 28 Sep 2005 08:18:17 +0200
Subject: [ALSA] sequencer: remove superfluous function parameter

Modules: ALSA sequencer

Remove the last parameter of snd_seq_timer_set_tick_resolution()
because it is always one.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/core/seq/seq_timer.c | 9 ++++-----
 1 file changed, 4 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c
index b57a3c07ff6..9c87d9f8ba9 100644
--- a/sound/core/seq/seq_timer.c
+++ b/sound/core/seq/seq_timer.c
@@ -37,7 +37,7 @@ extern int seq_default_timer_resolution;
 #define SKEW_BASE	0x10000	/* 16bit shift */
 
 static void snd_seq_timer_set_tick_resolution(seq_timer_tick_t *tick,
-					      int tempo, int ppq, int nticks)
+					      int tempo, int ppq)
 {
 	if (tempo < 1000000)
 		tick->resolution = (tempo * 1000) / ppq;
@@ -51,7 +51,6 @@ static void snd_seq_timer_set_tick_resolution(seq_timer_tick_t *tick,
 	}
 	if (tick->resolution <= 0)
 		tick->resolution = 1;
-	tick->resolution *= nticks;
 	snd_seq_timer_update_tick(tick, 0);
 }
 
@@ -100,7 +99,7 @@ void snd_seq_timer_defaults(seq_timer_t * tmr)
 	/* setup defaults */
 	tmr->ppq = 96;		/* 96 PPQ */
 	tmr->tempo = 500000;	/* 120 BPM */
-	snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq, 1);
+	snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq);
 	tmr->running = 0;
 
 	tmr->type = SNDRV_SEQ_TIMER_ALSA;
@@ -183,7 +182,7 @@ int snd_seq_timer_set_tempo(seq_timer_t * tmr, int tempo)
 	spin_lock_irqsave(&tmr->lock, flags);
 	if ((unsigned int)tempo != tmr->tempo) {
 		tmr->tempo = tempo;
-		snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq, 1);
+		snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq);
 	}
 	spin_unlock_irqrestore(&tmr->lock, flags);
 	return 0;
@@ -207,7 +206,7 @@ int snd_seq_timer_set_ppq(seq_timer_t * tmr, int ppq)
 	}
 
 	tmr->ppq = ppq;
-	snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq, 1);
+	snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq);
 	spin_unlock_irqrestore(&tmr->lock, flags);
 	return 0;
 }
-- 
cgit v1.2.3-18-g5258


From d602c885a6c30c78197b752c9241ce7e86cdf704 Mon Sep 17 00:00:00 2001
From: Glen Masgai <mimosius@gmx.de>
Date: Wed, 28 Sep 2005 08:19:05 +0200
Subject: [ALSA] ymfpci: add S/PDIF-in > S/PDIF-out loop

Modules: YMFPCI driver

This patch adds a new mixer control called 'IEC958 Loop' which makes
it possible to loop digital signals from S/PDIF-in to S/PDIF-out.

Signed-off-by: Glen Masgai <mimosius@gmx.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
---
 sound/pci/ymfpci/ymfpci_main.c | 39 ++++++++++++++++++++++++---------------
 1 file changed, 24 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 27fa523639a..2e671ee438c 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -1421,15 +1421,18 @@ static snd_kcontrol_new_t snd_ymfpci_drec_source __devinitdata = {
  *  Mixer controls
  */
 
-#define YMFPCI_SINGLE(xname, xindex, reg) \
+#define YMFPCI_SINGLE(xname, xindex, reg, shift) \
 { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
   .info = snd_ymfpci_info_single, \
   .get = snd_ymfpci_get_single, .put = snd_ymfpci_put_single, \
-  .private_value = reg }
+  .private_value = ((reg) | ((shift) << 16)) }
 
-static int snd_ymfpci_info_single(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
+static int snd_ymfpci_info_single(snd_kcontrol_t *kcontrol,
+				  snd_ctl_elem_info_t *uinfo)
 {
-	switch (kcontrol->private_value) {
+	int reg = kcontrol->private_value & 0xffff;
+
+	switch (reg) {
 	case YDSXGR_SPDIFOUTCTRL: break;
 	case YDSXGR_SPDIFINCTRL: break;
 	default: return -EINVAL;
@@ -1441,30 +1444,35 @@ static int snd_ymfpci_info_single(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t
 	return 0;
 }
 
-static int snd_ymfpci_get_single(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+static int snd_ymfpci_get_single(snd_kcontrol_t *kcontrol,
+				 snd_ctl_elem_value_t *ucontrol)
 {
 	ymfpci_t *chip = snd_kcontrol_chip(kcontrol);
-	int reg = kcontrol->private_value;
-	unsigned int shift = 0, mask = 1;
+	int reg = kcontrol->private_value & 0xffff;
+	unsigned int shift = (kcontrol->private_value >> 16) & 0xff;
+	unsigned int mask = 1;
 	
-	switch (kcontrol->private_value) {
+	switch (reg) {
 	case YDSXGR_SPDIFOUTCTRL: break;
 	case YDSXGR_SPDIFINCTRL: break;
 	default: return -EINVAL;
 	}
-	ucontrol->value.integer.value[0] = (snd_ymfpci_readl(chip, reg) >> shift) & mask;
+	ucontrol->value.integer.value[0] =
+		(snd_ymfpci_readl(chip, reg) >> shift) & mask;
 	return 0;
 }
 
-static int snd_ymfpci_put_single(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
+static int snd_ymfpci_put_single(snd_kcontrol_t *kcontrol,
+				 snd_ctl_elem_value_t *ucontrol)
 {
 	ymfpci_t *chip = snd_kcontrol_chip(kcontrol);
-	int reg = kcontrol->private_value;
-	unsigned int shift = 0, mask = 1;
+	int reg = kcontrol->private_value & 0xffff;
+	unsigned int shift = (kcontrol->private_value >> 16) & 0xff;
+ 	unsigned int mask = 1;
 	int change;
 	unsigned int val, oval;
 	
-	switch (kcontrol->private_value) {
+	switch (reg) {
 	case YDSXGR_SPDIFOUTCTRL: break;
 	case YDSXGR_SPDIFINCTRL: break;
 	default: return -EINVAL;
@@ -1583,8 +1591,9 @@ YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ", PLAYBACK,VOLUME), 0, YDSXGR_ZVOUTVO
 YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("", CAPTURE,VOLUME), 0, YDSXGR_ZVLOOPVOL),
 YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("AC97 ",PLAYBACK,VOLUME), 1, YDSXGR_SPDIFOUTVOL),
 YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,VOLUME), 1, YDSXGR_SPDIFLOOPVOL),
-YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, YDSXGR_SPDIFOUTCTRL),
-YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, YDSXGR_SPDIFINCTRL),
+YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, YDSXGR_SPDIFOUTCTRL, 0),
+YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, YDSXGR_SPDIFINCTRL, 0),
+YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("Loop",NONE,NONE), 0, YDSXGR_SPDIFINCTRL, 4),
 {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "4ch Duplication",
-- 
cgit v1.2.3-18-g5258


From 6632d198c6643294319a4ca3f614539dc1ad37a8 Mon Sep 17 00:00:00 2001
From: Sasha Khapyorsky <sashak@smlink.com>
Date: Thu, 29 Sep 2005 11:48:17 +0200
Subject: [ALSA] dev_class=SNDRV_PCM_CLASS_MODEM for modem PCMs

dev_class=SNDRV_PCM_CLASS_MODEM for all supported softmodem PCMs

Signed-off-by: Sasha Khapyorsky <sashak@smlink.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ali5451/ali5451.c | 5 +++--
 sound/pci/atiixp_modem.c    | 1 +
 sound/pci/hda/hda_intel.c   | 1 +
 sound/pci/intel8x0m.c       | 1 +
 sound/pci/via82xx_modem.c   | 1 +
 5 files changed, 7 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index f35b558c29b..01d971c89e4 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1795,6 +1795,7 @@ struct ali_pcm_description {
 	unsigned int capture_num;
 	snd_pcm_ops_t *playback_ops;
 	snd_pcm_ops_t *capture_ops;
+	unsigned short class;
 };
 
 
@@ -1818,7 +1819,6 @@ static int __devinit snd_ali_pcm(ali_t * codec, int device, struct ali_pcm_descr
 	}
 	pcm->private_data = codec;
 	pcm->private_free = snd_ali_pcm_free;
-	pcm->info_flags = 0;
 	if (desc->playback_ops)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, desc->playback_ops);
 	if (desc->capture_ops)
@@ -1828,6 +1828,7 @@ static int __devinit snd_ali_pcm(ali_t * codec, int device, struct ali_pcm_descr
 					      snd_dma_pci_data(codec->pci), 64*1024, 128*1024);
 
 	pcm->info_flags = 0;
+	pcm->dev_class = desc->class;
 	pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
 	strcpy(pcm->name, desc->name);
 	codec->pcm[0] = pcm;
@@ -1836,7 +1837,7 @@ static int __devinit snd_ali_pcm(ali_t * codec, int device, struct ali_pcm_descr
 
 static struct ali_pcm_description ali_pcms[] = {
 	{ "ALI 5451", ALI_CHANNELS, 1, &snd_ali_playback_ops, &snd_ali_capture_ops },
-	{ "ALI 5451 modem", 1, 1, &snd_ali_modem_playback_ops, &snd_ali_modem_capture_ops }
+	{ "ALI 5451 modem", 1, 1, &snd_ali_modem_playback_ops, &snd_ali_modem_capture_ops, SNDRV_PCM_CLASS_MODEM }
 };
 
 static int __devinit snd_ali_build_pcms(ali_t *codec)
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index c1a239a4dac..c020c53a0cd 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -989,6 +989,7 @@ static int __devinit snd_atiixp_pcm_new(atiixp_t *chip)
 		return err;
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_atiixp_playback_ops);
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_atiixp_capture_ops);
+	pcm->dev_class = SNDRV_PCM_CLASS_MODEM;
 	pcm->private_data = chip;
 	strcpy(pcm->name, "ATI IXP MC97");
 	chip->pcmdevs[ATI_PCMDEV_ANALOG] = pcm;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 2c5d4114771..faf9dd00ba6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1277,6 +1277,7 @@ static int __devinit azx_pcm_create(azx_t *chip)
 			err = create_codec_pcm(chip, codec, &codec->pcm_info[c], pcm_dev);
 			if (err < 0)
 				return err;
+			chip->pcm[pcm_dev]->dev_class = SNDRV_PCM_CLASS_MODEM;
 			pcm_dev++;
 		}
 	}
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 9e2060d56c2..2ac1fec5059 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -746,6 +746,7 @@ static int __devinit snd_intel8x0_pcm1(intel8x0_t *chip, int device, struct ich_
 
 	pcm->private_data = chip;
 	pcm->info_flags = 0;
+	pcm->dev_class = SNDRV_PCM_CLASS_MODEM;
 	if (rec->suffix)
 		sprintf(pcm->name, "%s - %s", chip->card->shortname, rec->suffix);
 	else
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 7eac6f6ac73..034dc1c6262 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -832,6 +832,7 @@ static int __devinit snd_via686_pcm_new(via82xx_t *chip)
 		return err;
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_via686_playback_ops);
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_via686_capture_ops);
+	pcm->dev_class = SNDRV_PCM_CLASS_MODEM;
 	pcm->private_data = chip;
 	strcpy(pcm->name, chip->card->shortname);
 	chip->pcms[0] = pcm;
-- 
cgit v1.2.3-18-g5258


From 2ba71978c04d4dba983b4fc658f82eae164c2bca Mon Sep 17 00:00:00 2001
From: Sasha Khapyorsky <sashak@smlink.com>
Date: Thu, 29 Sep 2005 12:58:24 +0200
Subject: [ALSA] Removing obsolete AC97_SHARED_TYPES

This patch cleans last ac97 audio/modem codec interception in
initialization procedures (ac97_mixer_new()) and removes obsolete
SHARED_TYPE 'locking' which prevents from AMC codecs to function
correctly.

Signed-off-by: Sasha Khapyorsky <sashak@smlink.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/ac97/ac97_codec.c | 38 ++------------------------------------
 sound/pci/atiixp.c          |  1 -
 sound/pci/atiixp_modem.c    |  1 -
 sound/pci/intel8x0.c        |  1 -
 sound/pci/intel8x0m.c       |  1 -
 sound/pci/via82xx.c         |  2 +-
 sound/pci/via82xx_modem.c   |  1 -
 7 files changed, 3 insertions(+), 42 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index 41fc290149e..56549add80a 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -220,12 +220,6 @@ const char *snd_ac97_stereo_enhancements[] =
   /*  31 */ "Reserved 31"
 };
 
-/*
- * Shared AC97 controllers (ICH, ATIIXP...)
- */
-static DECLARE_MUTEX(shared_codec_mutex);
-static ac97_t *shared_codec[AC97_SHARED_TYPES][4];
-
 
 /*
  *  I/O routines
@@ -996,14 +990,8