Merge master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa

* master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa:
  [ALSA] echoaudio - Remove kfree_nocheck()
  [ALSA] echoaudio - Fix Makefile
  [ALSA] Add Intel D965 board support
  [ALSA] Fix/add support of Realtek ALC883 / ALC888 and ALC861 codecs
  [ALSA] Fix a typo in echoaudio/midi.c
  [ALSA] snd-aoa: enable dual-edge in GPIOs
  [ALSA] snd-aoa: support iMac G5 iSight
  [ALSA] snd-aoa: not experimental
  [ALSA] Add echoaudio sound drivers
  [ALSA] ak4xxx-adda - Code clean-up
  [ALSA] Remove CONFIG_EXPERIMENTAL from intel8x0m driver
  [ALSA] Stereo controls for M-Audio Revolution cards
  [ALSA] Fix misuse of __list_add() in seq_ports.c
  [ALSA] hda-codec - Add model entry for Samsung X60 Chane
  [ALSA] make CONFIG_SND_DYNAMIC_MINORS non-experimental
  [ALSA] Fix wrong dependencies of snd-aoa driver
  [ALSA] fix build failure due to snd-aoa
  [ALSA] AD1888 mixer controls for DC mode
  [ALSA] Suppress irq handler mismatch messages in ALSA ISA drivers
  [ALSA] usb-audio support for Turtle Beach Roadie

53 files changed, 11393 insertions, 210 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 87d76a5c73d0..f61af23dd85d 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -472,6 +472,22 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     The power-management is supported.
 
+  Module snd-darla20
+  ------------------
+
+    Module for Echoaudio Darla20
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-darla24
+  ------------------
+
+    Module for Echoaudio Darla24
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
   Module snd-dt019x
   -----------------
 
@@ -499,6 +515,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     The power-management is supported.
 
+  Module snd-echo3g
+  -----------------
+
+    Module for Echoaudio 3G cards (Gina3G/Layla3G)
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
   Module snd-emu10k1
   ------------------
 
@@ -657,6 +681,22 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     
     The power-management is supported.
 
+  Module snd-gina20
+  -----------------
+
+    Module for Echoaudio Gina20
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-gina24
+  -----------------
+
+    Module for Echoaudio Gina24
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
   Module snd-gusclassic
   ---------------------
 
@@ -760,12 +800,18 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
           basic         fixed pin assignment w/o SPDIF
           auto          auto-config reading BIOS (default)
 
-        ALC882/883/885
+        ALC882/885
           3stack-dig    3-jack with SPDIF I/O
           6stck-dig     6-jack digital with SPDIF I/O
           auto          auto-config reading BIOS (default)
 
-        ALC861
+        ALC883/888
+          3stack-dig    3-jack with SPDIF I/O
+          6stack-dig    6-jack digital with SPDIF I/O
+          6stack-dig-demo  6-stack digital for Intel demo board
+          auto          auto-config reading BIOS (default)
+
+        ALC861/660
           3stack        3-jack
           3stack-dig    3-jack with SPDIF I/O
           6stack-dig    6-jack with SPDIF I/O
@@ -937,6 +983,30 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
           driver isn't configured properly or you want to try another
           type for testing.
 
+  Module snd-indigo
+  -----------------
+
+    Module for Echoaudio Indigo
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-indigodj
+  -------------------
+
+    Module for Echoaudio Indigo DJ
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-indigoio
+  -------------------
+
+    Module for Echoaudio Indigo IO
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
   Module snd-intel8x0
   -------------------
 
@@ -1036,6 +1106,22 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     This module supports multiple cards.
 
+  Module snd-layla20
+  ------------------
+
+    Module for Echoaudio Layla20
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
+  Module snd-layla24
+  ------------------
+
+    Module for Echoaudio Layla24
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
   Module snd-maestro3
   -------------------
 
@@ -1056,6 +1142,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
 
     The power-management is supported.
 
+  Module snd-mia
+  ---------------
+
+    Module for Echoaudio Mia
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
   Module snd-miro
   ---------------
 
@@ -1088,6 +1182,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
     When no hotplug fw loader is available, you need to load the
     firmware via mixartloader utility in alsa-tools package.
 
+  Module snd-mona
+  ---------------
+
+    Module for Echoaudio Mona
+
+    This module supports multiple cards.
+    The driver requires the firmware loader support on kernel.
+
   Module snd-mpu401
   -----------------
 
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 446afc3ea27f..758f8bf133c7 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -265,6 +265,7 @@
 
 /* specific - Analog Devices */
 #define AC97_AD_TEST            0x5a    /* test register */
+#define AC97_AD_TEST2           0x5c    /* undocumented test register 2 */
 #define AC97_AD_CODEC_CFG       0x70    /* codec configuration */
 #define AC97_AD_JACK_SPDIF      0x72    /* Jack Sense & S/PDIF */
 #define AC97_AD_SERIAL_CFG      0x74    /* Serial Configuration */
diff --git a/include/sound/ak4xxx-adda.h b/include/sound/ak4xxx-adda.h
index 3bf5911fe827..3d9888492026 100644
--- a/include/sound/ak4xxx-adda.h
+++ b/include/sound/ak4xxx-adda.h
@@ -32,8 +32,8 @@ struct snd_akm4xxx;
 struct snd_ak4xxx_ops {
         void (*lock)(struct snd_akm4xxx *ak, int chip);
         void (*unlock)(struct snd_akm4xxx *ak, int chip);
-        void (*write)(struct snd_akm4xxx *ak, int chip, unsigned char reg, unsigned char val);
-        // unsigned char (*read)(struct snd_akm4xxx *ak, int chip, unsigned char reg);
+        void (*write)(struct snd_akm4xxx *ak, int chip, unsigned char reg,
+                      unsigned char val);
         void (*set_rate_val)(struct snd_akm4xxx *ak, unsigned int rate);
 };
 
@@ -41,29 +41,40 @@ struct snd_ak4xxx_ops {
 
 struct snd_akm4xxx {
         struct snd_card *card;
-        unsigned int num_adcs;                          /* AK4524 or AK4528 ADCs */
-        unsigned int num_dacs;                          /* AK4524 or AK4528 DACs */
-        unsigned char images[AK4XXX_IMAGE_SIZE];        /* saved register image */
-        unsigned char ipga_gain[AK4XXX_MAX_CHIPS][2];   /* saved register image for IPGA (AK4528) */
+        unsigned int num_adcs;                  /* AK4524 or AK4528 ADCs */
+        unsigned int num_dacs;                  /* AK4524 or AK4528 DACs */
+        unsigned char images[AK4XXX_IMAGE_SIZE]; /* saved register image */
+        unsigned char ipga_gain[AK4XXX_MAX_CHIPS][2]; /* saved register image
+                                                       * for IPGA (AK4528)
+                                                       */
         unsigned long private_value[AK4XXX_MAX_CHIPS];  /* helper for driver */
         void *private_data[AK4XXX_MAX_CHIPS];           /* helper for driver */
         /* template should fill the following fields */
-        unsigned int idx_offset;                        /* control index offset */
+        unsigned int idx_offset;                /* control index offset */
         enum {
                 SND_AK4524, SND_AK4528, SND_AK4529,
                 SND_AK4355, SND_AK4358, SND_AK4381
         } type;
+        unsigned int *num_stereo;       /* array of combined counts
+                                         * for the mixer
+                                         */
+        char **channel_names;           /* array of mixer channel names */
         struct snd_ak4xxx_ops ops;
 };
 
-void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg, unsigned char val);
+void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg,
+                       unsigned char val);
 void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state);
 void snd_akm4xxx_init(struct snd_akm4xxx *ak);
 int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak);
 
-#define snd_akm4xxx_get(ak,chip,reg) (ak)->images[(chip) * 16 + (reg)]
-#define snd_akm4xxx_set(ak,chip,reg,val) ((ak)->images[(chip) * 16 + (reg)] = (val))
-#define snd_akm4xxx_get_ipga(ak,chip,reg) (ak)->ipga_gain[chip][(reg)-4]
-#define snd_akm4xxx_set_ipga(ak,chip,reg,val) ((ak)->ipga_gain[chip][(reg)-4] = (val))
+#define snd_akm4xxx_get(ak,chip,reg) \
+        (ak)->images[(chip) * 16 + (reg)]
+#define snd_akm4xxx_set(ak,chip,reg,val) \
+        ((ak)->images[(chip) * 16 + (reg)] = (val))
+#define snd_akm4xxx_get_ipga(ak,chip,reg) \
+        (ak)->ipga_gain[chip][(reg)-4]
+#define snd_akm4xxx_set_ipga(ak,chip,reg,val) \
+        ((ak)->ipga_gain[chip][(reg)-4] = (val))
 
 #endif /* __SOUND_AK4XXX_ADDA_H */
diff --git a/include/sound/initval.h b/include/sound/initval.h
index d29e3d31d149..d45170b9e0b7 100644
--- a/include/sound/initval.h
+++ b/include/sound/initval.h
@@ -62,7 +62,8 @@ static int snd_legacy_find_free_irq(int *irq_table)
 {
         while (*irq_table != -1) {
                 if (!request_irq(*irq_table, snd_legacy_empty_irq_handler,
-                                 SA_INTERRUPT, "ALSA Test IRQ", (void *) irq_table)) {
+                                 SA_INTERRUPT | SA_PROBEIRQ, "ALSA Test IRQ",
+                                 (void *) irq_table)) {
                         free_irq(*irq_table, (void *) irq_table);
                         return *irq_table;
                 }
diff --git a/sound/Makefile b/sound/Makefile
index a682ea30f0c9..1f60797afa8a 100644
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -4,7 +4,8 @@
 obj-$(CONFIG_SOUND) += soundcore.o
 obj-$(CONFIG_SOUND_PRIME) += oss/
 obj-$(CONFIG_DMASOUND) += oss/
-obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ aoa/
+obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/
+obj-$(CONFIG_SND_AOA) += aoa/
 
 ifeq ($(CONFIG_SND),y)
   obj-y += last.o
diff --git a/sound/aoa/Kconfig b/sound/aoa/Kconfig
index a85194fe0b06..2f4334d19ccd 100644
--- a/sound/aoa/Kconfig
+++ b/sound/aoa/Kconfig
@@ -3,7 +3,8 @@ menu "Apple Onboard Audio driver"
 
 config SND_AOA
         tristate "Apple Onboard Audio driver"
-        depends on SOUND && SND_PCM
+        depends on SND
+        select SND_PCM
         ---help---
         This option enables the new driver for the various
         Apple Onboard Audio components.
diff --git a/sound/aoa/core/snd-aoa-gpio-feature.c b/sound/aoa/core/snd-aoa-gpio-feature.c
index 2c6eb7784cc9..bab97547a052 100644
--- a/sound/aoa/core/snd-aoa-gpio-feature.c
+++ b/sound/aoa/core/snd-aoa-gpio-feature.c
@@ -207,6 +207,17 @@ static void ftr_handle_notify(void *data)
         mutex_unlock(&notif->mutex);
 }
 
+static void gpio_enable_dual_edge(int gpio)
+{
+        int v;
+
+        if (gpio == -1)
+                return;
+        v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0);
+        v |= 0x80; /* enable dual edge */
+        pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, gpio, v);
+}
+
 static void ftr_gpio_init(struct gpio_runtime *rt)
 {
         get_gpio("headphone-mute", NULL,
@@ -234,6 +245,10 @@ static void ftr_gpio_init(struct gpio_runtime *rt)
                                       &linein_detect_gpio,
                                       &linein_detect_gpio_activestate);
 
+        gpio_enable_dual_edge(headphone_detect_gpio);
+        gpio_enable_dual_edge(lineout_detect_gpio);
+        gpio_enable_dual_edge(linein_detect_gpio);
+
         get_irq(headphone_detect_node, &headphone_detect_irq);
         get_irq(lineout_detect_node, &lineout_detect_irq);
         get_irq(linein_detect_node, &linein_detect_irq);
diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
index 04a7238e9494..cbc8a3b5cea4 100644
--- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c
+++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
@@ -94,6 +94,7 @@ MODULE_ALIAS("sound-layout-82");
 MODULE_ALIAS("sound-layout-84");
 MODULE_ALIAS("sound-layout-86");
 MODULE_ALIAS("sound-layout-92");
+MODULE_ALIAS("sound-layout-96");
 
 /* onyx with all but microphone connected */
 static struct codec_connection onyx_connections_nomic[] = {
@@ -381,6 +382,13 @@ static struct layout layouts[] = {
                 .connections = toonie_connections,
           },
         },
+        {
+          .layout_id = 96,
+          .codecs[0] = {
+                .name = "onyx",
+                .connections = onyx_connections_noheadphones,
+          },
+        },
         /* unknown, untested, but this comes from Apple */
         { .layout_id = 41,
           .codecs[0] = {
@@ -479,12 +487,6 @@ static struct layout layouts[] = {
                 .connections = onyx_connections_noheadphones,
           },
         },
-        { .layout_id = 96,
-          .codecs[0] = {
-                .name = "onyx",
-                .connections = onyx_connections_noheadphones,
-          },
-        },
         { .layout_id = 98,
           .codecs[0] = {
                 .name = "toonie",
diff --git a/sound/aoa/soundbus/Kconfig b/sound/aoa/soundbus/Kconfig
index d532d27a9f54..7368b7ddfe0d 100644
--- a/sound/aoa/soundbus/Kconfig
+++ b/sound/aoa/soundbus/Kconfig
@@ -1,6 +1,7 @@
 config SND_AOA_SOUNDBUS
         tristate "Apple Soundbus support"
-        depends on SOUND && SND_PCM && EXPERIMENTAL
+        depends on SOUND
+        select SND_PCM
         ---help---
         This option enables the generic driver for the soundbus
         support on Apple machines.
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 4262a1c87731..b2927523d79d 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -122,8 +122,8 @@ config SND_SEQ_RTCTIMER_DEFAULT
           If in doubt, say Y.
 
 config SND_DYNAMIC_MINORS
-        bool "Dynamic device file minor numbers (EXPERIMENTAL)"
-        depends on SND && EXPERIMENTAL
+        bool "Dynamic device file minor numbers"
+        depends on SND
         help
           If you say Y here, the minor numbers of ALSA device files in
           /dev/snd/ are allocated dynamically.  This allows you to have
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
index 334579a9f268..d467b4f0ff2b 100644
--- a/sound/core/seq/seq_ports.c
+++ b/sound/core/seq/seq_ports.c
@@ -322,10 +322,8 @@ int snd_seq_delete_all_ports(struct snd_seq_client *client)
         mutex_lock(&client->ports_mutex);
         write_lock_irqsave(&client->ports_lock, flags);
         if (! list_empty(&client->ports_list_head)) {
-                __list_add(&deleted_list,
-                           client->ports_list_head.prev,
-                           client->ports_list_head.next);
-                INIT_LIST_HEAD(&client->ports_list_head);
+                list_add(&deleted_list, &client->ports_list_head);
+                list_del_init(&client->ports_list_head);
         } else {
                 INIT_LIST_HEAD(&deleted_list);
         }
diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
index 045e32a311e0..dc7cc2001b74 100644
--- a/sound/i2c/other/ak4xxx-adda.c
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -34,7 +34,8 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Takashi Iwai <tiwai@suse.de>");
 MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx  AD/DA converters");
 MODULE_LICENSE("GPL");
 
-void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg, unsigned char val)
+void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg,
+                       unsigned char val)
 {
         ak->ops.lock(ak, chip);
         ak->ops.write(ak, chip, reg, val);
@@ -52,6 +53,67 @@ void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg, unsi
         ak->ops.unlock(ak, chip);
 }
 
+EXPORT_SYMBOL(snd_akm4xxx_write);
+
+/* reset procedure for AK4524 and AK4528 */
+static void ak4524_reset(struct snd_akm4xxx *ak, int state)
+{
+        unsigned int chip;
+        unsigned char reg, maxreg;
+
+        if (ak->type == SND_AK4528)
+                maxreg = 0x06;
+        else
+                maxreg = 0x08;
+        for (chip = 0; chip < ak->num_dacs/2; chip++) {
+                snd_akm4xxx_write(ak, chip, 0x01, state ? 0x00 : 0x03);
+                if (state)
+                        continue;
+                /* DAC volumes */
+                for (reg = 0x04; reg < maxreg; reg++)
+                        snd_akm4xxx_write(ak, chip, reg,
+                                          snd_akm4xxx_get(ak, chip, reg));
+                if (ak->type == SND_AK4528)
+                        continue;
+                /* IPGA */
+                for (reg = 0x04; reg < 0x06; reg++)
+                        snd_akm4xxx_write(ak, chip, reg,
+                                          snd_akm4xxx_get_ipga(ak, chip, reg));
+        }
+}
+
+/* reset procedure for AK4355 and AK4358 */
+static void ak4355_reset(struct snd_akm4xxx *ak, int state)
+{
+        unsigned char reg;
+
+        if (state) {
+                snd_akm4xxx_write(ak, 0, 0x01, 0x02); /* reset and soft-mute */
+                return;
+        }
+        for (reg = 0x00; reg < 0x0b; reg++)
+                if (reg != 0x01)
+                        snd_akm4xxx_write(ak, 0, reg,
+                                          snd_akm4xxx_get(ak, 0, reg));
+        snd_akm4xxx_write(ak, 0, 0x01, 0x01); /* un-reset, unmute */
+}
+
+/* reset procedure for AK4381 */
+static void ak4381_reset(struct snd_akm4xxx *ak, int state)
+{
+        unsigned int chip;
+        unsigned char reg;
+
+        for (chip = 0; chip < ak->num_dacs/2; chip++) {
+                snd_akm4xxx_write(ak, chip, 0x00, state ? 0x0c : 0x0f);
+                if (state)
+                        continue;
+                for (reg = 0x01; reg < 0x05; reg++)
+                        snd_akm4xxx_write(ak, chip, reg,
+                                          snd_akm4xxx_get(ak, chip, reg));
+        }
+}
+
 /*
  * reset the AKM codecs
  * @state: 1 = reset codec, 0 = restore the registers
@@ -60,52 +122,26 @@ void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg, unsi
  */
 void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state)
 {
-        unsigned int chip;
-        unsigned char reg;
-        
         switch (ak->type) {
         case SND_AK4524:
         case SND_AK4528:
-                for (chip = 0; chip < ak->num_dacs/2; chip++) {
-                        snd_akm4xxx_write(ak, chip, 0x01, state ? 0x00 : 0x03);
-                        if (state)
-                                continue;
-                        /* DAC volumes */
-                        for (reg = 0x04; reg < (ak->type == SND_AK4528 ? 0x06 : 0x08); reg++)
-                                snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get(ak, chip, reg));
-                        if (ak->type == SND_AK4528)
-                                continue;
-                        /* IPGA */
-                        for (reg = 0x04; reg < 0x06; reg++)
-                                snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get_ipga(ak, chip, reg));
-                }
+                ak4524_reset(ak, state);
                 break;
         case SND_AK4529:
                 /* FIXME: needed for ak4529? */
                 break;
         case SND_AK4355:
         case SND_AK4358:
-                if (state) {
-                        snd_akm4xxx_write(ak, 0, 0x01, 0x02); /* reset and soft-mute */
-                        return;
-                }
-                for (reg = 0x00; reg < 0x0b; reg++)
-                        if (reg != 0x01)
-                                snd_akm4xxx_write(ak, 0, reg, snd_akm4xxx_get(ak, 0, reg));
-                snd_akm4xxx_write(ak, 0, 0x01, 0x01); /* un-reset, unmute */
+                ak4355_reset(ak, state);
                 break;
         case SND_AK4381:
-                for (chip = 0; chip < ak->num_dacs/2; chip++) {
-                        snd_akm4xxx_write(ak, chip, 0x00, state ? 0x0c : 0x0f);
-                        if (state)
-                                continue;
-                        for (reg = 0x01; reg < 0x05; reg++)
-                                snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get(ak, chip, reg));
-                }
+                ak4381_reset(ak, state);
                 break;
         }
 }
 
+EXPORT_SYMBOL(snd_akm4xxx_reset);
+
 /*
  * initialize all the ak4xxx chips
  */
@@ -153,7 +189,8 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak)
         };
         static unsigned char inits_ak4355[] = {
                 0x01, 0x02, /* 1: reset and soft-mute */
-                0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect, disable DZF, sharp roll-off, RSTN#=0 */
+                0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect,
+                             * disable DZF, sharp roll-off, RSTN#=0 */
                 0x02, 0x0e, /* 2: DA's power up, normal speed, RSTN#=0 */
                 // 0x02, 0x2e, /* quad speed */
                 0x03, 0x01, /* 3: de-emphasis off */
@@ -169,7 +206,8 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak)
         };
         static unsigned char inits_ak4358[] = {
                 0x01, 0x02, /* 1: reset and soft-mute */
-                0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect, disable DZF, sharp roll-off, RSTN#=0 */
+                0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect,
+                             * disable DZF, sharp roll-off, RSTN#=0 */
                 0x02, 0x0e, /* 2: DA's power up, normal speed, RSTN#=0 */
                 // 0x02, 0x2e, /* quad speed */
                 0x03, 0x01, /* 3: de-emphasis off */
@@ -187,7 +225,8 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak)
         };
         static unsigned char inits_ak4381[] = {
                 0x00, 0x0c, /* 0: mode3(i2s), disable auto-clock detect */
-                0x01, 0x02, /* 1: de-emphasis off, normal speed, sharp roll-off, DZF off */
+                0x01, 0x02, /* 1: de-emphasis off, normal speed,
+                             * sharp roll-off, DZF off */
                 // 0x01, 0x12, /* quad speed */
                 0x02, 0x00, /* 2: DZF disabled */
                 0x03, 0x00, /* 3: LATT 0 */
@@ -239,12 +278,15 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak)
         }
 }
 
+EXPORT_SYMBOL(snd_akm4xxx_init);
+
 #define AK_GET_CHIP(val)                (((val) >> 8) & 0xff)
 #define AK_GET_ADDR(val)                ((val) & 0xff)
 #define AK_GET_SHIFT(val)               (((val) >> 16) & 0x7f)
 #define AK_GET_INVERT(val)              (((val) >> 23) & 1)
 #define AK_GET_MASK(val)                (((val) >> 24) & 0xff)
-#define AK_COMPOSE(chip,addr,shift,mask) (((chip) << 8) | (addr) | ((shift) << 16) | ((mask) << 24))
+#define AK_COMPOSE(chip,addr,shift,mask) \
+        (((chip) << 8) | (addr) | ((shift) << 16) | ((mask) << 24))
 #define AK_INVERT                       (1<<23)
 
 static int snd_akm4xxx_volume_info(struct snd_kcontrol *kcontrol,
@@ -292,6 +334,64 @@ static int snd_akm4xxx_volume_put(struct snd_kcontrol *kcontrol,
         return change;
 }
 
+static int snd_akm4xxx_stereo_volume_info(struct snd_kcontrol *kcontrol,
+                                          struct snd_ctl_elem_info *uinfo)
+{
+        unsigned int mask = AK_GET_MASK(kcontrol->private_value);
+
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = 2;
+        uinfo->value.integer.min = 0;
+        uinfo->value.integer.max = mask;
+        return 0;
+}
+
+static int snd_akm4xxx_stereo_volume_get(struct snd_kcontrol *kcontrol,
+                                         struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+        int chip = AK_GET_CHIP(kcontrol->private_value);
+        int addr = AK_GET_ADDR(kcontrol->private_value);
+        int invert = AK_GET_INVERT(kcontrol->private_value);
+        unsigned int mask = AK_GET_MASK(kcontrol->private_value);
+        unsigned char val = snd_akm4xxx_get(ak, chip, addr);
+        
+        ucontrol->value.integer.value[0] = invert ? mask - val : val;
+
+        val = snd_akm4xxx_get(ak, chip, addr+1);
+        ucontrol->value.integer.value[1] = invert ? mask - val : val;
+
+        return 0;
+}
+
+static int snd_akm4xxx_stereo_volume_put(struct snd_kcontrol *kcontrol,
+                                         struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
+        int chip = AK_GET_CHIP(kcontrol->private_value);
+        int addr = AK_GET_ADDR(kcontrol->private_value);
+        int invert = AK_GET_INVERT(kcontrol->private_value);
+        unsigned int mask = AK_GET_MASK(kcontrol->private_value);
+        unsigned char nval = ucontrol->value.integer.value[0] % (mask+1);
+        int change0, change1;
+
+        if (invert)
+                nval = mask - nval;
+        change0 = snd_akm4xxx_get(ak, chip, addr) != nval;
+        if (change0)
+                snd_akm4xxx_write(ak, chip, addr, nval);
+
+        nval = ucontrol->value.integer.value[1] % (mask+1);
+        if (invert)
+                nval = mask - nval;
+        change1 = snd_akm4xxx_get(ak, chip, addr+1) != nval;
+        if (change1)
+                snd_akm4xxx_write(ak, chip, addr+1, nval);
+
+
+        return change0 || change1;
+}
+
 static int snd_akm4xxx_ipga_gain_info(struct snd_kcontrol *kcontrol,
                                       struct snd_ctl_elem_info *uinfo)
 {
@@ -308,7 +408,8 @@ static int snd_akm4xxx_ipga_gain_get(struct snd_kcontrol *kcontrol,
         struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
         int chip = AK_GET_CHIP(kcontrol->private_value);
         int addr = AK_GET_ADDR(kcontrol->private_value);
-        ucontrol->value.integer.value[0] = snd_akm4xxx_get_ipga(ak, chip, addr) & 0x7f;
+        ucontrol->value.integer.value[0] =
+                snd_akm4xxx_get_ipga(ak, chip, addr) & 0x7f;
         return 0;
 }
 
@@ -336,7 +437,8 @@ static int snd_akm4xxx_deemphasis_info(struct snd_kcontrol *kcontrol,
         uinfo->value.enumerated.items = 4;
         if (uinfo->value.enumerated.item >= 4)
                 uinfo->value.enumerated.item = 3;
-        strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+        strcpy(uinfo->value.enumerated.name,
+               texts[uinfo->value.enumerated.item]);
         return 0;
 }
 
@@ -347,7 +449,8 @@ static int snd_akm4xxx_deemphasis_get(struct snd_kcontrol *kcontrol,
         int chip = AK_GET_CHIP(kcontrol->private_value);
         int addr = AK_GET_ADDR(kcontrol->private_value);
         int shift = AK_GET_SHIFT(kcontrol->private_value);
-        ucontrol->value.enumerated.item[0] = (snd_akm4xxx_get(ak, chip, addr) >> shift) & 3;
+        ucontrol->value.enumerated.item[0] =
+                (snd_akm4xxx_get(ak, chip, addr) >> shift) & 3;
         return 0;
 }
 
@@ -361,7 +464,8 @@ static int snd_akm4xxx_deemphasis_put(struct snd_kcontrol *kcontrol,
         unsigned char nval = ucontrol->value.enumerated.item[0] & 3;
         int change;
         
-        nval = (nval << shift) | (snd_akm4xxx_get(ak, chip, addr) & ~(3 << shift));
+        nval = (nval << shift) |
+                (snd_akm4xxx_get(ak, chip, addr) & ~(3 << shift));
         change = snd_akm4xxx_get(ak, chip, addr) != nval;
         if (change)
                 snd_akm4xxx_write(ak, chip, addr, nval);
@@ -377,51 +481,86 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
         unsigned int idx, num_emphs;
         struct snd_kcontrol *ctl;
         int err;
+        int mixer_ch = 0;
+        int num_stereo;
 
         ctl = kmalloc(sizeof(*ctl), GFP_KERNEL);
         if (! ctl)
                 return -ENOMEM;
 
-        for (idx = 0; idx < ak->num_dacs; ++idx) {
+        for (idx = 0; idx < ak->num_dacs; ) {
                 memset(ctl, 0, sizeof(*ctl));
-                strcpy(ctl->id.name, "DAC Volume");
-                ctl->id.index = idx + ak->idx_offset * 2;
+                if (ak->channel_names == NULL) {
+                        strcpy(ctl->id.name, "DAC Volume");
+                        num_stereo = 1;
+                        ctl->id.index = mixer_ch + ak->idx_offset * 2;
+                } else {
+                        strcpy(ctl->id.name, ak->channel_names[mixer_ch]);
+                        num_stereo = ak->num_stereo[mixer_ch];
+                        ctl->id.index = 0;
+                }
                 ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
                 ctl->count = 1;
-                ctl->info = snd_akm4xxx_volume_info;
-                ctl->get = snd_akm4xxx_volume_get;
-                ctl->put = snd_akm4xxx_volume_put;
+                if (num_stereo == 2) {
+                        ctl->info = snd_akm4xxx_stereo_volume_info;
+                        ctl->get = snd_akm4xxx_stereo_volume_get;
+                        ctl->put = snd_akm4xxx_stereo_volume_put;
+                } else {
+                        ctl->info = snd_akm4xxx_volume_info;
+                        ctl->get = snd_akm4xxx_volume_get;
+                        ctl->put = snd_akm4xxx_volume_put;
+                }
                 switch (ak->type) {
                 case SND_AK4524:
-                        ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 6, 0, 127); /* register 6 & 7 */
+                        /* register 6 & 7 */
+                        ctl->private_value =
+                                AK_COMPOSE(idx/2, (idx%2) + 6, 0, 127);
                         break;
                 case SND_AK4528:
-                        ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 4, 0, 127); /* register 4 & 5 */
+                        /* register 4 & 5 */
+                        ctl->private_value =
+                                AK_COMPOSE(idx/2, (idx%2) + 4, 0, 127);
                         break;
                 case SND_AK4529: {
-                        int val = idx < 6 ? idx + 2 : (idx - 6) + 0xb; /* registers 2-7 and b,c */
-                        ctl->private_value = AK_COMPOSE(0, val, 0, 255) | AK_INVERT;
+                        /* registers 2-7 and b,c */
+                        int val = idx < 6 ? idx + 2 : (idx - 6) + 0xb;
+                        ctl->private_value =
+                                AK_COMPOSE(0, val, 0, 255) | AK_INVERT;
                         break;
                 }
                 case SND_AK4355:
-                        ctl->private_value = AK_COMPOSE(0, idx + 4, 0, 255); /* register 4-9, chip #0 only */
+                        /* register 4-9, chip #0 only */
+                        ctl->private_value = AK_COMPOSE(0, idx + 4, 0, 255);
                         break;
                 case SND_AK4358:
                         if (idx >= 6)
-                                ctl->private_value = AK_COMPOSE(0, idx + 5, 0, 255); /* register 4-9, chip #0 only */
+                                /* register 4-9, chip #0 only */
+                                ctl->private_value =
+                                        AK_COMPOSE(0, idx + 5, 0, 255);
                         else
-                                ctl->private_value = AK_COMPOSE(0, idx + 4, 0, 255); /* register 4-9, chip #0 only */
+                                /* register 4-9, chip #0 only */
+                                ctl->private_value =
+                                        AK_COMPOSE(0, idx + 4, 0, 255);
                         break;
                 case SND_AK4381:
-                        ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 3, 0, 255); /* register 3 & 4 */
+                        /* register 3 & 4 */
+                        ctl->private_value =
+                                AK_COMPOSE(idx/2, (idx%2) + 3, 0, 255);
                         break;
                 default:
                         err = -EINVAL;
                         goto __error;
                 }
+
                 ctl->private_data = ak;
-                if ((err = snd_ctl_add(ak->card, snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE))) < 0)
+                err = snd_ctl_add(ak->card,
+                                  snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|
+                                              SNDRV_CTL_ELEM_ACCESS_WRITE));
+                if (err < 0)
                         goto __error;
+
+                idx += num_stereo;
+                mixer_ch++;
         }
         for (idx = 0; idx < ak->num_adcs && ak->type == SND_AK4524; ++idx) {
                 memset(ctl, 0, sizeof(*ctl));
@@ -432,9 +571,14 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
                 ctl->info = snd_akm4xxx_volume_info;
                 ctl->get = snd_akm4xxx_volume_get;
                 ctl->put = snd_akm4xxx_volume_put;
-                ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 4, 0, 127); /* register 4 & 5 */
+                /* register 4 & 5 */
+                ctl->private_value =
+                        AK_COMPOSE(idx/2, (idx%2) + 4, 0, 127);
                 ctl->private_data = ak;
-                if ((err = snd_ctl_add(ak->card, snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE))) < 0)
+                err = snd_ctl_add(ak->card,
+                                  snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|
+                                              SNDRV_CTL_ELEM_ACCESS_WRITE));
+                if (err < 0)
                         goto __error;
 
                 memset(ctl, 0, sizeof(*ctl));
@@ -445,9 +589,13 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
                 ctl->info = snd_akm4xxx_ipga_gain_info;
                 ctl->get = snd_akm4xxx_ipga_gain_get;
                 ctl->put = snd_akm4xxx_ipga_gain_put;
-                ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 4, 0, 0); /* register 4 & 5 */
+                /* register 4 & 5 */
+                ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 4, 0, 0);
                 ctl->private_data = ak;
-                if ((err = snd_ctl_add(ak->card, snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE))) < 0)
+                err = snd_ctl_add(ak->card,
+                                  snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|
+                                              SNDRV_CTL_ELEM_ACCESS_WRITE));
+                if (err < 0)
                         goto __error;
         }
         if (ak->type == SND_AK4355 || ak->type == SND_AK4358)
@@ -466,11 +614,13 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
                 switch (ak->type) {
                 case SND_AK4524:
                 case SND_AK4528:
-                        ctl->private_value = AK_COMPOSE(idx, 3, 0, 0); /* register 3 */
+                        /* register 3 */
+                        ctl->private_value = AK_COMPOSE(idx, 3, 0, 0);
                         break;
                 case SND_AK4529: {
                         int shift = idx == 3 ? 6 : (2 - idx) * 2;
-                        ctl->private_value = AK_COMPOSE(0, 8, shift, 0); /* register 8 with shift */
+                        /* register 8 with shift */
+                        ctl->private_value = AK_COMPOSE(0, 8, shift, 0);
                         break;
                 }
                 case SND_AK4355:
@@ -482,7 +632,10 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
                         break;
                 }
                 ctl->private_data = ak;
-                if ((err = snd_ctl_add(ak->card, snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE))) < 0)
+                err = snd_ctl_add(ak->card,
+                                  snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|
+                                              SNDRV_CTL_ELEM_ACCESS_WRITE));
+                if (err < 0)
                         goto __error;
         }
         err = 0;
@@ -492,6 +645,8 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
         return err;
 }
 
+EXPORT_SYMBOL(snd_akm4xxx_build_controls);
+
 static int __init alsa_akm4xxx_module_init(void)
 {
         return 0;
@@ -503,8 +658,3 @@ static void __exit alsa_akm4xxx_module_exit(void)
         
 module_init(alsa_akm4xxx_module_init)
 module_exit(alsa_akm4xxx_module_exit)
-
-EXPORT_SYMBOL(snd_akm4xxx_write);
-EXPORT_SYMBOL(snd_akm4xxx_reset);
-EXPORT_SYMBOL(snd_akm4xxx_init);
-EXPORT_SYMBOL(snd_akm4xxx_build_controls);
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index d37346b12dc0..23e54cedfd4a 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -233,6 +233,143 @@ config SND_CS5535AUDIO
           To compile this driver as a module, choose M here: the module
           will be called snd-cs5535audio.
 
+config SND_DARLA20
+        tristate "(Echoaudio) Darla20"
+        depends on SND
+        depends on FW_LOADER
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Darla.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-darla20
+
+config SND_GINA20
+        tristate "(Echoaudio) Gina20"
+        depends on SND
+        depends on FW_LOADER
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Gina.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-gina20
+
+config SND_LAYLA20
+        tristate "(Echoaudio) Layla20"
+        depends on SND
+        depends on FW_LOADER
+        select SND_RAWMIDI
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Layla.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-layla20
+
+config SND_DARLA24
+        tristate "(Echoaudio) Darla24"
+        depends on SND
+        depends on FW_LOADER
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Darla24.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-darla24
+
+config SND_GINA24
+        tristate "(Echoaudio) Gina24"
+        depends on SND
+        depends on FW_LOADER
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Gina24.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-gina24
+
+config SND_LAYLA24
+        tristate "(Echoaudio) Layla24"
+        depends on SND
+        depends on FW_LOADER
+        select SND_RAWMIDI
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Layla24.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-layla24
+
+config SND_MONA
+        tristate "(Echoaudio) Mona"
+        depends on SND
+        depends on FW_LOADER
+        select SND_RAWMIDI
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Mona.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-mona
+
+config SND_MIA
+        tristate "(Echoaudio) Mia"
+        depends on SND
+        depends on FW_LOADER
+        select SND_RAWMIDI
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Mia and Mia-midi.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-mia
+
+config SND_ECHO3G
+        tristate "(Echoaudio) 3G cards"
+        depends on SND
+        depends on FW_LOADER
+        select SND_RAWMIDI
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Gina3G and Layla3G.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-echo3g
+
+config SND_INDIGO
+        tristate "(Echoaudio) Indigo"
+        depends on SND
+        depends on FW_LOADER
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Indigo.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-indigo
+
+config SND_INDIGOIO
+        tristate "(Echoaudio) Indigo IO"
+        depends on SND
+        depends on FW_LOADER
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Indigo IO.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-indigoio
+
+config SND_INDIGODJ
+        tristate "(Echoaudio) Indigo DJ"
+        depends on SND
+        depends on FW_LOADER
+        select SND_PCM
+        help
+          Say 'Y' or 'M' to include support for Echoaudio Indigo DJ.
+
+          To compile this driver as a module, choose M here: the module
+          will be called snd-indigodj
+
 config SND_EMU10K1
         tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)"
         depends on SND
@@ -420,8 +557,8 @@ config SND_INTEL8X0
           will be called snd-intel8x0.
 
 config SND_INTEL8X0M
-        tristate "Intel/SiS/nVidia/AMD MC97 Modem (EXPERIMENTAL)"
-        depends on SND && EXPERIMENTAL
+        tristate "Intel/SiS/nVidia/AMD MC97 Modem"
+        depends on SND
         select SND_AC97_CODEC
         help
           Say Y here to include support for the integrated MC97 modem on
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
index cba5105aafea..e06736da9ef1 100644
--- a/sound/pci/Makefile
+++ b/sound/pci/Makefile
@@ -57,6 +57,7 @@ obj-$(CONFIG_SND) += \
         ca0106/ \
         cs46xx/ \
         cs5535audio/ \
+        echoaudio/ \
         emu10k1/ \
         hda/ \
         ice1712/ \
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 7f197c780816..094cfc1f3a19 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1824,6 +1824,8 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = {
                 .get = snd_ac97_ad1888_lohpsel_get,
                 .put = snd_ac97_ad1888_lohpsel_put
         },
+        AC97_SINGLE("V_REFOUT Enable", AC97_AD_MISC, 2, 1, 1),
+        AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2, 12, 1, 1),
         AC97_SINGLE("Spread Front to Surround and Center/LFE", AC97_AD_MISC, 7, 1, 0),
         {
                 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile
new file mode 100644
index 000000000000..7b576aeb3f8d
--- /dev/null
+++ b/sound/pci/echoaudio/Makefile
@@ -0,0 +1,30 @@
+#
+# Makefile for ALSA Echoaudio soundcard drivers
+# Copyright (c) 2003 by Giuliano Pochini <pochini@shiny.it>
+#
+
+snd-darla20-objs := darla20.o
+snd-gina20-objs := gina20.o
+snd-layla20-objs := layla20.o
+snd-darla24-objs := darla24.o
+snd-gina24-objs := gina24.o
+snd-layla24-objs := layla24.o
+snd-mona-objs := mona.o
+snd-mia-objs := mia.o
+snd-echo3g-objs := echo3g.o
+snd-indigo-objs := indigo.o
+snd-indigoio-objs := indigoio.o
+snd-indigodj-objs := indigodj.o
+
+obj-$(CONFIG_SND_DARLA20) += snd-darla20.o
+obj-$(CONFIG_SND_GINA20) += snd-gina20.o
+obj-$(CONFIG_SND_LAYLA20) += snd-layla20.o
+obj-$(CONFIG_SND_DARLA24) += snd-darla24.o
+obj-$(CONFIG_SND_GINA24) += snd-gina24.o
+obj-$(CONFIG_SND_LAYLA24) += snd-layla24.o
+obj-$(CONFIG_SND_MONA) += snd-mona.o
+obj-$(CONFIG_SND_MIA) += snd-mia.o
+obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o
+obj-$(CONFIG_SND_INDIGO) += snd-indigo.o
+obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o
+obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o
diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c
new file mode 100644
index 000000000000..b7108e29a668
--- /dev/null
+++ b/sound/pci/echoaudio/darla20.c
@@ -0,0 +1,99 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHOGALS_FAMILY
+#define ECHOCARD_DARLA20
+#define ECHOCARD_NAME "Darla20"
+#define ECHOCARD_HAS_MONITOR
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 8 */
+#define PX_DIGITAL_OUT  8       /* 0 */
+#define PX_ANALOG_IN    8       /* 2 */
+#define PX_DIGITAL_IN   10      /* 0 */
+#define PX_NUM          10
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 8 */
+#define BX_DIGITAL_OUT  8       /* 0 */
+#define BX_ANALOG_IN    8       /* 2 */
+#define BX_DIGITAL_IN   10      /* 0 */
+#define BX_NUM          10
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_DARLA20_DSP  0
+
+static const struct firmware card_fw[] = {
+        {0, "darla20_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x1801, 0xECC0, 0x0010, 0, 0, 0},      /* DSP 56301 Darla20 rev.0 */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+        .rate_min = 44100,
+        .rate_max = 48000,
+        .channels_min = 1,
+        .channels_max = 2,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+        /* One page (4k) contains 512 instructions. I don't know if the hw
+        supports lists longer than this. In this case periods_max=220 is a
+        safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "darla20_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c
new file mode 100644
index 000000000000..4159e3bc186f
--- /dev/null
+++ b/sound/pci/echoaudio/darla20_dsp.c
@@ -0,0 +1,125 @@
+/***************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Darla20\n"));
+        snd_assert((subdevice_id & 0xfff0) == DARLA20, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_DARLA20_DSP];
+        chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
+        chip->clock_state = GD_CLOCK_UNDEF;
+        /* Since this card has no ASIC, mark it as loaded so everything
+           works OK */
+        chip->asic_loaded = TRUE;
+        chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+/* The Darla20 has no external clock sources */
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The Darla20 has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+        return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u8 clock_state, spdif_status;
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        switch (rate) {
+        case 44100:
+                clock_state = GD_CLOCK_44;
+                spdif_status = GD_SPDIF_STATUS_44;
+                break;
+        case 48000:
+                clock_state = GD_CLOCK_48;
+                spdif_status = GD_SPDIF_STATUS_48;
+                break;
+        default:
+                clock_state = GD_CLOCK_NOCHANGE;
+                spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+                break;
+        }
+
+        if (chip->clock_state == clock_state)
+                clock_state = GD_CLOCK_NOCHANGE;
+        if (spdif_status == chip->spdif_status)
+                spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+
+        chip->comm_page->sample_rate = cpu_to_le32(rate);
+        chip->comm_page->gd_clock_state = clock_state;
+        chip->comm_page->gd_spdif_status = spdif_status;
+        chip->comm_page->gd_resampler_state = 3;        /* magic number - should always be 3 */
+
+        /* Save the new audio state if it changed */
+        if (clock_state != GD_CLOCK_NOCHANGE)
+                chip->clock_state = clock_state;
+        if (spdif_status != GD_SPDIF_STATUS_NOCHANGE)
+                chip->spdif_status = spdif_status;
+        chip->sample_rate = rate;
+
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
+}
diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c
new file mode 100644
index 000000000000..e59a982ee361
--- /dev/null
+++ b/sound/pci/echoaudio/darla24.c
@@ -0,0 +1,106 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHOGALS_FAMILY
+#define ECHOCARD_DARLA24
+#define ECHOCARD_NAME "Darla24"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 8 */
+#define PX_DIGITAL_OUT  8       /* 0 */
+#define PX_ANALOG_IN    8       /* 2 */
+#define PX_DIGITAL_IN   10      /* 0 */
+#define PX_NUM          10
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 8 */
+#define BX_DIGITAL_OUT  8       /* 0 */
+#define BX_ANALOG_IN    8       /* 2 */
+#define BX_DIGITAL_IN   10      /* 0 */
+#define BX_NUM          10
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_DARLA24_DSP  0
+
+static const struct firmware card_fw[] = {
+        {0, "darla24_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x1801, 0xECC0, 0x0040, 0, 0, 0},      /* DSP 56301 Darla24 rev.0 */
+        {0x1057, 0x1801, 0xECC0, 0x0041, 0, 0, 0},      /* DSP 56301 Darla24 rev.1 */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates =        SNDRV_PCM_RATE_8000_48000 |
+                        SNDRV_PCM_RATE_88200 |
+                        SNDRV_PCM_RATE_96000,
+        .rate_min = 8000,
+        .rate_max = 96000,
+        .channels_min = 1,
+        .channels_max = 8,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+        /* One page (4k) contains 512 instructions. I don't know if the hw
+        supports lists longer than this. In this case periods_max=220 is a
+        safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "darla24_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c
new file mode 100644
index 000000000000..79938eed7e9c
--- /dev/null
+++ b/sound/pci/echoaudio/darla24_dsp.c
@@ -0,0 +1,156 @@
+/***************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Darla24\n"));
+        snd_assert((subdevice_id & 0xfff0) == DARLA24, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_DARLA24_DSP];
+        /* Since this card has no ASIC, mark it as loaded so everything
+           works OK */
+        chip->asic_loaded = TRUE;
+        chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
+                ECHO_CLOCK_BIT_ESYNC;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        u32 clocks_from_dsp, clock_bits;
+
+        /* Map the DSP clock detect bits to the generic driver clock
+           detect bits */
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+        if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_ESYNC)
+                clock_bits |= ECHO_CLOCK_BIT_ESYNC;
+
+        return clock_bits;
+}
+
+
+
+/* The Darla24 has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+        return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u8 clock;
+
+        switch (rate) {
+        case 96000:
+                clock = GD24_96000;
+                break;
+        case 88200:
+                clock = GD24_88200;
+                break;
+        case 48000:
+                clock = GD24_48000;
+                break;
+        case 44100:
+                clock = GD24_44100;
+                break;
+        case 32000:
+                clock = GD24_32000;
+                break;
+        case 22050:
+                clock = GD24_22050;
+                break;
+        case 16000:
+                clock = GD24_16000;
+                break;
+        case 11025:
+                clock = GD24_11025;
+                break;
+        case 8000:
+                clock = GD24_8000;
+                break;
+        default:
+                DE_ACT(("set_sample_rate: Error, invalid sample rate %d\n",
+                        rate));
+                return -EINVAL;
+        }
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+        chip->sample_rate = rate;
+
+        /* Override the sample rate if this card is set to Echo sync. */
+        if (chip->input_clock == ECHO_CLOCK_ESYNC)
+                clock = GD24_EXT_SYNC;
+
+        chip->comm_page->sample_rate = cpu_to_le32(rate);       /* ignored by the DSP ? */
+        chip->comm_page->gd_clock_state = clock;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+        snd_assert(clock == ECHO_CLOCK_INTERNAL ||
+                   clock == ECHO_CLOCK_ESYNC, return -EINVAL);
+        chip->input_clock = clock;
+        return set_sample_rate(chip, chip->sample_rate);
+}
+
diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c
new file mode 100644
index 000000000000..12099fe1547d
--- /dev/null
+++ b/sound/pci/echoaudio/echo3g.c
@@ -0,0 +1,118 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO3G_FAMILY
+#define ECHOCARD_ECHO3G
+#define ECHOCARD_NAME "Echo3G"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+#define ECHOCARD_HAS_ADAT       6
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+#define ECHOCARD_HAS_MIDI
+#define ECHOCARD_HAS_PHANTOM_POWER
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0
+#define PX_DIGITAL_OUT  chip->px_digital_out
+#define PX_ANALOG_IN    chip->px_analog_in
+#define PX_DIGITAL_IN   chip->px_digital_in
+#define PX_NUM          chip->px_num
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0
+#define BX_DIGITAL_OUT  chip->bx_digital_out
+#define BX_ANALOG_IN    chip->bx_analog_in
+#define BX_DIGITAL_IN   chip->bx_digital_in
+#define BX_NUM          chip->bx_num
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_361_LOADER   0
+#define FW_ECHO3G_DSP   1
+#define FW_3G_ASIC      2
+
+static const struct firmware card_fw[] = {
+        {0, "loader_dsp.fw"},
+        {0, "echo3g_dsp.fw"},
+        {0, "3g_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x3410, 0xECC0, 0x0100, 0, 0, 0},      /* Echo 3G */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates =        SNDRV_PCM_RATE_32000 |
+                        SNDRV_PCM_RATE_44100 |
+                        SNDRV_PCM_RATE_48000 |
+                        SNDRV_PCM_RATE_88200 |
+                        SNDRV_PCM_RATE_96000 |
+                        SNDRV_PCM_RATE_CONTINUOUS,
+        .rate_min = 32000,
+        .rate_max = 100000,
+        .channels_min = 1,
+        .channels_max = 8,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+};
+
+#include "echo3g_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio_3g.c"
+#include "echoaudio.c"
+#include "midi.c"
diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c
new file mode 100644
index 000000000000..d26a1d1f3ed1
--- /dev/null
+++ b/sound/pci/echoaudio/echo3g_dsp.c
@@ -0,0 +1,131 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+static int load_asic(struct echoaudio *chip);
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode);
+static int set_digital_mode(struct echoaudio *chip, u8 mode);
+static int check_asic_status(struct echoaudio *chip);
+static int set_sample_rate(struct echoaudio *chip, u32 rate);
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int set_phantom_power(struct echoaudio *chip, char on);
+static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq,
+                             char force);
+
+#include <linux/irq.h>
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        local_irq_enable();
+        DE_INIT(("init_hw() - Echo3G\n"));
+        snd_assert((subdevice_id & 0xfff0) == ECHO3G, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->comm_page->e3g_frq_register =
+                __constant_cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2);
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->has_midi = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_ECHO3G_DSP];
+
+        /* Load the DSP code and the ASIC on the PCI card and get
+        what type of external box is attached */
+        err = load_firmware(chip);
+
+        if (err < 0) {
+                return err;
+        } else if (err == E3G_GINA3G_BOX_TYPE) {
+                chip->input_clock_types =       ECHO_CLOCK_BIT_INTERNAL |
+                                                ECHO_CLOCK_BIT_SPDIF |
+                                                ECHO_CLOCK_BIT_ADAT;
+                chip->card_name = "Gina3G";
+                chip->px_digital_out = chip->bx_digital_out = 6;
+                chip->px_analog_in = chip->bx_analog_in = 14;
+                chip->px_digital_in = chip->bx_digital_in = 16;
+                chip->px_num = chip->bx_num = 24;
+                chip->has_phantom_power = TRUE;
+                chip->hasnt_input_nominal_level = TRUE;
+        } else if (err == E3G_LAYLA3G_BOX_TYPE) {
+                chip->input_clock_types =       ECHO_CLOCK_BIT_INTERNAL |
+                                                ECHO_CLOCK_BIT_SPDIF |
+                                                ECHO_CLOCK_BIT_ADAT |
+                                                ECHO_CLOCK_BIT_WORD;
+                chip->card_name = "Layla3G";
+                chip->px_digital_out = chip->bx_digital_out = 8;
+                chip->px_analog_in = chip->bx_analog_in = 16;
+                chip->px_digital_in = chip->bx_digital_in = 24;
+                chip->px_num = chip->bx_num = 32;
+        } else {
+                return -ENODEV;
+        }
+
+        chip->digital_modes =   ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+                                ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+                                ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
+        chip->digital_mode =    DIGITAL_MODE_SPDIF_RCA;
+        chip->professional_spdif = FALSE;
+        chip->non_audio_spdif = FALSE;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+        err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
+        snd_assert(err >= 0, return err);
+        err = set_phantom_power(chip, 0);
+        snd_assert(err >= 0, return err);
+        err = set_professional_spdif(chip, TRUE);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static int set_phantom_power(struct echoaudio *chip, char on)
+{
+        u32 control_reg = le32_to_cpu(chip->comm_page->control_register);
+
+        if (on)
+                control_reg |= E3G_PHANTOM_POWER;
+        else
+                control_reg &= ~E3G_PHANTOM_POWER;
+
+        chip->phantom_power = on;
+        return write_control_reg(chip, control_reg,
+                                 le32_to_cpu(chip->comm_page->e3g_frq_register),
+                                 0);
+}
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
new file mode 100644
index 000000000000..43b408ada1da
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -0,0 +1,2196 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+MODULE_AUTHOR("Giuliano Pochini <pochini@shiny.it>");
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Echoaudio " ECHOCARD_NAME " soundcards driver");
+MODULE_SUPPORTED_DEVICE("{{Echoaudio," ECHOCARD_NAME "}}");
+MODULE_DEVICE_TABLE(pci, snd_echo_ids);
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for " ECHOCARD_NAME " soundcard.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for " ECHOCARD_NAME " soundcard.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable " ECHOCARD_NAME " soundcard.");
+
+static unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999};
+
+static int get_firmware(const struct firmware **fw_entry,
+                        const struct firmware *frm, struct echoaudio *chip)
+{
+        int err;
+        char name[30];
+        DE_ACT(("firmware requested: %s\n", frm->data));
+        snprintf(name, sizeof(name), "ea/%s", frm->data);
+        if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0)
+                snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err);
+        return err;
+}
+
+static void free_firmware(const struct firmware *fw_entry)
+{
+        release_firmware(fw_entry);
+        DE_ACT(("firmware released\n"));
+}
+
+
+
+/******************************************************************************
+        PCM interface
+******************************************************************************/
+
+static void audiopipe_free(struct snd_pcm_runtime *runtime)
+{
+        struct audiopipe *pipe = runtime->private_data;
+
+        if (pipe->sgpage.area)
+                snd_dma_free_pages(&pipe->sgpage);
+        kfree(pipe);
+}
+
+
+
+static int hw_rule_capture_format_by_channels(struct snd_pcm_hw_params *params,
+                                              struct snd_pcm_hw_rule *rule)
+{
+        struct snd_interval *c = hw_param_interval(params,
+                                                   SNDRV_PCM_HW_PARAM_CHANNELS);
+        struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+        struct snd_mask fmt;
+
+        snd_mask_any(&fmt);
+
+#ifndef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+        /* >=2 channels cannot be S32_BE */
+        if (c->min == 2) {
+                fmt.bits[0] &= ~SNDRV_PCM_FMTBIT_S32_BE;
+                return snd_mask_refine(f, &fmt);
+        }
+#endif
+        /* > 2 channels cannot be U8 and S32_BE */
+        if (c->min > 2) {
+                fmt.bits[0] &= ~(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_BE);
+                return snd_mask_refine(f, &fmt);
+        }
+        /* Mono is ok with any format */
+        return 0;
+}
+
+
+
+static int hw_rule_capture_channels_by_format(struct snd_pcm_hw_params *params,
+                                              struct snd_pcm_hw_rule *rule)
+{
+        struct snd_interval *c = hw_param_interval(params,
+                                                   SNDRV_PCM_HW_PARAM_CHANNELS);
+        struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+        struct snd_interval ch;
+
+        snd_interval_any(&ch);
+
+        /* S32_BE is mono (and stereo) only */
+        if (f->bits[0] == SNDRV_PCM_FMTBIT_S32_BE) {
+                ch.min = 1;
+#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+                ch.max = 2;
+#else
+                ch.max = 1;
+#endif
+                ch.integer = 1;
+                return snd_interval_refine(c, &ch);
+        }
+        /* U8 can be only mono or stereo */
+        if (f->bits[0] == SNDRV_PCM_FMTBIT_U8) {
+                ch.min = 1;
+                ch.max = 2;
+                ch.integer = 1;
+                return snd_interval_refine(c, &ch);
+        }
+        /* S16_LE, S24_3LE and S32_LE support any number of channels. */
+        return 0;
+}
+
+
+
+static int hw_rule_playback_format_by_channels(struct snd_pcm_hw_params *params,
+                                               struct snd_pcm_hw_rule *rule)
+{
+        struct snd_interval *c = hw_param_interval(params,
+                                                   SNDRV_PCM_HW_PARAM_CHANNELS);
+        struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+        struct snd_mask fmt;
+        u64 fmask;
+        snd_mask_any(&fmt);
+
+        fmask = fmt.bits[0] + ((u64)fmt.bits[1] << 32);
+
+        /* >2 channels must be S16_LE, S24_3LE or S32_LE */
+        if (c->min > 2) {
+                fmask &= SNDRV_PCM_FMTBIT_S16_LE |
+                         SNDRV_PCM_FMTBIT_S24_3LE |
+                         SNDRV_PCM_FMTBIT_S32_LE;
+        /* 1 channel must be S32_BE or S32_LE */
+        } else if (c->max == 1)
+                fmask &= SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE;
+#ifndef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+        /* 2 channels cannot be S32_BE */
+        else if (c->min == 2 && c->max == 2)
+                fmask &= ~SNDRV_PCM_FMTBIT_S32_BE;
+#endif
+        else
+                return 0;
+
+        fmt.bits[0] &= (u32)fmask;
+        fmt.bits[1] &= (u32)(fmask >> 32);
+        return snd_mask_refine(f, &fmt);
+}
+
+
+
+static int hw_rule_playback_channels_by_format(struct snd_pcm_hw_params *params,
+                                               struct snd_pcm_hw_rule *rule)
+{
+        struct snd_interval *c = hw_param_interval(params,
+                                                   SNDRV_PCM_HW_PARAM_CHANNELS);
+        struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+        struct snd_interval ch;
+        u64 fmask;
+
+        snd_interval_any(&ch);
+        ch.integer = 1;
+        fmask = f->bits[0] + ((u64)f->bits[1] << 32);
+
+        /* S32_BE is mono (and stereo) only */
+        if (fmask == SNDRV_PCM_FMTBIT_S32_BE) {
+                ch.min = 1;
+#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+                ch.max = 2;
+#else
+                ch.max = 1;
+#endif
+        /* U8 is stereo only */
+        } else if (fmask == SNDRV_PCM_FMTBIT_U8)
+                ch.min = ch.max = 2;
+        /* S16_LE and S24_3LE must be at least stereo */
+        else if (!(fmask & ~(SNDRV_PCM_FMTBIT_S16_LE |
+                               SNDRV_PCM_FMTBIT_S24_3LE)))
+                ch.min = 2;
+        else
+                return 0;
+
+        return snd_interval_refine(c, &ch);
+}
+
+
+
+/* Since the sample rate is a global setting, do allow the user to change the
+sample rate only if there is only one pcm device open. */
+static int hw_rule_sample_rate(struct snd_pcm_hw_params *params,
+                               struct snd_pcm_hw_rule *rule)
+{
+        struct snd_interval *rate = hw_param_interval(params,
+                                                      SNDRV_PCM_HW_PARAM_RATE);
+        struct echoaudio *chip = rule->private;
+        struct snd_interval fixed;
+
+        if (!chip->can_set_rate) {
+                snd_interval_any(&fixed);
+                fixed.min = fixed.max = chip->sample_rate;
+                return snd_interval_refine(rate, &fixed);
+        }
+        return 0;
+}
+
+
+static int pcm_open(struct snd_pcm_substream *substream,
+                    signed char max_channels)
+{
+        struct echoaudio *chip;
+        struct snd_pcm_runtime *runtime;
+        struct audiopipe *pipe;
+        int err, i;
+
+        if (max_channels <= 0)
+                return -EAGAIN;
+
+        chip = snd_pcm_substream_chip(substream);
+        runtime = substream->runtime;
+
+        if (!(pipe = kmalloc(sizeof(struct audiopipe), GFP_KERNEL)))
+                return -ENOMEM;
+        memset(pipe, 0, sizeof(struct audiopipe));
+        pipe->index = -1;               /* Not configured yet */
+
+        /* Set up hw capabilities and contraints */
+        memcpy(&pipe->hw, &pcm_hardware_skel, sizeof(struct snd_pcm_hardware));
+        DE_HWP(("max_channels=%d\n", max_channels));
+        pipe->constr.list = channels_list;
+        pipe->constr.mask = 0;
+        for (i = 0; channels_list[i] <= max_channels; i++);
+        pipe->constr.count = i;
+        if (pipe->hw.channels_max > max_channels)
+                pipe->hw.channels_max = max_channels;
+        if (chip->digital_mode == DIGITAL_MODE_ADAT) {
+                pipe->hw.rate_max = 48000;
+                pipe->hw.rates &= SNDRV_PCM_RATE_8000_48000;
+        }
+
+        runtime->hw = pipe->hw;
+        runtime->private_data = pipe;
+        runtime->private_free = audiopipe_free;
+        snd_pcm_set_sync(substream);
+
+        /* Only mono and any even number of channels are allowed */
+        if ((err = snd_pcm_hw_constraint_list(runtime, 0,
+                                              SNDRV_PCM_HW_PARAM_CHANNELS,
+                                              &pipe->constr)) < 0)
+                return err;
+
+        /* All periods should have the same size */
+        if ((err = snd_pcm_hw_constraint_integer(runtime,
+                                                 SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
+                return err;
+
+        /* The hw accesses memory in chunks 32 frames long and they should be
+        32-bytes-aligned. It's not a requirement, but it seems that IRQs are
+        generated with a resolution of 32 frames. Thus we need the following */
+        if ((err = snd_pcm_hw_constraint_step(runtime, 0,
+                                              SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+                                              32)) < 0)
+                return err;
+        if ((err = snd_pcm_hw_constraint_step(runtime, 0,
+                                              SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+                                              32)) < 0)
+                return err;
+
+        if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+                                       SNDRV_PCM_HW_PARAM_RATE,
+                                        hw_rule_sample_rate, chip,
+                                       SNDRV_PCM_HW_PARAM_RATE, -1)) < 0)
+                return err;
+
+        /* Finally allocate a page for the scatter-gather list */
+        if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
+                                       snd_dma_pci_data(chip->pci),
+                                       PAGE_SIZE, &pipe->sgpage)) < 0) {
+                DE_HWP(("s-g list allocation failed\n"));
+                return err;
+        }
+
+        return 0;
+}
+
+
+
+static int pcm_analog_in_open(struct snd_pcm_substream *substream)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+        int err;
+
+        DE_ACT(("pcm_analog_in_open\n"));
+        if ((err = pcm_open(substream, num_analog_busses_in(chip) -
+                            substream->number)) < 0)
+                return err;
+        if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+                                       SNDRV_PCM_HW_PARAM_CHANNELS,
+                                       hw_rule_capture_channels_by_format, NULL,
+                                       SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0)
+                return err;
+        if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+                                       SNDRV_PCM_HW_PARAM_FORMAT,
+                                       hw_rule_capture_format_by_channels, NULL,
+                                       SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
+                return err;
+        atomic_inc(&chip->opencount);
+        if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
+                chip->can_set_rate=0;
+        DE_HWP(("pcm_analog_in_open  cs=%d  oc=%d  r=%d\n",
+                chip->can_set_rate, atomic_read(&chip->opencount),
+                chip->sample_rate));
+        return 0;
+}
+
+
+
+static int pcm_analog_out_open(struct snd_pcm_substream *substream)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+        int max_channels, err;
+
+#ifdef ECHOCARD_HAS_VMIXER
+        max_channels = num_pipes_out(chip);
+#else
+        max_channels = num_analog_busses_out(chip);
+#endif
+        DE_ACT(("pcm_analog_out_open\n"));
+        if ((err = pcm_open(substream, max_channels - substream->number)) < 0)
+                return err;
+        if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+                                       SNDRV_PCM_HW_PARAM_CHANNELS,
+                                       hw_rule_playback_channels_by_format,
+                                       NULL,
+                                       SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0)
+                return err;
+        if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+                                       SNDRV_PCM_HW_PARAM_FORMAT,
+                                       hw_rule_playback_format_by_channels,
+                                       NULL,
+                                       SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
+                return err;
+        atomic_inc(&chip->opencount);
+        if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
+                chip->can_set_rate=0;
+        DE_HWP(("pcm_analog_out_open  cs=%d  oc=%d  r=%d\n",
+                chip->can_set_rate, atomic_read(&chip->opencount),
+                chip->sample_rate));
+        return 0;
+}
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+
+static int pcm_digital_in_open(struct snd_pcm_substream *substream)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+        int err, max_channels;
+
+        DE_ACT(("pcm_digital_in_open\n"));
+        max_channels = num_digital_busses_in(chip) - substream->number;
+        down(&chip->mode_mutex);
+        if (chip->digital_mode == DIGITAL_MODE_ADAT)
+                err = pcm_open(substream, max_channels);
+        else    /* If the card has ADAT, subtract the 6 channels
+                 * that S/PDIF doesn't have
+                 */
+                err = pcm_open(substream, max_channels - ECHOCARD_HAS_ADAT);
+
+        if (err < 0)
+                goto din_exit;
+
+        if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+                                       SNDRV_PCM_HW_PARAM_CHANNELS,
+                                       hw_rule_capture_channels_by_format, NULL,
+                                       SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0)
+                goto din_exit;
+        if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+                                       SNDRV_PCM_HW_PARAM_FORMAT,
+                                       hw_rule_capture_format_by_channels, NULL,
+                                       SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
+                goto din_exit;
+
+        atomic_inc(&chip->opencount);
+        if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
+                chip->can_set_rate=0;
+
+din_exit:
+        up(&chip->mode_mutex);
+        return err;
+}
+
+
+
+#ifndef ECHOCARD_HAS_VMIXER     /* See the note in snd_echo_new_pcm() */
+
+static int pcm_digital_out_open(struct snd_pcm_substream *substream)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+        int err, max_channels;
+
+        DE_ACT(("pcm_digital_out_open\n"));
+        max_channels = num_digital_busses_out(chip) - substream->number;
+        down(&chip->mode_mutex);
+        if (chip->digital_mode == DIGITAL_MODE_ADAT)
+                err = pcm_open(substream, max_channels);
+        else    /* If the card has ADAT, subtract the 6 channels
+                 * that S/PDIF doesn't have
+                 */
+                err = pcm_open(substream, max_channels - ECHOCARD_HAS_ADAT);
+
+        if (err < 0)
+                goto dout_exit;
+
+        if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+                                       SNDRV_PCM_HW_PARAM_CHANNELS,
+                                       hw_rule_playback_channels_by_format,
+                                       NULL, SNDRV_PCM_HW_PARAM_FORMAT,
+                                       -1)) < 0)
+                goto dout_exit;
+        if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
+                                       SNDRV_PCM_HW_PARAM_FORMAT,
+                                       hw_rule_playback_format_by_channels,
+                                       NULL, SNDRV_PCM_HW_PARAM_CHANNELS,
+                                       -1)) < 0)
+                goto dout_exit;
+        atomic_inc(&chip->opencount);
+        if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
+                chip->can_set_rate=0;
+dout_exit:
+        up(&chip->mode_mutex);
+        return err;
+}
+
+#endif /* !ECHOCARD_HAS_VMIXER */
+
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+        int oc;
+
+        /* Nothing to do here. Audio is already off and pipe will be
+         * freed by its callback
+         */
+        DE_ACT(("pcm_close\n"));
+
+        atomic_dec(&chip->opencount);
+        oc = atomic_read(&chip->opencount);
+        DE_ACT(("pcm_close  oc=%d  cs=%d  rs=%d\n", oc,
+                chip->can_set_rate, chip->rate_set));
+        if (oc < 2)
+                chip->can_set_rate = 1;
+        if (oc == 0)
+                chip->rate_set = 0;
+        DE_ACT(("pcm_close2 oc=%d  cs=%d  rs=%d\n", oc,
+                chip->can_set_rate,chip->rate_set));
+
+        return 0;
+}
+
+
+
+/* Channel allocation and scatter-gather list setup */
+static int init_engine(struct snd_pcm_substream *substream,
+                       struct snd_pcm_hw_params *hw_params,
+                       int pipe_index, int interleave)
+{
+        struct echoaudio *chip;
+        int err, per, rest, page, edge, offs;
+        struct snd_sg_buf *sgbuf;
+        struct audiopipe *pipe;
+
+        chip = snd_pcm_substream_chip(substream);
+        pipe = (struct audiopipe *) substream->runtime->private_data;
+
+        /* Sets up che hardware. If it's already initialized, reset and
+         * redo with the new parameters
+         */
+        spin_lock_irq(&chip->lock);
+        if (pipe->index >= 0) {
+                DE_HWP(("hwp_ie free(%d)\n", pipe->index));
+                err = free_pipes(chip, pipe);
+                snd_assert(!err);
+                chip->substream[pipe->index] = NULL;
+        }
+
+        err = allocate_pipes(chip, pipe, pipe_index, interleave);
+        if (err < 0) {
+                spin_unlock_irq(&chip->lock);
+                DE_ACT((KERN_NOTICE "allocate_pipes(%d) err=%d\n",
+                        pipe_index, err));
+                return err;
+        }
+        spin_unlock_irq(&chip->lock);
+        DE_ACT((KERN_NOTICE "allocate_pipes()=%d\n", pipe_index));
+
+        DE_HWP(("pcm_hw_params (bufsize=%dB periods=%d persize=%dB)\n",
+                params_buffer_bytes(hw_params), params_periods(hw_params),
+                params_period_bytes(hw_params)));
+        err = snd_pcm_lib_malloc_pages(substream,
+                                       params_buffer_bytes(hw_params));
+        if (err < 0) {
+                snd_printk(KERN_ERR "malloc_pages err=%d\n", err);
+                spin_lock_irq(&chip->lock);
+                free_pipes(chip, pipe);
+                spin_unlock_irq(&chip->lock);
+                pipe->index = -1;
+                return err;
+        }
+
+        sgbuf = snd_pcm_substream_sgbuf(substream);
+
+        DE_HWP(("pcm_hw_params table size=%d pages=%d\n",
+                sgbuf->size, sgbuf->pages));
+        sglist_init(chip, pipe);
+        edge = PAGE_SIZE;
+        for (offs = page = per = 0; offs < params_buffer_bytes(hw_params);
+             per++) {
+                rest = params_period_bytes(hw_params);
+                if (offs + rest > params_buffer_bytes(hw_params))
+                        rest = params_buffer_bytes(hw_params) - offs;
+                while (rest) {
+                        if (rest <= edge - offs) {
+                                sglist_add_mapping(chip, pipe,
+                                                   snd_sgbuf_get_addr(sgbuf, offs),
+                                                   rest);
+                                sglist_add_irq(chip, pipe);
+                                offs += rest;
+                                rest = 0;
+                        } else {
+                                sglist_add_mapping(chip, pipe,
+                                                   snd_sgbuf_get_addr(sgbuf, offs),
+                                                   edge - offs);
+                                rest -= edge - offs;
+                                offs = edge;
+                        }
+                        if (offs == edge) {
+                                edge += PAGE_SIZE;
+                                page++;
+                        }
+                }
+        }
+
+        /* Close the ring buffer */
+        sglist_wrap(chip, pipe);
+
+        /* This stuff is used by the irq handler, so it must be
+         * initialized before chip->substream
+         */
+        chip->last_period[pipe_index] = 0;
+        pipe->last_counter = 0;
+        pipe->position = 0;
+        smp_wmb();
+        chip->substream[pipe_index] = substream;
+        chip->rate_set = 1;
+        spin_lock_irq(&chip->lock);
+        set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den);
+        spin_unlock_irq(&chip->lock);
+        DE_HWP(("pcm_hw_params ok\n"));
+        return 0;
+}
+
+
+
+static int pcm_analog_in_hw_params(struct snd_pcm_substream *substream,
+                                   struct snd_pcm_hw_params *hw_params)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+
+        return init_engine(substream, hw_params, px_analog_in(chip) +
+                        substream->number, params_channels(hw_params));
+}
+
+
+
+static int pcm_analog_out_hw_params(struct snd_pcm_substream *substream,
+                                    struct snd_pcm_hw_params *hw_params)
+{
+        return init_engine(substream, hw_params, substream->number,
+                           params_channels(hw_params));
+}
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+
+static int pcm_digital_in_hw_params(struct snd_pcm_substream *substream,
+                                    struct snd_pcm_hw_params *hw_params)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+
+        return init_engine(substream, hw_params, px_digital_in(chip) +
+                        substream->number, params_channels(hw_params));
+}
+
+
+
+#ifndef ECHOCARD_HAS_VMIXER     /* See the note in snd_echo_new_pcm() */
+static int pcm_digital_out_hw_params(struct snd_pcm_substream *substream,
+                                     struct snd_pcm_hw_params *hw_params)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+
+        return init_engine(substream, hw_params, px_digital_out(chip) +
+                        substream->number, params_channels(hw_params));
+}
+#endif /* !ECHOCARD_HAS_VMIXER */
+
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+
+
+static int pcm_hw_free(struct snd_pcm_substream *substream)
+{
+        struct echoaudio *chip;
+        struct audiopipe *pipe;
+
+        chip = snd_pcm_substream_chip(substream);
+        pipe = (struct audiopipe *) substream->runtime->private_data;
+
+        spin_lock_irq(&chip->lock);
+        if (pipe->index >= 0) {
+                DE_HWP(("pcm_hw_free(%d)\n", pipe->index));
+                free_pipes(chip, pipe);
+                chip->substream[pipe->index] = NULL;
+                pipe->index = -1;
+        }
+        spin_unlock_irq(&chip->lock);
+
+        DE_HWP(("pcm_hw_freed\n"));
+        snd_pcm_lib_free_pages(substream);
+        return 0;
+}
+
+
+
+static int pcm_prepare(struct snd_pcm_substream *substream)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct audioformat format;
+        int pipe_index = ((struct audiopipe *)runtime->private_data)->index;
+
+        DE_HWP(("Prepare rate=%d format=%d channels=%d\n",
+                runtime->rate, runtime->format, runtime->channels));
+        format.interleave = runtime->channels;
+        format.data_are_bigendian = 0;
+        format.mono_to_stereo = 0;
+        switch (runtime->format) {
+        case SNDRV_PCM_FORMAT_U8:
+                format.bits_per_sample = 8;
+                break;
+        case SNDRV_PCM_FORMAT_S16_LE:
+                format.bits_per_sample = 16;
+                break;
+        case SNDRV_PCM_FORMAT_S24_3LE:
+                format.bits_per_sample = 24;
+                break;
+        case SNDRV_PCM_FORMAT_S32_BE:
+                format.data_are_bigendian = 1;
+        case SNDRV_PCM_FORMAT_S32_LE:
+                format.bits_per_sample = 32;
+                break;
+        default:
+                DE_HWP(("Prepare error: unsupported format %d\n",
+                        runtime->format));
+                return -EINVAL;
+        }
+
+        snd_assert(pipe_index < px_num(chip), return -EINVAL);
+        snd_assert(is_pipe_allocated(chip, pipe_index), return -EINVAL);
+        set_audio_format(chip, pipe_index, &format);
+        return 0;
+}
+
+
+
+static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        struct echoaudio *chip = snd_pcm_substream_chip(substream);
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct audiopipe *pipe = runtime->private_data;
+        int i, err;
+        u32 channelmask = 0;
+        struct list_head *pos;
+        struct snd_pcm_substream *s;
+
+        snd_pcm_group_for_each(pos, substream) {
+                s = snd_pcm_group_substream_entry(pos);
+                for (i = 0; i < DSP_MAXPIPES; i++) {
+                        if (s == chip->substream[i]) {
+                                channelmask |= 1 << i;
+                                snd_pcm_trigger_done(s, substream);
+                        }
+                }
+        }
+
+        spin_lock(&chip->lock);
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+                DE_ACT(("pcm_trigger start\n"));
+                for (i = 0; i < DSP_MAXPIPES; i++) {
+                        if (channelmask & (1 << i)) {
+                                pipe = chip->substream[i]->runtime->private_data;
+                                switch (pipe->state) {
+                                case PIPE_STATE_STOPPED:
+                                        chip->last_period[i] = 0;
+                                        pipe->last_counter = 0;
+                                        pipe->position = 0;
+                                        *pipe->dma_counter = 0;
+                                case PIPE_STATE_PAUSED:
+                                        pipe->state = PIPE_STATE_STARTED;
+                                        break;
+                                case PIPE_STATE_STARTED:
+                                        break;
+                                }
+                        }
+                }
+                err = start_transport(chip, channelmask,
+                                      chip->pipe_cyclic_mask);
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+                DE_ACT(("pcm_trigger stop\n"));
+                for (i = 0; i < DSP_MAXPIPES; i++) {
+                        if (channelmask & (1 << i)) {
+                                pipe = chip->substream[i]->runtime->private_data;
+                                pipe->state = PIPE_STATE_STOPPED;
+                        }
+                }
+                err = stop_transport(chip, channelmask);
+                break;
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                DE_ACT(("pcm_trigger pause\n"));
+                for (i = 0; i < DSP_MAXPIPES; i++) {
+                        if (channelmask & (1 << i)) {
+                                pipe = chip->substream[i]->runtime->private_data;
+                                pipe->state = PIPE_STATE_PAUSED;
+                        }
+                }
+                err = pause_transport(chip, channelmask);
+                break;
+        default:
+                err = -EINVAL;
+        }
+        spin_unlock(&chip->lock);
+        return err;
+}
+
+
+
+static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        struct audiopipe *pipe = runtime->private_data;
+        size_t cnt, bufsize, pos;
+
+        cnt = le32_to_cpu(*pipe->dma_counter);
+        pipe->position += cnt - pipe->last_counter;
+        pipe->last_counter = cnt;
+        bufsize = substream->runtime->buffer_size;
+        pos = bytes_to_frames(substream->runtime, pipe->position);
+
+        while (pos >= bufsize) {
+                pipe->position -= frames_to_bytes(substream->runtime, bufsize);
+                pos -= bufsize;
+        }
+        return pos;
+}
+
+
+
+/* pcm *_ops structures */
+static struct snd_pcm_ops analog_playback_ops = {
+        .open = pcm_analog_out_open,
+        .close = pcm_close,
+        .ioctl = snd_pcm_lib_ioctl,
+        .hw_params = pcm_analog_out_hw_params,
+        .hw_free = pcm_hw_free,
+        .prepare = pcm_prepare,
+        .trigger = pcm_trigger,
+        .pointer = pcm_pointer,
+        .page = snd_pcm_sgbuf_ops_page,
+};
+static struct snd_pcm_ops analog_capture_ops = {
+        .open = pcm_analog_in_open,
+        .close = pcm_close,
+        .ioctl = snd_pcm_lib_ioctl,
+        .hw_params = pcm_analog_in_hw_params,
+        .hw_free = pcm_hw_free,
+        .prepare = pcm_prepare,
+        .trigger = pcm_trigger,
+        .pointer = pcm_pointer,
+        .page = snd_pcm_sgbuf_ops_page,
+};
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+#ifndef ECHOCARD_HAS_VMIXER
+static struct snd_pcm_ops digital_playback_ops = {
+        .open = pcm_digital_out_open,
+        .close = pcm_close,
+        .ioctl = snd_pcm_lib_ioctl,
+        .hw_params = pcm_digital_out_hw_params,
+        .hw_free = pcm_hw_free,
+        .prepare = pcm_prepare,
+        .trigger = pcm_trigger,
+        .pointer = pcm_pointer,
+        .page = snd_pcm_sgbuf_ops_page,
+};
+#endif /* !ECHOCARD_HAS_VMIXER */
+static struct snd_pcm_ops digital_capture_ops = {
+        .open = pcm_digital_in_open,
+        .close = pcm_close,
+        .ioctl = snd_pcm_lib_ioctl,
+        .hw_params = pcm_digital_in_hw_params,
+        .hw_free = pcm_hw_free,
+        .prepare = pcm_prepare,
+        .trigger = pcm_trigger,
+        .pointer = pcm_pointer,
+        .page = snd_pcm_sgbuf_ops_page,
+};
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+
+
+/* Preallocate memory only for the first substream because it's the most
+ * used one
+ */
+static int snd_echo_preallocate_pages(struct snd_pcm *pcm, struct device *dev)
+{
+        struct snd_pcm_substream *ss;
+        int stream, err;
+
+        for (stream = 0; stream < 2; stream++)
+                for (ss = pcm->streams[stream].substream; ss; ss = ss->next) {
+                        err = snd_pcm_lib_preallocate_pages(ss, SNDRV_DMA_TYPE_DEV_SG,
+                                                            dev,
+                                                            ss->number ? 0 : 128<<10,
+                                                            256<<10);
+                        if (err < 0)
+                                return err;
+                }
+        return 0;
+}
+
+
+
+/*<--snd_echo_probe() */
+static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
+{
+        struct snd_pcm *pcm;
+        int err;
+
+#ifdef ECHOCARD_HAS_VMIXER
+        /* This card has a Vmixer, that is there is no direct mapping from PCM
+        streams to physical outputs. The user can mix the streams as he wishes
+        via control interface and it's possible to send any stream to any
+        output, thus it makes no sense to keep analog and digital outputs
+        separated */
+
+        /* PCM#0 Virtual outputs and analog inputs */
+        if ((err = snd_pcm_new(chip->card, "PCM", 0, num_pipes_out(chip),
+                                num_analog_busses_in(chip), &pcm)) < 0)
+                return err;
+        pcm->private_data = chip;
+        chip->analog_pcm = pcm;
+        strcpy(pcm->name, chip->card->shortname);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
+        if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
+                return err;
+        DE_INIT(("Analog PCM ok\n"));
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+        /* PCM#1 Digital inputs, no outputs */
+        if ((err = snd_pcm_new(chip->card, "Digital PCM", 1, 0,
+                               num_digital_busses_in(chip), &pcm)) < 0)
+                return err;
+        pcm->private_data = chip;
+        chip->digital_pcm = pcm;
+        strcpy(pcm->name, chip->card->shortname);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
+        if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
+                return err;
+        DE_INIT(("Digital PCM ok\n"));
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+#else /* ECHOCARD_HAS_VMIXER */
+
+        /* The card can manage substreams formed by analog and digital channels
+        at the same time, but I prefer to keep analog and digital channels
+        separated, because that mixed thing is confusing and useless. So we
+        register two PCM devices: */
+
+        /* PCM#0 Analog i/o */
+        if ((err = snd_pcm_new(chip->card, "Analog PCM", 0,
+                               num_analog_busses_out(chip),
+                               num_analog_busses_in(chip), &pcm)) < 0)
+                return err;
+        pcm->private_data = chip;
+        chip->analog_pcm = pcm;
+        strcpy(pcm->name, chip->card->shortname);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
+        if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
+                return err;
+        DE_INIT(("Analog PCM ok\n"));
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+        /* PCM#1 Digital i/o */
+        if ((err = snd_pcm_new(chip->card, "Digital PCM", 1,
+                               num_digital_busses_out(chip),
+                               num_digital_busses_in(chip), &pcm)) < 0)
+                return err;
+        pcm->private_data = chip;
+        chip->digital_pcm = pcm;
+        strcpy(pcm->name, chip->card->shortname);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &digital_playback_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
+        if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
+                return err;
+        DE_INIT(("Digital PCM ok\n"));
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+#endif /* ECHOCARD_HAS_VMIXER */
+
+        return 0;
+}
+
+
+
+
+/******************************************************************************
+        Control interface
+******************************************************************************/
+
+/******************* PCM output volume *******************/
+static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
+                                     struct snd_ctl_elem_info *uinfo)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = num_busses_out(chip);
+        uinfo->value.integer.min = ECHOGAIN_MINOUT;
+        uinfo->value.integer.max = ECHOGAIN_MAXOUT;
+        return 0;
+}
+
+static int snd_echo_output_gain_get(struct snd_kcontrol *kcontrol,
+                                    struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int c;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        for (c = 0; c < num_busses_out(chip); c++)
+                ucontrol->value.integer.value[c] = chip->output_gain[c];
+        return 0;
+}
+
+static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
+                                    struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int c, changed, gain;
+
+        changed = 0;
+        chip = snd_kcontrol_chip(kcontrol);
+        spin_lock_irq(&chip->lock);
+        for (c = 0; c < num_busses_out(chip); c++) {
+                gain = ucontrol->value.integer.value[c];
+                /* Ignore out of range values */
+                if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
+                        continue;
+                if (chip->output_gain[c] != gain) {
+                        set_output_gain(chip, c, gain);
+                        changed = 1;
+                }
+        }
+        if (changed)
+                update_output_line_level(chip);
+        spin_unlock_irq(&chip->lock);
+        return changed;
+}
+
+#ifdef ECHOCARD_HAS_VMIXER
+/* On Vmixer cards this one controls the line-out volume */
+static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
+        .name = "Line Playback Volume",
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .info = snd_echo_output_gain_info,
+        .get = snd_echo_output_gain_get,
+        .put = snd_echo_output_gain_put,
+};
+#else
+static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
+        .name = "PCM Playback Volume",
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .info = snd_echo_output_gain_info,
+        .get = snd_echo_output_gain_get,
+        .put = snd_echo_output_gain_put,
+};
+#endif
+
+
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+
+/******************* Analog input volume *******************/
+static int snd_echo_input_gain_info(struct snd_kcontrol *kcontrol,
+                                    struct snd_ctl_elem_info *uinfo)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = num_analog_busses_in(chip);
+        uinfo->value.integer.min = ECHOGAIN_MININP;
+        uinfo->value.integer.max = ECHOGAIN_MAXINP;
+        return 0;
+}
+
+static int snd_echo_input_gain_get(struct snd_kcontrol *kcontrol,
+                                   struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int c;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        for (c = 0; c < num_analog_busses_in(chip); c++)
+                ucontrol->value.integer.value[c] = chip->input_gain[c];
+        return 0;
+}
+
+static int snd_echo_input_gain_put(struct snd_kcontrol *kcontrol,
+                                   struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int c, gain, changed;
+
+        changed = 0;
+        chip = snd_kcontrol_chip(kcontrol);
+        spin_lock_irq(&chip->lock);
+        for (c = 0; c < num_analog_busses_in(chip); c++) {
+                gain = ucontrol->value.integer.value[c];
+                /* Ignore out of range values */
+                if (gain < ECHOGAIN_MININP || gain > ECHOGAIN_MAXINP)
+                        continue;
+                if (chip->input_gain[c] != gain) {
+                        set_input_gain(chip, c, gain);
+                        changed = 1;
+                }
+        }
+        if (changed)
+                update_input_line_level(chip);
+        spin_unlock_irq(&chip->lock);
+        return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_line_input_gain __devinitdata = {
+        .name = "Line Capture Volume",
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .info = snd_echo_input_gain_info,
+        .get = snd_echo_input_gain_get,
+        .put = snd_echo_input_gain_put,
+};
+
+#endif /* ECHOCARD_HAS_INPUT_GAIN */
+
+
+
+#ifdef ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+
+/************ Analog output nominal level (+4dBu / -10dBV) ***************/
+static int snd_echo_output_nominal_info (struct snd_kcontrol *kcontrol,
+                                         struct snd_ctl_elem_info *uinfo)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+        uinfo->count = num_analog_busses_out(chip);
+        uinfo->value.integer.min = 0;
+        uinfo->value.integer.max = 1;
+        return 0;
+}
+
+static int snd_echo_output_nominal_get(struct snd_kcontrol *kcontrol,
+                                       struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int c;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        for (c = 0; c < num_analog_busses_out(chip); c++)
+                ucontrol->value.integer.value[c] = chip->nominal_level[c];
+        return 0;
+}
+
+static int snd_echo_output_nominal_put(struct snd_kcontrol *kcontrol,
+                                       struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int c, changed;
+
+        changed = 0;
+        chip = snd_kcontrol_chip(kcontrol);
+        spin_lock_irq(&chip->lock);
+        for (c = 0; c < num_analog_busses_out(chip); c++) {
+                if (chip->nominal_level[c] != ucontrol->value.integer.value[c]) {
+                        set_nominal_level(chip, c,
+                                          ucontrol->value.integer.value[c]);
+                        changed = 1;
+                }
+        }
+        if (changed)
+                update_output_line_level(chip);
+        spin_unlock_irq(&chip->lock);
+        return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_output_nominal_level __devinitdata = {
+        .name = "Line Playback Switch (-10dBV)",
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .info = snd_echo_output_nominal_info,
+        .get = snd_echo_output_nominal_get,
+        .put = snd_echo_output_nominal_put,
+};
+
+#endif /* ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL */
+
+
+
+#ifdef ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+
+/*************** Analog input nominal level (+4dBu / -10dBV) ***************/
+static int snd_echo_input_nominal_info(struct snd_kcontrol *kcontrol,
+                                       struct snd_ctl_elem_info *uinfo)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+        uinfo->count = num_analog_busses_in(chip);
+        uinfo->value.integer.min = 0;
+        uinfo->value.integer.max = 1;
+        return 0;
+}
+
+static int snd_echo_input_nominal_get(struct snd_kcontrol *kcontrol,
+                                      struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int c;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        for (c = 0; c < num_analog_busses_in(chip); c++)
+                ucontrol->value.integer.value[c] =
+                        chip->nominal_level[bx_analog_in(chip) + c];
+        return 0;
+}
+
+static int snd_echo_input_nominal_put(struct snd_kcontrol *kcontrol,
+                                      struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int c, changed;
+
+        changed = 0;
+        chip = snd_kcontrol_chip(kcontrol);
+        spin_lock_irq(&chip->lock);
+        for (c = 0; c < num_analog_busses_in(chip); c++) {
+                if (chip->nominal_level[bx_analog_in(chip) + c] !=
+                    ucontrol->value.integer.value[c]) {
+                        set_nominal_level(chip, bx_analog_in(chip) + c,
+                                          ucontrol->value.integer.value[c]);
+                        changed = 1;
+                }
+        }
+        if (changed)
+                update_output_line_level(chip); /* "Output" is not a mistake
+                                                 * here.
+                                                 */
+        spin_unlock_irq(&chip->lock);
+        return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_intput_nominal_level __devinitdata = {
+        .name = "Line Capture Switch (-10dBV)",
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .info = snd_echo_input_nominal_info,
+        .get = snd_echo_input_nominal_get,
+        .put = snd_echo_input_nominal_put,
+};
+
+#endif /* ECHOCARD_HAS_INPUT_NOMINAL_LEVEL */
+
+
+
+#ifdef ECHOCARD_HAS_MONITOR
+
+/******************* Monitor mixer *******************/
+static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol,
+                               struct snd_ctl_elem_info *uinfo)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = 1;
+        uinfo->value.integer.min = ECHOGAIN_MINOUT;
+        uinfo->value.integer.max = ECHOGAIN_MAXOUT;
+        uinfo->dimen.d[0] = num_busses_out(chip);
+        uinfo->dimen.d[1] = num_busses_in(chip);
+        return 0;
+}
+
+static int snd_echo_mixer_get(struct snd_kcontrol *kcontrol,
+                              struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        ucontrol->value.integer.value[0] =
+                chip->monitor_gain[ucontrol->id.index / num_busses_in(chip)]
+                        [ucontrol->id.index % num_busses_in(chip)];
+        return 0;
+}
+
+static int snd_echo_mixer_put(struct snd_kcontrol *kcontrol,
+                              struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int changed,  gain;
+        short out, in;
+
+        changed = 0;
+        chip = snd_kcontrol_chip(kcontrol);
+        out = ucontrol->id.index / num_busses_in(chip);
+        in = ucontrol->id.index % num_busses_in(chip);
+        gain = ucontrol->value.integer.value[0];
+        if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
+                return -EINVAL;
+        if (chip->monitor_gain[out][in] != gain) {
+                spin_lock_irq(&chip->lock);
+                set_monitor_gain(chip, out, in, gain);
+                update_output_line_level(chip);
+                spin_unlock_irq(&chip->lock);
+                changed = 1;
+        }
+        return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_monitor_mixer __devinitdata = {
+        .name = "Monitor Mixer Volume",
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .info = snd_echo_mixer_info,
+        .get = snd_echo_mixer_get,
+        .put = snd_echo_mixer_put,
+};
+
+#endif /* ECHOCARD_HAS_MONITOR */
+
+
+
+#ifdef ECHOCARD_HAS_VMIXER
+
+/******************* Vmixer *******************/
+static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_info *uinfo)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = 1;
+        uinfo->value.integer.min = ECHOGAIN_MINOUT;
+        uinfo->value.integer.max = ECHOGAIN_MAXOUT;
+        uinfo->dimen.d[0] = num_busses_out(chip);
+        uinfo->dimen.d[1] = num_pipes_out(chip);
+        return 0;
+}
+
+static int snd_echo_vmixer_get(struct snd_kcontrol *kcontrol,
+                               struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        ucontrol->value.integer.value[0] =
+                chip->vmixer_gain[ucontrol->id.index / num_pipes_out(chip)]
+                        [ucontrol->id.index % num_pipes_out(chip)];
+        return 0;
+}
+
+static int snd_echo_vmixer_put(struct snd_kcontrol *kcontrol,
+                               struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int gain, changed;
+        short vch, out;
+
+        changed = 0;
+        chip = snd_kcontrol_chip(kcontrol);
+        out = ucontrol->id.index / num_pipes_out(chip);
+        vch = ucontrol->id.index % num_pipes_out(chip);
+        gain = ucontrol->value.integer.value[0];
+        if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
+                return -EINVAL;
+        if (chip->vmixer_gain[out][vch] != ucontrol->value.integer.value[0]) {
+                spin_lock_irq(&chip->lock);
+                set_vmixer_gain(chip, out, vch, ucontrol->value.integer.value[0]);
+                update_vmixer_level(chip);
+                spin_unlock_irq(&chip->lock);
+                changed = 1;
+        }
+        return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_vmixer __devinitdata = {
+        .name = "VMixer Volume",
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .info = snd_echo_vmixer_info,
+        .get = snd_echo_vmixer_get,
+        .put = snd_echo_vmixer_put,
+};
+
+#endif /* ECHOCARD_HAS_VMIXER */
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+
+/******************* Digital mode switch *******************/
+static int snd_echo_digital_mode_info(struct snd_kcontrol *kcontrol,
+                                      struct snd_ctl_elem_info *uinfo)
+{
+        static char *names[4] = {
+                "S/PDIF Coaxial", "S/PDIF Optical", "ADAT Optical",
+                "S/PDIF Cdrom"
+        };
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+        uinfo->value.enumerated.items = chip->num_digital_modes;
+        uinfo->count = 1;
+        if (uinfo->value.enumerated.item >= chip->num_digital_modes)
+                uinfo->value.enumerated.item = chip->num_digital_modes - 1;
+        strcpy(uinfo->value.enumerated.name, names[
+                        chip->digital_mode_list[uinfo->value.enumerated.item]]);
+        return 0;
+}
+
+static int snd_echo_digital_mode_get(struct snd_kcontrol *kcontrol,
+                                     struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int i, mode;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        mode = chip->digital_mode;
+        for (i = chip->num_digital_modes - 1; i >= 0; i--)
+                if (mode == chip->digital_mode_list[i]) {
+                        ucontrol->value.enumerated.item[0] = i;
+                        break;
+                }
+        return 0;
+}
+
+static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol,
+                                     struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int changed;
+        unsigned short emode, dmode;
+
+        changed = 0;
+        chip = snd_kcontrol_chip(kcontrol);
+
+        emode = ucontrol->value.enumerated.item[0];
+        if (emode >= chip->num_digital_modes)
+                return -EINVAL;
+        dmode = chip->digital_mode_list[emode];
+
+        if (dmode != chip->digital_mode) {
+                /* mode_mutex is required to make this operation atomic wrt
+                pcm_digital_*_open() and set_input_clock() functions. */
+                down(&chip->mode_mutex);
+
+                /* Do not allow the user to change the digital mode when a pcm
+                device is open because it also changes the number of channels
+                and the allowed sample rates */
+                if (atomic_read(&chip->opencount)) {
+                        changed = -EAGAIN;
+                } else {
+                        changed = set_digital_mode(chip, dmode);
+                        /* If we had to change the clock source, report it */
+                        if (changed > 0 && chip->clock_src_ctl) {
+                                snd_ctl_notify(chip->card,
+                                               SNDRV_CTL_EVENT_MASK_VALUE,
+                                               &chip->clock_src_ctl->id);
+                                DE_ACT(("SDM() =%d\n", changed));
+                        }
+                        if (changed >= 0)
+                                changed = 1;    /* No errors */
+                }
+                up(&chip->mode_mutex);
+        }
+        return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_digital_mode_switch __devinitdata = {
+        .name = "Digital mode Switch",
+        .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+        .info = snd_echo_digital_mode_info,
+        .get = snd_echo_digital_mode_get,
+        .put = snd_echo_digital_mode_put,
+};
+
+#endif /* ECHOCARD_HAS_DIGITAL_MODE_SWITCH */
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+
+/******************* S/PDIF mode switch *******************/
+static int snd_echo_spdif_mode_info(struct snd_kcontrol *kcontrol,
+                                    struct snd_ctl_elem_info *uinfo)
+{
+        static char *names[2] = {"Consumer", "Professional"};
+
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+        uinfo->value.enumerated.items = 2;
+        uinfo->count = 1;
+        if (uinfo->value.enumerated.item)
+                uinfo->value.enumerated.item = 1;
+        strcpy(uinfo->value.enumerated.name,
+               names[uinfo->value.enumerated.item]);
+        return 0;
+}
+
+static int snd_echo_spdif_mode_get(struct snd_kcontrol *kcontrol,
+                                   struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        ucontrol->value.enumerated.item[0] = !!chip->professional_spdif;
+        return 0;
+}
+
+static int snd_echo_spdif_mode_put(struct snd_kcontrol *kcontrol,
+                                   struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int mode;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        mode = !!ucontrol->value.enumerated.item[0];
+        if (mode != chip->professional_spdif) {
+                spin_lock_irq(&chip->lock);
+                set_professional_spdif(chip, mode);
+                spin_unlock_irq(&chip->lock);
+                return 1;
+        }
+        return 0;
+}
+
+static struct snd_kcontrol_new snd_echo_spdif_mode_switch __devinitdata = {
+        .name = "S/PDIF mode Switch",
+        .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+        .info = snd_echo_spdif_mode_info,
+        .get = snd_echo_spdif_mode_get,
+        .put = snd_echo_spdif_mode_put,
+};
+
+#endif /* ECHOCARD_HAS_DIGITAL_IO */
+
+
+
+#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
+
+/******************* Select input clock source *******************/
+static int snd_echo_clock_source_info(struct snd_kcontrol *kcontrol,
+                                      struct snd_ctl_elem_info *uinfo)
+{
+        static char *names[8] = {
+                "Internal", "Word", "Super", "S/PDIF", "ADAT", "ESync",
+                "ESync96", "MTC"
+        };
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+        uinfo->value.enumerated.items = chip->num_clock_sources;
+        uinfo->count = 1;
+        if (uinfo->value.enumerated.item >= chip->num_clock_sources)
+                uinfo->value.enumerated.item = chip->num_clock_sources - 1;
+        strcpy(uinfo->value.enumerated.name, names[
+                        chip->clock_source_list[uinfo->value.enumerated.item]]);
+        return 0;
+}
+
+static int snd_echo_clock_source_get(struct snd_kcontrol *kcontrol,
+                                     struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int i, clock;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        clock = chip->input_clock;
+
+        for (i = 0; i < chip->num_clock_sources; i++)
+                if (clock == chip->clock_source_list[i])
+                        ucontrol->value.enumerated.item[0] = i;
+
+        return 0;
+}
+
+static int snd_echo_clock_source_put(struct snd_kcontrol *kcontrol,
+                                     struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int changed;
+        unsigned int eclock, dclock;
+
+        changed = 0;
+        chip = snd_kcontrol_chip(kcontrol);
+        eclock = ucontrol->value.enumerated.item[0];
+        if (eclock >= chip->input_clock_types)
+                return -EINVAL;
+        dclock = chip->clock_source_list[eclock];
+        if (chip->input_clock != dclock) {
+                down(&chip->mode_mutex);
+                spin_lock_irq(&chip->lock);
+                if ((changed = set_input_clock(chip, dclock)) == 0)
+                        changed = 1;    /* no errors */
+                spin_unlock_irq(&chip->lock);
+                up(&chip->mode_mutex);
+        }
+
+        if (changed < 0)
+                DE_ACT(("seticlk val%d err 0x%x\n", dclock, changed));
+
+        return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_clock_source_switch __devinitdata = {
+        .name = "Sample Clock Source",
+        .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+        .info = snd_echo_clock_source_info,
+        .get = snd_echo_clock_source_get,
+        .put = snd_echo_clock_source_put,
+};
+
+#endif /* ECHOCARD_HAS_EXTERNAL_CLOCK */
+
+
+
+#ifdef ECHOCARD_HAS_PHANTOM_POWER
+
+/******************* Phantom power switch *******************/
+static int snd_echo_phantom_power_info(struct snd_kcontrol *kcontrol,
+                                       struct snd_ctl_elem_info *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 snd_echo_phantom_power_get(struct snd_kcontrol *kcontrol,
+                                      struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
+
+        ucontrol->value.integer.value[0] = chip->phantom_power;
+        return 0;
+}
+
+static int snd_echo_phantom_power_put(struct snd_kcontrol *kcontrol,
+                                      struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
+        int power, changed = 0;
+
+        power = !!ucontrol->value.integer.value[0];
+        if (chip->phantom_power != power) {
+                spin_lock_irq(&chip->lock);
+                changed = set_phantom_power(chip, power);
+                spin_unlock_irq(&chip->lock);
+                if (changed == 0)
+                        changed = 1;    /* no errors */
+        }
+        return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_phantom_power_switch __devinitdata = {
+        .name = "Phantom power Switch",
+        .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+        .info = snd_echo_phantom_power_info,
+        .get = snd_echo_phantom_power_get,
+        .put = snd_echo_phantom_power_put,
+};
+
+#endif /* ECHOCARD_HAS_PHANTOM_POWER */
+
+
+
+#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+
+/******************* Digital input automute switch *******************/
+static int snd_echo_automute_info(struct snd_kcontrol *kcontrol,
+                                  struct snd_ctl_elem_info *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 snd_echo_automute_get(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
+
+        ucontrol->value.integer.value[0] = chip->digital_in_automute;
+        return 0;
+}
+
+static int snd_echo_automute_put(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
+        int automute, changed = 0;
+
+        automute = !!ucontrol->value.integer.value[0];
+        if (chip->digital_in_automute != automute) {
+                spin_lock_irq(&chip->lock);
+                changed = set_input_auto_mute(chip, automute);
+                spin_unlock_irq(&chip->lock);
+                if (changed == 0)
+                        changed = 1;    /* no errors */
+        }
+        return changed;
+}
+
+static struct snd_kcontrol_new snd_echo_automute_switch __devinitdata = {
+        .name = "Digital Capture Switch (automute)",
+        .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+        .info = snd_echo_automute_info,
+        .get = snd_echo_automute_get,
+        .put = snd_echo_automute_put,
+};
+
+#endif /* ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE */
+
+
+
+/******************* VU-meters switch *******************/
+static int snd_echo_vumeters_switch_info(struct snd_kcontrol *kcontrol,
+                                         struct snd_ctl_elem_info *uinfo)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+        uinfo->count = 1;
+        uinfo->value.integer.min = 0;
+        uinfo->value.integer.max = 1;
+        return 0;
+}
+
+static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol,
+                                        struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        spin_lock_irq(&chip->lock);
+        set_meters_on(chip, ucontrol->value.integer.value[0]);
+        spin_unlock_irq(&chip->lock);
+        return 1;
+}
+
+static struct snd_kcontrol_new snd_echo_vumeters_switch __devinitdata = {
+        .name = "VU-meters Switch",
+        .iface = SNDRV_CTL_ELEM_IFACE_CARD,
+        .access = SNDRV_CTL_ELEM_ACCESS_WRITE,
+        .info = snd_echo_vumeters_switch_info,
+        .put = snd_echo_vumeters_switch_put,
+};
+
+
+
+/***** Read VU-meters (input, output, analog and digital together) *****/
+static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol,
+                                  struct snd_ctl_elem_info *uinfo)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = 96;
+        uinfo->value.integer.min = ECHOGAIN_MINOUT;
+        uinfo->value.integer.max = 0;
+#ifdef ECHOCARD_HAS_VMIXER
+        uinfo->dimen.d[0] = 3;  /* Out, In, Virt */
+#else
+        uinfo->dimen.d[0] = 2;  /* Out, In */
+#endif
+        uinfo->dimen.d[1] = 16; /* 16 channels */
+        uinfo->dimen.d[2] = 2;  /* 0=level, 1=peak */
+        return 0;
+}
+
+static int snd_echo_vumeters_get(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        get_audio_meters(chip, ucontrol->value.integer.value);
+        return 0;
+}
+
+static struct snd_kcontrol_new snd_echo_vumeters __devinitdata = {
+        .name = "VU-meters",
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+        .info = snd_echo_vumeters_info,
+        .get = snd_echo_vumeters_get,
+};
+
+
+
+/*** Channels info - it exports informations about the number of channels ***/
+static int snd_echo_channels_info_info(struct snd_kcontrol *kcontrol,
+                                       struct snd_ctl_elem_info *uinfo)
+{
+        struct echoaudio *chip;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = 6;
+        uinfo->value.integer.min = 0;
+        uinfo->value.integer.max = 1 << ECHO_CLOCK_NUMBER;
+        return 0;
+}
+
+static int snd_echo_channels_info_get(struct snd_kcontrol *kcontrol,
+                                      struct snd_ctl_elem_value *ucontrol)
+{
+        struct echoaudio *chip;
+        int detected, clocks, bit, src;
+
+        chip = snd_kcontrol_chip(kcontrol);
+        ucontrol->value.integer.value[0] = num_busses_in(chip);
+        ucontrol->value.integer.value[1] = num_analog_busses_in(chip);
+        ucontrol->value.integer.value[2] = num_busses_out(chip);
+        ucontrol->value.integer.value[3] = num_analog_busses_out(chip);
+        ucontrol->value.integer.value[4] = num_pipes_out(chip);
+
+        /* Compute the bitmask of the currently valid input clocks */
+        detected = detect_input_clocks(chip);
+        clocks = 0;
+        src = chip->num_clock_sources - 1;
+        for (bit = ECHO_CLOCK_NUMBER - 1; bit >= 0; bit--)
+                if (detected & (1 << bit))
+                        for (; src >= 0; src--)
+                                if (bit == chip->clock_source_list[src]) {
+                                        clocks |= 1 << src;
+                                        break;
+                                }
+        ucontrol->value.integer.value[5] = clocks;
+
+        return 0;
+}
+
+static struct snd_kcontrol_new snd_echo_channels_info __devinitdata = {
+        .name = "Channels info",
+        .iface = SNDRV_CTL_ELEM_IFACE_HWDEP,
+        .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
+        .info = snd_echo_channels_info_info,
+        .get = snd_echo_channels_info_get,
+};
+
+
+
+
+/******************************************************************************
+        IRQ Handler
+******************************************************************************/
+
+static irqreturn_t snd_echo_interrupt(int irq, void *dev_id,
+                                      struct pt_regs *regs)
+{
+        struct echoaudio *chip = dev_id;
+        struct snd_pcm_substream *substream;
+        int period, ss, st;
+
+        spin_lock(&chip->lock);
+        st = service_irq(chip);
+        if (st < 0) {
+                spin_unlock(&chip->lock);
+                return IRQ_NONE;
+        }
+        /* The hardware doesn't tell us which substream caused the irq,
+        thus we have to check all running substreams. */
+        for (ss = 0; ss < DSP_MAXPIPES; ss++) {
+                if ((substream = chip->substream[ss])) {
+                        period = pcm_pointer(substream) /
+                                substream->runtime->period_size;
+                        if (period != chip->last_period[ss]) {
+                                chip->last_period[ss] = period;
+                                spin_unlock(&chip->lock);
+                                snd_pcm_period_elapsed(substream);
+                                spin_lock(&chip->lock);
+                        }
+                }
+        }
+        spin_unlock(&chip->lock);
+
+#ifdef ECHOCARD_HAS_MIDI
+        if (st > 0 && chip->midi_in) {
+                snd_rawmidi_receive(chip->midi_in, chip->midi_buffer, st);
+                DE_MID(("rawmidi_iread=%d\n", st));
+        }
+#endif
+        return IRQ_HANDLED;
+}
+
+
+
+
+/******************************************************************************
+        Module construction / destruction
+******************************************************************************/
+
+static int snd_echo_free(struct echoaudio *chip)
+{
+        DE_INIT(("Stop DSP...\n"));
+        if (chip->comm_page) {
+                rest_in_peace(chip);
+                snd_dma_free_pages(&chip->commpage_dma_buf);
+        }
+        DE_INIT(("Stopped.\n"));
+
+        if (chip->irq >= 0)
+                free_irq(chip->irq, (void *)chip);
+
+        if (chip->dsp_registers)
+                iounmap(chip->dsp_registers);
+
+        if (chip->iores)
+                release_and_free_resource(chip->iores);
+
+        DE_INIT(("MMIO freed.\n"));
+
+        pci_disable_device(chip->pci);
+
+        /* release chip data */
+        kfree(chip);
+        DE_INIT(("Chip freed.\n"));
+        return 0;
+}
+
+
+
+static int snd_echo_dev_free(struct snd_device *device)
+{
+        struct echoaudio *chip = device->device_data;
+
+        DE_INIT(("snd_echo_dev_free()...\n"));
+        return snd_echo_free(chip);
+}
+
+
+
+/* <--snd_echo_probe() */
+static __devinit int snd_echo_create(struct snd_card *card,
+                                     struct pci_dev *pci,
+                                     struct echoaudio **rchip)
+{
+        struct echoaudio *chip;
+        int err;
+        size_t sz;
+        static struct snd_device_ops ops = {
+                .dev_free = snd_echo_dev_free,
+        };
+
+        *rchip = NULL;
+
+        pci_write_config_byte(pci, PCI_LATENCY_TIMER, 0xC0);
+
+        if ((err = pci_enable_device(pci)) < 0)
+                return err;
+        pci_set_master(pci);
+
+        /* allocate a chip-specific data */
+        chip = kzalloc(sizeof(*chip), GFP_KERNEL);
+        if (!chip) {
+                pci_disable_device(pci);
+                return -ENOMEM;
+        }
+        DE_INIT(("chip=%p\n", chip));
+
+        spin_lock_init(&chip->lock);
+        chip->card = card;
+        chip->pci = pci;
+        chip->irq = -1;
+
+        /* PCI resource allocation */
+        chip->dsp_registers_phys = pci_resource_start(pci, 0);
+        sz = pci_resource_len(pci, 0);
+        if (sz > PAGE_SIZE)
+                sz = PAGE_SIZE;         /* We map only the required part */
+
+        if ((chip->iores = request_mem_region(chip->dsp_registers_phys, sz,
+                                              ECHOCARD_NAME)) == NULL) {
+                snd_echo_free(chip);
+                snd_printk(KERN_ERR "cannot get memory region\n");
+                return -EBUSY;
+        }
+        chip->dsp_registers = (volatile u32 __iomem *)
+                ioremap_nocache(chip->dsp_registers_phys, sz);
+
+        if (request_irq(pci->irq, snd_echo_interrupt, SA_INTERRUPT | SA_SHIRQ,
+                                                ECHOCARD_NAME, (void *)chip)) {
+                snd_echo_free(chip);
+                snd_printk(KERN_ERR "cannot grab irq\n");
+                return -EBUSY;
+        }
+        chip->irq = pci->irq;
+        DE_INIT(("pci=%p irq=%d subdev=%04x Init hardware...\n",
+                 chip->pci, chip->irq, chip->pci->subsystem_device));
+
+        /* Create the DSP comm page - this is the area of memory used for most
+        of the communication with the DSP, which accesses it via bus mastering */
+        if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+                                sizeof(struct comm_page),
+                                &chip->commpage_dma_buf) < 0) {
+                snd_echo_free(chip);
+                snd_printk(KERN_ERR "cannot allocate the comm page\n");
+                return -ENOMEM;
+        }
+        chip->comm_page_phys = chip->commpage_dma_buf.addr;
+        chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area;
+
+        err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device);
+        if (err) {
+                DE_INIT(("init_hw err=%d\n", err));
+                snd_echo_free(chip);
+                return err;
+        }
+        DE_INIT(("Card init OK\n"));
+
+        if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
+                snd_echo_free(chip);
+                return err;
+        }
+        atomic_set(&chip->opencount, 0);
+        init_MUTEX(&chip->mode_mutex);
+        chip->can_set_rate = 1;
+        *rchip = chip;
+        /* Init done ! */
+        return 0;
+}
+
+
+
+/* constructor */
+static int __devinit snd_echo_probe(struct pci_dev *pci,
+                                    const struct pci_device_id *pci_id)
+{
+        static int dev;
+        struct snd_card *card;
+        struct echoaudio *chip;
+        char *dsp;
+        int i, err;
+
+        if (dev >= SNDRV_CARDS)
+                return -ENODEV;
+        if (!enable[dev]) {
+                dev++;
+                return -ENOENT;
+        }
+
+        DE_INIT(("Echoaudio driver starting...\n"));
+        i = 0;
+        card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+        if (card == NULL)
+                return -ENOMEM;
+
+        if ((err = snd_echo_create(card, pci, &chip)) < 0) {
+                snd_card_free(card);
+                return err;
+        }
+
+        strcpy(card->driver, "Echo_" ECHOCARD_NAME);
+        strcpy(card->shortname, chip->card_name);
+
+        dsp = "56301";
+        if (pci_id->device == 0x3410)
+                dsp = "56361";
+
+        sprintf(card->longname, "%s rev.%d (DSP%s) at 0x%lx irq %i",
+                card->shortname, pci_id->subdevice & 0x000f, dsp,
+                chip->dsp_registers_phys, chip->irq);
+
+        if ((err = snd_echo_new_pcm(chip)) < 0) {
+                snd_printk(KERN_ERR "new pcm error %d\n", err);
+                snd_card_free(card);
+                return err;
+        }
+
+#ifdef ECHOCARD_HAS_MIDI
+        if (chip->has_midi) {   /* Some Mia's do not have midi */
+                if ((err = snd_echo_midi_create(card, chip)) < 0) {
+                        snd_printk(KERN_ERR "new midi error %d\n", err);
+                        snd_card_free(card);
+                        return err;
+                }
+        }
+#endif
+
+#ifdef ECHOCARD_HAS_VMIXER
+        snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0)
+                goto ctl_error;
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
+                goto ctl_error;
+#else
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0)
+                goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0)
+                goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+        if (!chip->hasnt_input_nominal_level)
+                if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_intput_nominal_level, chip))) < 0)
+                        goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_output_nominal_level, chip))) < 0)
+                goto ctl_error;
+#endif
+
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vumeters_switch, chip))) < 0)
+                goto ctl_error;
+
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vumeters, chip))) < 0)
+                goto ctl_error;
+
+#ifdef ECHOCARD_HAS_MONITOR
+        snd_echo_monitor_mixer.count = num_busses_in(chip) * num_busses_out(chip);
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_monitor_mixer, chip))) < 0)
+                goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_automute_switch, chip))) < 0)
+                goto ctl_error;
+#endif
+
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_channels_info, chip))) < 0)
+                goto ctl_error;
+
+#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+        /* Creates a list of available digital modes */
+        chip->num_digital_modes = 0;
+        for (i = 0; i < 6; i++)
+                if (chip->digital_modes & (1 << i))
+                        chip->digital_mode_list[chip->num_digital_modes++] = i;
+
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_digital_mode_switch, chip))) < 0)
+                goto ctl_error;
+#endif /* ECHOCARD_HAS_DIGITAL_MODE_SWITCH */
+
+#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
+        /* Creates a list of available clock sources */
+        chip->num_clock_sources = 0;
+        for (i = 0; i < 10; i++)
+                if (chip->input_clock_types & (1 << i))
+                        chip->clock_source_list[chip->num_clock_sources++] = i;
+
+        if (chip->num_clock_sources > 1) {
+                chip->clock_src_ctl = snd_ctl_new1(&snd_echo_clock_source_switch, chip);
+                if ((err = snd_ctl_add(chip->card, chip->clock_src_ctl)) < 0)
+                        goto ctl_error;
+        }
+#endif /* ECHOCARD_HAS_EXTERNAL_CLOCK */
+
+#ifdef ECHOCARD_HAS_DIGITAL_IO
+        if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_spdif_mode_switch, chip))) < 0)
+                goto ctl_error;
+#endif
+
+#ifdef ECHOCARD_HAS_PHANTOM_POWER
+        if (chip->has_phantom_power)
+                if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_phantom_power_switch, chip))) < 0)
+                        goto ctl_error;
+#endif
+
+        if ((err = snd_card_register(card)) < 0) {
+                snd_card_free(card);
+                goto ctl_error;
+        }
+        snd_printk(KERN_INFO "Card registered: %s\n", card->longname);
+
+        pci_set_drvdata(pci, chip);
+        dev++;
+        return 0;
+
+ctl_error:
+        snd_printk(KERN_ERR "new control error %d\n", err);
+        snd_card_free(card);
+        return err;
+}
+
+
+
+static void __devexit snd_echo_remove(struct pci_dev *pci)
+{
+        struct echoaudio *chip;
+
+        chip = pci_get_drvdata(pci);
+        if (chip)
+                snd_card_free(chip->card);
+        pci_set_drvdata(pci, NULL);
+}
+
+
+
+/******************************************************************************
+        Everything starts and ends here
+******************************************************************************/
+
+/* pci_driver definition */
+static struct pci_driver driver = {
+        .name = "Echoaudio " ECHOCARD_NAME,
+        .id_table = snd_echo_ids,
+        .probe = snd_echo_probe,
+        .remove = __devexit_p(snd_echo_remove),
+};
+
+
+
+/* initialization of the module */
+static int __init alsa_card_echo_init(void)
+{
+        return pci_register_driver(&driver);
+}
+
+
+
+/* clean up the module */
+static void __exit alsa_card_echo_exit(void)
+{
+        pci_unregister_driver(&driver);
+}
+
+
+module_init(alsa_card_echo_init)
+module_exit(alsa_card_echo_exit)
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
new file mode 100644
index 000000000000..7e88c968e22f
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -0,0 +1,590 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+ ****************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+ ****************************************************************************
+
+
+   Here's a block diagram of how most of the cards work:
+
+                  +-----------+
+           record |           |<-------------------- Inputs
+          <-------|           |        |
+     PCI          | Transport |        |
+     bus          |  engine   |       \|/
+          ------->|           |    +-------+
+            play  |           |--->|monitor|-------> Outputs
+                  +-----------+    | mixer |
+                                   +-------+
+
+   The lines going to and from the PCI bus represent "pipes".  A pipe performs
+   audio transport - moving audio data to and from buffers on the host via
+   bus mastering.
+
+   The inputs and outputs on the right represent input and output "busses."
+   A bus is a physical, real connection to the outside world.  An example
+   of a bus would be the 1/4" analog connectors on the back of Layla or
+   an RCA S/PDIF connector.
+
+   For most cards, there is a one-to-one correspondence between outputs
+   and busses; that is, each individual pipe is hard-wired to a single bus.
+
+   Cards that work this way are Darla20, Gina20, Layla20, Darla24, Gina24,
+   Layla24, Mona, and Indigo.
+
+
+   Mia has a feature called "virtual outputs."
+
+
+                  +-----------+
+           record |           |<----------------------------- Inputs
+          <-------|           |                  |
+     PCI          | Transport |                  |
+     bus          |  engine   |                 \|/
+          ------->|           |   +------+   +-------+
+            play  |           |-->|vmixer|-->|monitor|-------> Outputs
+                  +-----------+   +------+   | mixer |
+                                             +-------+
+
+
+   Obviously, the difference here is the box labeled "vmixer."  Vmixer is
+   short for "virtual output mixer."  For Mia, pipes are *not* hard-wired
+   to a single bus; the vmixer lets you mix any pipe to any bus in any
+   combination.
+
+   Note, however, that the left-hand side of the diagram is unchanged.
+   Transport works exactly the same way - the difference is in the mixer stage.
+
+
+   Pipes and busses are numbered starting at zero.
+
+
+
+   Pipe index
+   ==========
+
+   A number of calls in CEchoGals refer to a "pipe index".  A pipe index is
+   a unique number for a pipe that unambiguously refers to a playback or record
+   pipe.  Pipe indices are numbered starting with analog outputs, followed by
+   digital outputs, then analog inputs, then digital inputs.
+
+   Take Gina24 as an example:
+
+   Pipe index
+
+   0-7            Analog outputs (0 .. FirstDigitalBusOut-1)
+   8-15           Digital outputs (FirstDigitalBusOut .. NumBussesOut-1)
+   16-17          Analog inputs
+   18-25          Digital inputs
+
+
+   You get the pipe index by calling CEchoGals::OpenAudio; the other transport
+   functions take the pipe index as a parameter.  If you need a pipe index for
+   some other reason, use the handy Makepipe_index method.
+
+
+   Some calls take a CChannelMask parameter; CChannelMask is a handy way to
+   group pipe indices.
+
+
+
+   Digital mode switch
+   ===================
+
+   Some cards (right now, Gina24, Layla24, and Mona) have a Digital Mode Switch
+   or DMS.  Cards with a DMS can be set to one of three mutually exclusive
+   digital modes: S/PDIF RCA, S/PDIF optical, or ADAT optical.
+
+   This may create some confusion since ADAT optical is 8 channels wide and
+   S/PDIF is only two channels wide.  Gina24, Layla24, and Mona handle this
+   by acting as if they always have 8 digital outs and ins.  If you are in
+   either S/PDIF mode, the last 6 channels don't do anything - data sent
+   out these channels is thrown away and you will always record zeros.
+
+   Note that with Gina24, Layla24, and Mona, sample rates above 50 kHz are
+   only available if you have the card configured for S/PDIF optical or S/PDIF
+   RCA.
+
+
+
+   Double speed mode
+   =================
+
+   Some of the cards support 88.2 kHz and 96 kHz sampling (Darla24, Gina24,
+   Layla24, Mona, Mia, and Indigo).  For these cards, the driver sometimes has
+   to worry about "double speed mode"; double speed mode applies whenever the
+   sampling rate is above 50 kHz.
+
+   For instance, Mona and Layla24 support word clock sync.  However, they
+   actually support two different word clock modes - single speed (below
+   50 kHz) and double speed (above 50 kHz).  The hardware detects if a single
+   or double speed word clock signal is present; the generic code uses that
+   information to determine which mode to use.
+
+   The generic code takes care of all this for you.
+*/
+
+
+#ifndef _ECHOAUDIO_H_
+#define _ECHOAUDIO_H_
+
+
+#define TRUE 1
+#define FALSE 0
+
+#include "echoaudio_dsp.h"
+
+
+
+/***********************************************************************
+
+        PCI configuration space
+
+***********************************************************************/
+
+/*
+ * PCI vendor ID and device IDs for the hardware
+ */
+#define VENDOR_ID               0x1057
+#define DEVICE_ID_56301         0x1801
+#define DEVICE_ID_56361         0x3410
+#define SUBVENDOR_ID            0xECC0
+
+
+/*
+ * Valid Echo PCI subsystem card IDs
+ */
+#define DARLA20                 0x0010
+#define GINA20                  0x0020
+#define LAYLA20                 0x0030
+#define DARLA24                 0x0040
+#define GINA24                  0x0050
+#define LAYLA24                 0x0060
+#define MONA                    0x0070
+#define MIA                     0x0080
+#define INDIGO                  0x0090
+#define INDIGO_IO               0x00a0
+#define INDIGO_DJ               0x00b0
+#define ECHO3G                  0x0100
+
+
+/************************************************************************
+
+        Array sizes and so forth
+
+***********************************************************************/
+
+/*
+ * Sizes
+ */
+#define ECHO_MAXAUDIOINPUTS     32      /* Max audio input channels */
+#define ECHO_MAXAUDIOOUTPUTS    32      /* Max audio output channels */
+#define ECHO_MAXAUDIOPIPES      32      /* Max number of input and output
+                                         * pipes */
+#define E3G_MAX_OUTPUTS         16
+#define ECHO_MAXMIDIJACKS       1       /* Max MIDI ports */
+#define ECHO_MIDI_QUEUE_SZ      512     /* Max MIDI input queue entries */
+#define ECHO_MTC_QUEUE_SZ       32      /* Max MIDI time code input queue
+                                         * entries */
+
+/*
+ * MIDI activity indicator timeout
+ */
+#define MIDI_ACTIVITY_TIMEOUT_USEC      200000
+
+
+/****************************************************************************
+ 
+   Clocks
+
+*****************************************************************************/
+
+/*
+ * Clock numbers
+ */
+#define ECHO_CLOCK_INTERNAL             0
+#define ECHO_CLOCK_WORD                 1
+#define ECHO_CLOCK_SUPER                2
+#define ECHO_CLOCK_SPDIF                3
+#define ECHO_CLOCK_ADAT                 4
+#define ECHO_CLOCK_ESYNC                5
+#define ECHO_CLOCK_ESYNC96              6
+#define ECHO_CLOCK_MTC                  7
+#define ECHO_CLOCK_NUMBER               8
+#define ECHO_CLOCKS                     0xffff
+
+/*
+ * Clock bit numbers - used to report capabilities and whatever clocks
+ * are being detected dynamically.
+ */
+#define ECHO_CLOCK_BIT_INTERNAL         (1 << ECHO_CLOCK_INTERNAL)
+#define ECHO_CLOCK_BIT_WORD             (1 << ECHO_CLOCK_WORD)
+#define ECHO_CLOCK_BIT_SUPER            (1 << ECHO_CLOCK_SUPER)
+#define ECHO_CLOCK_BIT_SPDIF            (1 << ECHO_CLOCK_SPDIF)
+#define ECHO_CLOCK_BIT_ADAT             (1 << ECHO_CLOCK_ADAT)
+#define ECHO_CLOCK_BIT_ESYNC            (1 << ECHO_CLOCK_ESYNC)
+#define ECHO_CLOCK_BIT_ESYNC96          (1 << ECHO_CLOCK_ESYNC96)
+#define ECHO_CLOCK_BIT_MTC              (1<<ECHO_CLOCK_MTC)
+
+
+/***************************************************************************
+
+   Digital modes
+
+****************************************************************************/
+
+/*
+ * Digital modes for Mona, Layla24, and Gina24
+ */
+#define DIGITAL_MODE_NONE                       0xFF
+#define DIGITAL_MODE_SPDIF_RCA                  0
+#define DIGITAL_MODE_SPDIF_OPTICAL              1
+#define DIGITAL_MODE_ADAT                       2
+#define DIGITAL_MODE_SPDIF_CDROM                3
+#define DIGITAL_MODES                           4
+
+/*
+ * Digital mode capability masks
+ */
+#define ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA     (1 << DIGITAL_MODE_SPDIF_RCA)
+#define ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL (1 << DIGITAL_MODE_SPDIF_OPTICAL)
+#define ECHOCAPS_HAS_DIGITAL_MODE_ADAT          (1 << DIGITAL_MODE_ADAT)
+#define ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_CDROM   (1 << DIGITAL_MODE_SPDIF_CDROM)
+
+
+#define EXT_3GBOX_NC                    0x01    /* 3G box not connected */
+#define EXT_3GBOX_NOT_SET               0x02    /* 3G box not detected yet */
+
+
+#define ECHOGAIN_MUTED          (-128)  /* Minimum possible gain */
+#define ECHOGAIN_MINOUT         (-128)  /* Min output gain (dB) */
+#define ECHOGAIN_MAXOUT         (6)     /* Max output gain (dB) */
+#define ECHOGAIN_MININP         (-50)   /* Min input gain (0.5 dB) */
+#define ECHOGAIN_MAXINP         (50)    /* Max input gain (0.5 dB) */
+
+#define PIPE_STATE_STOPPED      0       /* Pipe has been reset */
+#define PIPE_STATE_PAUSED       1       /* Pipe has been stopped */
+#define PIPE_STATE_STARTED      2       /* Pipe has been started */
+#define PIPE_STATE_PENDING      3       /* Pipe has pending start */
+
+
+/* Debug initialization */
+#ifdef CONFIG_SND_DEBUG
+#define DE_INIT(x) snd_printk x
+#else
+#define DE_INIT(x)
+#endif
+
+/* Debug hw_params callbacks */
+#ifdef CONFIG_SND_DEBUG
+#define DE_HWP(x) snd_printk x
+#else
+#define DE_HWP(x)
+#endif
+
+/* Debug normal activity (open, start, stop...) */
+#ifdef CONFIG_SND_DEBUG
+#define DE_ACT(x) snd_printk x
+#else
+#define DE_ACT(x)
+#endif
+
+/* Debug midi activity */
+#ifdef CONFIG_SND_DEBUG
+#define DE_MID(x) snd_printk x
+#else
+#define DE_MID(x)
+#endif
+
+
+struct audiopipe {
+        volatile u32 *dma_counter;      /* Commpage register that contains
+                                         * the current dma position
+                                         * (lower 32 bits only)
+                                         */
+        u32 last_counter;               /* The last position, which is used
+                                         * to compute...
+                                         */
+        u32 position;                   /* ...the number of bytes tranferred
+                                         * by the DMA engine, modulo the
+                                         * buffer size
+                                         */
+        short index;                    /* Index of the first channel or <0
+                                         * if hw is not configured yet
+                                         */
+        short interleave;
+        struct snd_dma_buffer sgpage;   /* Room for the scatter-gather list */
+        struct snd_pcm_hardware hw;
+        struct snd_pcm_hw_constraint_list constr;
+        short sglist_head;
+        char state;                     /* pipe state */
+};
+
+
+struct audioformat {
+        u8 interleave;                  /* How the data is arranged in memory:
+                                         * mono = 1, stereo = 2, ...
+                                         */
+        u8 bits_per_sample;             /* 8, 16, 24, 32 (24 bits left aligned) */
+        char mono_to_stereo;            /* Only used if interleave is 1 and
+                                         * if this is an output pipe.
+                                         */
+        char data_are_bigendian;        /* 1 = big endian, 0 = little endian */
+};
+
+
+struct echoaudio {
+        spinlock_t lock;
+        struct snd_pcm_substream *substream[DSP_MAXPIPES];
+        int last_period[DSP_MAXPIPES];
+        struct semaphore mode_mutex;
+        u16 num_digital_modes, digital_mode_list[6];
+        u16 num_clock_sources, clock_source_list[10];
+        atomic_t opencount;
+        struct snd_kcontrol *clock_src_ctl;
+        struct snd_pcm *analog_pcm, *digital_pcm;
+        struct snd_card *card;
+        const char *card_name;
+        struct pci_dev *pci;
+        unsigned long dsp_registers_phys;
+        struct resource *iores;
+        struct snd_dma_buffer commpage_dma_buf;
+        int irq;
+#ifdef ECHOCARD_HAS_MIDI
+        struct snd_rawmidi *rmidi;
+        struct snd_rawmidi_substream *midi_in, *midi_out;
+#endif
+        struct timer_list timer;
+        char tinuse;                            /* Timer in use */
+        char midi_full;                         /* MIDI output buffer is full */
+        char can_set_rate;
+        char rate_set;
+
+        /* This stuff is used mainly by the lowlevel code */
+        struct comm_page *comm_page;    /* Virtual address of the memory
+                                         * seen by DSP
+                                         */
+        u32 pipe_alloc_mask;            /* Bitmask of allocated pipes */
+        u32 pipe_cyclic_mask;           /* Bitmask of pipes with cyclic
+                                         * buffers
+                                         */
+        u32 sample_rate;                /* Card sample rate in Hz */
+        u8 digital_mode;                /* Current digital mode
+                                         * (see DIGITAL_MODE_*)
+                                         */
+        u8 spdif_status;                /* Gina20, Darla20, Darla24 - only */
+        u8 clock_state;                 /* Gina20, Darla20, Darla24 - only */
+        u8 input_clock;                 /* Currently selected sample clock
+                                         * source
+                                         */
+        u8 output_clock;                /* Layla20 only */
+        char meters_enabled;            /* VU-meters status */
+        char asic_loaded;               /* Set TRUE when ASIC loaded */
+        char bad_board;                 /* Set TRUE if DSP won't load */
+        char professional_spdif;        /* 0 = consumer; 1 = professional */
+        char non_audio_spdif;           /* 3G - only */
+        char digital_in_automute;       /* Gina24, Layla24, Mona - only */
+        char has_phantom_power;
+        char hasnt_input_nominal_level; /* Gina3G */
+        char phantom_power;             /* Gina3G - only */
+        char has_midi;
+        char midi_input_enabled;
+
+#ifdef ECHOCARD_ECHO3G
+        /* External module -dependent pipe and bus indexes */
+        char px_digital_out, px_analog_in, px_digital_in, px_num;
+        char bx_digital_out, bx_analog_in, bx_digital_in, bx_num;
+#endif
+
+        char nominal_level[ECHO_MAXAUDIOPIPES]; /* True == -10dBV
+                                                 * False == +4dBu */
+        s8 input_gain[ECHO_MAXAUDIOINPUTS];     /* Input level -50..+50
+                                                 * unit is 0.5dB */
+        s8 output_gain[ECHO_MAXAUDIOOUTPUTS];   /* Output level -128..+6 dB
+                                                 * (-128=muted) */
+        s8 monitor_gain[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOINPUTS];
+                /* -128..+6 dB */
+        s8 vmixer_gain[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOOUTPUTS];
+                /* -128..+6 dB */
+
+        u16 digital_modes;              /* Bitmask of supported modes
+                                         * (see ECHOCAPS_HAS_DIGITAL_MODE_*) */
+        u16 input_clock_types;          /* Suppoted input clock types */
+        u16 output_clock_types;         /* Suppoted output clock types -
+                                         * Layla20 only */
+        u16 device_id, subdevice_id;
+        u16 *dsp_code;                  /* Current DSP code loaded,
+                                         * NULL if nothing loaded */
+        const struct firmware *dsp_code_to_load;/* DSP code to load */
+        const struct firmware *asic_code;       /* Current ASIC code */
+        u32 comm_page_phys;                     /* Physical address of the
+                                                 * memory seen by DSP */
+        volatile u32 __iomem *dsp_registers;    /* DSP's register base */
+        u32 active_mask;                        /* Chs. active mask or
+                                                 * punks out */
+
+#ifdef ECHOCARD_HAS_MIDI
+        u16 mtc_state;                          /* State for MIDI input parsing state machine */
+        u8 midi_buffer[MIDI_IN_BUFFER_SIZE];
+#endif
+};
+
+
+static int init_dsp_comm_page(struct echoaudio *chip);
+static int init_line_levels(struct echoaudio *chip);
+static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe);
+static int load_firmware(struct echoaudio *chip);
+static int wait_handshake(struct echoaudio *chip);
+static int send_vector(struct echoaudio *chip, u32 command);
+static int get_firmware(const struct firmware **fw_entry,
+                        const struct firmware *frm, struct echoaudio *chip);
+static void free_firmware(const struct firmware *fw_entry);
+
+#ifdef ECHOCARD_HAS_MIDI
+static int enable_midi_input(struct echoaudio *chip, char enable);
+static int midi_service_irq(struct echoaudio *chip);
+static int __devinit snd_echo_midi_create(struct snd_card *card,
+                                          struct echoaudio *chip);
+#endif
+
+
+static inline void clear_handshake(struct echoaudio *chip)
+{
+        chip->comm_page->handshake = 0;
+}
+
+static inline u32 get_dsp_register(struct echoaudio *chip, u32 index)
+{
+        return readl(&chip->dsp_registers[index]);
+}
+
+static inline void set_dsp_register(struct echoaudio *chip, u32 index,
+                                    u32 value)
+{
+        writel(value, &chip->dsp_registers[index]);
+}
+
+
+/* Pipe and bus indexes. PX_* and BX_* are defined as chip->px_* and chip->bx_*
+for 3G cards because they depend on the external box. They are integer
+constants for all other cards.
+Never use those defines directly, use the following functions instead. */
+
+static inline int px_digital_out(const struct echoaudio *chip)
+{
+        return PX_DIGITAL_OUT;
+}
+
+static inline int px_analog_in(const struct echoaudio *chip)
+{
+        return PX_ANALOG_IN;
+}
+
+static inline int px_digital_in(const struct echoaudio *chip)
+{
+        return PX_DIGITAL_IN;
+}
+
+static inline int px_num(const struct echoaudio *chip)
+{
+        return PX_NUM;
+}
+
+static inline int bx_digital_out(const struct echoaudio *chip)
+{
+        return BX_DIGITAL_OUT;
+}
+
+static inline int bx_analog_in(const struct echoaudio *chip)
+{
+        return BX_ANALOG_IN;
+}
+
+static inline int bx_digital_in(const struct echoaudio *chip)
+{
+        return BX_DIGITAL_IN;
+}
+
+static inline int bx_num(const struct echoaudio *chip)
+{
+        return BX_NUM;
+}
+
+static inline int num_pipes_out(const struct echoaudio *chip)
+{
+        return px_analog_in(chip);
+}
+
+static inline int num_pipes_in(const struct echoaudio *chip)
+{
+        return px_num(chip) - px_analog_in(chip);
+}
+
+static inline int num_busses_out(const struct echoaudio *chip)
+{
+        return bx_analog_in(chip);
+}
+
+static inline int num_busses_in(const struct echoaudio *chip)
+{
+        return bx_num(chip) - bx_analog_in(chip);
+}
+
+static inline int num_analog_busses_out(const struct echoaudio *chip)
+{
+        return bx_digital_out(chip);
+}
+
+static inline int num_analog_busses_in(const struct echoaudio *chip)
+{
+        return bx_digital_in(chip) - bx_analog_in(chip);
+}
+
+static inline int num_digital_busses_out(const struct echoaudio *chip)
+{
+        return num_busses_out(chip) - num_analog_busses_out(chip);
+}
+
+static inline int num_digital_busses_in(const struct echoaudio *chip)
+{
+        return num_busses_in(chip) - num_analog_busses_in(chip);
+}
+
+/* The monitor array is a one-dimensional array; compute the offset
+ * into the array */
+static inline int monitor_index(const struct echoaudio *chip, int out, int in)
+{
+        return out * num_busses_in(chip) + in;
+}
+
+
+#ifndef pci_device
+#define pci_device(chip) (&chip->pci->dev)
+#endif
+
+
+#endif /* _ECHOAUDIO_H_ */
diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c
new file mode 100644
index 000000000000..9f439ea459f4
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_3g.c
@@ -0,0 +1,431 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+
+/* These functions are common for all "3G" cards */
+
+
+static int check_asic_status(struct echoaudio *chip)
+{
+        u32 box_status;
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->comm_page->ext_box_status =
+                __constant_cpu_to_le32(E3G_ASIC_NOT_LOADED);
+        chip->asic_loaded = FALSE;
+        clear_handshake(chip);
+        send_vector(chip, DSP_VC_TEST_ASIC);
+
+        if (wait_handshake(chip)) {
+                chip->dsp_code = NULL;
+                return -EIO;
+        }
+
+        box_status = le32_to_cpu(chip->comm_page->ext_box_status);
+        DE_INIT(("box_status=%x\n", box_status));
+        if (box_status == E3G_ASIC_NOT_LOADED)
+                return -ENODEV;
+
+        chip->asic_loaded = TRUE;
+        return box_status & E3G_BOX_TYPE_MASK;
+}
+
+
+
+static inline u32 get_frq_reg(struct echoaudio *chip)
+{
+        return le32_to_cpu(chip->comm_page->e3g_frq_register);
+}
+
+
+
+/* Most configuration of 3G cards is accomplished by writing the control
+register. write_control_reg sends the new control register value to the DSP. */
+static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq,
+                             char force)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+
+        DE_ACT(("WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq));
+
+        ctl = cpu_to_le32(ctl);
+        frq = cpu_to_le32(frq);
+
+        if (ctl != chip->comm_page->control_register ||
+            frq != chip->comm_page->e3g_frq_register || force) {
+                chip->comm_page->e3g_frq_register = frq;
+                chip->comm_page->control_register = ctl;
+                clear_handshake(chip);
+                return send_vector(chip, DSP_VC_WRITE_CONTROL_REG);
+        }
+
+        DE_ACT(("WriteControlReg: not written, no change\n"));
+        return 0;
+}
+
+
+
+/* Set the digital mode - currently for Gina24, Layla24, Mona, 3G */
+static int set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+        u8 previous_mode;
+        int err, i, o;
+
+        /* All audio channels must be closed before changing the digital mode */
+        snd_assert(!chip->pipe_alloc_mask, return -EAGAIN);
+
+        snd_assert(chip->digital_modes & (1 << mode), return -EINVAL);
+
+        previous_mode = chip->digital_mode;
+        err = dsp_set_digital_mode(chip, mode);
+
+        /* If we successfully changed the digital mode from or to ADAT,
+         * then make sure all output, input and monitor levels are
+         * updated by the DSP comm object. */
+        if (err >= 0 && previous_mode != mode &&
+            (previous_mode == DIGITAL_MODE_ADAT || mode == DIGITAL_MODE_ADAT)) {
+                spin_lock_irq(&chip->lock);
+                for (o = 0; o < num_busses_out(chip); o++)
+                        for (i = 0; i < num_busses_in(chip); i++)
+                                set_monitor_gain(chip, o, i,
+                                                 chip->monitor_gain[o][i]);
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+                for (i = 0; i < num_busses_in(chip); i++)
+                        set_input_gain(chip, i, chip->input_gain[i]);
+                update_input_line_level(chip);
+#endif
+
+                for (o = 0; o < num_busses_out(chip); o++)
+                        set_output_gain(chip, o, chip->output_gain[o]);
+                update_output_line_level(chip);
+                spin_unlock_irq(&chip->lock);
+        }
+
+        return err;
+}
+
+
+
+static u32 set_spdif_bits(struct echoaudio *chip, u32 control_reg, u32 rate)
+{
+        control_reg &= E3G_SPDIF_FORMAT_CLEAR_MASK;
+
+        switch (rate) {
+        case 32000 :
+                control_reg |= E3G_SPDIF_SAMPLE_RATE0 | E3G_SPDIF_SAMPLE_RATE1;
+                break;
+        case 44100 :
+                if (chip->professional_spdif)
+                        control_reg |= E3G_SPDIF_SAMPLE_RATE0;
+                break;
+        case 48000 :
+                control_reg |= E3G_SPDIF_SAMPLE_RATE1;
+                break;
+        }
+
+        if (chip->professional_spdif)
+                control_reg |= E3G_SPDIF_PRO_MODE;
+
+        if (chip->non_audio_spdif)
+                control_reg |= E3G_SPDIF_NOT_AUDIO;
+
+        control_reg |= E3G_SPDIF_24_BIT | E3G_SPDIF_TWO_CHANNEL |
+                E3G_SPDIF_COPY_PERMIT;
+
+        return control_reg;
+}
+
+
+
+/* Set the S/PDIF output format */
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+        u32 control_reg;
+
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        chip->professional_spdif = prof;
+        control_reg = set_spdif_bits(chip, control_reg, chip->sample_rate);
+        return write_control_reg(chip, control_reg, get_frq_reg(chip), 0);
+}
+
+
+
+/* detect_input_clocks() returns a bitmask consisting of all the input clocks
+currently connected to the hardware; this changes as the user connects and
+disconnects clock inputs. You should use this information to determine which
+clocks the user is allowed to select. */
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        u32 clocks_from_dsp, clock_bits;
+
+        /* Map the DSP clock detect bits to the generic driver clock
+         * detect bits */
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+        if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD)
+                clock_bits |= ECHO_CLOCK_BIT_WORD;
+
+        switch(chip->digital_mode) {
+        case DIGITAL_MODE_SPDIF_RCA:
+        case DIGITAL_MODE_SPDIF_OPTICAL:
+                if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF)
+                        clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+                break;
+        case DIGITAL_MODE_ADAT:
+                if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_ADAT)
+                        clock_bits |= ECHO_CLOCK_BIT_ADAT;
+                break;
+        }
+
+        return clock_bits;
+}
+
+
+
+static int load_asic(struct echoaudio *chip)
+{
+        int box_type, err;
+
+        if (chip->asic_loaded)
+                return 0;
+
+        /* Give the DSP a few milliseconds to settle down */
+        mdelay(2);
+
+        err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC,
+                                &card_fw[FW_3G_ASIC]);
+        if (err < 0)
+                return err;
+
+        chip->asic_code = &card_fw[FW_3G_ASIC];
+
+        /* Now give the new ASIC a little time to set up */
+        mdelay(2);
+        /* See if it worked */
+        box_type = check_asic_status(chip);
+
+        /* Set up the control register if the load succeeded -
+         * 48 kHz, internal clock, S/PDIF RCA mode */
+        if (box_type >= 0) {
+                err = write_control_reg(chip, E3G_48KHZ,
+                                        E3G_FREQ_REG_DEFAULT, TRUE);
+                if (err < 0)
+                        return err;
+        }
+
+        return box_type;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u32 control_reg, clock, base_rate, frq_reg;
+
+        /* Only set the clock for internal mode. */
+        if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+                DE_ACT(("set_sample_rate: Cannot set sample rate - "
+                        "clock not set to CLK_CLOCKININTERNAL\n"));
+                /* Save the rate anyhow */
+                chip->comm_page->sample_rate = cpu_to_le32(rate);
+                chip->sample_rate = rate;
+                set_input_clock(chip, chip->input_clock);
+                return 0;
+        }
+
+        snd_assert(rate < 50000 || chip->digital_mode != DIGITAL_MODE_ADAT,
+                   return -EINVAL);
+
+        clock = 0;
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        control_reg &= E3G_CLOCK_CLEAR_MASK;
+
+        switch (rate) {
+        case 96000:
+                clock = E3G_96KHZ;
+                break;
+        case 88200:
+                clock = E3G_88KHZ;
+                break;
+        case 48000:
+                clock = E3G_48KHZ;
+                break;
+        case 44100:
+                clock = E3G_44KHZ;
+                break;
+        case 32000:
+                clock = E3G_32KHZ;
+                break;
+        default:
+                clock = E3G_CONTINUOUS_CLOCK;
+                if (rate > 50000)
+                        clock |= E3G_DOUBLE_SPEED_MODE;
+                break;
+        }
+
+        control_reg |= clock;
+        control_reg = set_spdif_bits(chip, control_reg, rate);
+
+        base_rate = rate;
+        if (base_rate > 50000)
+                base_rate /= 2;
+        if (base_rate < 32000)
+                base_rate = 32000;
+
+        frq_reg = E3G_MAGIC_NUMBER / base_rate - 2;
+        if (frq_reg > E3G_FREQ_REG_MAX)
+                frq_reg = E3G_FREQ_REG_MAX;
+
+        chip->comm_page->sample_rate = cpu_to_le32(rate);       /* ignored by the DSP */
+        chip->sample_rate = rate;
+        DE_ACT(("SetSampleRate: %d clock %x\n", rate, control_reg));
+
+        /* Tell the DSP about it - DSP reads both control reg & freq reg */
+        return write_control_reg(chip, control_reg, frq_reg, 0);
+}
+
+
+
+/* Set the sample clock source to internal, S/PDIF, ADAT */
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+        u32 control_reg, clocks_from_dsp;
+
+        DE_ACT(("set_input_clock:\n"));
+
+        /* Mask off the clock select bits */
+        control_reg = le32_to_cpu(chip->comm_page->control_register) &
+                E3G_CLOCK_CLEAR_MASK;
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        switch (clock) {
+        case ECHO_CLOCK_INTERNAL:
+                DE_ACT(("Set Echo3G clock to INTERNAL\n"));
+                chip->input_clock = ECHO_CLOCK_INTERNAL;
+                return set_sample_rate(chip, chip->sample_rate);
+        case ECHO_CLOCK_SPDIF:
+                if (chip->digital_mode == DIGITAL_MODE_ADAT)
+                        return -EAGAIN;
+                DE_ACT(("Set Echo3G clock to SPDIF\n"));
+                control_reg |= E3G_SPDIF_CLOCK;
+                if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96)
+                        control_reg |= E3G_DOUBLE_SPEED_MODE;
+                else
+                        control_reg &= ~E3G_DOUBLE_SPEED_MODE;
+                break;
+        case ECHO_CLOCK_ADAT:
+                if (chip->digital_mode != DIGITAL_MODE_ADAT)
+                        return -EAGAIN;
+                DE_ACT(("Set Echo3G clock to ADAT\n"));
+                control_reg |= E3G_ADAT_CLOCK;
+                control_reg &= ~E3G_DOUBLE_SPEED_MODE;
+                break;
+        case ECHO_CLOCK_WORD:
+                DE_ACT(("Set Echo3G clock to WORD\n"));
+                control_reg |= E3G_WORD_CLOCK;
+                if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96)
+                        control_reg |= E3G_DOUBLE_SPEED_MODE;
+                else
+                        control_reg &= ~E3G_DOUBLE_SPEED_MODE;
+                break;
+        default:
+                DE_ACT(("Input clock 0x%x not supported for Echo3G\n", clock));
+                return -EINVAL;
+        }
+
+        chip->input_clock = clock;
+        return write_control_reg(chip, control_reg, get_frq_reg(chip), 1);
+}
+
+
+
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+        u32 control_reg;
+        int err, incompatible_clock;
+
+        /* Set clock to "internal" if it's not compatible with the new mode */
+        incompatible_clock = FALSE;
+        switch (mode) {
+        case DIGITAL_MODE_SPDIF_OPTICAL:
+        case DIGITAL_MODE_SPDIF_RCA:
+                if (chip->input_clock == ECHO_CLOCK_ADAT)
+                        incompatible_clock = TRUE;
+                break;
+        case DIGITAL_MODE_ADAT:
+                if (chip->input_clock == ECHO_CLOCK_SPDIF)
+                        incompatible_clock = TRUE;
+                break;
+        default:
+                DE_ACT(("Digital mode not supported: %d\n", mode));
+                return -EINVAL;
+        }
+
+        spin_lock_irq(&chip->lock);
+
+        if (incompatible_clock) {
+                chip->sample_rate = 48000;
+                set_input_clock(chip, ECHO_CLOCK_INTERNAL);
+        }
+
+        /* Clear the current digital mode */
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        control_reg &= E3G_DIGITAL_MODE_CLEAR_MASK;
+
+        /* Tweak the control reg */
+        switch (mode) {
+        case DIGITAL_MODE_SPDIF_OPTICAL:
+                control_reg |= E3G_SPDIF_OPTICAL_MODE;
+                break;
+        case DIGITAL_MODE_SPDIF_RCA:
+                /* E3G_SPDIF_OPTICAL_MODE bit cleared */
+                break;
+        case DIGITAL_MODE_ADAT:
+                control_reg |= E3G_ADAT_MODE;
+                control_reg &= ~E3G_DOUBLE_SPEED_MODE;  /* @@ useless */
+                break;
+        }
+
+        err = write_control_reg(chip, control_reg, get_frq_reg(chip), 1);
+        spin_unlock_irq(&chip->lock);
+        if (err < 0)
+                return err;
+        chip->digital_mode = mode;
+
+        DE_ACT(("set_digital_mode(%d)\n", chip->digital_mode));
+        return incompatible_clock;
+}
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
new file mode 100644
index 000000000000..42afa837d9b4
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -0,0 +1,1125 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+#if PAGE_SIZE < 4096
+#error PAGE_SIZE is < 4k
+#endif
+
+static int restore_dsp_rettings(struct echoaudio *chip);
+
+
+/* Some vector commands involve the DSP reading or writing data to and from the
+comm page; if you send one of these commands to the DSP, it will complete the
+command and then write a non-zero value to the Handshake field in the
+comm page.  This function waits for the handshake to show up. */
+static int wait_handshake(struct echoaudio *chip)
+{
+        int i;
+
+        /* Wait up to 10ms for the handshake from the DSP */
+        for (i = 0; i < HANDSHAKE_TIMEOUT; i++) {
+                /* Look for the handshake value */
+                if (chip->comm_page->handshake) {
+                        /*if (i)  DE_ACT(("Handshake time: %d\n", i));*/
+                        return 0;
+                }
+                udelay(1);
+        }
+
+        snd_printk(KERN_ERR "wait_handshake(): Timeout waiting for DSP\n");
+        return -EBUSY;
+}
+
+
+
+/* Much of the interaction between the DSP and the driver is done via vector
+commands; send_vector writes a vector command to the DSP.  Typically, this
+causes the DSP to read or write fields in the comm page.
+PCI posting is not required thanks to the handshake logic. */
+static int send_vector(struct echoaudio *chip, u32 command)
+{
+        int i;
+
+        wmb();  /* Flush all pending writes before sending the command */
+
+        /* Wait up to 100ms for the "vector busy" bit to be off */
+        for (i = 0; i < VECTOR_BUSY_TIMEOUT; i++) {
+                if (!(get_dsp_register(chip, CHI32_VECTOR_REG) &
+                      CHI32_VECTOR_BUSY)) {
+                        set_dsp_register(chip, CHI32_VECTOR_REG, command);
+                        /*if (i)  DE_ACT(("send_vector time: %d\n", i));*/
+                        return 0;
+                }
+                udelay(1);
+        }
+
+        DE_ACT((KERN_ERR "timeout on send_vector\n"));
+        return -EBUSY;
+}
+
+
+
+/* write_dsp writes a 32-bit value to the DSP; this is used almost
+exclusively for loading the DSP. */
+static int write_dsp(struct echoaudio *chip, u32 data)
+{
+        u32 status, i;
+
+        for (i = 0; i < 10000000; i++) {        /* timeout = 10s */
+                status = get_dsp_register(chip, CHI32_STATUS_REG);
+                if ((status & CHI32_STATUS_HOST_WRITE_EMPTY) != 0) {
+                        set_dsp_register(chip, CHI32_DATA_REG, data);
+                        wmb();                  /* write it immediately */
+                        return 0;
+                }
+                udelay(1);
+                cond_resched();
+        }
+
+        chip->bad_board = TRUE;         /* Set TRUE until DSP re-loaded */
+        DE_ACT((KERN_ERR "write_dsp: Set bad_board to TRUE\n"));
+        return -EIO;
+}
+
+
+
+/* read_dsp reads a 32-bit value from the DSP; this is used almost
+exclusively for loading the DSP and checking the status of the ASIC. */
+static int read_dsp(struct echoaudio *chip, u32 *data)
+{
+        u32 status, i;
+
+        for (i = 0; i < READ_DSP_TIMEOUT; i++) {
+                status = get_dsp_register(chip, CHI32_STATUS_REG);
+                if ((status & CHI32_STATUS_HOST_READ_FULL) != 0) {
+                        *data = get_dsp_register(chip, CHI32_DATA_REG);
+                        return 0;
+                }
+                udelay(1);
+                cond_resched();
+        }
+
+        chip->bad_board = TRUE;         /* Set TRUE until DSP re-loaded */
+        DE_INIT((KERN_ERR "read_dsp: Set bad_board to TRUE\n"));
+        return -EIO;
+}
+
+
+
+/****************************************************************************
+        Firmware loading functions
+ ****************************************************************************/
+
+/* This function is used to read back the serial number from the DSP;
+this is triggered by the SET_COMMPAGE_ADDR command.
+Only some early Echogals products have serial numbers in the ROM;
+the serial number is not used, but you still need to do this as
+part of the DSP load process. */
+static int read_sn(struct echoaudio *chip)
+{
+        int i;
+        u32 sn[6];
+
+        for (i = 0; i < 5; i++) {
+                if (read_dsp(chip, &sn[i])) {
+                        snd_printk(KERN_ERR "Failed to read serial number\n");
+                        return -EIO;
+                }
+        }
+        DE_INIT(("Read serial number %08x %08x %08x %08x %08x\n",
+                 sn[0], sn[1], sn[2], sn[3], sn[4]));
+        return 0;
+}
+
+
+
+#ifndef ECHOCARD_HAS_ASIC
+/* This card has no ASIC, just return ok */
+static inline int check_asic_status(struct echoaudio *chip)
+{
+        chip->asic_loaded = TRUE;
+        return 0;
+}
+
+#endif /* !ECHOCARD_HAS_ASIC */
+
+
+
+#ifdef ECHOCARD_HAS_ASIC
+
+/* Load ASIC code - done after the DSP is loaded */
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+                             const struct firmware *asic)
+{
+        const struct firmware *fw;
+        int err;
+        u32 i, size;
+        u8 *code;
+
+        if ((err = get_firmware(&fw, asic, chip)) < 0) {
+                snd_printk(KERN_WARNING "Firmware not found !\n");
+                return err;
+        }
+
+        code = (u8 *)fw->data;
+        size = fw->size;
+
+        /* Send the "Here comes the ASIC" command */
+        if (write_dsp(chip, cmd) < 0)
+                goto la_error;
+
+        /* Write length of ASIC file in bytes */
+        if (write_dsp(chip, size) < 0)
+                goto la_error;
+
+        for (i = 0; i < size; i++) {
+                if (write_dsp(chip, code[i]) < 0)
+                        goto la_error;
+        }
+
+        DE_INIT(("ASIC loaded\n"));
+        free_firmware(fw);
+        return 0;
+
+la_error:
+        DE_INIT(("failed on write_dsp\n"));
+        free_firmware(fw);
+        return -EIO;
+}
+
+#endif /* ECHOCARD_HAS_ASIC */
+
+
+
+#ifdef DSP_56361
+
+/* Install the resident loader for 56361 DSPs;  The resident loader is on
+the EPROM on the board for 56301 DSP. The resident loader is a tiny little
+program that is used to load the real DSP code. */
+static int install_resident_loader(struct echoaudio *chip)
+{
+        u32 address;
+        int index, words, i;
+        u16 *code;
+        u32 status;
+        const struct firmware *fw;
+
+        /* 56361 cards only!  This check is required by the old 56301-based
+        Mona and Gina24 */
+        if (chip->device_id != DEVICE_ID_56361)
+                return 0;
+
+        /* Look to see if the resident loader is present.  If the resident
+        loader is already installed, host flag 5 will be on. */
+        status = get_dsp_register(chip, CHI32_STATUS_REG);
+        if (status & CHI32_STATUS_REG_HF5) {
+                DE_INIT(("Resident loader already installed; status is 0x%x\n",
+                         status));
+                return 0;
+        }
+
+        if ((i = get_firmware(&fw, &card_fw[FW_361_LOADER], chip)) < 0) {
+                snd_printk(KERN_WARNING "Firmware not found !\n");
+                return i;
+        }
+
+        /* The DSP code is an array of 16 bit words.  The array is divided up
+        into sections.  The first word of each section is the size in words,
+        followed by the section type.
+        Since DSP addresses and data are 24 bits wide, they each take up two
+        16 bit words in the array.
+        This is a lot like the other loader loop, but it's not a loop, you
+        don't write the memory type, and you don't write a zero at the end. */
+
+        /* Set DSP format bits for 24 bit mode */
+        set_dsp_register(chip, CHI32_CONTROL_REG,
+                         get_dsp_register(chip, CHI32_CONTROL_REG) | 0x900);
+
+        code = (u16 *)fw->data;
+
+        /* Skip the header section; the first word in the array is the size
+        of the first section, so the first real section of code is pointed
+        to by Code[0]. */
+        index = code[0];
+
+        /* Skip the section size, LRS block type, and DSP memory type */
+        index += 3;
+
+        /* Get the number of DSP words to write */
+        words = code[index++];
+
+        /* Get the DSP address for this block; 24 bits, so build from two words */
+        address = ((u32)code[index] << 16) + code[index + 1];
+        index += 2;
+
+        /* Write the count to the DSP */
+        if (write_dsp(chip, words)) {
+                DE_INIT(("install_resident_loader: Failed to write word count!\n"));
+                goto irl_error;
+        }
+        /* Write the DSP address */
+        if (write_dsp(chip, address)) {
+                DE_INIT(("install_resident_loader: Failed to write DSP address!\n"));
+                goto irl_error;
+        }
+        /* Write out this block of code to the DSP */
+        for (i = 0; i < words; i++) {
+                u32 data;
+
+                data = ((u32)code[index] << 16) + code[index + 1];
+                if (write_dsp(chip, data)) {
+                        DE_INIT(("install_resident_loader: Failed to write DSP code\n"));
+                        goto irl_error;
+                }
+                index += 2;
+        }
+
+        /* Wait for flag 5 to come up */
+        for (i = 0; i < 200; i++) {     /* Timeout is 50us * 200 = 10ms */
+                udelay(50);
+                status = get_dsp_register(chip, CHI32_STATUS_REG);
+                if (status & CHI32_STATUS_REG_HF5)
+                        break;
+        }
+
+        if (i == 200) {
+                DE_INIT(("Resident loader failed to set HF5\n"));
+                goto irl_error;
+        }
+
+        DE_INIT(("Resident loader successfully installed\n"));
+        free_firmware(fw);
+        return 0;
+
+irl_error:
+        free_firmware(fw);
+        return -EIO;
+}
+
+#endif /* DSP_56361 */
+
+
+static int load_dsp(struct echoaudio *chip, u16 *code)
+{
+        u32 address, data;
+        int index, words, i;
+
+        if (chip->dsp_code == code) {
+                DE_INIT(("DSP is already loaded!\n"));
+                return 0;
+        }
+        chip->bad_board = TRUE;         /* Set TRUE until DSP loaded */
+        chip->dsp_code = NULL;          /* Current DSP code not loaded */
+        chip->asic_loaded = FALSE;      /* Loading the DSP code will reset the ASIC */
+
+        DE_INIT(("load_dsp: Set bad_board to TRUE\n"));
+
+        /* If this board requires a resident loader, install it. */
+#ifdef DSP_56361
+        if ((i = install_resident_loader(chip)) < 0)
+                return i;
+#endif
+
+        /* Send software reset command */
+        if (send_vector(chip, DSP_VC_RESET) < 0) {
+                DE_INIT(("LoadDsp: send_vector DSP_VC_RESET failed, Critical Failure\n"));
+                return -EIO;
+        }
+        /* Delay 10us */
+        udelay(10);
+
+        /* Wait 10ms for HF3 to indicate that software reset is complete */
+        for (i = 0; i < 1000; i++) {    /* Timeout is 10us * 1000 = 10ms */
+                if (get_dsp_register(chip, CHI32_STATUS_REG) &
+                    CHI32_STATUS_REG_HF3)
+                        break;
+                udelay(10);
+        }
+
+        if (i == 1000) {
+                DE_INIT(("load_dsp: Timeout waiting for CHI32_STATUS_REG_HF3\n"));
+                return -EIO;
+        }
+
+        /* Set DSP format bits for 24 bit mode now that soft reset is done */
+        set_dsp_register(chip, CHI32_CONTROL_REG,
+                         get_dsp_register(chip, CHI32_CONTROL_REG) | 0x900);
+
+        /* Main loader loop */
+
+        index = code[0];
+        for (;;) {
+                int block_type, mem_type;
+
+                /* Total Block Size */
+                index++;
+
+                /* Block Type */
+                block_type = code[index];
+                if (block_type == 4)    /* We're finished */
+                        break;
+
+                index++;
+
+                /* Memory Type  P=0,X=1,Y=2 */
+                mem_type = code[index++];
+
+                /* Block Code Size */
+                words = code[index++];
+                if (words == 0)         /* We're finished */
+                        break;
+
+                /* Start Address */
+                address = ((u32)code[index] << 16) + code[index + 1];
+                index += 2;
+
+                if (write_dsp(chip, words) < 0) {
+                        DE_INIT(("load_dsp: failed to write number of DSP words\n"));
+                        return -EIO;
+                }
+                if (write_dsp(chip, address) < 0) {
+                        DE_INIT(("load_dsp: failed to write DSP address\n"));
+                        return -EIO;
+                }
+                if (write_dsp(chip, mem_type) < 0) {
+                        DE_INIT(("load_dsp: failed to write DSP memory type\n"));
+                        return -EIO;
+                }
+                /* Code */
+                for (i = 0; i < words; i++, index+=2) {
+                        data = ((u32)code[index] << 16) + code[index + 1];
+                        if (write_dsp(chip, data) < 0) {
+                                DE_INIT(("load_dsp: failed to write DSP data\n"));
+                                return -EIO;
+                        }
+                }
+        }
+
+        if (write_dsp(chip, 0) < 0) {   /* We're done!!! */
+                DE_INIT(("load_dsp: Failed to write final zero\n"));
+                return -EIO;
+        }
+        udelay(10);
+
+        for (i = 0; i < 5000; i++) {    /* Timeout is 100us * 5000 = 500ms */
+                /* Wait for flag 4 - indicates that the DSP loaded OK */
+                if (get_dsp_register(chip, CHI32_STATUS_REG) &
+                    CHI32_STATUS_REG_HF4) {
+                        set_dsp_register(chip, CHI32_CONTROL_REG,
+                                         get_dsp_register(chip, CHI32_CONTROL_REG) & ~0x1b00);
+
+                        if (write_dsp(chip, DSP_FNC_SET_COMMPAGE_ADDR) < 0) {
+                                DE_INIT(("load_dsp: Failed to write DSP_FNC_SET_COMMPAGE_ADDR\n"));
+                                return -EIO;
+                        }
+
+                        if (write_dsp(chip, chip->comm_page_phys) < 0) {
+                                DE_INIT(("load_dsp: Failed to write comm page address\n"));
+                                return -EIO;
+                        }
+
+                        /* Get the serial number via slave mode.
+                        This is triggered by the SET_COMMPAGE_ADDR command.
+                        We don't actually use the serial number but we have to
+                        get it as part of the DSP init voodoo. */
+                        if (read_sn(chip) < 0) {
+                                DE_INIT(("load_dsp: Failed to read serial number\n"));
+                                return -EIO;
+                        }
+
+                        chip->dsp_code = code;          /* Show which DSP code loaded */
+                        chip->bad_board = FALSE;        /* DSP OK */
+                        DE_INIT(("load_dsp: OK!\n"));
+                        return 0;
+                }
+                udelay(100);
+        }
+
+        DE_INIT(("load_dsp: DSP load timed out waiting for HF4\n"));
+        return -EIO;
+}
+
+
+
+/* load_firmware takes care of loading the DSP and any ASIC code. */
+static int load_firmware(struct echoaudio *chip)
+{
+        const struct firmware *fw;
+        int box_type, err;
+
+        snd_assert(chip->dsp_code_to_load && chip->comm_page, return -EPERM);
+
+        /* See if the ASIC is present and working - only if the DSP is already loaded */
+        if (chip->dsp_code) {
+                if ((box_type = check_asic_status(chip)) >= 0)
+                        return box_type;
+                /* ASIC check failed; force the DSP to reload */
+                chip->dsp_code = NULL;
+        }
+
+        if ((err = get_firmware(&fw, chip->dsp_code_to_load, chip)) < 0)
+                return err;
+        err = load_dsp(chip, (u16 *)fw->data);
+        free_firmware(fw);
+        if (err < 0)
+                return err;
+
+        if ((box_type = load_asic(chip)) < 0)
+                return box_type;        /* error */
+
+        if ((err = restore_dsp_rettings(chip)) < 0)
+                return err;
+
+        return box_type;
+}
+
+
+
+/****************************************************************************
+        Mixer functions
+ ****************************************************************************/
+
+#if defined(ECHOCARD_HAS_INPUT_NOMINAL_LEVEL) || \
+        defined(ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL)
+
+/* Set the nominal level for an input or output bus (true = -10dBV, false = +4dBu) */
+static int set_nominal_level(struct echoaudio *chip, u16 index, char consumer)
+{
+        snd_assert(index < num_busses_out(chip) + num_busses_in(chip),
+                   return -EINVAL);
+
+        /* Wait for the handshake (OK even if ASIC is not loaded) */
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->nominal_level[index] = consumer;
+
+        if (consumer)
+                chip->comm_page->nominal_level_mask |= cpu_to_le32(1 << index);
+        else
+                chip->comm_page->nominal_level_mask &= ~cpu_to_le32(1 << index);
+
+        return 0;
+}
+
+#endif /* ECHOCARD_HAS_*_NOMINAL_LEVEL */
+
+
+
+/* Set the gain for a single physical output channel (dB). */
+static int set_output_gain(struct echoaudio *chip, u16 channel, s8 gain)
+{
+        snd_assert(channel < num_busses_out(chip), return -EINVAL);
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        /* Save the new value */
+        chip->output_gain[channel] = gain;
+        chip->comm_page->line_out_level[channel] = gain;
+        return 0;
+}
+
+
+
+#ifdef ECHOCARD_HAS_MONITOR
+/* Set the monitor level from an input bus to an output bus. */
+static int set_monitor_gain(struct echoaudio *chip, u16 output, u16 input,
+                            s8 gain)
+{
+        snd_assert(output < num_busses_out(chip) &&
+                   input < num_busses_in(chip), return -EINVAL);
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->monitor_gain[output][input] = gain;
+        chip->comm_page->monitors[monitor_index(chip, output, input)] = gain;
+        return 0;
+}
+#endif /* ECHOCARD_HAS_MONITOR */
+
+
+/* Tell the DSP to read and update output, nominal & monitor levels in comm page. */
+static int update_output_line_level(struct echoaudio *chip)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_UPDATE_OUTVOL);
+}
+
+
+
+/* Tell the DSP to read and update input levels in comm page */
+static int update_input_line_level(struct echoaudio *chip)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_UPDATE_INGAIN);
+}
+
+
+
+/* set_meters_on turns the meters on or off.  If meters are turned on, the DSP
+will write the meter and clock detect values to the comm page at about 30Hz */
+static void set_meters_on(struct echoaudio *chip, char on)
+{
+        if (on && !chip->meters_enabled) {
+                send_vector(chip, DSP_VC_METERS_ON);
+                chip->meters_enabled = 1;
+        } else if (!on && chip->meters_enabled) {
+                send_vector(chip, DSP_VC_METERS_OFF);
+                chip->meters_enabled = 0;
+                memset((s8 *)chip->comm_page->vu_meter, ECHOGAIN_MUTED,
+                       DSP_MAXPIPES);
+                memset((s8 *)chip->comm_page->peak_meter, ECHOGAIN_MUTED,
+                       DSP_MAXPIPES);
+        }
+}
+
+
+
+/* Fill out an the given array using the current values in the comm page.
+Meters are written in the comm page by the DSP in this order:
+ Output busses
+ Input busses
+ Output pipes (vmixer cards only)
+
+This function assumes there are no more than 16 in/out busses or pipes
+Meters is an array [3][16][2] of long. */
+static void get_audio_meters(struct echoaudio *chip, long *meters)
+{
+        int i, m, n;
+
+        m = 0;
+        n = 0;
+        for (i = 0; i < num_busses_out(chip); i++, m++) {
+                meters[n++] = chip->comm_page->vu_meter[m];
+                meters[n++] = chip->comm_page->peak_meter[m];
+        }
+        for (; n < 32; n++)
+                meters[n] = 0;
+
+#ifdef ECHOCARD_ECHO3G
+        m = E3G_MAX_OUTPUTS;    /* Skip unused meters */
+#endif
+
+        for (i = 0; i < num_busses_in(chip); i++, m++) {
+                meters[n++] = chip->comm_page->vu_meter[m];
+                meters[n++] = chip->comm_page->peak_meter[m];
+        }
+        for (; n < 64; n++)
+                meters[n] = 0;
+
+#ifdef ECHOCARD_HAS_VMIXER
+        for (i = 0; i < num_pipes_out(chip); i++, m++) {
+                meters[n++] = chip->comm_page->vu_meter[m];
+                meters[n++] = chip->comm_page->peak_meter[m];
+        }
+#endif
+        for (; n < 96; n++)
+                meters[n] = 0;
+}
+
+
+
+static int restore_dsp_rettings(struct echoaudio *chip)
+{
+        int err;
+        DE_INIT(("restore_dsp_settings\n"));
+
+        if ((err = check_asic_status(chip)) < 0)
+                return err;
+
+        /* @ Gina20/Darla20 only. Should be harmless for other cards. */
+        chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF;
+        chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF;
+        chip->comm_page->handshake = 0xffffffff;
+
+        if ((err = set_sample_rate(chip, chip->sample_rate)) < 0)
+                return err;
+
+        if (chip->meters_enabled)
+                if (send_vector(chip, DSP_VC_METERS_ON) < 0)
+                        return -EIO;
+
+#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
+        if (set_input_clock(chip, chip->input_clock) < 0)
+                return -EIO;
+#endif
+
+#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH
+        if (set_output_clock(chip, chip->output_clock) < 0)
+                return -EIO;
+#endif
+
+        if (update_output_line_level(chip) < 0)
+                return -EIO;
+
+        if (update_input_line_level(chip) < 0)
+                return -EIO;
+
+#ifdef ECHOCARD_HAS_VMIXER
+        if (update_vmixer_level(chip) < 0)
+                return -EIO;
+#endif
+
+        if (wait_handshake(chip) < 0)
+                return -EIO;
+        clear_handshake(chip);
+
+        DE_INIT(("restore_dsp_rettings done\n"));
+        return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+/****************************************************************************
+        Transport functions
+ ****************************************************************************/
+
+/* set_audio_format() sets the format of the audio data in host memory for
+this pipe.  Note that _MS_ (mono-to-stereo) playback modes are not used by ALSA
+but they are here because they are just mono while capturing */
+static void set_audio_format(struct echoaudio *chip, u16 pipe_index,
+                             const struct audioformat *format)
+{
+        u16 dsp_format;
+
+        dsp_format = DSP_AUDIOFORM_SS_16LE;
+
+        /* Look for super-interleave (no big-endian and 8 bits) */
+        if (format->interleave > 2) {
+                switch (format->bits_per_sample) {
+                case 16:
+                        dsp_format = DSP_AUDIOFORM_SUPER_INTERLEAVE_16LE;
+                        break;
+                case 24:
+                        dsp_format = DSP_AUDIOFORM_SUPER_INTERLEAVE_24LE;
+                        break;
+                case 32:
+                        dsp_format = DSP_AUDIOFORM_SUPER_INTERLEAVE_32LE;
+                        break;
+                }
+                dsp_format |= format->interleave;
+        } else if (format->data_are_bigendian) {
+                /* For big-endian data, only 32 bit samples are supported */
+                switch (format->interleave) {
+                case 1:
+                        dsp_format = DSP_AUDIOFORM_MM_32BE;
+                        break;
+#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+                case 2:
+                        dsp_format = DSP_AUDIOFORM_SS_32BE;
+                        break;
+#endif
+                }
+        } else if (format->interleave == 1 &&
+                   format->bits_per_sample == 32 && !format->mono_to_stereo) {
+                /* 32 bit little-endian mono->mono case */
+                dsp_format = DSP_AUDIOFORM_MM_32LE;
+        } else {
+                /* Handle the other little-endian formats */
+                switch (format->bits_per_sample) {
+                case 8:
+                        if (format->interleave == 2)
+                                dsp_format = DSP_AUDIOFORM_SS_8;
+                        else
+                                dsp_format = DSP_AUDIOFORM_MS_8;
+                        break;
+                default:
+                case 16:
+                        if (format->interleave == 2)
+                                dsp_format = DSP_AUDIOFORM_SS_16LE;
+                        else
+                                dsp_format = DSP_AUDIOFORM_MS_16LE;
+                        break;
+                case 24:
+                        if (format->interleave == 2)
+                                dsp_format = DSP_AUDIOFORM_SS_24LE;
+                        else
+                                dsp_format = DSP_AUDIOFORM_MS_24LE;
+                        break;
+                case 32:
+                        if (format->interleave == 2)
+                                dsp_format = DSP_AUDIOFORM_SS_32LE;
+                        else
+                                dsp_format = DSP_AUDIOFORM_MS_32LE;
+                        break;
+                }
+        }
+        DE_ACT(("set_audio_format[%d] = %x\n", pipe_index, dsp_format));
+        chip->comm_page->audio_format[pipe_index] = cpu_to_le16(dsp_format);
+}
+
+
+
+/* start_transport starts transport for a set of pipes.
+The bits 1 in channel_mask specify what pipes to start. Only the bit of the
+first channel must be set, regardless its interleave.
+Same thing for pause_ and stop_ -trasport below. */
+static int start_transport(struct echoaudio *chip, u32 channel_mask,
+                           u32 cyclic_mask)
+{
+        DE_ACT(("start_transport %x\n", channel_mask));
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->comm_page->cmd_start |= cpu_to_le32(channel_mask);
+
+        if (chip->comm_page->cmd_start) {
+                clear_handshake(chip);
+                send_vector(chip, DSP_VC_START_TRANSFER);
+                if (wait_handshake(chip))
+                        return -EIO;
+                /* Keep track of which pipes are transporting */
+                chip->active_mask |= channel_mask;
+                chip->comm_page->cmd_start = 0;
+                return 0;
+        }
+
+        DE_ACT(("start_transport: No pipes to start!\n"));
+        return -EINVAL;
+}
+
+
+
+static int pause_transport(struct echoaudio *chip, u32 channel_mask)
+{
+        DE_ACT(("pause_transport %x\n", channel_mask));
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->comm_page->cmd_stop |= cpu_to_le32(channel_mask);
+        chip->comm_page->cmd_reset = 0;
+        if (chip->comm_page->cmd_stop) {
+                clear_handshake(chip);
+                send_vector(chip, DSP_VC_STOP_TRANSFER);
+                if (wait_handshake(chip))
+                        return -EIO;
+                /* Keep track of which pipes are transporting */
+                chip->active_mask &= ~channel_mask;
+                chip->comm_page->cmd_stop = 0;
+                chip->comm_page->cmd_reset = 0;
+                return 0;
+        }
+
+        DE_ACT(("pause_transport: No pipes to stop!\n"));
+        return 0;
+}
+
+
+
+static int stop_transport(struct echoaudio *chip, u32 channel_mask)
+{
+        DE_ACT(("stop_transport %x\n", channel_mask));
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->comm_page->cmd_stop |= cpu_to_le32(channel_mask);
+        chip->comm_page->cmd_reset |= cpu_to_le32(channel_mask);
+        if (chip->comm_page->cmd_reset) {
+                clear_handshake(chip);
+                send_vector(chip, DSP_VC_STOP_TRANSFER);
+                if (wait_handshake(chip))
+                        return -EIO;
+                /* Keep track of which pipes are transporting */
+                chip->active_mask &= ~channel_mask;
+                chip->comm_page->cmd_stop = 0;
+                chip->comm_page->cmd_reset = 0;
+                return 0;
+        }
+
+        DE_ACT(("stop_transport: No pipes to stop!\n"));
+        return 0;
+}
+
+
+
+static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index)
+{
+        return (chip->pipe_alloc_mask & (1 << pipe_index));
+}
+
+
+
+/* Stops everything and turns off the DSP. All pipes should be already
+stopped and unallocated. */
+static int rest_in_peace(struct echoaudio *chip)
+{
+        DE_ACT(("rest_in_peace() open=%x\n", chip->pipe_alloc_mask));
+
+        /* Stops all active pipes (just to be sure) */
+        stop_transport(chip, chip->active_mask);
+
+        set_meters_on(chip, FALSE);
+
+#ifdef ECHOCARD_HAS_MIDI
+        enable_midi_input(chip, FALSE);
+#endif
+
+        /* Go to sleep */
+        if (chip->dsp_code) {
+                /* Make load_firmware do a complete reload */
+                chip->dsp_code = NULL;
+                /* Put the DSP to sleep */
+                return send_vector(chip, DSP_VC_GO_COMATOSE);
+        }
+        return 0;
+}
+
+
+
+/* Fills the comm page with default values */
+static int init_dsp_comm_page(struct echoaudio *chip)
+{
+        /* Check if the compiler added extra padding inside the structure */
+        if (offsetof(struct comm_page, midi_output) != 0xbe0) {
+                DE_INIT(("init_dsp_comm_page() - Invalid struct comm_page structure\n"));
+                return -EPERM;
+        }
+
+        /* Init all the basic stuff */
+        chip->card_name = ECHOCARD_NAME;
+        chip->bad_board = TRUE; /* Set TRUE until DSP loaded */
+        chip->dsp_code = NULL;  /* Current DSP code not loaded */
+        chip->digital_mode = DIGITAL_MODE_NONE;
+        chip->input_clock = ECHO_CLOCK_INTERNAL;
+        chip->output_clock = ECHO_CLOCK_WORD;
+        chip->asic_loaded = FALSE;
+        memset(chip->comm_page, 0, sizeof(struct comm_page));
+
+        /* Init the comm page */
+        chip->comm_page->comm_size =
+                __constant_cpu_to_le32(sizeof(struct comm_page));
+        chip->comm_page->handshake = 0xffffffff;
+        chip->comm_page->midi_out_free_count =
+                __constant_cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE);
+        chip->comm_page->sample_rate = __constant_cpu_to_le32(44100);
+        chip->sample_rate = 44100;
+
+        /* Set line levels so we don't blast any inputs on startup */
+        memset(chip->comm_page->monitors, ECHOGAIN_MUTED, MONITOR_ARRAY_SIZE);
+        memset(chip->comm_page->vmixer, ECHOGAIN_MUTED, VMIXER_ARRAY_SIZE);
+
+        return 0;
+}
+
+
+
+/* This function initializes the several volume controls for busses and pipes.
+This MUST be called after the DSP is up and running ! */
+static int init_line_levels(struct echoaudio *chip)
+{
+        int st, i, o;
+
+        DE_INIT(("init_line_levels\n"));
+
+        /* Mute output busses */
+        for (i = 0; i < num_busses_out(chip); i++)
+                if ((st = set_output_gain(chip, i, ECHOGAIN_MUTED)))
+                        return st;
+        if ((st = update_output_line_level(chip)))
+                return st;
+
+#ifdef ECHOCARD_HAS_VMIXER
+        /* Mute the Vmixer */
+        for (i = 0; i < num_pipes_out(chip); i++)
+                for (o = 0; o < num_busses_out(chip); o++)
+                        if ((st = set_vmixer_gain(chip, o, i, ECHOGAIN_MUTED)))
+                                return st;
+        if ((st = update_vmixer_level(chip)))
+                return st;
+#endif /* ECHOCARD_HAS_VMIXER */
+
+#ifdef ECHOCARD_HAS_MONITOR
+        /* Mute the monitor mixer */
+        for (o = 0; o < num_busses_out(chip); o++)
+                for (i = 0; i < num_busses_in(chip); i++)
+                        if ((st = set_monitor_gain(chip, o, i, ECHOGAIN_MUTED)))
+                                return st;
+        if ((st = update_output_line_level(chip)))
+                return st;
+#endif /* ECHOCARD_HAS_MONITOR */
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+        for (i = 0; i < num_busses_in(chip); i++)
+                if ((st = set_input_gain(chip, i, ECHOGAIN_MUTED)))
+                        return st;
+        if ((st = update_input_line_level(chip)))
+                return st;
+#endif /* ECHOCARD_HAS_INPUT_GAIN */
+
+        return 0;
+}
+
+
+
+/* This is low level part of the interrupt handler.
+It returns -1 if the IRQ is not ours, or N>=0 if it is, where N is the number
+of midi data in the input queue. */
+static int service_irq(struct echoaudio *chip)
+{
+        int st;
+
+        /* Read the DSP status register and see if this DSP generated this interrupt */
+        if (get_dsp_register(chip, CHI32_STATUS_REG) & CHI32_STATUS_IRQ) {
+                st = 0;
+#ifdef ECHOCARD_HAS_MIDI
+                /* Get and parse midi data if present */
+                if (chip->comm_page->midi_input[0])     /* The count is at index 0 */
+                        st = midi_service_irq(chip);    /* Returns how many midi bytes we received */
+#endif
+                /* Clear the hardware interrupt */
+                chip->comm_page->midi_input[0] = 0;
+                send_vector(chip, DSP_VC_ACK_INT);
+                return st;
+        }
+        return -1;
+}
+
+
+
+
+/******************************************************************************
+        Functions for opening and closing pipes
+ ******************************************************************************/
+
+/* allocate_pipes is used to reserve audio pipes for your exclusive use.
+The call will fail if some pipes are already allocated. */
+static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe,
+                          int pipe_index, int interleave)
+{
+        int i;
+        u32 channel_mask;
+        char is_cyclic;
+
+        DE_ACT(("allocate_pipes: ch=%d int=%d\n", pipe_index, interleave));
+
+        if (chip->bad_board)
+                return -EIO;
+
+        is_cyclic = 1;  /* This driver uses cyclic buffers only */
+
+        for (channel_mask = i = 0; i < interleave; i++)
+                channel_mask |= 1 << (pipe_index + i);
+        if (chip->pipe_alloc_mask & channel_mask) {
+                DE_ACT(("allocate_pipes: channel already open\n"));
+                return -EAGAIN;
+        }
+
+        chip->comm_page->position[pipe_index] = 0;
+        chip->pipe_alloc_mask |= channel_mask;
+        if (is_cyclic)
+                chip->pipe_cyclic_mask |= channel_mask;
+        pipe->index = pipe_index;
+        pipe->interleave = interleave;
+        pipe->state = PIPE_STATE_STOPPED;
+
+        /* The counter register is where the DSP writes the 32 bit DMA
+        position for a pipe.  The DSP is constantly updating this value as
+        it moves data. The DMA counter is in units of bytes, not samples. */
+        pipe->dma_counter = &chip->comm_page->position[pipe_index];
+        *pipe->dma_counter = 0;
+        DE_ACT(("allocate_pipes: ok\n"));
+        return pipe_index;
+}
+
+
+
+static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe)
+{
+        u32 channel_mask;
+        int i;
+
+        DE_ACT(("free_pipes: Pipe %d\n", pipe->index));
+        snd_assert(is_pipe_allocated(chip, pipe->index), return -EINVAL);
+        snd_assert(pipe->state == PIPE_STATE_STOPPED, return -EINVAL);
+
+        for (channel_mask = i = 0; i < pipe->interleave; i++)
+                channel_mask |= 1 << (pipe->index + i);
+
+        chip->pipe_alloc_mask &= ~channel_mask;
+        chip->pipe_cyclic_mask &= ~channel_mask;
+        return 0;
+}
+
+
+
+/******************************************************************************
+        Functions for managing the scatter-gather list
+******************************************************************************/
+
+static int sglist_init(struct echoaudio *chip, struct audiopipe *pipe)
+{
+        pipe->sglist_head = 0;
+        memset(pipe->sgpage.area, 0, PAGE_SIZE);
+        chip->comm_page->sglist_addr[pipe->index].addr =
+                cpu_to_le32(pipe->sgpage.addr);
+        return 0;
+}
+
+
+
+static int sglist_add_mapping(struct echoaudio *chip, struct audiopipe *pipe,
+                                dma_addr_t address, size_t length)
+{
+        int head = pipe->sglist_head;
+        struct sg_entry *list = (struct sg_entry *)pipe->sgpage.area;
+
+        if (head < MAX_SGLIST_ENTRIES - 1) {
+                list[head].addr = cpu_to_le32(address);
+                list[head].size = cpu_to_le32(length);
+                pipe->sglist_head++;
+        } else {
+                DE_ACT(("SGlist: too many fragments\n"));
+                return -ENOMEM;
+        }
+        return 0;
+}
+
+
+
+static inline int sglist_add_irq(struct echoaudio *chip, struct audiopipe *pipe)
+{
+        return sglist_add_mapping(chip, pipe, 0, 0);
+}
+
+
+
+static inline int sglist_wrap(struct echoaudio *chip, struct audiopipe *pipe)
+{
+        return sglist_add_mapping(chip, pipe, pipe->sgpage.addr, 0);
+}
diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h
new file mode 100644
index 000000000000..e55ee00991ac
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_dsp.h
@@ -0,0 +1,694 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+#ifndef _ECHO_DSP_
+#define _ECHO_DSP_
+
+
+/**** Echogals: Darla20, Gina20, Layla20, and Darla24 ****/
+#if defined(ECHOGALS_FAMILY)
+
+#define NUM_ASIC_TESTS          5
+#define READ_DSP_TIMEOUT        1000000L        /* one second */
+
+/**** Echo24: Gina24, Layla24, Mona, Mia, Mia-midi ****/
+#elif defined(ECHO24_FAMILY)
+
+#define DSP_56361                       /* Some Echo24 cards use the 56361 DSP */
+#define READ_DSP_TIMEOUT        100000L         /* .1 second */
+
+/**** 3G: Gina3G, Layla3G ****/
+#elif defined(ECHO3G_FAMILY)
+
+#define DSP_56361
+#define READ_DSP_TIMEOUT        100000L         /* .1 second */
+#define MIN_MTC_1X_RATE         32000
+
+/**** Indigo: Indigo, Indigo IO, Indigo DJ ****/
+#elif defined(INDIGO_FAMILY)
+
+#define DSP_56361
+#define READ_DSP_TIMEOUT        100000L         /* .1 second */
+
+#else
+
+#error No family is defined
+
+#endif
+
+
+
+/*
+ *
+ *  Max inputs and outputs
+ *
+ */
+
+#define DSP_MAXAUDIOINPUTS              16      /* Max audio input channels */
+#define DSP_MAXAUDIOOUTPUTS             16      /* Max audio output channels */
+#define DSP_MAXPIPES                    32      /* Max total pipes (input + output) */
+
+
+/*
+ *
+ * These are the offsets for the memory-mapped DSP registers; the DSP base
+ * address is treated as the start of a u32 array.
+ */
+
+#define CHI32_CONTROL_REG               4
+#define CHI32_STATUS_REG                5
+#define CHI32_VECTOR_REG                6
+#define CHI32_DATA_REG                  7
+
+
+/*
+ *
+ * Interesting bits within the DSP registers
+ *
+ */
+
+#define CHI32_VECTOR_BUSY               0x00000001
+#define CHI32_STATUS_REG_HF3            0x00000008
+#define CHI32_STATUS_REG_HF4            0x00000010
+#define CHI32_STATUS_REG_HF5            0x00000020
+#define CHI32_STATUS_HOST_READ_FULL     0x00000004
+#define CHI32_STATUS_HOST_WRITE_EMPTY   0x00000002
+#define CHI32_STATUS_IRQ                0x00000040
+
+
+/* 
+ *
+ * DSP commands sent via slave mode; these are sent to the DSP by write_dsp()
+ *
+ */
+
+#define DSP_FNC_SET_COMMPAGE_ADDR               0x02
+#define DSP_FNC_LOAD_LAYLA_ASIC                 0xa0
+#define DSP_FNC_LOAD_GINA24_ASIC                0xa0
+#define DSP_FNC_LOAD_MONA_PCI_CARD_ASIC         0xa0
+#define DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC      0xa0
+#define DSP_FNC_LOAD_MONA_EXTERNAL_ASIC         0xa1
+#define DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC      0xa1
+#define DSP_FNC_LOAD_3G_ASIC                    0xa0
+
+
+/*
+ *
+ * Defines to handle the MIDI input state engine; these are used to properly
+ * extract MIDI time code bytes and their timestamps from the MIDI input stream.
+ *
+ */
+
+#define MIDI_IN_STATE_NORMAL    0
+#define MIDI_IN_STATE_TS_HIGH   1
+#define MIDI_IN_STATE_TS_LOW    2
+#define MIDI_IN_STATE_F1_DATA   3
+#define MIDI_IN_SKIP_DATA       (-1)
+
+
+/*----------------------------------------------------------------------------
+
+Setting the sample rates on Layla24 is somewhat schizophrenic.
+
+For standard rates, it works exactly like Mona and Gina24.  That is, for
+8, 11.025, 16, 22.05, 32, 44.1, 48, 88.2, and 96 kHz, you just set the
+appropriate bits in the control register and write the control register.
+
+In order to support MIDI time code sync (and possibly SMPTE LTC sync in
+the future), Layla24 also has "continuous sample rate mode".  In this mode,
+Layla24 can generate any sample rate between 25 and 50 kHz inclusive, or
+50 to 100 kHz inclusive for double speed mode.
+
+To use continuous mode:
+
+-Set the clock select bits in the control register to 0xe (see the #define
+ below)
+
+-Set double-speed mode if you want to use sample rates above 50 kHz
+
+-Write the control register as you would normally
+
+-Now, you need to set the frequency register. First, you need to determine the
+ value for the frequency register.  This is given by the following formula:
+
+frequency_reg = (LAYLA24_MAGIC_NUMBER / sample_rate) - 2
+
+Note the #define below for the magic number
+
+-Wait for the DSP handshake
+-Write the frequency_reg value to the .SampleRate field of the comm page
+-Send the vector command SET_LAYLA24_FREQUENCY_REG (see vmonkey.h)
+
+Once you have set the control register up for continuous mode, you can just
+write the frequency register to change the sample rate.  This could be
+used for MIDI time code sync. For MTC sync, the control register is set for
+continuous mode.  The driver then just keeps writing the
+SET_LAYLA24_FREQUENCY_REG command.
+
+-----------------------------------------------------------------------------*/
+
+#define LAYLA24_MAGIC_NUMBER                    677376000
+#define LAYLA24_CONTINUOUS_CLOCK                0x000e
+
+
+/*
+ *
+ * DSP vector commands
+ *
+ */
+
+#define DSP_VC_RESET                            0x80ff
+
+#ifndef DSP_56361
+
+#define DSP_VC_ACK_INT                          0x8073
+#define DSP_VC_SET_VMIXER_GAIN                  0x0000  /* Not used, only for compile */
+#define DSP_VC_START_TRANSFER                   0x0075  /* Handshke rqd. */
+#define DSP_VC_METERS_ON                        0x0079
+#define DSP_VC_METERS_OFF                       0x007b
+#define DSP_VC_UPDATE_OUTVOL                    0x007d  /* Handshke rqd. */
+#define DSP_VC_UPDATE_INGAIN                    0x007f  /* Handshke rqd. */
+#define DSP_VC_ADD_AUDIO_BUFFER                 0x0081  /* Handshke rqd. */
+#define DSP_VC_TEST_ASIC                        0x00eb
+#define DSP_VC_UPDATE_CLOCKS                    0x00ef  /* Handshke rqd. */
+#define DSP_VC_SET_LAYLA_SAMPLE_RATE            0x00f1  /* Handshke rqd. */
+#define DSP_VC_SET_GD_AUDIO_STATE               0x00f1  /* Handshke rqd. */
+#define DSP_VC_WRITE_CONTROL_REG                0x00f1  /* Handshke rqd. */
+#define DSP_VC_MIDI_WRITE                       0x00f5  /* Handshke rqd. */
+#define DSP_VC_STOP_TRANSFER                    0x00f7  /* Handshke rqd. */
+#define DSP_VC_UPDATE_FLAGS                     0x00fd  /* Handshke rqd. */
+#define DSP_VC_GO_COMATOSE                      0x00f9
+
+#else /* !DSP_56361 */
+
+/* Vector commands for families that use either the 56301 or 56361 */
+#define DSP_VC_ACK_INT                          0x80F5
+#define DSP_VC_SET_VMIXER_GAIN                  0x00DB  /* Handshke rqd. */
+#define DSP_VC_START_TRANSFER                   0x00DD  /* Handshke rqd. */
+#define DSP_VC_METERS_ON                        0x00EF
+#define DSP_VC_METERS_OFF                       0x00F1
+#define DSP_VC_UPDATE_OUTVOL                    0x00E3  /* Handshke rqd. */
+#define DSP_VC_UPDATE_INGAIN                    0x00E5  /* Handshke rqd. */
+#define DSP_VC_ADD_AUDIO_BUFFER                 0x00E1  /* Handshke rqd. */
+#define DSP_VC_TEST_ASIC                        0x00ED
+#define DSP_VC_UPDATE_CLOCKS                    0x00E9  /* Handshke rqd. */
+#define DSP_VC_SET_LAYLA24_FREQUENCY_REG        0x00E9  /* Handshke rqd. */
+#define DSP_VC_SET_LAYLA_SAMPLE_RATE            0x00EB  /* Handshke rqd. */
+#define DSP_VC_SET_GD_AUDIO_STATE               0x00EB  /* Handshke rqd. */
+#define DSP_VC_WRITE_CONTROL_REG                0x00EB  /* Handshke rqd. */
+#define DSP_VC_MIDI_WRITE                       0x00E7  /* Handshke rqd. */
+#define DSP_VC_STOP_TRANSFER                    0x00DF  /* Handshke rqd. */
+#define DSP_VC_UPDATE_FLAGS                     0x00FB  /* Handshke rqd. */
+#define DSP_VC_GO_COMATOSE                      0x00d9
+
+#endif /* !DSP_56361 */
+
+
+/*
+ *
+ * Timeouts
+ *
+ */
+
+#define HANDSHAKE_TIMEOUT               20000   /* send_vector command timeout (20ms) */
+#define VECTOR_BUSY_TIMEOUT             100000  /* 100ms */
+#define MIDI_OUT_DELAY_USEC             2000    /* How long to wait after MIDI fills up */
+
+
+/*
+ *
+ * Flags for .Flags field in the comm page
+ *
+ */
+
+#define DSP_FLAG_MIDI_INPUT             0x0001  /* Enable MIDI input */
+#define DSP_FLAG_SPDIF_NONAUDIO         0x0002  /* Sets the "non-audio" bit
+                                                 * in the S/PDIF out status
+                                                 * bits.  Clear this flag for
+                                                 * audio data;
+                                                 * set it for AC3 or WMA or
+                                                 * some such */
+#define DSP_FLAG_PROFESSIONAL_SPDIF     0x0008  /* 1 Professional, 0 Consumer */
+
+
+/*
+ *
+ * Clock detect bits reported by the DSP for Gina20, Layla20, Darla24, and Mia
+ *
+ */
+
+#define GLDM_CLOCK_DETECT_BIT_WORD      0x0002
+#define GLDM_CLOCK_DETECT_BIT_SUPER     0x0004
+#define GLDM_CLOCK_DETECT_BIT_SPDIF     0x0008
+#define GLDM_CLOCK_DETECT_BIT_ESYNC     0x0010
+
+
+/*
+ *
+ * Clock detect bits reported by the DSP for Gina24, Mona, and Layla24
+ *
+ */
+
+#define GML_CLOCK_DETECT_BIT_WORD96     0x0002
+#define GML_CLOCK_DETECT_BIT_WORD48     0x0004
+#define GML_CLOCK_DETECT_BIT_SPDIF48    0x0008
+#define GML_CLOCK_DETECT_BIT_SPDIF96    0x0010
+#define GML_CLOCK_DETECT_BIT_WORD       (GML_CLOCK_DETECT_BIT_WORD96 | GML_CLOCK_DETECT_BIT_WORD48)
+#define GML_CLOCK_DETECT_BIT_SPDIF      (GML_CLOCK_DETECT_BIT_SPDIF48 | GML_CLOCK_DETECT_BIT_SPDIF96)
+#define GML_CLOCK_DETECT_BIT_ESYNC      0x0020
+#define GML_CLOCK_DETECT_BIT_ADAT       0x0040
+
+
+/*
+ *
+ * Layla clock numbers to send to DSP
+ *
+ */
+
+#define LAYLA20_CLOCK_INTERNAL          0
+#define LAYLA20_CLOCK_SPDIF             1
+#define LAYLA20_CLOCK_WORD              2
+#define LAYLA20_CLOCK_SUPER             3
+
+
+/*
+ *
+ * Gina/Darla clock states
+ *
+ */
+
+#define GD_CLOCK_NOCHANGE               0
+#define GD_CLOCK_44                     1
+#define GD_CLOCK_48                     2
+#define GD_CLOCK_SPDIFIN                3
+#define GD_CLOCK_UNDEF                  0xff
+
+
+/*
+ *
+ * Gina/Darla S/PDIF status bits
+ *
+ */
+
+#define GD_SPDIF_STATUS_NOCHANGE        0
+#define GD_SPDIF_STATUS_44              1
+#define GD_SPDIF_STATUS_48              2
+#define GD_SPDIF_STATUS_UNDEF           0xff
+
+
+/*
+ *
+ * Layla20 output clocks
+ *
+ */
+
+#define LAYLA20_OUTPUT_CLOCK_SUPER      0
+#define LAYLA20_OUTPUT_CLOCK_WORD       1
+
+
+/****************************************************************************
+
+   Magic constants for the Darla24 hardware
+
+ ****************************************************************************/
+
+#define GD24_96000      0x0
+#define GD24_48000      0x1
+#define GD24_44100      0x2
+#define GD24_32000      0x3
+#define GD24_22050      0x4
+#define GD24_16000      0x5
+#define GD24_11025      0x6
+#define GD24_8000       0x7
+#define GD24_88200      0x8
+#define GD24_EXT_SYNC   0x9
+
+
+/*
+ *
+ * Return values from the DSP when ASIC is loaded
+ *
+ */
+
+#define ASIC_ALREADY_LOADED     0x1
+#define ASIC_NOT_LOADED         0x0
+
+
+/*
+ *
+ * DSP Audio formats
+ *
+ * These are the audio formats that the DSP can transfer
+ * via input and output pipes.  LE means little-endian,
+ * BE means big-endian.
+ *
+ * DSP_AUDIOFORM_MS_8   
+ *
+ *    8-bit mono unsigned samples.  For playback,
+ *    mono data is duplicated out the left and right channels
+ *    of the output bus.  The "MS" part of the name
+ *    means mono->stereo.
+ *
+ * DSP_AUDIOFORM_MS_16LE
+ *
+ *    16-bit signed little-endian mono samples.  Playback works
+ *    like the previous code.
+ *
+ * DSP_AUDIOFORM_MS_24LE
+ *
+ *    24-bit signed little-endian mono samples.  Data is packed
+ *    three bytes per sample; if you had two samples 0x112233 and 0x445566
+ *    they would be stored in memory like this: 33 22 11 66 55 44.
+ *
+ * DSP_AUDIOFORM_MS_32LE
+ * 
+ *    24-bit signed little-endian mono samples in a 32-bit 
+ *    container.  In other words, each sample is a 32-bit signed 
+ *    integer, where the actual audio data is left-justified 
+ *    in the 32 bits and only the 24 most significant bits are valid.
+ *
+ * DSP_AUDIOFORM_SS_8
+ * DSP_AUDIOFORM_SS_16LE
+ * DSP_AUDIOFORM_SS_24LE
+ * DSP_AUDIOFORM_SS_32LE
+ *
+ *    Like the previous ones, except now with stereo interleaved
+ *    data.  "SS" means stereo->stereo.
+ *
+ * DSP_AUDIOFORM_MM_32LE
+ *
+ *    Similar to DSP_AUDIOFORM_MS_32LE, except that the mono
+ *    data is not duplicated out both the left and right outputs.
+ *    This mode is used by the ASIO driver.  Here, "MM" means
+ *    mono->mono.
+ *
+ * DSP_AUDIOFORM_MM_32BE
+ *
+ *    Just like DSP_AUDIOFORM_MM_32LE, but now the data is
+ *    in big-endian format.
+ *
+ */
+
+#define DSP_AUDIOFORM_MS_8      0       /* 8 bit mono */
+#define DSP_AUDIOFORM_MS_16LE   1       /* 16 bit mono */
+#define DSP_AUDIOFORM_MS_24LE   2       /* 24 bit mono */
+#define DSP_AUDIOFORM_MS_32LE   3       /* 32 bit mono */
+#define DSP_AUDIOFORM_SS_8      4       /* 8 bit stereo */
+#define DSP_AUDIOFORM_SS_16LE   5       /* 16 bit stereo */
+#define DSP_AUDIOFORM_SS_24LE   6       /* 24 bit stereo */
+#define DSP_AUDIOFORM_SS_32LE   7       /* 32 bit stereo */
+#define DSP_AUDIOFORM_MM_32LE   8       /* 32 bit mono->mono little-endian */
+#define DSP_AUDIOFORM_MM_32BE   9       /* 32 bit mono->mono big-endian */
+#define DSP_AUDIOFORM_SS_32BE   10      /* 32 bit stereo big endian */
+#define DSP_AUDIOFORM_INVALID   0xFF    /* Invalid audio format */
+
+
+/*
+ *
+ * Super-interleave is defined as interleaving by 4 or more.  Darla20 and Gina20
+ * do not support super interleave.
+ *
+ * 16 bit, 24 bit, and 32 bit little endian samples are supported for super 
+ * interleave.  The interleave factor must be even.  16 - way interleave is the 
+ * current maximum, so you can interleave by 4, 6, 8, 10, 12, 14, and 16.
+ *
+ * The actual format code is derived by taking the define below and or-ing with
+ * the interleave factor.  So, 32 bit interleave by 6 is 0x86 and
+ * 16 bit interleave by 16 is (0x40 | 0x10) = 0x50.
+ *
+ */
+
+#define DSP_AUDIOFORM_SUPER_INTERLEAVE_16LE     0x40
+#define DSP_AUDIOFORM_SUPER_INTERLEAVE_24LE     0xc0
+#define DSP_AUDIOFORM_SUPER_INTERLEAVE_32LE     0x80
+
+
+/*
+ *
+ * Gina24, Mona, and Layla24 control register defines
+ *
+ */
+
+#define GML_CONVERTER_ENABLE    0x0010
+#define GML_SPDIF_PRO_MODE      0x0020  /* Professional S/PDIF == 1,
+                                           consumer == 0 */
+#define GML_SPDIF_SAMPLE_RATE0  0x0040
+#define GML_SPDIF_SAMPLE_RATE1  0x0080
+#define GML_SPDIF_TWO_CHANNEL   0x0100  /* 1 == two channels,
+                                           0 == one channel */
+#define GML_SPDIF_NOT_AUDIO     0x0200
+#define GML_SPDIF_COPY_PERMIT   0x0400
+#define GML_SPDIF_24_BIT        0x0800  /* 1 == 24 bit, 0 == 20 bit */
+#define GML_ADAT_MODE           0x1000  /* 1 == ADAT mode, 0 == S/PDIF mode */
+#define GML_SPDIF_OPTICAL_MODE  0x2000  /* 1 == optical mode, 0 == RCA mode */
+#define GML_SPDIF_CDROM_MODE    0x3000  /* 1 == CDROM mode,
+                                         * 0 == RCA or optical mode */
+#define GML_DOUBLE_SPEED_MODE   0x4000  /* 1 == double speed,
+                                           0 == single speed */
+
+#define GML_DIGITAL_IN_AUTO_MUTE 0x800000
+
+#define GML_96KHZ               (0x0 | GML_DOUBLE_SPEED_MODE)
+#define GML_88KHZ               (0x1 | GML_DOUBLE_SPEED_MODE)
+#define GML_48KHZ               0x2
+#define GML_44KHZ               0x3
+#define GML_32KHZ               0x4
+#define GML_22KHZ               0x5
+#define GML_16KHZ               0x6
+#define GML_11KHZ               0x7
+#define GML_8KHZ                0x8
+#define GML_SPDIF_CLOCK         0x9
+#define GML_ADAT_CLOCK          0xA
+#define GML_WORD_CLOCK          0xB
+#define GML_ESYNC_CLOCK         0xC
+#define GML_ESYNCx2_CLOCK       0xD
+
+#define GML_CLOCK_CLEAR_MASK            0xffffbff0
+#define GML_SPDIF_RATE_CLEAR_MASK       (~(GML_SPDIF_SAMPLE_RATE0|GML_SPDIF_SAMPLE_RATE1))
+#define GML_DIGITAL_MODE_CLEAR_MASK     0xffffcfff
+#define GML_SPDIF_FORMAT_CLEAR_MASK     0xfffff01f
+
+
+/*
+ *
+ * Mia sample rate and clock setting constants
+ *
+ */
+
+#define MIA_32000       0x0040
+#define MIA_44100       0x0042
+#define MIA_48000       0x0041
+#define MIA_88200       0x0142
+#define MIA_96000       0x0141
+
+#define MIA_SPDIF       0x00000044
+#define MIA_SPDIF96     0x00000144
+
+#define MIA_MIDI_REV    1       /* Must be Mia rev 1 for MIDI support */
+
+
+/*
+ *
+ * 3G register bits
+ *
+ */
+
+#define E3G_CONVERTER_ENABLE    0x0010
+#define E3G_SPDIF_PRO_MODE      0x0020  /* Professional S/PDIF == 1,
+                                           consumer == 0 */
+#define E3G_SPDIF_SAMPLE_RATE0  0x0040
+#define E3G_SPDIF_SAMPLE_RATE1  0x0080
+#define E3G_SPDIF_TWO_CHANNEL   0x0100  /* 1 == two channels,
+                                           0 == one channel */
+#define E3G_SPDIF_NOT_AUDIO     0x0200
+#define E3G_SPDIF_COPY_PERMIT   0x0400
+#define E3G_SPDIF_24_BIT        0x0800  /* 1 == 24 bit, 0 == 20 bit */
+#define E3G_DOUBLE_SPEED_MODE   0x4000  /* 1 == double speed,
+                                           0 == single speed */
+#define E3G_PHANTOM_POWER       0x8000  /* 1 == phantom power on,
+                                           0 == phantom power off */
+
+#define E3G_96KHZ               (0x0 | E3G_DOUBLE_SPEED_MODE)
+#define E3G_88KHZ               (0x1 | E3G_DOUBLE_SPEED_MODE)
+#define E3G_48KHZ               0x2
+#define E3G_44KHZ               0x3
+#define E3G_32KHZ               0x4
+#define E3G_22KHZ               0x5
+#define E3G_16KHZ               0x6
+#define E3G_11KHZ               0x7
+#define E3G_8KHZ                0x8
+#define E3G_SPDIF_CLOCK         0x9
+#define E3G_ADAT_CLOCK          0xA
+#define E3G_WORD_CLOCK          0xB
+#define E3G_CONTINUOUS_CLOCK    0xE
+
+#define E3G_ADAT_MODE           0x1000
+#define E3G_SPDIF_OPTICAL_MODE  0x2000
+
+#define E3G_CLOCK_CLEAR_MASK            0xbfffbff0
+#define E3G_DIGITAL_MODE_CLEAR_MASK     0xffffcfff
+#define E3G_SPDIF_FORMAT_CLEAR_MASK     0xfffff01f
+
+/* Clock detect bits reported by the DSP */
+#define E3G_CLOCK_DETECT_BIT_WORD96     0x0001
+#define E3G_CLOCK_DETECT_BIT_WORD48     0x0002
+#define E3G_CLOCK_DETECT_BIT_SPDIF48    0x0004
+#define E3G_CLOCK_DETECT_BIT_ADAT       0x0004
+#define E3G_CLOCK_DETECT_BIT_SPDIF96    0x0008
+#define E3G_CLOCK_DETECT_BIT_WORD       (E3G_CLOCK_DETECT_BIT_WORD96|E3G_CLOCK_DETECT_BIT_WORD48)
+#define E3G_CLOCK_DETECT_BIT_SPDIF      (E3G_CLOCK_DETECT_BIT_SPDIF48|E3G_CLOCK_DETECT_BIT_SPDIF96)
+
+/* Frequency control register */
+#define E3G_MAGIC_NUMBER                677376000
+#define E3G_FREQ_REG_DEFAULT            (E3G_MAGIC_NUMBER / 48000 - 2)
+#define E3G_FREQ_REG_MAX                0xffff
+
+/* 3G external box types */
+#define E3G_GINA3G_BOX_TYPE             0x00
+#define E3G_LAYLA3G_BOX_TYPE            0x10
+#define E3G_ASIC_NOT_LOADED             0xffff
+#define E3G_BOX_TYPE_MASK               0xf0
+
+#define EXT_3GBOX_NC                    0x01
+#define EXT_3GBOX_NOT_SET               0x02
+
+
+/*
+ *
+ * Gina20 & Layla20 have input gain controls for the analog inputs;
+ * this is the magic number for the hardware that gives you 0 dB at -10.
+ *
+ */
+
+#define GL20_INPUT_GAIN_MAGIC_NUMBER    0xC8
+
+
+/*
+ *
+ * Defines how much time must pass between DSP load attempts
+ *
+ */
+
+#define DSP_LOAD_ATTEMPT_PERIOD         1000000L        /* One second */
+
+
+/*
+ *
+ * Size of arrays for the comm page.  MAX_PLAY_TAPS and MAX_REC_TAPS are
+ * no longer used, but the sizes must still be right for the DSP to see
+ * the comm page correctly.
+ *
+ */
+
+#define MONITOR_ARRAY_SIZE      0x180
+#define VMIXER_ARRAY_SIZE       0x40
+#define MIDI_OUT_BUFFER_SIZE    32
+#define MIDI_IN_BUFFER_SIZE     256
+#define MAX_PLAY_TAPS           168
+#define MAX_REC_TAPS            192
+#define DSP_MIDI_OUT_FIFO_SIZE  64
+
+
+/* sg_entry is a single entry for the scatter-gather list.  The array of struct
+sg_entry struct is read by the DSP, so all values must be little-endian. */
+
+#define MAX_SGLIST_ENTRIES 512
+
+struct sg_entry {
+        u32 addr;
+        u32 size;
+};
+
+
+/****************************************************************************
+
+  The comm page.  This structure is read and written by the DSP; the
+  DSP code is a firm believer in the byte offsets written in the comments
+  at the end of each line.  This structure should not be changed.
+
+  Any reads from or writes to this structure should be in little-endian format.
+
+ ****************************************************************************/
+
+struct comm_page {              /*                              Base    Length*/
+        u32 comm_size;          /* size of this object          0x000   4 */
+        u32 flags;              /* See Appendix A below         0x004   4 */
+        u32 unused;             /* Unused entry                 0x008   4 */
+        u32 sample_rate;        /* Card sample rate in Hz       0x00c   4 */
+        volatile u32 handshake; /* DSP command handshake        0x010   4 */
+        u32 cmd_start;          /* Chs. to start mask           0x014   4 */
+        u32 cmd_stop;           /* Chs. to stop mask            0x018   4 */
+        u32 cmd_reset;          /* Chs. to reset mask           0x01c   4 */
+        u16 audio_format[DSP_MAXPIPES]; /* Chs. audio format    0x020   32*2 */
+        struct sg_entry sglist_addr[DSP_MAXPIPES];
+                                /* Chs. Physical sglist addrs   0x060   32*8 */
+        volatile u32 position[DSP_MAXPIPES];
+                                /* Positions for ea. ch.        0x160   32*4 */
+        volatile s8 vu_meter[DSP_MAXPIPES];
+                                /* VU meters                    0x1e0   32*1 */
+        volatile s8 peak_meter[DSP_MAXPIPES];
+                                /* Peak meters                  0x200   32*1 */
+        s8 line_out_level[DSP_MAXAUDIOOUTPUTS];
+                                /* Output gain                  0x220   16*1 */
+        s8 line_in_level[DSP_MAXAUDIOINPUTS];
+                                /* Input gain                   0x230   16*1 */
+        s8 monitors[MONITOR_ARRAY_SIZE];
+                                /* Monitor map                  0x240   0x180 */
+        u32 play_coeff[MAX_PLAY_TAPS];
+                        /* Gina/Darla play filters - obsolete   0x3c0   168*4 */
+        u32 rec_coeff[MAX_REC_TAPS];
+                        /* Gina/Darla record filters - obsolete 0x660   192*4 */
+        volatile u16 midi_input[MIDI_IN_BUFFER_SIZE];
+                        /* MIDI input data transfer buffer      0x960   256*2 */
+        u8 gd_clock_state;      /* Chg Gina/Darla clock state   0xb60   1 */
+        u8 gd_spdif_status;     /* Chg. Gina/Darla S/PDIF state 0xb61   1 */
+        u8 gd_resampler_state;  /* Should always be 3           0xb62   1 */
+        u8 filler2;             /*                              0xb63   1 */
+        u32 nominal_level_mask; /* -10 level enable mask        0xb64   4 */
+        u16 input_clock;        /* Chg. Input clock state       0xb68   2 */
+        u16 output_clock;       /* Chg. Output clock state      0xb6a   2 */
+        volatile u32 status_clocks;
+                                /* Current Input clock state    0xb6c   4 */
+        u32 ext_box_status;     /* External box status          0xb70   4 */
+        u32 cmd_add_buffer;     /* Pipes to add (obsolete)      0xb74   4 */
+        volatile u32 midi_out_free_count;
+                        /* # of bytes free in MIDI output FIFO  0xb78   4 */
+        u32 unused2;            /* Cyclic pipes                 0xb7c   4 */
+        u32 control_register;
+                        /* Mona, Gina24, Layla24, 3G ctrl reg   0xb80   4 */
+        u32 e3g_frq_register;   /* 3G frequency register        0xb84   4 */
+        u8 filler[24];          /* filler                       0xb88   24*1 */
+        s8 vmixer[VMIXER_ARRAY_SIZE];
+                                /* Vmixer levels                0xba0   64*1 */
+        u8 midi_output[MIDI_OUT_BUFFER_SIZE];
+                                /* MIDI output data             0xbe0   32*1 */
+};
+
+#endif /* _ECHO_DSP_ */
diff --git a/sound/pci/echoaudio/echoaudio_gml.c b/sound/pci/echoaudio/echoaudio_gml.c
new file mode 100644
index 000000000000..3aa37e76ebab
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_gml.c
@@ -0,0 +1,198 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+/* These functions are common for Gina24, Layla24 and Mona cards */
+
+
+/* ASIC status check - some cards have one or two ASICs that need to be
+loaded.  Once that load is complete, this function is called to see if
+the load was successful.
+If this load fails, it does not necessarily mean that the hardware is
+defective - the external box may be disconnected or turned off. */
+static int check_asic_status(struct echoaudio *chip)
+{
+        u32 asic_status;
+
+        send_vector(chip, DSP_VC_TEST_ASIC);
+
+        /* The DSP will return a value to indicate whether or not the
+           ASIC is currently loaded */
+        if (read_dsp(chip, &asic_status) < 0) {
+                DE_INIT(("check_asic_status: failed on read_dsp\n"));
+                chip->asic_loaded = FALSE;
+                return -EIO;
+        }
+
+        chip->asic_loaded = (asic_status == ASIC_ALREADY_LOADED);
+        return chip->asic_loaded ? 0 : -EIO;
+}
+
+
+
+/* Most configuration of Gina24, Layla24, or Mona is accomplished by writing
+the control register.  write_control_reg sends the new control register
+value to the DSP. */
+static int write_control_reg(struct echoaudio *chip, u32 value, char force)
+{
+        /* Handle the digital input auto-mute */
+        if (chip->digital_in_automute)
+                value |= GML_DIGITAL_IN_AUTO_MUTE;
+        else
+                value &= ~GML_DIGITAL_IN_AUTO_MUTE;
+
+        DE_ACT(("write_control_reg: 0x%x\n", value));
+
+        /* Write the control register */
+        value = cpu_to_le32(value);
+        if (value != chip->comm_page->control_register || force) {
+                if (wait_handshake(chip))
+                        return -EIO;
+                chip->comm_page->control_register = value;
+                clear_handshake(chip);
+                return send_vector(chip, DSP_VC_WRITE_CONTROL_REG);
+        }
+        return 0;
+}
+
+
+
+/* Gina24, Layla24, and Mona support digital input auto-mute.  If the digital
+input auto-mute is enabled, the DSP will only enable the digital inputs if
+the card is syncing to a valid clock on the ADAT or S/PDIF inputs.
+If the auto-mute is disabled, the digital inputs are enabled regardless of
+what the input clock is set or what is connected. */
+static int set_input_auto_mute(struct echoaudio *chip, int automute)
+{
+        DE_ACT(("set_input_auto_mute %d\n", automute));
+
+        chip->digital_in_automute = automute;
+
+        /* Re-set the input clock to the current value - indirectly causes
+        the auto-mute flag to be sent to the DSP */
+        return set_input_clock(chip, chip->input_clock);
+}
+
+
+
+/* S/PDIF coax / S/PDIF optical / ADAT - switch */
+static int set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+        u8 previous_mode;
+        int err, i, o;
+
+        if (chip->bad_board)
+                return -EIO;
+
+        /* All audio channels must be closed before changing the digital mode */
+        snd_assert(!chip->pipe_alloc_mask, return -EAGAIN);
+
+        snd_assert(chip->digital_modes & (1 << mode), return -EINVAL);
+
+        previous_mode = chip->digital_mode;
+        err = dsp_set_digital_mode(chip, mode);
+
+        /* If we successfully changed the digital mode from or to ADAT,
+           then make sure all output, input and monitor levels are
+           updated by the DSP comm object. */
+        if (err >= 0 && previous_mode != mode &&
+            (previous_mode == DIGITAL_MODE_ADAT || mode == DIGITAL_MODE_ADAT)) {
+                spin_lock_irq(&chip->lock);
+                for (o = 0; o < num_busses_out(chip); o++)
+                        for (i = 0; i < num_busses_in(chip); i++)
+                                set_monitor_gain(chip, o, i,
+                                                 chip->monitor_gain[o][i]);
+
+#ifdef ECHOCARD_HAS_INPUT_GAIN
+                for (i = 0; i < num_busses_in(chip); i++)
+                        set_input_gain(chip, i, chip->input_gain[i]);
+                update_input_line_level(chip);
+#endif
+
+                for (o = 0; o < num_busses_out(chip); o++)
+                        set_output_gain(chip, o, chip->output_gain[o]);
+                update_output_line_level(chip);
+                spin_unlock_irq(&chip->lock);
+        }
+
+        return err;
+}
+
+
+
+/* Set the S/PDIF output format */
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+        u32 control_reg;
+        int err;
+
+        /* Clear the current S/PDIF flags */
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        control_reg &= GML_SPDIF_FORMAT_CLEAR_MASK;
+
+        /* Set the new S/PDIF flags depending on the mode */
+        control_reg |= GML_SPDIF_TWO_CHANNEL | GML_SPDIF_24_BIT |
+                GML_SPDIF_COPY_PERMIT;
+        if (prof) {
+                /* Professional mode */
+                control_reg |= GML_SPDIF_PRO_MODE;
+
+                switch (chip->sample_rate) {
+                case 32000:
+                        control_reg |= GML_SPDIF_SAMPLE_RATE0 |
+                                GML_SPDIF_SAMPLE_RATE1;
+                        break;
+                case 44100:
+                        control_reg |= GML_SPDIF_SAMPLE_RATE0;
+                        break;
+                case 48000:
+                        control_reg |= GML_SPDIF_SAMPLE_RATE1;
+                        break;
+                }
+        } else {
+                /* Consumer mode */
+                switch (chip->sample_rate) {
+                case 32000:
+                        control_reg |= GML_SPDIF_SAMPLE_RATE0 |
+                                GML_SPDIF_SAMPLE_RATE1;
+                        break;
+                case 48000:
+                        control_reg |= GML_SPDIF_SAMPLE_RATE1;
+                        break;
+                }
+        }
+
+        if ((err = write_control_reg(chip, control_reg, FALSE)))
+                return err;
+        chip->professional_spdif = prof;
+        DE_ACT(("set_professional_spdif to %s\n",
+                prof ? "Professional" : "Consumer"));
+        return 0;
+}
diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c
new file mode 100644
index 000000000000..29d6d12f80ca
--- /dev/null
+++ b/sound/pci/echoaudio/gina20.c
@@ -0,0 +1,103 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHOGALS_FAMILY
+#define ECHOCARD_GINA20
+#define ECHOCARD_NAME "Gina20"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_INPUT_GAIN
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT       FALSE
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 8 */
+#define PX_DIGITAL_OUT  8       /* 2 */
+#define PX_ANALOG_IN    10      /* 2 */
+#define PX_DIGITAL_IN   12      /* 2 */
+#define PX_NUM          14
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 8 */
+#define BX_DIGITAL_OUT  8       /* 2 */
+#define BX_ANALOG_IN    10      /* 2 */
+#define BX_DIGITAL_IN   12      /* 2 */
+#define BX_NUM          14
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_GINA20_DSP   0
+
+static const struct firmware card_fw[] = {
+        {0, "gina20_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x1801, 0xECC0, 0x0020, 0, 0, 0},      /* DSP 56301 Gina20 rev.0 */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+        .rate_min = 44100,
+        .rate_max = 48000,
+        .channels_min = 1,
+        .channels_max = 2,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+        /* One page (4k) contains 512 instructions. I don't know if the hw
+        supports lists longer than this. In this case periods_max=220 is a
+        safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "gina20_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c
new file mode 100644
index 000000000000..2757c8960843
--- /dev/null
+++ b/sound/pci/echoaudio/gina20_dsp.c
@@ -0,0 +1,215 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int update_flags(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Gina20\n"));
+        snd_assert((subdevice_id & 0xfff0) == GINA20, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_GINA20_DSP];
+        chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
+        chip->clock_state = GD_CLOCK_UNDEF;
+        /* Since this card has no ASIC, mark it as loaded so everything
+           works OK */
+        chip->asic_loaded = TRUE;
+        chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
+                ECHO_CLOCK_BIT_SPDIF;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+
+        err = set_professional_spdif(chip, TRUE);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        u32 clocks_from_dsp, clock_bits;
+
+        /* Map the DSP clock detect bits to the generic driver clock
+           detect bits */
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+        if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
+                clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+        return clock_bits;
+}
+
+
+
+/* The Gina20 has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+        return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u8 clock_state, spdif_status;
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        switch (rate) {
+        case 44100:
+                clock_state = GD_CLOCK_44;
+                spdif_status = GD_SPDIF_STATUS_44;
+                break;
+        case 48000:
+                clock_state = GD_CLOCK_48;
+                spdif_status = GD_SPDIF_STATUS_48;
+                break;
+        default:
+                clock_state = GD_CLOCK_NOCHANGE;
+                spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+                break;
+        }
+
+        if (chip->clock_state == clock_state)
+                clock_state = GD_CLOCK_NOCHANGE;
+        if (spdif_status == chip->spdif_status)
+                spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+
+        chip->comm_page->sample_rate = cpu_to_le32(rate);
+        chip->comm_page->gd_clock_state = clock_state;
+        chip->comm_page->gd_spdif_status = spdif_status;
+        chip->comm_page->gd_resampler_state = 3;        /* magic number - should always be 3 */
+
+        /* Save the new audio state if it changed */
+        if (clock_state != GD_CLOCK_NOCHANGE)
+                chip->clock_state = clock_state;
+        if (spdif_status != GD_SPDIF_STATUS_NOCHANGE)
+                chip->spdif_status = spdif_status;
+        chip->sample_rate = rate;
+
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+        DE_ACT(("set_input_clock:\n"));
+
+        switch (clock) {
+        case ECHO_CLOCK_INTERNAL:
+                /* Reset the audio state to unknown (just in case) */
+                chip->clock_state = GD_CLOCK_UNDEF;
+                chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
+                set_sample_rate(chip, chip->sample_rate);
+                chip->input_clock = clock;
+                DE_ACT(("Set Gina clock to INTERNAL\n"));
+                break;
+        case ECHO_CLOCK_SPDIF:
+                chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN;
+                chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_NOCHANGE;
+                clear_handshake(chip);
+                send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
+                chip->clock_state = GD_CLOCK_SPDIFIN;
+                DE_ACT(("Set Gina20 clock to SPDIF\n"));
+                chip->input_clock = clock;
+                break;
+        default:
+                return -EINVAL;
+        }
+
+        return 0;
+}
+
+
+
+/* Set input bus gain (one unit is 0.5dB !) */
+static int set_input_gain(struct echoaudio *chip, u16 input, int gain)
+{
+        snd_assert(input < num_busses_in(chip), return -EINVAL);
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->input_gain[input] = gain;
+        gain += GL20_INPUT_GAIN_MAGIC_NUMBER;
+        chip->comm_page->line_in_level[input] = gain;
+        return 0;
+}
+
+
+
+/* Tell the DSP to reread the flags from the comm page */
+static int update_flags(struct echoaudio *chip)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+        DE_ACT(("set_professional_spdif %d\n", prof));
+        if (prof)
+                chip->comm_page->flags |=
+                        __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+        else
+                chip->comm_page->flags &=
+                        ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+        chip->professional_spdif = prof;
+        return update_flags(chip);
+}
diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c
new file mode 100644
index 000000000000..e464d720d0bd
--- /dev/null
+++ b/sound/pci/echoaudio/gina24.c
@@ -0,0 +1,123 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO24_FAMILY
+#define ECHOCARD_GINA24
+#define ECHOCARD_NAME "Gina24"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT       6
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 8 */
+#define PX_DIGITAL_OUT  8       /* 8 */
+#define PX_ANALOG_IN    16      /* 2 */
+#define PX_DIGITAL_IN   18      /* 8 */
+#define PX_NUM          26
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 8 */
+#define BX_DIGITAL_OUT  8       /* 8 */
+#define BX_ANALOG_IN    16      /* 2 */
+#define BX_DIGITAL_IN   18      /* 8 */
+#define BX_NUM          26
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_361_LOADER           0
+#define FW_GINA24_301_DSP       1
+#define FW_GINA24_361_DSP       2
+#define FW_GINA24_301_ASIC      3
+#define FW_GINA24_361_ASIC      4
+
+static const struct firmware card_fw[] = {
+        {0, "loader_dsp.fw"},
+        {0, "gina24_301_dsp.fw"},
+        {0, "gina24_361_dsp.fw"},
+        {0, "gina24_301_asic.fw"},
+        {0, "gina24_361_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x1801, 0xECC0, 0x0050, 0, 0, 0},      /* DSP 56301 Gina24 rev.0 */
+        {0x1057, 0x1801, 0xECC0, 0x0051, 0, 0, 0},      /* DSP 56301 Gina24 rev.1 */
+        {0x1057, 0x3410, 0xECC0, 0x0050, 0, 0, 0},      /* DSP 56361 Gina24 rev.0 */
+        {0x1057, 0x3410, 0xECC0, 0x0051, 0, 0, 0},      /* DSP 56361 Gina24 rev.1 */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates =        SNDRV_PCM_RATE_8000_48000 |
+                        SNDRV_PCM_RATE_88200 |
+                        SNDRV_PCM_RATE_96000,
+        .rate_min = 8000,
+        .rate_max = 96000,
+        .channels_min = 1,
+        .channels_max = 8,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+        /* One page (4k) contains 512 instructions. I don't know if the hw
+        supports lists longer than this. In this case periods_max=220 is a
+        safe limit to make sure the list never exceeds 512 instructions.
+        220 ~= (512 - 1 - (BUFFER_BYTES_MAX / PAGE_SIZE)) / 2 */
+};
+
+#include "gina24_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio_gml.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c
new file mode 100644
index 000000000000..144fc567becf
--- /dev/null
+++ b/sound/pci/echoaudio/gina24_dsp.c
@@ -0,0 +1,346 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int write_control_reg(struct echoaudio *chip, u32 value, char force);
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int set_digital_mode(struct echoaudio *chip, u8 mode);
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+                             const struct firmware *asic);
+static int check_asic_status(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Gina24\n"));
+        snd_assert((subdevice_id & 0xfff0) == GINA24, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->input_clock_types =
+                ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
+                ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 |
+                ECHO_CLOCK_BIT_ADAT;
+        chip->professional_spdif = FALSE;
+        chip->digital_in_automute = TRUE;
+        chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
+
+        /* Gina24 comes in both '301 and '361 flavors */
+        if (chip->device_id == DEVICE_ID_56361) {
+                chip->dsp_code_to_load = &card_fw[FW_GINA24_361_DSP];
+                chip->digital_modes =
+                        ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+                        ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+                        ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
+        } else {
+                chip->dsp_code_to_load = &card_fw[FW_GINA24_301_DSP];
+                chip->digital_modes =
+                        ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+                        ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+                        ECHOCAPS_HAS_DIGITAL_MODE_ADAT |
+                        ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_CDROM;
+        }
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+        err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
+        snd_assert(err >= 0, return err);
+        err = set_professional_spdif(chip, TRUE);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        u32 clocks_from_dsp, clock_bits;
+
+        /* Map the DSP clock detect bits to the generic driver clock
+           detect bits */
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+        if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF)
+                clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+        if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ADAT)
+                clock_bits |= ECHO_CLOCK_BIT_ADAT;
+
+        if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ESYNC)
+                clock_bits |= ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96;
+
+        return clock_bits;
+}
+
+
+
+/* Gina24 has an ASIC on the PCI card which must be loaded for anything
+interesting to happen. */
+static int load_asic(struct echoaudio *chip)
+{
+        u32 control_reg;
+        int err;
+        const struct firmware *fw;
+
+        if (chip->asic_loaded)
+                return 1;
+
+        /* Give the DSP a few milliseconds to settle down */
+        mdelay(10);
+
+        /* Pick the correct ASIC for '301 or '361 Gina24 */
+        if (chip->device_id == DEVICE_ID_56361)
+                fw = &card_fw[FW_GINA24_361_ASIC];
+        else
+                fw = &card_fw[FW_GINA24_301_ASIC];
+
+        if ((err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, fw)) < 0)
+                return err;
+
+        chip->asic_code = fw;
+
+        /* Now give the new ASIC a little time to set up */
+        mdelay(10);
+        /* See if it worked */
+        err = check_asic_status(chip);
+
+        /* Set up the control register if the load succeeded -
+           48 kHz, internal clock, S/PDIF RCA mode */
+        if (!err) {
+                control_reg = GML_CONVERTER_ENABLE | GML_48KHZ;
+                err = write_control_reg(chip, control_reg, TRUE);
+        }
+        DE_INIT(("load_asic() done\n"));
+        return err;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u32 control_reg, clock;
+
+        snd_assert(rate < 50000 || chip->digital_mode != DIGITAL_MODE_ADAT,
+                   return -EINVAL);
+
+        /* Only set the clock for internal mode. */
+        if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+                DE_ACT(("set_sample_rate: Cannot set sample rate - "
+                        "clock not set to CLK_CLOCKININTERNAL\n"));
+                /* Save the rate anyhow */
+                chip->comm_page->sample_rate = cpu_to_le32(rate);
+                chip->sample_rate = rate;
+                return 0;
+        }
+
+        clock = 0;
+
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        control_reg &= GML_CLOCK_CLEAR_MASK & GML_SPDIF_RATE_CLEAR_MASK;
+
+        switch (rate) {
+        case 96000:
+                clock = GML_96KHZ;
+                break;
+        case 88200:
+                clock = GML_88KHZ;
+                break;
+        case 48000:
+                clock = GML_48KHZ | GML_SPDIF_SAMPLE_RATE1;
+                break;
+        case 44100:
+                clock = GML_44KHZ;
+                /* Professional mode ? */
+                if (control_reg & GML_SPDIF_PRO_MODE)
+                        clock |= GML_SPDIF_SAMPLE_RATE0;
+                break;
+        case 32000:
+                clock = GML_32KHZ | GML_SPDIF_SAMPLE_RATE0 |
+                        GML_SPDIF_SAMPLE_RATE1;
+                break;
+        case 22050:
+                clock = GML_22KHZ;
+                break;
+        case 16000:
+                clock = GML_16KHZ;
+                break;
+        case 11025:
+                clock = GML_11KHZ;
+                break;
+        case 8000:
+                clock = GML_8KHZ;
+                break;
+        default:
+                DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+                return -EINVAL;
+        }
+
+        control_reg |= clock;
+
+        chip->comm_page->sample_rate = cpu_to_le32(rate);       /* ignored by the DSP */
+        chip->sample_rate = rate;
+        DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+
+        return write_control_reg(chip, control_reg, FALSE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+        u32 control_reg, clocks_from_dsp;
+
+        DE_ACT(("set_input_clock:\n"));
+
+        /* Mask off the clock select bits */
+        control_reg = le32_to_cpu(chip->comm_page->control_register) &
+                GML_CLOCK_CLEAR_MASK;
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        switch (clock) {
+        case ECHO_CLOCK_INTERNAL:
+                DE_ACT(("Set Gina24 clock to INTERNAL\n"));
+                chip->input_clock = ECHO_CLOCK_INTERNAL;
+                return set_sample_rate(chip, chip->sample_rate);
+        case ECHO_CLOCK_SPDIF:
+                if (chip->digital_mode == DIGITAL_MODE_ADAT)
+                        return -EAGAIN;
+                DE_ACT(("Set Gina24 clock to SPDIF\n"));
+                control_reg |= GML_SPDIF_CLOCK;
+                if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
+                        control_reg |= GML_DOUBLE_SPEED_MODE;
+                else
+                        control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                break;
+        case ECHO_CLOCK_ADAT:
+                if (chip->digital_mode != DIGITAL_MODE_ADAT)
+                        return -EAGAIN;
+                DE_ACT(("Set Gina24 clock to ADAT\n"));
+                control_reg |= GML_ADAT_CLOCK;
+                control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                break;
+        case ECHO_CLOCK_ESYNC:
+                DE_ACT(("Set Gina24 clock to ESYNC\n"));
+                control_reg |= GML_ESYNC_CLOCK;
+                control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                break;
+        case ECHO_CLOCK_ESYNC96:
+                DE_ACT(("Set Gina24 clock to ESYNC96\n"));
+                control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE;
+                break;
+        default:
+                DE_ACT(("Input clock 0x%x not supported for Gina24\n", clock));
+                return -EINVAL;
+        }
+
+        chip->input_clock = clock;
+        return write_control_reg(chip, control_reg, TRUE);
+}
+
+
+
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+        u32 control_reg;
+        int err, incompatible_clock;
+
+        /* Set clock to "internal" if it's not compatible with the new mode */
+        incompatible_clock = FALSE;
+        switch (mode) {
+        case DIGITAL_MODE_SPDIF_OPTICAL:
+        case DIGITAL_MODE_SPDIF_CDROM:
+        case DIGITAL_MODE_SPDIF_RCA:
+                if (chip->input_clock == ECHO_CLOCK_ADAT)
+                        incompatible_clock = TRUE;
+                break;
+        case DIGITAL_MODE_ADAT:
+                if (chip->input_clock == ECHO_CLOCK_SPDIF)
+                        incompatible_clock = TRUE;
+                break;
+        default:
+                DE_ACT(("Digital mode not supported: %d\n", mode));
+                return -EINVAL;
+        }
+
+        spin_lock_irq(&chip->lock);
+
+        if (incompatible_clock) {       /* Switch to 48KHz, internal */
+                chip->sample_rate = 48000;
+                set_input_clock(chip, ECHO_CLOCK_INTERNAL);
+        }
+
+        /* Clear the current digital mode */
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        control_reg &= GML_DIGITAL_MODE_CLEAR_MASK;
+
+        /* Tweak the control reg */
+        switch (mode) {
+        case DIGITAL_MODE_SPDIF_OPTICAL:
+                control_reg |= GML_SPDIF_OPTICAL_MODE;
+                break;
+        case DIGITAL_MODE_SPDIF_CDROM:
+                /* '361 Gina24 cards do not have the S/PDIF CD-ROM mode */
+                if (chip->device_id == DEVICE_ID_56301)
+                        control_reg |= GML_SPDIF_CDROM_MODE;
+                break;
+        case DIGITAL_MODE_SPDIF_RCA:
+                /* GML_SPDIF_OPTICAL_MODE bit cleared */
+                break;
+        case DIGITAL_MODE_ADAT:
+                control_reg |= GML_ADAT_MODE;
+                control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                break;
+        }
+
+        err = write_control_reg(chip, control_reg, TRUE);
+        spin_unlock_irq(&chip->lock);
+        if (err < 0)
+                return err;
+        chip->digital_mode = mode;
+
+        DE_ACT(("set_digital_mode to %d\n", chip->digital_mode));
+        return incompatible_clock;
+}
diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c
new file mode 100644
index 000000000000..bfd2467099ac
--- /dev/null
+++ b/sound/pci/echoaudio/indigo.c
@@ -0,0 +1,104 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO
+#define ECHOCARD_NAME "Indigo"
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 8 */
+#define PX_DIGITAL_OUT  8       /* 0 */
+#define PX_ANALOG_IN    8       /* 0 */
+#define PX_DIGITAL_IN   8       /* 0 */
+#define PX_NUM          8
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 2 */
+#define BX_DIGITAL_OUT  2       /* 0 */
+#define BX_ANALOG_IN    2       /* 0 */
+#define BX_DIGITAL_IN   2       /* 0 */
+#define BX_NUM          2
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_361_LOADER   0
+#define FW_INDIGO_DSP   1
+
+static const struct firmware card_fw[] = {
+        {0, "loader_dsp.fw"},
+        {0, "indigo_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x3410, 0xECC0, 0x0090, 0, 0, 0},      /* Indigo */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates =        SNDRV_PCM_RATE_32000 |
+                        SNDRV_PCM_RATE_44100 |
+                        SNDRV_PCM_RATE_48000 |
+                        SNDRV_PCM_RATE_88200 |
+                        SNDRV_PCM_RATE_96000,
+        .rate_min = 32000,
+        .rate_max = 96000,
+        .channels_min = 1,
+        .channels_max = 8,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+};
+
+#include "indigo_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+
diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c
new file mode 100644
index 000000000000..d6ac7734609e
--- /dev/null
+++ b/sound/pci/echoaudio/indigo_dsp.c
@@ -0,0 +1,170 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+                           int gain);
+static int update_vmixer_level(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Indigo\n"));
+        snd_assert((subdevice_id & 0xfff0) == INDIGO, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_INDIGO_DSP];
+        /* Since this card has no ASIC, mark it as loaded so everything
+           works OK */
+        chip->asic_loaded = TRUE;
+        chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+
+        /* Default routing of the virtual channels: all vchannels are routed
+        to the stereo output */
+        set_vmixer_gain(chip, 0, 0, 0);
+        set_vmixer_gain(chip, 1, 1, 0);
+        set_vmixer_gain(chip, 0, 2, 0);
+        set_vmixer_gain(chip, 1, 3, 0);
+        set_vmixer_gain(chip, 0, 4, 0);
+        set_vmixer_gain(chip, 1, 5, 0);
+        set_vmixer_gain(chip, 0, 6, 0);
+        set_vmixer_gain(chip, 1, 7, 0);
+        err = update_vmixer_level(chip);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The Indigo has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+        return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u32 control_reg;
+
+        switch (rate) {
+        case 96000:
+                control_reg = MIA_96000;
+                break;
+        case 88200:
+                control_reg = MIA_88200;
+                break;
+        case 48000:
+                control_reg = MIA_48000;
+                break;
+        case 44100:
+                control_reg = MIA_44100;
+                break;
+        case 32000:
+                control_reg = MIA_32000;
+                break;
+        default:
+                DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+                return -EINVAL;
+        }
+
+        /* Set the control register if it has changed */
+        if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
+                if (wait_handshake(chip))
+                        return -EIO;
+
+                chip->comm_page->sample_rate = cpu_to_le32(rate);       /* ignored by the DSP */
+                chip->comm_page->control_register = cpu_to_le32(control_reg);
+                chip->sample_rate = rate;
+
+                clear_handshake(chip);
+                return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+        }
+        return 0;
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+                           int gain)
+{
+        int index;
+
+        snd_assert(pipe < num_pipes_out(chip) &&
+                   output < num_busses_out(chip), return -EINVAL);
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->vmixer_gain[output][pipe] = gain;
+        index = output * num_pipes_out(chip) + pipe;
+        chip->comm_page->vmixer[index] = gain;
+
+        DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+        return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c
new file mode 100644
index 000000000000..8ed7ff1fd875
--- /dev/null
+++ b/sound/pci/echoaudio/indigodj.c
@@ -0,0 +1,104 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO_DJ
+#define ECHOCARD_NAME "Indigo DJ"
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 8 */
+#define PX_DIGITAL_OUT  8       /* 0 */
+#define PX_ANALOG_IN    8       /* 0 */
+#define PX_DIGITAL_IN   8       /* 0 */
+#define PX_NUM          8
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 4 */
+#define BX_DIGITAL_OUT  4       /* 0 */
+#define BX_ANALOG_IN    4       /* 0 */
+#define BX_DIGITAL_IN   4       /* 0 */
+#define BX_NUM          4
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_361_LOADER           0
+#define FW_INDIGO_DJ_DSP        1
+
+static const struct firmware card_fw[] = {
+        {0, "loader_dsp.fw"},
+        {0, "indigo_dj_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0},      /* Indigo DJ*/
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates =        SNDRV_PCM_RATE_32000 |
+                        SNDRV_PCM_RATE_44100 |
+                        SNDRV_PCM_RATE_48000 |
+                        SNDRV_PCM_RATE_88200 |
+                        SNDRV_PCM_RATE_96000,
+        .rate_min = 32000,
+        .rate_max = 96000,
+        .channels_min = 1,
+        .channels_max = 4,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+};
+
+#include "indigodj_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+
diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c
new file mode 100644
index 000000000000..500e150b49fc
--- /dev/null
+++ b/sound/pci/echoaudio/indigodj_dsp.c
@@ -0,0 +1,170 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+                           int gain);
+static int update_vmixer_level(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Indigo DJ\n"));
+        snd_assert((subdevice_id & 0xfff0) == INDIGO_DJ, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP];
+        /* Since this card has no ASIC, mark it as loaded so everything
+           works OK */
+        chip->asic_loaded = TRUE;
+        chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+
+        /* Default routing of the virtual channels: vchannels 0-3 and
+        vchannels 4-7 are routed to real channels 0-4 */
+        set_vmixer_gain(chip, 0, 0, 0);
+        set_vmixer_gain(chip, 1, 1, 0);
+        set_vmixer_gain(chip, 2, 2, 0);
+        set_vmixer_gain(chip, 3, 3, 0);
+        set_vmixer_gain(chip, 0, 4, 0);
+        set_vmixer_gain(chip, 1, 5, 0);
+        set_vmixer_gain(chip, 2, 6, 0);
+        set_vmixer_gain(chip, 3, 7, 0);
+        err = update_vmixer_level(chip);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The IndigoDJ has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+        return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u32 control_reg;
+
+        switch (rate) {
+        case 96000:
+                control_reg = MIA_96000;
+                break;
+        case 88200:
+                control_reg = MIA_88200;
+                break;
+        case 48000:
+                control_reg = MIA_48000;
+                break;
+        case 44100:
+                control_reg = MIA_44100;
+                break;
+        case 32000:
+                control_reg = MIA_32000;
+                break;
+        default:
+                DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+                return -EINVAL;
+        }
+
+        /* Set the control register if it has changed */
+        if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
+                if (wait_handshake(chip))
+                        return -EIO;
+
+                chip->comm_page->sample_rate = cpu_to_le32(rate);       /* ignored by the DSP */
+                chip->comm_page->control_register = cpu_to_le32(control_reg);
+                chip->sample_rate = rate;
+
+                clear_handshake(chip);
+                return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+        }
+        return 0;
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+                           int gain)
+{
+        int index;
+
+        snd_assert(pipe < num_pipes_out(chip) &&
+                   output < num_busses_out(chip), return -EINVAL);
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->vmixer_gain[output][pipe] = gain;
+        index = output * num_pipes_out(chip) + pipe;
+        chip->comm_page->vmixer[index] = gain;
+
+        DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+        return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c
new file mode 100644
index 000000000000..a8788e959171
--- /dev/null
+++ b/sound/pci/echoaudio/indigoio.c
@@ -0,0 +1,105 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO_IO
+#define ECHOCARD_NAME "Indigo IO"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 8 */
+#define PX_DIGITAL_OUT  8       /* 0 */
+#define PX_ANALOG_IN    8       /* 2 */
+#define PX_DIGITAL_IN   10      /* 0 */
+#define PX_NUM          10
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 2 */
+#define BX_DIGITAL_OUT  2       /* 0 */
+#define BX_ANALOG_IN    2       /* 2 */
+#define BX_DIGITAL_IN   4       /* 0 */
+#define BX_NUM          4
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_361_LOADER           0
+#define FW_INDIGO_IO_DSP        1
+
+static const struct firmware card_fw[] = {
+        {0, "loader_dsp.fw"},
+        {0, "indigo_io_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0},      /* Indigo IO*/
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates =        SNDRV_PCM_RATE_32000 |
+                        SNDRV_PCM_RATE_44100 |
+                        SNDRV_PCM_RATE_48000 |
+                        SNDRV_PCM_RATE_88200 |
+                        SNDRV_PCM_RATE_96000,
+        .rate_min = 32000,
+        .rate_max = 96000,
+        .channels_min = 1,
+        .channels_max = 8,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+};
+
+#include "indigoio_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+
diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c
new file mode 100644
index 000000000000..f3ad13d06be0
--- /dev/null
+++ b/sound/pci/echoaudio/indigoio_dsp.c
@@ -0,0 +1,141 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+                           int gain);
+static int update_vmixer_level(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Indigo IO\n"));
+        snd_assert((subdevice_id & 0xfff0) == INDIGO_IO, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP];
+        /* Since this card has no ASIC, mark it as loaded so everything
+           works OK */
+        chip->asic_loaded = TRUE;
+        chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+
+        /* Default routing of the virtual channels: all vchannels are routed
+        to the stereo output */
+        set_vmixer_gain(chip, 0, 0, 0);
+        set_vmixer_gain(chip, 1, 1, 0);
+        set_vmixer_gain(chip, 0, 2, 0);
+        set_vmixer_gain(chip, 1, 3, 0);
+        set_vmixer_gain(chip, 0, 4, 0);
+        set_vmixer_gain(chip, 1, 5, 0);
+        set_vmixer_gain(chip, 0, 6, 0);
+        set_vmixer_gain(chip, 1, 7, 0);
+        err = update_vmixer_level(chip);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The IndigoIO has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+        return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->sample_rate = rate;
+        chip->comm_page->sample_rate = cpu_to_le32(rate);
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+                           int gain)
+{
+        int index;
+
+        snd_assert(pipe < num_pipes_out(chip) &&
+                   output < num_busses_out(chip), return -EINVAL);
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->vmixer_gain[output][pipe] = gain;
+        index = output * num_pipes_out(chip) + pipe;
+        chip->comm_page->vmixer[index] = gain;
+
+        DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+        return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c
new file mode 100644
index 000000000000..e503d74b3ba9
--- /dev/null
+++ b/sound/pci/echoaudio/layla20.c
@@ -0,0 +1,112 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHOGALS_FAMILY
+#define ECHOCARD_LAYLA20
+#define ECHOCARD_NAME "Layla20"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_INPUT_GAIN
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT       FALSE
+#define ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH
+#define ECHOCARD_HAS_MIDI
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 10 */
+#define PX_DIGITAL_OUT  10      /*  2 */
+#define PX_ANALOG_IN    12      /*  8 */
+#define PX_DIGITAL_IN   20      /*  2 */
+#define PX_NUM          22
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 10 */
+#define BX_DIGITAL_OUT  10      /*  2 */
+#define BX_ANALOG_IN    12      /*  8 */
+#define BX_DIGITAL_IN   20      /*  2 */
+#define BX_NUM          22
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_LAYLA20_DSP  0
+#define FW_LAYLA20_ASIC 1
+
+static const struct firmware card_fw[] = {
+        {0, "layla20_dsp.fw"},
+        {0, "layla20_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x1801, 0xECC0, 0x0030, 0, 0, 0},      /* DSP 56301 Layla20 rev.0 */
+        {0x1057, 0x1801, 0xECC0, 0x0031, 0, 0, 0},      /* DSP 56301 Layla20 rev.1 */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS,
+        .rate_min = 8000,
+        .rate_max = 50000,
+        .channels_min = 1,
+        .channels_max = 10,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+        /* One page (4k) contains 512 instructions. I don't know if the hw
+        supports lists longer than this. In this case periods_max=220 is a
+        safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+#include "layla20_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+#include "midi.c"
diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c
new file mode 100644
index 000000000000..990c9a60a0a8
--- /dev/null
+++ b/sound/pci/echoaudio/layla20_dsp.c
@@ -0,0 +1,290 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int read_dsp(struct echoaudio *chip, u32 *data);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+                             const struct firmware *asic);
+static int check_asic_status(struct echoaudio *chip);
+static int update_flags(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Layla20\n"));
+        snd_assert((subdevice_id & 0xfff0) == LAYLA20, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->has_midi = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_LAYLA20_DSP];
+        chip->input_clock_types =
+                ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
+                ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER;
+        chip->output_clock_types =
+                ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+
+        err = set_professional_spdif(chip, TRUE);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        u32 clocks_from_dsp, clock_bits;
+
+        /* Map the DSP clock detect bits to the generic driver clock detect bits */
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+        if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
+                clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+        if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_WORD) {
+                if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SUPER)
+                        clock_bits |= ECHO_CLOCK_BIT_SUPER;
+                else
+                        clock_bits |= ECHO_CLOCK_BIT_WORD;
+        }
+
+        return clock_bits;
+}
+
+
+
+/* ASIC status check - some cards have one or two ASICs that need to be
+loaded.  Once that load is complete, this function is called to see if
+the load was successful.
+If this load fails, it does not necessarily mean that the hardware is
+defective - the external box may be disconnected or turned off.
+This routine sometimes fails for Layla20; for Layla20, the loop runs
+5 times and succeeds if it wins on three of the loops. */
+static int check_asic_status(struct echoaudio *chip)
+{
+        u32 asic_status;
+        int goodcnt, i;
+
+        chip->asic_loaded = FALSE;
+        for (i = goodcnt = 0; i < 5; i++) {
+                send_vector(chip, DSP_VC_TEST_ASIC);
+
+                /* The DSP will return a value to indicate whether or not
+                   the ASIC is currently loaded */
+                if (read_dsp(chip, &asic_status) < 0) {
+                        DE_ACT(("check_asic_status: failed on read_dsp\n"));
+                        return -EIO;
+                }
+
+                if (asic_status == ASIC_ALREADY_LOADED) {
+                        if (++goodcnt == 3) {
+                                chip->asic_loaded = TRUE;
+                                return 0;
+                        }
+                }
+        }
+        return -EIO;
+}
+
+
+
+/* Layla20 has an ASIC in the external box */
+static int load_asic(struct echoaudio *chip)
+{
+        int err;
+
+        if (chip->asic_loaded)
+                return 0;
+
+        err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA_ASIC,
+                                &card_fw[FW_LAYLA20_ASIC]);
+        if (err < 0)
+                return err;
+
+        /* Check if ASIC is alive and well. */
+        return check_asic_status(chip);
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        snd_assert(rate >= 8000 && rate <= 50000, return -EINVAL);
+
+        /* Only set the clock for internal mode. Do not return failure,
+           simply treat it as a non-event. */
+        if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+                DE_ACT(("set_sample_rate: Cannot set sample rate - "
+                        "clock not set to CLK_CLOCKININTERNAL\n"));
+                chip->comm_page->sample_rate = cpu_to_le32(rate);
+                chip->sample_rate = rate;
+                return 0;
+        }
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        DE_ACT(("set_sample_rate(%d)\n", rate));
+        chip->sample_rate = rate;
+        chip->comm_page->sample_rate = cpu_to_le32(rate);
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_SET_LAYLA_SAMPLE_RATE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock_source)
+{
+        u16 clock;
+        u32 rate;
+
+        DE_ACT(("set_input_clock:\n"));
+        rate = 0;
+        switch (clock_source) {
+        case ECHO_CLOCK_INTERNAL:
+                DE_ACT(("Set Layla20 clock to INTERNAL\n"));
+                rate = chip->sample_rate;
+                clock = LAYLA20_CLOCK_INTERNAL;
+                break;
+        case ECHO_CLOCK_SPDIF:
+                DE_ACT(("Set Layla20 clock to SPDIF\n"));
+                clock = LAYLA20_CLOCK_SPDIF;
+                break;
+        case ECHO_CLOCK_WORD:
+                DE_ACT(("Set Layla20 clock to WORD\n"));
+                clock = LAYLA20_CLOCK_WORD;
+                break;
+        case ECHO_CLOCK_SUPER:
+                DE_ACT(("Set Layla20 clock to SUPER\n"));
+                clock = LAYLA20_CLOCK_SUPER;
+                break;
+        default:
+                DE_ACT(("Input clock 0x%x not supported for Layla24\n",
+                        clock_source));
+                return -EINVAL;
+        }
+        chip->input_clock = clock_source;
+
+        chip->comm_page->input_clock = cpu_to_le16(clock);
+        clear_handshake(chip);
+        send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+
+        if (rate)
+                set_sample_rate(chip, rate);
+
+        return 0;
+}
+
+
+
+static int set_output_clock(struct echoaudio *chip, u16 clock)
+{
+        DE_ACT(("set_output_clock: %d\n", clock));
+        switch (clock) {
+        case ECHO_CLOCK_SUPER:
+                clock = LAYLA20_OUTPUT_CLOCK_SUPER;
+                break;
+        case ECHO_CLOCK_WORD:
+                clock = LAYLA20_OUTPUT_CLOCK_WORD;
+                break;
+        default:
+                DE_ACT(("set_output_clock wrong clock\n"));
+                return -EINVAL;
+        }
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->comm_page->output_clock = cpu_to_le16(clock);
+        chip->output_clock = clock;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+}
+
+
+
+/* Set input bus gain (one unit is 0.5dB !) */
+static int set_input_gain(struct echoaudio *chip, u16 input, int gain)
+{
+        snd_assert(input < num_busses_in(chip), return -EINVAL);
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->input_gain[input] = gain;
+        gain += GL20_INPUT_GAIN_MAGIC_NUMBER;
+        chip->comm_page->line_in_level[input] = gain;
+        return 0;
+}
+
+
+
+/* Tell the DSP to reread the flags from the comm page */
+static int update_flags(struct echoaudio *chip)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+        DE_ACT(("set_professional_spdif %d\n", prof));
+        if (prof)
+                chip->comm_page->flags |=
+                        __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+        else
+                chip->comm_page->flags &=
+                        ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+        chip->professional_spdif = prof;
+        return update_flags(chip);
+}
diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c
new file mode 100644
index 000000000000..d4581fdc841c
--- /dev/null
+++ b/sound/pci/echoaudio/layla24.c
@@ -0,0 +1,121 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO24_FAMILY
+#define ECHOCARD_LAYLA24
+#define ECHOCARD_NAME "Layla24"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT       6
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+#define ECHOCARD_HAS_MIDI
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 8 */
+#define PX_DIGITAL_OUT  8       /* 8 */
+#define PX_ANALOG_IN    16      /* 8 */
+#define PX_DIGITAL_IN   24      /* 8 */
+#define PX_NUM          32
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 8 */
+#define BX_DIGITAL_OUT  8       /* 8 */
+#define BX_ANALOG_IN    16      /* 8 */
+#define BX_DIGITAL_IN   24      /* 8 */
+#define BX_NUM          32
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_361_LOADER           0
+#define FW_LAYLA24_DSP          1
+#define FW_LAYLA24_1_ASIC       2
+#define FW_LAYLA24_2A_ASIC      3
+#define FW_LAYLA24_2S_ASIC      4
+
+static const struct firmware card_fw[] = {
+        {0, "loader_dsp.fw"},
+        {0, "layla24_dsp.fw"},
+        {0, "layla24_1_asic.fw"},
+        {0, "layla24_2A_asic.fw"},
+        {0, "layla24_2S_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x3410, 0xECC0, 0x0060, 0, 0, 0},      /* DSP 56361 Layla24 rev.0 */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates =        SNDRV_PCM_RATE_8000_96000,
+        .rate_min = 8000,
+        .rate_max = 100000,
+        .channels_min = 1,
+        .channels_max = 8,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+        /* One page (4k) contains 512 instructions. I don't know if the hw
+        supports lists longer than this. In this case periods_max=220 is a
+        safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "layla24_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio_gml.c"
+#include "echoaudio.c"
+#include "midi.c"
diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c
new file mode 100644
index 000000000000..7ec5b63d0dce
--- /dev/null
+++ b/sound/pci/echoaudio/layla24_dsp.c
@@ -0,0 +1,394 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int write_control_reg(struct echoaudio *chip, u32 value, char force);
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int set_digital_mode(struct echoaudio *chip, u8 mode);
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+                             const struct firmware *asic);
+static int check_asic_status(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Layla24\n"));
+        snd_assert((subdevice_id & 0xfff0) == LAYLA24, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->has_midi = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_LAYLA24_DSP];
+        chip->input_clock_types =
+                ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
+                ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT;
+        chip->digital_modes =
+                ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+                ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+                ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
+        chip->digital_mode =            DIGITAL_MODE_SPDIF_RCA;
+        chip->professional_spdif = FALSE;
+        chip->digital_in_automute = TRUE;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+
+        err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
+        snd_assert(err >= 0, return err);
+        err = set_professional_spdif(chip, TRUE);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        u32 clocks_from_dsp, clock_bits;
+
+        /* Map the DSP clock detect bits to the generic driver clock detect bits */
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+        if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF)
+                clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+        if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ADAT)
+                clock_bits |= ECHO_CLOCK_BIT_ADAT;
+
+        if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD)
+                clock_bits |= ECHO_CLOCK_BIT_WORD;
+
+        return clock_bits;
+}
+
+
+
+/* Layla24 has an ASIC on the PCI card and another ASIC in the external box;
+both need to be loaded. */
+static int load_asic(struct echoaudio *chip)
+{
+        int err;
+
+        if (chip->asic_loaded)
+                return 1;
+
+        DE_INIT(("load_asic\n"));
+
+        /* Give the DSP a few milliseconds to settle down */
+        mdelay(10);
+
+        /* Load the ASIC for the PCI card */
+        err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC,
+                                &card_fw[FW_LAYLA24_1_ASIC]);
+        if (err < 0)
+                return err;
+
+        chip->asic_code = &card_fw[FW_LAYLA24_2S_ASIC];
+
+        /* Now give the new ASIC a little time to set up */
+        mdelay(10);
+
+        /* Do the external one */
+        err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC,
+                                &card_fw[FW_LAYLA24_2S_ASIC]);
+        if (err < 0)
+                return FALSE;
+
+        /* Now give the external ASIC a little time to set up */
+        mdelay(10);
+
+        /* See if it worked */
+        err = check_asic_status(chip);
+
+        /* Set up the control register if the load succeeded -
+           48 kHz, internal clock, S/PDIF RCA mode */
+        if (!err)
+                err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ,
+                                        TRUE);
+        
+        DE_INIT(("load_asic() done\n"));
+        return err;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u32 control_reg, clock, base_rate;
+
+        snd_assert(rate < 50000 || chip->digital_mode != DIGITAL_MODE_ADAT,
+                   return -EINVAL);
+
+        /* Only set the clock for internal mode. */
+        if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+                DE_ACT(("set_sample_rate: Cannot set sample rate - "
+                        "clock not set to CLK_CLOCKININTERNAL\n"));
+                /* Save the rate anyhow */
+                chip->comm_page->sample_rate = cpu_to_le32(rate);
+                chip->sample_rate = rate;
+                return 0;
+        }
+
+        /* Get the control register & clear the appropriate bits */
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        control_reg &= GML_CLOCK_CLEAR_MASK & GML_SPDIF_RATE_CLEAR_MASK;
+
+        clock = 0;
+
+        switch (rate) {
+        case 96000:
+                clock = GML_96KHZ;
+                break;
+        case 88200:
+                clock = GML_88KHZ;
+                break;
+        case 48000:
+                clock = GML_48KHZ | GML_SPDIF_SAMPLE_RATE1;
+                break;
+        case 44100:
+                clock = GML_44KHZ;
+                /* Professional mode */
+                if (control_reg & GML_SPDIF_PRO_MODE)
+                        clock |= GML_SPDIF_SAMPLE_RATE0;
+                break;
+        case 32000:
+                clock = GML_32KHZ | GML_SPDIF_SAMPLE_RATE0 |
+                        GML_SPDIF_SAMPLE_RATE1;
+                break;
+        case 22050:
+                clock = GML_22KHZ;
+                break;
+        case 16000:
+                clock = GML_16KHZ;
+                break;
+        case 11025:
+                clock = GML_11KHZ;
+                break;
+        case 8000:
+                clock = GML_8KHZ;
+                break;
+        default:
+                /* If this is a non-standard rate, then the driver needs to
+                use Layla24's special "continuous frequency" mode */
+                clock = LAYLA24_CONTINUOUS_CLOCK;
+                if (rate > 50000) {
+                        base_rate = rate >> 1;
+                        control_reg |= GML_DOUBLE_SPEED_MODE;
+                } else {
+                        base_rate = rate;
+                }
+
+                if (base_rate < 25000)
+                        base_rate = 25000;
+
+                if (wait_handshake(chip))
+                        return -EIO;
+
+                chip->comm_page->sample_rate =
+                        cpu_to_le32(LAYLA24_MAGIC_NUMBER / base_rate - 2);
+
+                clear_handshake(chip);
+                send_vector(chip, DSP_VC_SET_LAYLA24_FREQUENCY_REG);
+        }
+
+        control_reg |= clock;
+
+        chip->comm_page->sample_rate = cpu_to_le32(rate);       /* ignored by the DSP ? */
+        chip->sample_rate = rate;
+        DE_ACT(("set_sample_rate: %d clock %d\n", rate, control_reg));
+
+        return write_control_reg(chip, control_reg, FALSE);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+        u32 control_reg, clocks_from_dsp;
+
+        /* Mask off the clock select bits */
+        control_reg = le32_to_cpu(chip->comm_page->control_register) &
+                GML_CLOCK_CLEAR_MASK;
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        /* Pick the new clock */
+        switch (clock) {
+        case ECHO_CLOCK_INTERNAL:
+                DE_ACT(("Set Layla24 clock to INTERNAL\n"));
+                chip->input_clock = ECHO_CLOCK_INTERNAL;
+                return set_sample_rate(chip, chip->sample_rate);
+        case ECHO_CLOCK_SPDIF:
+                if (chip->digital_mode == DIGITAL_MODE_ADAT)
+                        return -EAGAIN;
+                control_reg |= GML_SPDIF_CLOCK;
+                /* Layla24 doesn't support 96KHz S/PDIF */
+                control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                DE_ACT(("Set Layla24 clock to SPDIF\n"));
+                break;
+        case ECHO_CLOCK_WORD:
+                control_reg |= GML_WORD_CLOCK;
+                if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96)
+                        control_reg |= GML_DOUBLE_SPEED_MODE;
+                else
+                        control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                DE_ACT(("Set Layla24 clock to WORD\n"));
+                break;
+        case ECHO_CLOCK_ADAT:
+                if (chip->digital_mode != DIGITAL_MODE_ADAT)
+                        return -EAGAIN;
+                control_reg |= GML_ADAT_CLOCK;
+                control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                DE_ACT(("Set Layla24 clock to ADAT\n"));
+                break;
+        default:
+                DE_ACT(("Input clock 0x%x not supported for Layla24\n", clock));
+                return -EINVAL;
+        }
+
+        chip->input_clock = clock;
+        return write_control_reg(chip, control_reg, TRUE);
+}
+
+
+
+/* Depending on what digital mode you want, Layla24 needs different ASICs
+loaded.  This function checks the ASIC needed for the new mode and sees
+if it matches the one already loaded. */
+static int switch_asic(struct echoaudio *chip, const struct firmware *asic)
+{
+        s8 *monitors;
+
+        /*  Check to see if this is already loaded */
+        if (asic != chip->asic_code) {
+                monitors = kmalloc(MONITOR_ARRAY_SIZE, GFP_KERNEL);
+                if (! monitors)
+                        return -ENOMEM;
+
+                memcpy(monitors, chip->comm_page->monitors, MONITOR_ARRAY_SIZE);
+                memset(chip->comm_page->monitors, ECHOGAIN_MUTED,
+                       MONITOR_ARRAY_SIZE);
+
+                /* Load the desired ASIC */
+                if (load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC,
+                                      asic) < 0) {
+                        memcpy(chip->comm_page->monitors, monitors,
+                               MONITOR_ARRAY_SIZE);
+                        kfree(monitors);
+                        return -EIO;
+                }
+                chip->asic_code = asic;
+                memcpy(chip->comm_page->monitors, monitors, MONITOR_ARRAY_SIZE);
+                kfree(monitors);
+        }
+
+        return 0;
+}
+
+
+
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+        u32 control_reg;
+        int err, incompatible_clock;
+        const struct firmware *asic;
+
+        /* Set clock to "internal" if it's not compatible with the new mode */
+        incompatible_clock = FALSE;
+        switch (mode) {
+        case DIGITAL_MODE_SPDIF_OPTICAL:
+        case DIGITAL_MODE_SPDIF_RCA:
+                if (chip->input_clock == ECHO_CLOCK_ADAT)
+                        incompatible_clock = TRUE;
+                asic = &card_fw[FW_LAYLA24_2S_ASIC];
+                break;
+        case DIGITAL_MODE_ADAT:
+                if (chip->input_clock == ECHO_CLOCK_SPDIF)
+                        incompatible_clock = TRUE;
+                asic = &card_fw[FW_LAYLA24_2A_ASIC];
+                break;
+        default:
+                DE_ACT(("Digital mode not supported: %d\n", mode));
+                return -EINVAL;
+        }
+
+        if (incompatible_clock) {       /* Switch to 48KHz, internal */
+                chip->sample_rate = 48000;
+                spin_lock_irq(&chip->lock);
+                set_input_clock(chip, ECHO_CLOCK_INTERNAL);
+                spin_unlock_irq(&chip->lock);
+        }
+
+        /* switch_asic() can sleep */
+        if (switch_asic(chip, asic) < 0)
+                return -EIO;
+
+        spin_lock_irq(&chip->lock);
+
+        /* Tweak the control register */
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        control_reg &= GML_DIGITAL_MODE_CLEAR_MASK;
+
+        switch (mode) {
+        case DIGITAL_MODE_SPDIF_OPTICAL:
+                control_reg |= GML_SPDIF_OPTICAL_MODE;
+                break;
+        case DIGITAL_MODE_SPDIF_RCA:
+                /* GML_SPDIF_OPTICAL_MODE bit cleared */
+                break;
+        case DIGITAL_MODE_ADAT:
+                control_reg |= GML_ADAT_MODE;
+                control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                break;
+        }
+
+        err = write_control_reg(chip, control_reg, TRUE);
+        spin_unlock_irq(&chip->lock);
+        if (err < 0)
+                return err;
+        chip->digital_mode = mode;
+
+        DE_ACT(("set_digital_mode to %d\n", mode));
+        return incompatible_clock;
+}
diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c
new file mode 100644
index 000000000000..be40c64263d2
--- /dev/null
+++ b/sound/pci/echoaudio/mia.c
@@ -0,0 +1,117 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO24_FAMILY
+#define ECHOCARD_MIA
+#define ECHOCARD_NAME "Mia"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT       FALSE
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+#define ECHOCARD_HAS_MIDI
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 8 */
+#define PX_DIGITAL_OUT  8       /* 0 */
+#define PX_ANALOG_IN    8       /* 2 */
+#define PX_DIGITAL_IN   10      /* 2 */
+#define PX_NUM          12
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 2 */
+#define BX_DIGITAL_OUT  2       /* 2 */
+#define BX_ANALOG_IN    4       /* 2 */
+#define BX_DIGITAL_IN   6       /* 2 */
+#define BX_NUM          8
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <sound/rawmidi.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_361_LOADER   0
+#define FW_MIA_DSP      1
+
+static const struct firmware card_fw[] = {
+        {0, "loader_dsp.fw"},
+        {0, "mia_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x3410, 0xECC0, 0x0080, 0, 0, 0},      /* DSP 56361 Mia rev.0 */
+        {0x1057, 0x3410, 0xECC0, 0x0081, 0, 0, 0},      /* DSP 56361 Mia rev.1 */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates =        SNDRV_PCM_RATE_32000 |
+                        SNDRV_PCM_RATE_44100 |
+                        SNDRV_PCM_RATE_48000 |
+                        SNDRV_PCM_RATE_88200 |
+                        SNDRV_PCM_RATE_96000,
+        .rate_min = 8000,
+        .rate_max = 96000,
+        .channels_min = 1,
+        .channels_max = 8,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+        /* One page (4k) contains 512 instructions. I don't know if the hw
+        supports lists longer than this. In this case periods_max=220 is a
+        safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "mia_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+#include "midi.c"
diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c
new file mode 100644
index 000000000000..891c70519096
--- /dev/null
+++ b/sound/pci/echoaudio/mia_dsp.c
@@ -0,0 +1,229 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int update_flags(struct echoaudio *chip);
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+                           int gain);
+static int update_vmixer_level(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Mia\n"));
+        snd_assert((subdevice_id & 0xfff0) == MIA, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->dsp_code_to_load = &card_fw[FW_MIA_DSP];
+        /* Since this card has no ASIC, mark it as loaded so everything
+           works OK */
+        chip->asic_loaded = TRUE;
+        if ((subdevice_id & 0x0000f) == MIA_MIDI_REV)
+                chip->has_midi = TRUE;
+        chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
+                ECHO_CLOCK_BIT_SPDIF;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)))
+                return err;
+
+        /* Default routing of the virtual channels: vchannels 0-3 go to analog
+        outputs and vchannels 4-7 go to S/PDIF outputs */
+        set_vmixer_gain(chip, 0, 0, 0);
+        set_vmixer_gain(chip, 1, 1, 0);
+        set_vmixer_gain(chip, 0, 2, 0);
+        set_vmixer_gain(chip, 1, 3, 0);
+        set_vmixer_gain(chip, 2, 4, 0);
+        set_vmixer_gain(chip, 3, 5, 0);
+        set_vmixer_gain(chip, 2, 6, 0);
+        set_vmixer_gain(chip, 3, 7, 0);
+        err = update_vmixer_level(chip);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        u32 clocks_from_dsp, clock_bits;
+
+        /* Map the DSP clock detect bits to the generic driver clock
+           detect bits */
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+        if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
+                clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+        return clock_bits;
+}
+
+
+
+/* The Mia has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+        return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u32 control_reg;
+
+        switch (rate) {
+        case 96000:
+                control_reg = MIA_96000;
+                break;
+        case 88200:
+                control_reg = MIA_88200;
+                break;
+        case 48000:
+                control_reg = MIA_48000;
+                break;
+        case 44100:
+                control_reg = MIA_44100;
+                break;
+        case 32000:
+                control_reg = MIA_32000;
+                break;
+        default:
+                DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+                return -EINVAL;
+        }
+
+        /* Override the clock setting if this Mia is set to S/PDIF clock */
+        if (chip->input_clock == ECHO_CLOCK_SPDIF)
+                control_reg |= MIA_SPDIF;
+
+        /* Set the control register if it has changed */
+        if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
+                if (wait_handshake(chip))
+                        return -EIO;
+
+                chip->comm_page->sample_rate = cpu_to_le32(rate);       /* ignored by the DSP */
+                chip->comm_page->control_register = cpu_to_le32(control_reg);
+                chip->sample_rate = rate;
+
+                clear_handshake(chip);
+                return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+        }
+        return 0;
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+        DE_ACT(("set_input_clock(%d)\n", clock));
+        snd_assert(clock == ECHO_CLOCK_INTERNAL || clock == ECHO_CLOCK_SPDIF,
+                   return -EINVAL);
+
+        chip->input_clock = clock;
+        return set_sample_rate(chip, chip->sample_rate);
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+                           int gain)
+{
+        int index;
+
+        snd_assert(pipe < num_pipes_out(chip) &&
+                   output < num_busses_out(chip), return -EINVAL);
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        chip->vmixer_gain[output][pipe] = gain;
+        index = output * num_pipes_out(chip) + pipe;
+        chip->comm_page->vmixer[index] = gain;
+
+        DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+        return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
+
+
+/* Tell the DSP to reread the flags from the comm page */
+static int update_flags(struct echoaudio *chip)
+{
+        if (wait_handshake(chip))
+                return -EIO;
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+static int set_professional_spdif(struct echoaudio *chip, char prof)
+{
+        DE_ACT(("set_professional_spdif %d\n", prof));
+        if (prof)
+                chip->comm_page->flags |=
+                        __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+        else
+                chip->comm_page->flags &=
+                        ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+        chip->professional_spdif = prof;
+        return update_flags(chip);
+}
+
diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c
new file mode 100644
index 000000000000..e31f0f11e3a8
--- /dev/null
+++ b/sound/pci/echoaudio/midi.c
@@ -0,0 +1,327 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+/******************************************************************************
+        MIDI lowlevel code
+******************************************************************************/
+
+/* Start and stop Midi input */
+static int enable_midi_input(struct echoaudio *chip, char enable)
+{
+        DE_MID(("enable_midi_input(%d)\n", enable));
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        if (enable) {
+                chip->mtc_state = MIDI_IN_STATE_NORMAL;
+                chip->comm_page->flags |=
+                        __constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
+        } else
+                chip->comm_page->flags &=
+                        ~__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
+
+        clear_handshake(chip);
+        return send_vector(chip, DSP_VC_UPDATE_FLAGS);
+}
+
+
+
+/* Send a buffer full of MIDI data to the DSP
+Returns how many actually written or < 0 on error */
+static int write_midi(struct echoaudio *chip, u8 *data, int bytes)
+{
+        snd_assert(bytes > 0 && bytes < MIDI_OUT_BUFFER_SIZE, return -EINVAL);
+
+        if (wait_handshake(chip))
+                return -EIO;
+
+        /* HF4 indicates that it is safe to write MIDI output data */
+        if (! (get_dsp_register(chip, CHI32_STATUS_REG) & CHI32_STATUS_REG_HF4))
+                return 0;
+
+        chip->comm_page->midi_output[0] = bytes;
+        memcpy(&chip->comm_page->midi_output[1], data, bytes);
+        chip->comm_page->midi_out_free_count = 0;
+        clear_handshake(chip);
+        send_vector(chip, DSP_VC_MIDI_WRITE);
+        DE_MID(("write_midi: %d\n", bytes));
+        return bytes;
+}
+
+
+
+/* Run the state machine for MIDI input data
+MIDI time code sync isn't supported by this code right now, but you still need
+this state machine to parse the incoming MIDI data stream.  Every time the DSP
+sees a 0xF1 byte come in, it adds the DSP sample position to the MIDI data
+stream. The DSP sample position is represented as a 32 bit unsigned value,
+with the high 16 bits first, followed by the low 16 bits. Since these aren't
+real MIDI bytes, the following logic is needed to skip them. */
+static inline int mtc_process_data(struct echoaudio *chip, short midi_byte)
+{
+        switch (chip->mtc_state) {
+        case MIDI_IN_STATE_NORMAL:
+                if (midi_byte == 0xF1)
+                        chip->mtc_state = MIDI_IN_STATE_TS_HIGH;
+                break;
+        case MIDI_IN_STATE_TS_HIGH:
+                chip->mtc_state = MIDI_IN_STATE_TS_LOW;
+                return MIDI_IN_SKIP_DATA;
+                break;
+        case MIDI_IN_STATE_TS_LOW:
+                chip->mtc_state = MIDI_IN_STATE_F1_DATA;
+                return MIDI_IN_SKIP_DATA;
+                break;
+        case MIDI_IN_STATE_F1_DATA:
+                chip->mtc_state = MIDI_IN_STATE_NORMAL;
+                break;
+        }
+        return 0;
+}
+
+
+
+/* This function is called from the IRQ handler and it reads the midi data
+from the DSP's buffer.  It returns the number of bytes received. */
+static int midi_service_irq(struct echoaudio *chip)
+{
+        short int count, midi_byte, i, received;
+
+        /* The count is at index 0, followed by actual data */
+        count = le16_to_cpu(chip->comm_page->midi_input[0]);
+
+        snd_assert(count < MIDI_IN_BUFFER_SIZE, return 0);
+
+        /* Get the MIDI data from the comm page */
+        i = 1;
+        received = 0;
+        for (i = 1; i <= count; i++) {
+                /* Get the MIDI byte */
+                midi_byte = le16_to_cpu(chip->comm_page->midi_input[i]);
+
+                /* Parse the incoming MIDI stream. The incoming MIDI data
+                consists of MIDI bytes and timestamps for the MIDI time code
+                0xF1 bytes. mtc_process_data() is a little state machine that
+                parses the stream. If you get MIDI_IN_SKIP_DATA back, then
+                this is a timestamp byte, not a MIDI byte, so don't store it
+                in the MIDI input buffer. */
+                if (mtc_process_data(chip, midi_byte) == MIDI_IN_SKIP_DATA)
+                        continue;
+
+                chip->midi_buffer[received++] = (u8)midi_byte;
+        }
+
+        return received;
+}
+
+
+
+
+/******************************************************************************
+        MIDI interface
+******************************************************************************/
+
+static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream)
+{
+        struct echoaudio *chip = substream->rmidi->private_data;
+
+        chip->midi_in = substream;
+        DE_MID(("rawmidi_iopen\n"));
+        return 0;
+}
+
+
+
+static void snd_echo_midi_input_trigger(struct snd_rawmidi_substream *substream,
+                                        int up)
+{
+        struct echoaudio *chip = substream->rmidi->private_data;
+
+        if (up != chip->midi_input_enabled) {
+                spin_lock_irq(&chip->lock);
+                enable_midi_input(chip, up);
+                spin_unlock_irq(&chip->lock);
+                chip->midi_input_enabled = up;
+        }
+}
+
+
+
+static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream)
+{
+        struct echoaudio *chip = substream->rmidi->private_data;
+
+        chip->midi_in = NULL;
+        DE_MID(("rawmidi_iclose\n"));
+        return 0;
+}
+
+
+
+static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream)
+{
+        struct echoaudio *chip = substream->rmidi->private_data;
+
+        chip->tinuse = 0;
+        chip->midi_full = 0;
+        chip->midi_out = substream;
+        DE_MID(("rawmidi_oopen\n"));
+        return 0;
+}
+
+
+
+static void snd_echo_midi_output_write(unsigned long data)
+{
+        struct echoaudio *chip = (struct echoaudio *)data;
+        unsigned long flags;
+        int bytes, sent, time;
+        unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1];
+
+        DE_MID(("snd_echo_midi_output_write\n"));
+        /* No interrupts are involved: we have to check at regular intervals
+        if the card's output buffer has room for new data. */
+        sent = bytes = 0;
+        spin_lock_irqsave(&chip->lock, flags);
+        chip->midi_full = 0;
+        if (chip->midi_out && !snd_rawmidi_transmit_empty(chip->midi_out)) {
+                bytes = snd_rawmidi_transmit_peek(chip->midi_out, buf,
+                                                  MIDI_OUT_BUFFER_SIZE - 1);
+                DE_MID(("Try to send %d bytes...\n", bytes));
+                sent = write_midi(chip, buf, bytes);
+                if (sent < 0) {
+                        snd_printk(KERN_ERR "write_midi() error %d\n", sent);
+                        /* retry later */
+                        sent = 9000;
+                        chip->midi_full = 1;
+                } else if (sent > 0) {
+                        DE_MID(("%d bytes sent\n", sent));
+                        snd_rawmidi_transmit_ack(chip->midi_out, sent);
+                } else {
+                        /* Buffer is full. DSP's internal buffer is 64 (128 ?)
+                        bytes long. Let's wait until half of them are sent */
+                        DE_MID(("Full\n"));
+                        sent = 32;
+                        chip->midi_full = 1;
+                }
+        }
+
+        /* We restart the timer only if there is some data left to send */
+        if (!snd_rawmidi_transmit_empty(chip->midi_out) && chip->tinuse) {
+                /* The timer will expire slightly after the data has been
+                   sent */
+                time = (sent << 3) / 25 + 1;    /* 8/25=0.32ms to send a byte */
+                mod_timer(&chip->timer, jiffies + (time * HZ + 999) / 1000);
+                DE_MID(("Timer armed(%d)\n", ((time * HZ + 999) / 1000)));
+        }
+        spin_unlock_irqrestore(&chip->lock, flags);
+}
+
+
+
+static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream,
+                                         int up)
+{
+        struct echoaudio *chip = substream->rmidi->private_data;
+
+        DE_MID(("snd_echo_midi_output_trigger(%d)\n", up));
+        spin_lock_irq(&chip->lock);
+        if (up) {
+                if (!chip->tinuse) {
+                        init_timer(&chip->timer);
+                        chip->timer.function = snd_echo_midi_output_write;
+                        chip->timer.data = (unsigned long)chip;
+                        chip->tinuse = 1;
+                }
+        } else {
+                if (chip->tinuse) {
+                        del_timer(&chip->timer);
+                        chip->tinuse = 0;
+                        DE_MID(("Timer removed\n"));
+                }
+        }
+        spin_unlock_irq(&chip->lock);
+
+        if (up && !chip->midi_full)
+                snd_echo_midi_output_write((unsigned long)chip);
+}
+
+
+
+static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream)
+{
+        struct echoaudio *chip = substream->rmidi->private_data;
+
+        chip->midi_out = NULL;
+        DE_MID(("rawmidi_oclose\n"));
+        return 0;
+}
+
+
+
+static struct snd_rawmidi_ops snd_echo_midi_input = {
+        .open = snd_echo_midi_input_open,
+        .close = snd_echo_midi_input_close,
+        .trigger = snd_echo_midi_input_trigger,
+};
+
+static struct snd_rawmidi_ops snd_echo_midi_output = {
+        .open = snd_echo_midi_output_open,
+        .close = snd_echo_midi_output_close,
+        .trigger = snd_echo_midi_output_trigger,
+};
+
+
+
+/* <--snd_echo_probe() */
+static int __devinit snd_echo_midi_create(struct snd_card *card,
+                                          struct echoaudio *chip)
+{
+        int err;
+
+        if ((err = snd_rawmidi_new(card, card->shortname, 0, 1, 1,
+                                   &chip->rmidi)) < 0)
+                return err;
+
+        strcpy(chip->rmidi->name, card->shortname);
+        chip->rmidi->private_data = chip;
+
+        snd_rawmidi_set_ops(chip->rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+                            &snd_echo_midi_input);
+        snd_rawmidi_set_ops(chip->rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+                            &snd_echo_midi_output);
+
+        chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
+                SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX;
+        DE_INIT(("MIDI ok\n"));
+        return 0;
+}
diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c
new file mode 100644
index 000000000000..5dc512add372
--- /dev/null
+++ b/sound/pci/echoaudio/mona.c
@@ -0,0 +1,129 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define ECHO24_FAMILY
+#define ECHOCARD_MONA
+#define ECHOCARD_NAME "Mona"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_ASIC
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_DIGITAL_IO
+#define ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
+#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
+#define ECHOCARD_HAS_EXTERNAL_CLOCK
+#define ECHOCARD_HAS_ADAT       6
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT   0       /* 6 */
+#define PX_DIGITAL_OUT  6       /* 8 */
+#define PX_ANALOG_IN    14      /* 4 */
+#define PX_DIGITAL_IN   18      /* 8 */
+#define PX_NUM          26
+
+/* Bus indexes */
+#define BX_ANALOG_OUT   0       /* 6 */
+#define BX_DIGITAL_OUT  6       /* 8 */
+#define BX_ANALOG_IN    14      /* 4 */
+#define BX_DIGITAL_IN   18      /* 8 */
+#define BX_NUM          26
+
+
+#include <sound/driver.h>
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/io.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+#define FW_361_LOADER           0
+#define FW_MONA_301_DSP         1
+#define FW_MONA_361_DSP         2
+#define FW_MONA_301_1_ASIC48    3
+#define FW_MONA_301_1_ASIC96    4
+#define FW_MONA_361_1_ASIC48    5
+#define FW_MONA_361_1_ASIC96    6
+#define FW_MONA_2_ASIC          7
+
+static const struct firmware card_fw[] = {
+        {0, "loader_dsp.fw"},
+        {0, "mona_301_dsp.fw"},
+        {0, "mona_361_dsp.fw"},
+        {0, "mona_301_1_asic_48.fw"},
+        {0, "mona_301_1_asic_96.fw"},
+        {0, "mona_361_1_asic_48.fw"},
+        {0, "mona_361_1_asic_96.fw"},
+        {0, "mona_2_asic.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+        {0x1057, 0x1801, 0xECC0, 0x0070, 0, 0, 0},      /* DSP 56301 Mona rev.0 */
+        {0x1057, 0x1801, 0xECC0, 0x0071, 0, 0, 0},      /* DSP 56301 Mona rev.1 */
+        {0x1057, 0x1801, 0xECC0, 0x0072, 0, 0, 0},      /* DSP 56301 Mona rev.2 */
+        {0x1057, 0x3410, 0xECC0, 0x0070, 0, 0, 0},      /* DSP 56361 Mona rev.0 */
+        {0x1057, 0x3410, 0xECC0, 0x0071, 0, 0, 0},      /* DSP 56361 Mona rev.1 */
+        {0x1057, 0x3410, 0xECC0, 0x0072, 0, 0, 0},      /* DSP 56361 Mona rev.2 */
+        {0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+        .info = SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_BLOCK_TRANSFER |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_PAUSE |
+                SNDRV_PCM_INFO_SYNC_START,
+        .formats =      SNDRV_PCM_FMTBIT_U8 |
+                        SNDRV_PCM_FMTBIT_S16_LE |
+                        SNDRV_PCM_FMTBIT_S24_3LE |
+                        SNDRV_PCM_FMTBIT_S32_LE |
+                        SNDRV_PCM_FMTBIT_S32_BE,
+        .rates =        SNDRV_PCM_RATE_8000_48000 |
+                        SNDRV_PCM_RATE_88200 |
+                        SNDRV_PCM_RATE_96000,
+        .rate_min = 8000,
+        .rate_max = 96000,
+        .channels_min = 1,
+        .channels_max = 8,
+        .buffer_bytes_max = 262144,
+        .period_bytes_min = 32,
+        .period_bytes_max = 131072,
+        .periods_min = 2,
+        .periods_max = 220,
+        /* One page (4k) contains 512 instructions. I don't know if the hw
+        supports lists longer than this. In this case periods_max=220 is a
+        safe limit to make sure the list never exceeds 512 instructions. */
+};
+
+
+#include "mona_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio_gml.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c
new file mode 100644
index 000000000000..c0b4bf0be7d1
--- /dev/null
+++ b/sound/pci/echoaudio/mona_dsp.c
@@ -0,0 +1,428 @@
+/****************************************************************************
+
+   Copyright Echo Digital Audio Corporation (c) 1998 - 2004
+   All rights reserved
+   www.echoaudio.com
+
+   This file is part of Echo Digital Audio's generic driver library.
+
+   Echo Digital Audio's generic driver library is free software;
+   you can redistribute it and/or modify it under the terms of
+   the GNU General Public License as published by the Free Software
+   Foundation.
+
+   This program is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License
+   along with this program; if not, write to the Free Software
+   Foundation, Inc., 59 Temple Place - Suite 330, Boston,
+   MA  02111-1307, USA.
+
+   *************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+****************************************************************************/
+
+
+static int write_control_reg(struct echoaudio *chip, u32 value, char force);
+static int set_input_clock(struct echoaudio *chip, u16 clock);
+static int set_professional_spdif(struct echoaudio *chip, char prof);
+static int set_digital_mode(struct echoaudio *chip, u8 mode);
+static int load_asic_generic(struct echoaudio *chip, u32 cmd,
+                             const struct firmware *asic);
+static int check_asic_status(struct echoaudio *chip);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+        int err;
+
+        DE_INIT(("init_hw() - Mona\n"));
+        snd_assert((subdevice_id & 0xfff0) == MONA, return -ENODEV);
+
+        if ((err = init_dsp_comm_page(chip))) {
+                DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+                return err;
+        }
+
+        chip->device_id = device_id;
+        chip->subdevice_id = subdevice_id;
+        chip->bad_board = TRUE;
+        chip->input_clock_types =
+                ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
+                ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT;
+        chip->digital_modes =
+                ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
+                ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
+                ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
+
+        /* Mona comes in both '301 and '361 flavors */
+        if (chip->device_id == DEVICE_ID_56361)
+                chip->dsp_code_to_load = &card_fw[FW_MONA_361_DSP];
+        else
+                chip->dsp_code_to_load = &card_fw[FW_MONA_301_DSP];
+
+        chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
+        chip->professional_spdif = FALSE;
+        chip->digital_in_automute = TRUE;
+
+        if ((err = load_firmware(chip)) < 0)
+                return err;
+        chip->bad_board = FALSE;
+
+        if ((err = init_line_levels(chip)) < 0)
+                return err;
+
+        err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
+        snd_assert(err >= 0, return err);
+        err = set_professional_spdif(chip, TRUE);
+
+        DE_INIT(("init_hw done\n"));
+        return err;
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+        u32 clocks_from_dsp, clock_bits;
+
+        /* Map the DSP clock detect bits to the generic driver clock
+           detect bits */
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        clock_bits = ECHO_CLOCK_BIT_INTERNAL;
+
+        if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF)
+                clock_bits |= ECHO_CLOCK_BIT_SPDIF;
+
+        if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ADAT)
+                clock_bits |= ECHO_CLOCK_BIT_ADAT;
+
+        if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD)
+                clock_bits |= ECHO_CLOCK_BIT_WORD;
+
+        return clock_bits;
+}
+
+
+
+/* Mona has an ASIC on the PCI card and another ASIC in the external box; 
+both need to be loaded. */
+static int load_asic(struct echoaudio *chip)
+{
+        u32 control_reg;
+        int err;
+        const struct firmware *asic;
+
+        if (chip->asic_loaded)
+                return 0;
+
+        mdelay(10);
+
+        if (chip->device_id == DEVICE_ID_56361)
+                asic = &card_fw[FW_MONA_361_1_ASIC48];
+        else
+                asic = &card_fw[FW_MONA_301_1_ASIC48];
+
+        err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC, asic);
+        if (err < 0)
+                return err;
+
+        chip->asic_code = asic;
+        mdelay(10);
+
+        /* Do the external one */
+        err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_EXTERNAL_ASIC,
+                                &card_fw[FW_MONA_2_ASIC]);
+        if (err < 0)
+                return err;
+
+        mdelay(10);
+        err = check_asic_status(chip);
+
+        /* Set up the control register if the load succeeded -
+           48 kHz, internal clock, S/PDIF RCA mode */
+        if (!err) {
+                control_reg = GML_CONVERTER_ENABLE | GML_48KHZ;
+                err = write_control_reg(chip, control_reg, TRUE);
+        }
+
+        return err;
+}
+
+
+
+/* Depending on what digital mode you want, Mona needs different ASICs
+loaded.  This function checks the ASIC needed for the new mode and sees
+if it matches the one already loaded. */
+static int switch_asic(struct echoaudio *chip, char double_speed)
+{
+        const struct firmware *asic;
+        int err;
+
+        /* Check the clock detect bits to see if this is
+        a single-speed clock or a double-speed clock; load
+        a new ASIC if necessary. */
+        if (chip->device_id == DEVICE_ID_56361) {
+                if (double_speed)
+                        asic = &card_fw[FW_MONA_361_1_ASIC96];
+                else
+                        asic = &card_fw[FW_MONA_361_1_ASIC48];
+        } else {
+                if (double_speed)
+                        asic = &card_fw[FW_MONA_301_1_ASIC96];
+                else
+                        asic = &card_fw[FW_MONA_301_1_ASIC48];
+        }
+
+        if (asic != chip->asic_code) {
+                /* Load the desired ASIC */
+                err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC,
+                                        asic);
+                if (err < 0)
+                        return err;
+                chip->asic_code = asic;
+        }
+
+        return 0;
+}
+
+
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+        u32 control_reg, clock;
+        const struct firmware *asic;
+        char force_write;
+
+        /* Only set the clock for internal mode. */
+        if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
+                DE_ACT(("set_sample_rate: Cannot set sample rate - "
+                        "clock not set to CLK_CLOCKININTERNAL\n"));
+                /* Save the rate anyhow */
+                chip->comm_page->sample_rate = cpu_to_le32(rate);
+                chip->sample_rate = rate;
+                return 0;
+        }
+
+        /* Now, check to see if the required ASIC is loaded */
+        if (rate >= 88200) {
+                if (chip->digital_mode == DIGITAL_MODE_ADAT)
+                        return -EINVAL;
+                if (chip->device_id == DEVICE_ID_56361)
+                        asic = &card_fw[FW_MONA_361_1_ASIC96];
+                else
+                        asic = &card_fw[FW_MONA_301_1_ASIC96];
+        } else {
+                if (chip->device_id == DEVICE_ID_56361)
+                        asic = &card_fw[FW_MONA_361_1_ASIC48];
+                else
+                        asic = &card_fw[FW_MONA_301_1_ASIC48];
+        }
+
+        force_write = 0;
+        if (asic != chip->asic_code) {
+                int err;
+                /* Load the desired ASIC (load_asic_generic() can sleep) */
+                spin_unlock_irq(&chip->lock);
+                err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC,
+                                        asic);
+                spin_lock_irq(&chip->lock);
+
+                if (err < 0)
+                        return err;
+                chip->asic_code = asic;
+                force_write = 1;
+        }
+
+        /* Compute the new control register value */
+        clock = 0;
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        control_reg &= GML_CLOCK_CLEAR_MASK;
+        control_reg &= GML_SPDIF_RATE_CLEAR_MASK;
+
+        switch (rate) {
+        case 96000:
+                clock = GML_96KHZ;
+                break;
+        case 88200:
+                clock = GML_88KHZ;
+                break;
+        case 48000:
+                clock = GML_48KHZ | GML_SPDIF_SAMPLE_RATE1;
+                break;
+        case 44100:
+                clock = GML_44KHZ;
+                /* Professional mode */
+                if (control_reg & GML_SPDIF_PRO_MODE)
+                        clock |= GML_SPDIF_SAMPLE_RATE0;
+                break;
+        case 32000:
+                clock = GML_32KHZ | GML_SPDIF_SAMPLE_RATE0 |
+                        GML_SPDIF_SAMPLE_RATE1;
+                break;
+        case 22050:
+                clock = GML_22KHZ;
+                break;
+        case 16000:
+                clock = GML_16KHZ;
+                break;
+        case 11025:
+                clock = GML_11KHZ;
+                break;
+        case 8000:
+                clock = GML_8KHZ;
+                break;
+        default:
+                DE_ACT(("set_sample_rate: %d invalid!\n", rate));
+                return -EINVAL;
+        }
+
+        control_reg |= clock;
+
+        chip->comm_page->sample_rate = cpu_to_le32(rate);       /* ignored by the DSP */
+        chip->sample_rate = rate;
+        DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
+
+        return write_control_reg(chip, control_reg, force_write);
+}
+
+
+
+static int set_input_clock(struct echoaudio *chip, u16 clock)
+{
+        u32 control_reg, clocks_from_dsp;
+        int err;
+
+        DE_ACT(("set_input_clock:\n"));
+
+        /* Prevent two simultaneous calls to switch_asic() */
+        if (atomic_read(&chip->opencount))
+                return -EAGAIN;
+
+        /* Mask off the clock select bits */
+        control_reg = le32_to_cpu(chip->comm_page->control_register) &
+                GML_CLOCK_CLEAR_MASK;
+        clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
+
+        switch (clock) {
+        case ECHO_CLOCK_INTERNAL:
+                DE_ACT(("Set Mona clock to INTERNAL\n"));
+                chip->input_clock = ECHO_CLOCK_INTERNAL;
+                return set_sample_rate(chip, chip->sample_rate);
+        case ECHO_CLOCK_SPDIF:
+                if (chip->digital_mode == DIGITAL_MODE_ADAT)
+                        return -EAGAIN;
+                spin_unlock_irq(&chip->lock);
+                err = switch_asic(chip, clocks_from_dsp &
+                                  GML_CLOCK_DETECT_BIT_SPDIF96);
+                spin_lock_irq(&chip->lock);
+                if (err < 0)
+                        return err;
+                DE_ACT(("Set Mona clock to SPDIF\n"));
+                control_reg |= GML_SPDIF_CLOCK;
+                if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
+                        control_reg |= GML_DOUBLE_SPEED_MODE;
+                else
+                        control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                break;
+        case ECHO_CLOCK_WORD:
+                DE_ACT(("Set Mona clock to WORD\n"));
+                spin_unlock_irq(&chip->lock);
+                err = switch_asic(chip, clocks_from_dsp &
+                                  GML_CLOCK_DETECT_BIT_WORD96);
+                spin_lock_irq(&chip->lock);
+                if (err < 0)
+                        return err;
+                control_reg |= GML_WORD_CLOCK;
+                if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96)
+                        control_reg |= GML_DOUBLE_SPEED_MODE;
+                else
+                        control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                break;
+        case ECHO_CLOCK_ADAT:
+                DE_ACT(("Set Mona clock to ADAT\n"));
+                if (chip->digital_mode != DIGITAL_MODE_ADAT)
+                        return -EAGAIN;
+                control_reg |= GML_ADAT_CLOCK;
+                control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                break;
+        default:
+                DE_ACT(("Input clock 0x%x not supported for Mona\n", clock));
+                return -EINVAL;
+        }
+
+        chip->input_clock = clock;
+        return write_control_reg(chip, control_reg, TRUE);
+}
+
+
+
+static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
+{
+        u32 control_reg;
+        int err, incompatible_clock;
+
+        /* Set clock to "internal" if it's not compatible with the new mode */
+        incompatible_clock = FALSE;
+        switch (mode) {
+        case DIGITAL_MODE_SPDIF_OPTICAL:
+        case DIGITAL_MODE_SPDIF_RCA:
+                if (chip->input_clock == ECHO_CLOCK_ADAT)
+                        incompatible_clock = TRUE;
+                break;
+        case DIGITAL_MODE_ADAT:
+                if (chip->input_clock == ECHO_CLOCK_SPDIF)
+                        incompatible_clock = TRUE;
+                break;
+        default:
+                DE_ACT(("Digital mode not supported: %d\n", mode));
+                return -EINVAL;
+        }
+
+        spin_lock_irq(&chip->lock);
+
+        if (incompatible_clock) {       /* Switch to 48KHz, internal */
+                chip->sample_rate = 48000;
+                set_input_clock(chip, ECHO_CLOCK_INTERNAL);
+        }
+
+        /* Clear the current digital mode */
+        control_reg = le32_to_cpu(chip->comm_page->control_register);
+        control_reg &= GML_DIGITAL_MODE_CLEAR_MASK;
+
+        /* Tweak the control reg */
+        switch (mode) {
+        case DIGITAL_MODE_SPDIF_OPTICAL:
+                control_reg |= GML_SPDIF_OPTICAL_MODE;
+                break;
+        case DIGITAL_MODE_SPDIF_RCA:
+                /* GML_SPDIF_OPTICAL_MODE bit cleared */
+                break;
+        case DIGITAL_MODE_ADAT:
+                /* If the current ASIC is the 96KHz ASIC, switch the ASIC
+                   and set to 48 KHz */
+                if (chip->asic_code == &card_fw[FW_MONA_361_1_ASIC96] ||
+                    chip->asic_code == &card_fw[FW_MONA_301_1_ASIC96]) {
+                        set_sample_rate(chip, 48000);
+                }
+                control_reg |= GML_ADAT_MODE;
+                control_reg &= ~GML_DOUBLE_SPEED_MODE;
+                break;
+        }
+
+        err = write_control_reg(chip, control_reg, FALSE);
+        spin_unlock_irq(&chip->lock);
+        if (err < 0)
+                return err;
+        chip->digital_mode = mode;
+
+        DE_ACT(("set_digital_mode to %d\n", mode));
+        return incompatible_clock;
+}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8c2a8174ece1..23201f3eeb12 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -408,7 +408,9 @@ static const struct hda_codec_preset *find_codec_preset(struct hda_codec *codec)
                         u32 mask = preset->mask;
                         if (! mask)
                                 mask = ~0;
-                        if (preset->id == (codec->vendor_id & mask))
+                        if (preset->id == (codec->vendor_id & mask) &&
+                            (! preset->rev ||
+                             preset->rev == codec->revision_id))
                                 return preset;
                 }
         }
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index dd4e00a82b55..33b7d5806469 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -799,6 +799,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = {
         { .pci_subvendor = 0x1043, .pci_subdevice = 0x818f,
           .config = AD1986A_LAPTOP }, /* ASUS P5GV-MX */
         { .modelname = "laptop-eapd",   .config = AD1986A_LAPTOP_EAPD },
+        { .pci_subvendor = 0x144d, .pci_subdevice = 0xc023,
+          .config = AD1986A_LAPTOP_EAPD }, /* Samsung X60 Chane */
         { .pci_subvendor = 0x144d, .pci_subdevice = 0xc024,
           .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */
         { .pci_subvendor = 0x1043, .pci_subdevice = 0x1153,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 98b9f16c26ff..18d105263fea 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -78,6 +78,7 @@ enum {
 enum {
         ALC262_BASIC,
         ALC262_FUJITSU,
+        ALC262_HP_BPC,
         ALC262_AUTO,
         ALC262_MODEL_LAST /* last tag */
 };
@@ -85,6 +86,7 @@ enum {
 /* ALC861 models */
 enum {
         ALC861_3ST,
+        ALC660_3ST,
         ALC861_3ST_DIG,
         ALC861_6ST_DIG,
         ALC861_AUTO,
@@ -99,6 +101,17 @@ enum {
         ALC882_MODEL_LAST,
 };
 
+/* ALC883 models */
+enum {
+        ALC883_3ST_2ch_DIG,
+        ALC883_3ST_6ch_DIG,
+        ALC883_3ST_6ch,
+        ALC883_6ST_DIG,
+        ALC888_DEMO_BOARD,
+        ALC883_AUTO,
+        ALC883_MODEL_LAST,
+};
+
 /* for GPIO Poll */
 #define GPIO_MASK       0x03
 
@@ -108,7 +121,8 @@ struct alc_spec {
         unsigned int num_mixers;
 
         const struct hda_verb *init_verbs[5];   /* initialization verbs
-                                                 * don't forget NULL termination!
+                                                 * don't forget NULL
+                                                 * termination!
                                                  */
         unsigned int num_init_verbs;
 
@@ -163,7 +177,9 @@ struct alc_spec {
  * configuration template - to be copied to the spec instance
  */
 struct alc_config_preset {
-        struct snd_kcontrol_new *mixers[5]; /* should be identical size with spec */
+        struct snd_kcontrol_new *mixers[5]; /* should be identical size
+                                             * with spec
+                                             */
         const struct hda_verb *init_verbs[5];
         unsigned int num_dacs;
         hda_nid_t *dac_nids;
@@ -184,7 +200,8 @@ struct alc_config_preset {
 /*
  * input MUX handling
  */
-static int alc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+static int alc_mux_enum_info(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_info *uinfo)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         struct alc_spec *spec = codec->spec;
@@ -194,7 +211,8 @@ static int alc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
         return snd_hda_input_mux_info(&spec->input_mux[mux_idx], uinfo);
 }
 
-static int alc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_mux_enum_get(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         struct alc_spec *spec = codec->spec;
@@ -204,21 +222,24 @@ static int alc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
         return 0;
 }
 
-static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         struct alc_spec *spec = codec->spec;
         unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
         unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
         return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol,
-                                     spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]);
+                                     spec->adc_nids[adc_idx],
+                                     &spec->cur_mux[adc_idx]);
 }
 
 
 /*
  * channel mode setting
  */
-static int alc_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_info *uinfo)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         struct alc_spec *spec = codec->spec;
@@ -226,20 +247,24 @@ static int alc_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_i
                                     spec->num_channel_mode);
 }
 
-static int alc_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
+                           struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         struct alc_spec *spec = codec->spec;
         return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
-                                   spec->num_channel_mode, spec->multiout.max_channels);
+                                   spec->num_channel_mode,
+                                   spec->multiout.max_channels);
 }
 
-static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
+                           struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         struct alc_spec *spec = codec->spec;
         return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
-                                   spec->num_channel_mode, &spec->multiout.max_channels);
+                                   spec->num_channel_mode,
+                                   &spec->multiout.max_channels);
 }
 
 /*
@@ -290,7 +315,8 @@ static signed char alc_pin_mode_dir_info[5][2] = {
 #define alc_pin_mode_n_items(_dir) \
         (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
 
-static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_info *uinfo)
 {
         unsigned int item_num = uinfo->value.enumerated.item;
         unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
@@ -305,40 +331,46 @@ static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
         return 0;
 }
 
-static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
 {
         unsigned int i;
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = kcontrol->private_value & 0xffff;
         unsigned char dir = (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);
+        unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
+                                                 AC_VERB_GET_PIN_WIDGET_CONTROL,
+                                                 0x00);
 
         /* Find enumerated value for current pinctl setting */
         i = alc_pin_mode_min(dir);
-        while (alc_pin_mode_values[i]!=pinctl && i<=alc_pin_mode_max(dir))
+        while (alc_pin_mode_values[i] != pinctl && i <= alc_pin_mode_max(dir))
                 i++;
-        *valp = i<=alc_pin_mode_max(dir)?i:alc_pin_mode_min(dir);
+        *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
         return 0;
 }
 
-static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_value *ucontrol)
 {
         signed int change;
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = kcontrol->private_value & 0xffff;
         unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
         long val = *ucontrol->value.integer.value;
-        unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00);
+        unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
+                                                 AC_VERB_GET_PIN_WIDGET_CONTROL,
+                                                 0x00);
 
-        if (val<alc_pin_mode_min(dir) || val>alc_pin_mode_max(dir)) 
+        if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir)) 
                 val = alc_pin_mode_min(dir);
 
         change = pinctl != alc_pin_mode_values[val];
         if (change) {
                 /* Set pin mode to that requested */
                 snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL,
-                        alc_pin_mode_values[val]);
+                                    alc_pin_mode_values[val]);
 
                 /* Also enable the retasking pin's input/output as required 
                  * for the requested pin mode.  Enum values of 2 or less are
@@ -351,15 +383,19 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
                  * this turns out to be necessary in the future.
                  */
                 if (val <= 2) {
-                        snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE,
-                                AMP_OUT_MUTE);
-                        snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE,
-                                AMP_IN_UNMUTE(0));
+                        snd_hda_codec_write(codec, nid, 0,
+                                            AC_VERB_SET_AMP_GAIN_MUTE,
+                                            AMP_OUT_MUTE);
+                        snd_hda_codec_write(codec, nid, 0,
+                                            AC_VERB_SET_AMP_GAIN_MUTE,
+                                            AMP_IN_UNMUTE(0));
                 } else {
-                        snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE,
-                                AMP_IN_MUTE(0));
-                        snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE,
-                                AMP_OUT_UNMUTE);
+                        snd_hda_codec_write(codec, nid, 0,
+                                            AC_VERB_SET_AMP_GAIN_MUTE,
+                                            AMP_IN_MUTE(0));
+                        snd_hda_codec_write(codec, nid, 0,
+                                            AC_VERB_SET_AMP_GAIN_MUTE,
+                                            AMP_OUT_UNMUTE);
                 }
         }
         return change;
@@ -378,7 +414,8 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
  * needed for any "production" models.
  */
 #ifdef CONFIG_SND_DEBUG
-static int alc_gpio_data_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+static int alc_gpio_data_info(struct snd_kcontrol *kcontrol,
+                              struct snd_ctl_elem_info *uinfo)
 {
         uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
         uinfo->count = 1;
@@ -386,33 +423,38 @@ static int alc_gpio_data_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
         uinfo->value.integer.max = 1;
         return 0;
 }                                
-static int alc_gpio_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = kcontrol->private_value & 0xffff;
         unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
         long *valp = ucontrol->value.integer.value;
-        unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00);
+        unsigned int val = snd_hda_codec_read(codec, nid, 0,
+                                              AC_VERB_GET_GPIO_DATA, 0x00);
 
         *valp = (val & mask) != 0;
         return 0;
 }
-static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
+                             struct snd_ctl_elem_value *ucontrol)
 {
         signed int change;
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = kcontrol->private_value & 0xffff;
         unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
         long val = *ucontrol->value.integer.value;
-        unsigned int gpio_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00);
+        unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
+                                                    AC_VERB_GET_GPIO_DATA,
+                                                    0x00);
 
         /* Set/unset the masked GPIO bit(s) as needed */
-        change = (val==0?0:mask) != (gpio_data & mask);
-        if (val==0)
+        change = (val == 0 ? 0 : mask) != (gpio_data & mask);
+        if (val == 0)
                 gpio_data &= ~mask;
         else
                 gpio_data |= mask;
-        snd_hda_codec_write(codec,nid,0,AC_VERB_SET_GPIO_DATA,gpio_data);
+        snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_GPIO_DATA, gpio_data);
 
         return change;
 }
@@ -432,7 +474,8 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
  * necessary.
  */
 #ifdef CONFIG_SND_DEBUG
-static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol,
+                               struct snd_ctl_elem_info *uinfo)
 {
         uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
         uinfo->count = 1;
@@ -440,33 +483,39 @@ static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ele
         uinfo->value.integer.max = 1;
         return 0;
 }                                
-static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
+                              struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = kcontrol->private_value & 0xffff;
         unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
         long *valp = ucontrol->value.integer.value;
-        unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00);
+        unsigned int val = snd_hda_codec_read(codec, nid, 0,
+                                              AC_VERB_GET_DIGI_CONVERT, 0x00);
 
         *valp = (val & mask) != 0;
         return 0;
 }
-static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
+                              struct snd_ctl_elem_value *ucontrol)
 {
         signed int change;
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = kcontrol->private_value & 0xffff;
         unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
         long val = *ucontrol->value.integer.value;
-        unsigned int ctrl_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00);
+        unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
+                                                    AC_VERB_GET_DIGI_CONVERT,
+                                                    0x00);
 
         /* Set/unset the masked control bit(s) as needed */
-        change = (val==0?0:mask) != (ctrl_data & mask);
+        change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
         if (val==0)
                 ctrl_data &= ~mask;
         else
                 ctrl_data |= mask;
-        snd_hda_codec_write(codec,nid,0,AC_VERB_SET_DIGI_CONVERT_1,ctrl_data);
+        snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
+                            ctrl_data);
 
         return change;
 }
@@ -481,14 +530,17 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 /*
  * set up from the preset table
  */
-static void setup_preset(struct alc_spec *spec, const struct alc_config_preset *preset)
+static void setup_preset(struct alc_spec *spec,
+                         const struct alc_config_preset *preset)
 {
         int i;
 
         for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
                 spec->mixers[spec->num_mixers++] = preset->mixers[i];
-        for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i]; i++)
-                spec->init_verbs[spec->num_init_verbs++] = preset->init_verbs[i];
+        for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
+             i++)
+                spec->init_verbs[spec->num_init_verbs++] =
+                        preset->init_verbs[i];
         
         spec->channel_mode = preset->channel_mode;
         spec->num_channel_mode = preset->num_channel_mode;
@@ -517,8 +569,8 @@ static void setup_preset(struct alc_spec *spec, const struct alc_config_preset *
  * ALC880 3-stack model
  *
  * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
- * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18, F-Mic = 0x1b
- *                 HP = 0x19
+ * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18,
+ *                 F-Mic = 0x1b, HP = 0x19
  */
 
 static hda_nid_t alc880_dac_nids[4] = {
@@ -662,7 +714,8 @@ static struct snd_kcontrol_new alc880_capture_alt_mixer[] = {
 /*
  * ALC880 5-stack model
  *
- * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d), Side = 0x02 (0xd)
+ * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d),
+ *      Side = 0x02 (0xd)
  * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
  *                 Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
  */
@@ -700,7 +753,8 @@ static struct hda_channel_mode alc880_fivestack_modes[2] = {
 /*
  * ALC880 6-stack model
  *
- * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e), Side = 0x05 (0x0f)
+ * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e),
+ *      Side = 0x05 (0x0f)
  * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
  *   Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
  */
@@ -811,7 +865,8 @@ static struct snd_kcontrol_new alc880_w810_base_mixer[] = {
  * Z710V model
  *
  * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
- * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?), Line = 0x1a
+ * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?),
+ *                 Line = 0x1a
  */
 
 static hda_nid_t alc880_z71v_dac_nids[1] = {
@@ -966,7 +1021,8 @@ static int alc_build_controls(struct hda_codec *codec)
         }
 
         if (spec->multiout.dig_out_nid) {
-                err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+                err = snd_hda_create_spdif_out_ctls(codec,
+                                                    spec->multiout.dig_out_nid);
                 if (err < 0)
                         return err;
         }
@@ -999,8 +1055,8 @@ static struct hda_verb alc880_volume_init_verbs[] = {
 
         /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
          * mixer widget
-         * Note: PASD motherboards uses the Line In 2 as the input for front panel
-         * mic (mic 2)
+         * Note: PASD motherboards uses the Line In 2 as the input for front
+         * panel mic (mic 2)
          */
         /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
         {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -1154,8 +1210,8 @@ static struct hda_verb alc880_pin_z71v_init_verbs[] = {
 
 /*
  * 6-stack pin configuration:
- * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, f-mic = 0x19,
- * line = 0x1a, HP = 0x1b
+ * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
+ * f-mic = 0x19, line = 0x1a, HP = 0x1b
  */
 static struct hda_verb alc880_pin_6stack_init_verbs[] = {
         {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
@@ -1587,8 +1643,8 @@ static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
                                        struct snd_pcm_substream *substream)
 {
         struct alc_spec *spec = codec->spec;
-        return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
-                                                format, substream);
+        return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+                                                stream_tag, format, substream);
 }
 
 static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
@@ -1640,7 +1696,8 @@ static int alc880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 {
         struct alc_spec *spec = codec->spec;
 
-        snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0);
+        snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+                                   0, 0, 0);
         return 0;
 }
 
@@ -1822,7 +1879,8 @@ static struct hda_channel_mode alc880_test_modes[4] = {
         { 8, NULL },
 };
 
-static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_info *uinfo)
 {
         static char *texts[] = {
                 "N/A", "Line Out", "HP Out",
@@ -1837,7 +1895,8 @@ static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_e
         return 0;
 }
 
-static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
@@ -1863,7 +1922,8 @@ static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_el
         return 0;
 }
 
-static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
@@ -1881,15 +1941,18 @@ static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el
                                      AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
         new_ctl = ctls[ucontrol->value.enumerated.item[0]];
         if (old_ctl != new_ctl) {
-                snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl);
+                snd_hda_codec_write(codec, nid, 0,
+                                    AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl);
                 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-                                    ucontrol->value.enumerated.item[0] >= 3 ? 0xb080 : 0xb000);
+                                    (ucontrol->value.enumerated.item[0] >= 3 ?
+                                     0xb080 : 0xb000));
                 return 1;
         }
         return 0;
 }
 
-static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
+static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
+                                 struct snd_ctl_elem_info *uinfo)
 {
         static char *texts[] = {
                 "Front", "Surround", "CLFE", "Side"
@@ -1903,7 +1966,8 @@ static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_e
         return 0;
 }
 
-static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
@@ -1914,7 +1978,8 @@ static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol, struct snd_ctl_el
         return 0;
 }
 
-static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
 {
         struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
         hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
@@ -2739,7 +2804,8 @@ static int patch_alc880(struct hda_codec *codec)
 
         board_config = snd_hda_check_board_config(codec, alc880_cfg_tbl);
         if (board_config < 0 || board_config >= ALC880_MODEL_LAST) {
-                printk(KERN_INFO "hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...\n");
+                printk(KERN_INFO "hda_codec: Unknown model for ALC880, "
+                       "trying auto-probe from BIOS...\n");
                 board_config = ALC880_AUTO;
         }
 
@@ -2750,7 +2816,9 @@ static int patch_alc880(struct hda_codec *codec)
                         alc_free(codec);
                         return err;
                 } else if (! err) {
-                        printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS.  Using 3-stack mode...\n");
+                        printk(KERN_INFO
+                               "hda_codec: Cannot set up configuration "
+                               "from BIOS.  Using 3-stack mode...\n");
                         board_config = ALC880_3ST;
                 }
         }
@@ -3947,7 +4015,8 @@ static int patch_alc260(struct hda_codec *codec)
 
         board_config = snd_hda_check_board_config(codec, alc260_cfg_tbl);
         if (board_config < 0 || board_config >= ALC260_MODEL_LAST) {
-                snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260\n");
+                snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260, "
+                           "trying auto-probe from BIOS...\n");
                 board_config = ALC260_AUTO;
         }
 
@@ -3958,7 +4027,9 @@ static int patch_alc260(struct hda_codec *codec)
                         alc_free(codec);
                         return err;
                 } else if (! err) {
-                        printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS.  Using base mode...\n");
+                        printk(KERN_INFO
+                               "hda_codec: Cannot set up configuration "
+                               "from BIOS.  Using base mode...\n");
                         board_config = ALC260_BASIC;
                 }
         }
@@ -4320,9 +4391,12 @@ static struct snd_kcontrol_new alc882_capture_mixer[] = {
 static struct hda_board_config alc882_cfg_tbl[] = {
         { .modelname = "3stack-dig", .config = ALC882_3ST_DIG },
         { .modelname = "6stack-dig", .config = ALC882_6ST_DIG },
-        { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* MSI  */
-        { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* Foxconn */
-        { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* ECS */
+        { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668,
+          .config = ALC882_6ST_DIG }, /* MSI  */
+        { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668,
+          .config = ALC882_6ST_DIG }, /* Foxconn */
+        { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668,
+          .config = ALC882_6ST_DIG }, /* ECS to Intel*/
         { .modelname = "auto", .config = ALC882_AUTO },
         {}
 };
@@ -4439,10 +4513,6 @@ static void alc882_auto_init(struct hda_codec *codec)
         alc882_auto_init_analog_input(codec);
 }
 
-/*
- *  ALC882 Headphone poll in 3.5.1a or 3.5.2
- */
-
 static int patch_alc882(struct hda_codec *codec)
 {
         struct alc_spec *spec;
@@ -4457,7 +4527,8 @@ static int patch_alc882(struct hda_codec *codec)
         board_config = snd_hda_check_board_config(codec, alc882_cfg_tbl);
 
         if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
-                printk(KERN_INFO "hda_codec: Unknown model for ALC882, trying auto-probe from BIOS...\n");
+                printk(KERN_INFO "hda_codec: Unknown model for ALC882, "
+                       "trying auto-probe from BIOS...\n");
                 board_config = ALC882_AUTO;
         }
 
@@ -4468,7 +4539,9 @@ static int patch_alc882(struct hda_codec *codec)
                         alc_free(codec);
                         return err;
                 } else if (! err) {
-                        printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS.  Using base mode...\n");
+                        printk(KERN_INFO
+                               "hda_codec: Cannot set up configuration "
+                               "from BIOS.  Using base mode...\n");
                         board_config = ALC882_3ST_DIG;
                 }
         }
@@ -4509,6 +4582,652 @@ static int patch_alc882(struct hda_codec *codec)
 }
 
 /*
+ * ALC883 support
+ *
+ * ALC883 is almost identical with ALC880 but has cleaner and more flexible
+ * configuration.  Each pin widget can choose any input DACs and a mixer.
+ * Each ADC is connected from a mixer of all inputs.  This makes possible
+ * 6-channel independent captures.
+ *
+ * In addition, an independent DAC for the multi-playback (not used in this
+ * driver yet).
+ */
+#define ALC883_DIGOUT_NID       0x06
+#define ALC883_DIGIN_NID        0x0a
+
+static hda_nid_t alc883_dac_nids[4] = {
+        /* front, rear, clfe, rear_surr */
+        0x02, 0x04, 0x03, 0x05
+};
+
+static hda_nid_t alc883_adc_nids[2] = {
+        /* ADC1-2 */
+        0x08, 0x09,
+};
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+
+static struct hda_input_mux alc883_capture_source = {
+        .num_items = 4,
+        .items = {
+                { "Mic", 0x0 },
+                { "Front Mic", 0x1 },
+                { "Line", 0x2 },
+                { "CD", 0x4 },
+        },
+};
+#define alc883_mux_enum_info alc_mux_enum_info
+#define alc883_mux_enum_get alc_mux_enum_get
+
+static int alc883_mux_enum_put(struct snd_kcontrol *kcontrol,
+                               struct snd_ctl_elem_value *ucontrol)
+{
+        struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+        struct alc_spec *spec = codec->spec;
+        const struct hda_input_mux *imux = spec->input_mux;
+        unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+        static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
+        hda_nid_t nid = capture_mixers[adc_idx];
+        unsigned int *cur_val = &spec->cur_mux[adc_idx];
+        unsigned int i, idx;
+
+        idx = ucontrol->value.enumerated.item[0];
+        if (idx >= imux->num_items)
+                idx = imux->num_items - 1;
+        if (*cur_val == idx && ! codec->in_resume)
+                return 0;
+        for (i = 0; i < imux->num_items; i++) {
+                unsigned int v = (i == idx) ? 0x7000 : 0x7080;
+                snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+                                    v | (imux->items[i].index << 8));
+        }
+        *cur_val = idx;
+        return 1;
+}
+/*
+ * 2ch mode
+ */
+static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
+        { 2, NULL }
+};
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_3ST_ch2_init[] = {
+        { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+        { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+        { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+        { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+        { } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_3ST_ch6_init[] = {
+        { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+        { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+        { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+        { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+        { } /* end */
+};
+
+static struct hda_channel_mode alc883_3ST_6ch_modes[2] = {
+        { 2, alc883_3ST_ch2_init },
+        { 6, alc883_3ST_ch6_init },
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_sixstack_ch6_init[] = {
+        { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+        { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        { } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc883_sixstack_ch8_init[] = {
+        { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        { } /* end */
+};
+
+static struct hda_channel_mode alc883_sixstack_modes[2] = {
+        { 6, alc883_sixstack_ch6_init },
+        { 8, alc883_sixstack_ch8_init },
+};
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+
+static struct snd_kcontrol_new alc883_base_mixer[] = {
+        HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+        HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+        HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+        HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+        HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+        HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+        HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                /* .name = "Capture Source", */
+                .name = "Input Source",
+                .count = 2,
+                .info = alc883_mux_enum_info,
+                .get = alc883_mux_enum_get,
+                .put = alc883_mux_enum_put,
+        },
+        { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
+        HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+        HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+        HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+        HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+        HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+        HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                /* .name = "Capture Source", */
+                .name = "Input Source",
+                .count = 2,
+                .info = alc883_mux_enum_info,
+                .get = alc883_mux_enum_get,
+                .put = alc883_mux_enum_put,
+        },
+        { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
+        HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+        HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+        HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+        HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+        HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+        HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+        HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+        HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+        HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                /* .name = "Capture Source", */
+                .name = "Input Source",
+                .count = 2,
+                .info = alc883_mux_enum_info,
+                .get = alc883_mux_enum_get,
+                .put = alc883_mux_enum_put,
+        },
+        { } /* end */
+};
+
+static struct snd_kcontrol_new alc883_chmode_mixer[] = {
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "Channel Mode",
+                .info = alc_ch_mode_info,
+                .get = alc_ch_mode_get,
+                .put = alc_ch_mode_put,
+        },
+        { } /* end */
+};
+
+static struct hda_verb alc883_init_verbs[] = {
+        /* ADC1: mute amp left and right */
+        {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+        /* ADC2: mute amp left and right */
+        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+        /* Front mixer: unmute input/output amp left and right (volume = 0) */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+        /* Rear mixer */
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+        /* CLFE mixer */
+        {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+        /* Side mixer */
+        {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+        /* Front Pin: output 0 (0x0c) */
+        {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+        {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+        /* Rear Pin: output 1 (0x0d) */
+        {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+        {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+        /* CLFE Pin: output 2 (0x0e) */
+        {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+        {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+        /* Side Pin: output 3 (0x0f) */
+        {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+        {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+        /* Mic (rear) pin: input vref at 80% */
+        {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+        {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* Front Mic pin: input vref at 80% */
+        {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+        {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* Line In pin: input */
+        {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+        /* Line-2 In: Headphone output (output 0 - 0x0c) */
+        {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+        {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+        /* CD pin widget for input */
+        {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+
+        /* FIXME: use matrix-type input source selection */
+        /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+        /* Input mixer2 */
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+        /* Input mixer3 */
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+        { }
+};
+
+/*
+ * generic initialization of ADC, input mixers and output mixers
+ */
+static struct hda_verb alc883_auto_init_verbs[] = {
+        /*
+         * Unmute ADC0-2 and set the default input to mic-in
+         */
+        {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+        {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+        /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+         * mixer widget
+         * Note: PASD motherboards uses the Line In 2 as the input for front panel
+         * mic (mic 2)
+         */
+        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+        /*
+         * Set up output mixers (0x0c - 0x0f)
+         */
+        /* set vol=0 to output mixers */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        /* set up input amps for analog loopback */
+        /* Amp Indices: DAC = 0, mixer = 1 */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+        /* FIXME: use matrix-type input source selection */
+        /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+        /* Input mixer1 */
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+        //{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+        /* Input mixer2 */
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+        //{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+
+        { }
+};
+
+/* capture mixer elements */
+static struct snd_kcontrol_new alc883_capture_mixer[] = {
+        HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                /* The multiple "Capture Source" controls confuse alsamixer
+                 * So call somewhat different..
+                 * FIXME: the controls appear in the "playback" view!
+                 */
+                /* .name = "Capture Source", */
+                .name = "Input Source",
+                .count = 2,
+                .info = alc882_mux_enum_info,
+                .get = alc882_mux_enum_get,
+                .put = alc882_mux_enum_put,
+        },
+        { } /* end */
+};
+
+/* pcm configuration: identiacal with ALC880 */
+#define alc883_pcm_analog_playback      alc880_pcm_analog_playback
+#define alc883_pcm_analog_capture       alc880_pcm_analog_capture
+#define alc883_pcm_digital_playback     alc880_pcm_digital_playback
+#define alc883_pcm_digital_capture      alc880_pcm_digital_capture
+
+/*
+ * configuration and preset
+ */
+static struct hda_board_config alc883_cfg_tbl[] = {
+        { .modelname = "3stack-dig", .config = ALC883_3ST_2ch_DIG },
+        { .modelname = "6stack-dig", .config = ALC883_6ST_DIG },
+        { .modelname = "6stack-dig-demo", .config = ALC888_DEMO_BOARD },
+        { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668,
+          .config = ALC883_6ST_DIG }, /* MSI  */
+        { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668,
+          .config = ALC883_6ST_DIG }, /* Foxconn */
+        { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668,
+          .config = ALC883_3ST_6ch_DIG }, /* ECS to Intel*/
+        { .pci_subvendor = 0x108e, .pci_subdevice = 0x534d,
+          .config = ALC883_3ST_6ch },
+        { .modelname = "auto", .config = ALC883_AUTO },
+        {}
+};
+
+static struct alc_config_preset alc883_presets[] = {
+        [ALC883_3ST_2ch_DIG] = {
+                .mixers = { alc883_3ST_2ch_mixer },
+                .init_verbs = { alc883_init_verbs },
+                .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+                .dac_nids = alc883_dac_nids,
+                .dig_out_nid = ALC883_DIGOUT_NID,
+                .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+                .adc_nids = alc883_adc_nids,
+                .dig_in_nid = ALC883_DIGIN_NID,
+                .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+                .channel_mode = alc883_3ST_2ch_modes,
+                .input_mux = &alc883_capture_source,
+        },
+        [ALC883_3ST_6ch_DIG] = {
+                .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+                .init_verbs = { alc883_init_verbs },
+                .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+                .dac_nids = alc883_dac_nids,
+                .dig_out_nid = ALC883_DIGOUT_NID,
+                .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+                .adc_nids = alc883_adc_nids,
+                .dig_in_nid = ALC883_DIGIN_NID,
+                .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+                .channel_mode = alc883_3ST_6ch_modes,
+                .input_mux = &alc883_capture_source,
+        },      
+        [ALC883_3ST_6ch] = {
+                .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+                .init_verbs = { alc883_init_verbs },
+                .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+                .dac_nids = alc883_dac_nids,
+                .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+                .adc_nids = alc883_adc_nids,
+                .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+                .channel_mode = alc883_3ST_6ch_modes,
+                .input_mux = &alc883_capture_source,
+        },      
+        [ALC883_6ST_DIG] = {
+                .mixers = { alc883_base_mixer, alc883_chmode_mixer },
+                .init_verbs = { alc883_init_verbs },
+                .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+                .dac_nids = alc883_dac_nids,
+                .dig_out_nid = ALC883_DIGOUT_NID,
+                .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+                .adc_nids = alc883_adc_nids,
+                .dig_in_nid = ALC883_DIGIN_NID,
+                .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+                .channel_mode = alc883_sixstack_modes,
+                .input_mux = &alc883_capture_source,
+        },
+        [ALC888_DEMO_BOARD] = {
+                .mixers = { alc883_base_mixer, alc883_chmode_mixer },
+                .init_verbs = { alc883_init_verbs },
+                .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+                .dac_nids = alc883_dac_nids,
+                .dig_out_nid = ALC883_DIGOUT_NID,
+                .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
+                .adc_nids = alc883_adc_nids,
+                .dig_in_nid = ALC883_DIGIN_NID,
+                .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
+                .channel_mode = alc883_sixstack_modes,
+                .input_mux = &alc883_capture_source,
+        },
+};
+
+
+/*
+ * BIOS auto configuration
+ */
+static void alc883_auto_set_output_and_unmute(struct hda_codec *codec,
+                                              hda_nid_t nid, int pin_type,
+                                              int dac_idx)
+{
+        /* set as output */
+        struct alc_spec *spec = codec->spec;
+        int idx; 
+        
+        if (spec->multiout.dac_nids[dac_idx] == 0x25)
+                idx = 4;
+        else
+                idx = spec->multiout.dac_nids[dac_idx] - 2;
+
+        snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+                            pin_type);
+        snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+                            AMP_OUT_UNMUTE);
+        snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
+
+}
+
+static void alc883_auto_init_multi_out(struct hda_codec *codec)
+{
+        struct alc_spec *spec = codec->spec;
+        int i;
+
+        for (i = 0; i <= HDA_SIDE; i++) {
+                hda_nid_t nid = spec->autocfg.line_out_pins[i]; 
+                if (nid)
+                        alc883_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
+        }
+}
+
+static void alc883_auto_init_hp_out(struct hda_codec *codec)
+{
+        struct alc_spec *spec = codec->spec;
+        hda_nid_t pin;
+
+        pin = spec->autocfg.hp_pin;
+        if (pin) /* connect to front */
+                /* use dac 0 */
+                alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+}
+
+#define alc883_is_input_pin(nid)        alc880_is_input_pin(nid)
+#define ALC883_PIN_CD_NID               ALC880_PIN_CD_NID
+
+static void alc883_auto_init_analog_input(struct hda_codec *codec)
+{
+        struct alc_spec *spec = codec->spec;
+        int i;
+
+        for (i = 0; i < AUTO_PIN_LAST; i++) {
+                hda_nid_t nid = spec->autocfg.input_pins[i];
+                if (alc883_is_input_pin(nid)) {
+                        snd_hda_codec_write(codec, nid, 0,
+                                            AC_VERB_SET_PIN_WIDGET_CONTROL,
+                                            (i <= AUTO_PIN_FRONT_MIC ?
+                                             PIN_VREF80 : PIN_IN));
+                        if (nid != ALC883_PIN_CD_NID)
+                                snd_hda_codec_write(codec, nid, 0,
+                                                    AC_VERB_SET_AMP_GAIN_MUTE,
+                                                    AMP_OUT_MUTE);
+                }
+        }
+}
+
+/* almost identical with ALC880 parser... */
+static int alc883_parse_auto_config(struct hda_codec *codec)
+{
+        struct alc_spec *spec = codec->spec;
+        int err = alc880_parse_auto_config(codec);
+
+        if (err < 0)
+                return err;
+        else if (err > 0)
+                /* hack - override the init verbs */
+                spec->init_verbs[0] = alc883_auto_init_verbs;
+                spec->mixers[spec->num_mixers] = alc883_capture_mixer;
+                spec->num_mixers++;
+        return err;
+}
+
+/* additional initialization for auto-configuration model */
+static void alc883_auto_init(struct hda_codec *codec)
+{
+        alc883_auto_init_multi_out(codec);
+        alc883_auto_init_hp_out(codec);
+        alc883_auto_init_analog_input(codec);
+}
+
+static int patch_alc883(struct hda_codec *codec)
+{
+        struct alc_spec *spec;
+        int err, board_config;
+
+        spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+        if (spec == NULL)
+                return -ENOMEM;
+
+        codec->spec = spec;
+
+        board_config = snd_hda_check_board_config(codec, alc883_cfg_tbl);
+        if (board_config < 0 || board_config >= ALC883_MODEL_LAST) {
+                printk(KERN_INFO "hda_codec: Unknown model for ALC883, "
+                       "trying auto-probe from BIOS...\n");
+                board_config = ALC883_AUTO;
+        }
+
+        if (board_config == ALC883_AUTO) {
+                /* automatic parse from the BIOS config */
+                err = alc883_parse_auto_config(codec);
+                if (err < 0) {
+                        alc_free(codec);
+                        return err;
+                } else if (! err) {
+                        printk(KERN_INFO
+                               "hda_codec: Cannot set up configuration "
+                               "from BIOS.  Using base mode...\n");
+                        board_config = ALC883_3ST_2ch_DIG;
+                }
+        }
+
+        if (board_config != ALC883_AUTO)
+                setup_preset(spec, &alc883_presets[board_config]);
+
+        spec->stream_name_analog = "ALC883 Analog";
+        spec->stream_analog_playback = &alc883_pcm_analog_playback;
+        spec->stream_analog_capture = &alc883_pcm_analog_capture;
+
+        spec->stream_name_digital = "ALC883 Digital";
+        spec->stream_digital_playback = &alc883_pcm_digital_playback;
+        spec->stream_digital_capture = &alc883_pcm_digital_capture;
+
+        spec->adc_nids = alc883_adc_nids;
+        spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+
+        codec->patch_ops = alc_patch_ops;
+        if (board_config == ALC883_AUTO)
+                spec->init_hook = alc883_auto_init;
+
+        return 0;
+}
+
+/*
  * ALC262 support
  */
 
@@ -4542,6 +5261,28 @@ static struct snd_kcontrol_new alc262_base_mixer[] = {
         { } /* end */
 };
 
+static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
+        HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+        HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+
+        HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+        HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+        HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+        HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+        HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
+        HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT),
+        HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
+        HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
+        { } /* end */
+};
+
 #define alc262_capture_mixer            alc882_capture_mixer
 #define alc262_capture_alt_mixer        alc882_capture_alt_mixer
 
@@ -4645,6 +5386,17 @@ static struct hda_input_mux alc262_fujitsu_capture_source = {
         },
 };
 
+static struct hda_input_mux alc262_HP_capture_source = {
+        .num_items = 5,
+        .items = {
+                { "Mic", 0x0 },
+                { "Front Mic", 0x3 },
+                { "Line", 0x2 },
+                { "CD", 0x4 },
+                { "AUX IN", 0x6 },
+        },
+};
+
 /* mute/unmute internal speaker according to the hp jack and mute state */
 static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
 {
@@ -4868,6 +5620,93 @@ static struct hda_verb alc262_volume_init_verbs[] = {
         { }
 };
 
+static struct hda_verb alc262_HP_BPC_init_verbs[] = {
+        /*
+         * Unmute ADC0-2 and set the default input to mic-in
+         */
+        {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+        {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+        {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+        {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+        /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
+         * mixer widget
+         * Note: PASD motherboards uses the Line In 2 as the input for front panel
+         * mic (mic 2)
+         */
+        /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
+        {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
+        
+        /*
+         * Set up output mixers (0x0c - 0x0e)
+         */
+        /* set vol=0 to output mixers */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+        /* set up input amps for analog loopback */
+        /* Amp Indices: DAC = 0, mixer = 1 */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+        {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+        {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+
+        {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
+        {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+        {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+
+        {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+        {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+
+        {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+        {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+        {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+        {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+        {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+        {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+        {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+        {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+        {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+        {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
+        {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+        {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
+
+
+        /* FIXME: use matrix-type input source selection */
+        /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+        /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+        /* Input mixer2 */
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+        {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+        /* Input mixer3 */
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+        {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+
+        { }
+};
+
 /* pcm configuration: identiacal with ALC880 */
 #define alc262_pcm_analog_playback      alc880_pcm_analog_playback
 #define alc262_pcm_analog_capture       alc880_pcm_analog_capture
@@ -4928,7 +5767,16 @@ static void alc262_auto_init(struct hda_codec *codec)
 static struct hda_board_config alc262_cfg_tbl[] = {
         { .modelname = "basic", .config = ALC262_BASIC },
         { .modelname = "fujitsu", .config = ALC262_FUJITSU },
-        { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397, .config = ALC262_FUJITSU },
+        { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397,
+          .config = ALC262_FUJITSU },
+        { .pci_subvendor = 0x103c, .pci_subdevice = 0x208c,
+          .config = ALC262_HP_BPC }, /* xw4400 */
+        { .pci_subvendor = 0x103c, .pci_subdevice = 0x3014,
+          .config = ALC262_HP_BPC }, /* xw6400 */
+        { .pci_subvendor = 0x103c, .pci_subdevice = 0x3015,
+          .config = ALC262_HP_BPC }, /* xw8400 */
+        { .pci_subvendor = 0x103c, .pci_subdevice = 0x12fe,
+          .config = ALC262_HP_BPC }, /* xw9400 */
         { .modelname = "auto", .config = ALC262_AUTO },
         {}
 };
@@ -4956,6 +5804,16 @@ static struct alc_config_preset alc262_presets[] = {
                 .input_mux = &alc262_fujitsu_capture_source,
                 .unsol_event = alc262_fujitsu_unsol_event,
         },
+        [ALC262_HP_BPC] = {
+                .mixers = { alc262_HP_BPC_mixer },
+                .init_verbs = { alc262_HP_BPC_init_verbs },
+                .num_dacs = ARRAY_SIZE(alc262_dac_nids),
+                .dac_nids = alc262_dac_nids,
+                .hp_nid = 0x03,
+                .num_channel_mode = ARRAY_SIZE(alc262_modes),
+                .channel_mode = alc262_modes,
+                .input_mux = &alc262_HP_capture_source,
+        },      
 };
 
 static int patch_alc262(struct hda_codec *codec)
@@ -4981,8 +5839,10 @@ static int patch_alc262(struct hda_codec *codec)
 #endif
 
         board_config = snd_hda_check_board_config(codec, alc262_cfg_tbl);
+        
         if (board_config < 0 || board_config >= ALC262_MODEL_LAST) {
-                printk(KERN_INFO "hda_codec: Unknown model for ALC262, trying auto-probe from BIOS...\n");
+                printk(KERN_INFO "hda_codec: Unknown model for ALC262, "
+                       "trying auto-probe from BIOS...\n");
                 board_config = ALC262_AUTO;
         }
 
@@ -4993,7 +5853,9 @@ static int patch_alc262(struct hda_codec *codec)
                         alc_free(codec);
                         return err;
                 } else if (! err) {
-                        printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS.  Using base mode...\n");
+                        printk(KERN_INFO
+                               "hda_codec: Cannot set up configuration "
+                               "from BIOS.  Using base mode...\n");
                         board_config = ALC262_BASIC;
                 }
         }
@@ -5034,7 +5896,6 @@ static int patch_alc262(struct hda_codec *codec)
         return 0;
 }
 
-
 /*
  *  ALC861 channel source setting (2/6 channel selection for 3-stack)
  */
@@ -5049,9 +5910,11 @@ static struct hda_verb alc861_threestack_ch2_init[] = {
         /* set pin widget 18h (mic1/2) for input, for mic also enable the vref */
         { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
 
-        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
-        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, //mic
-        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, //line in
+        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
+#if 0
+        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
+        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
+#endif
         { } /* end */
 };
 /*
@@ -5065,11 +5928,13 @@ static struct hda_verb alc861_threestack_ch6_init[] = {
         { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
 
         { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
-        { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
+        { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
 
-        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
-        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, //mic
-        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, //line in
+        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+#if 0
+        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
+        { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
+#endif
         { } /* end */
 };
 
@@ -5353,6 +6218,11 @@ static hda_nid_t alc861_dac_nids[4] = {
         0x03, 0x06, 0x05, 0x04
 };
 
+static hda_nid_t alc660_dac_nids[3] = {
+        /* front, clfe, surround */
+        0x03, 0x05, 0x06
+};
+
 static hda_nid_t alc861_adc_nids[1] = {
         /* ADC0-2 */
         0x08,
@@ -5605,7 +6475,10 @@ static void alc861_auto_init(struct hda_codec *codec)
  */
 static struct hda_board_config alc861_cfg_tbl[] = {
         { .modelname = "3stack", .config = ALC861_3ST },
-        { .pci_subvendor = 0x8086, .pci_subdevice = 0xd600, .config = ALC861_3ST },
+        { .pci_subvendor = 0x8086, .pci_subdevice = 0xd600,
+          .config = ALC861_3ST },
+        { .pci_subvendor = 0x1043, .pci_subdevice = 0x81e7,
+          .config = ALC660_3ST },
         { .modelname = "3stack-dig", .config = ALC861_3ST_DIG },
         { .modelname = "6stack-dig", .config = ALC861_6ST_DIG },
         { .modelname = "auto", .config = ALC861_AUTO },
@@ -5648,6 +6521,17 @@ static struct alc_config_preset alc861_presets[] = {
                 .adc_nids = alc861_adc_nids,
                 .input_mux = &alc861_capture_source,
         },
+        [ALC660_3ST] = {
+                .mixers = { alc861_3ST_mixer },
+                .init_verbs = { alc861_threestack_init_verbs },
+                .num_dacs = ARRAY_SIZE(alc660_dac_nids),
+                .dac_nids = alc660_dac_nids,
+                .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
+                .channel_mode = alc861_threestack_modes,
+                .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
+                .adc_nids = alc861_adc_nids,
+                .input_mux = &alc861_capture_source,
+        },
 };      
 
 
@@ -5664,8 +6548,10 @@ static int patch_alc861(struct hda_codec *codec)
         codec->spec = spec;     
 
         board_config = snd_hda_check_board_config(codec, alc861_cfg_tbl);
+
         if (board_config < 0 || board_config >= ALC861_MODEL_LAST) {
-                printk(KERN_INFO "hda_codec: Unknown model for ALC861, trying auto-probe from BIOS...\n");
+                printk(KERN_INFO "hda_codec: Unknown model for ALC861, "
+                       "trying auto-probe from BIOS...\n");
                 board_config = ALC861_AUTO;
         }
 
@@ -5676,7 +6562,9 @@ static int patch_alc861(struct hda_codec *codec)
                         alc_free(codec);
                         return err;
                 } else if (! err) {
-                        printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS.  Using base mode...\n");
+                        printk(KERN_INFO
+                               "hda_codec: Cannot set up configuration "
+                               "from BIOS.  Using base mode...\n");
                    board_config = ALC861_3ST_DIG;
                 }
         }
@@ -5707,8 +6595,12 @@ struct hda_codec_preset snd_hda_preset_realtek[] = {
         { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
         { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
         { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
-        { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 },
+        { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 },
         { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
-        { .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 },
+        { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
+        { .id = 0x10ec0861, .rev = 0x100300, .name = "ALC861",
+          .patch = patch_alc861 },
+        { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
+          .patch = patch_alc861 },
         {} /* terminator */
 };
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 36f199442fdc..fb4bed0759d1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -42,6 +42,9 @@
 #define STAC_D945GTP3           1
 #define STAC_D945GTP5           2
 #define STAC_MACMINI            3
+#define STAC_D965_2112          4
+#define STAC_D965_284B          5
+#define STAC_922X_MODELS        6       /* number of 922x models */
 
 struct sigmatel_spec {
         struct snd_kcontrol_new *mixers[4];
@@ -107,10 +110,24 @@ static hda_nid_t stac922x_adc_nids[2] = {
         0x06, 0x07,
 };
 
+static hda_nid_t stac9227_adc_nids[2] = {
+        0x07, 0x08,
+};
+
+#if 0
+static hda_nid_t d965_2112_dac_nids[3] = {
+        0x02, 0x03, 0x05,
+};
+#endif
+
 static hda_nid_t stac922x_mux_nids[2] = {
         0x12, 0x13,
 };
 
+static hda_nid_t stac9227_mux_nids[2] = {
+        0x15, 0x16,
+};
+
 static hda_nid_t stac927x_adc_nids[3] = {
         0x07, 0x08, 0x09
 };
@@ -173,6 +190,24 @@ static struct hda_verb stac922x_core_init[] = {
         {}
 };
 
+static struct hda_verb stac9227_core_init[] = {
+        /* set master volume and direct control */      
+        { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+        /* unmute node 0x1b */
+        { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+        {}
+};
+
+static struct hda_verb d965_2112_core_init[] = {
+        /* set master volume and direct control */      
+        { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
+        /* unmute node 0x1b */
+        { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+        /* select node 0x03 as DAC */   
+        { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01},
+        {}
+};
+
 static struct hda_verb stac927x_core_init[] = {
         /* set master volume and direct control */      
         { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -212,6 +247,21 @@ static struct snd_kcontrol_new stac922x_mixer[] = {
         { } /* end */
 };
 
+/* This needs to be generated dynamically based on sequence */
+static struct snd_kcontrol_new stac9227_mixer[] = {
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+                .name = "Input Source",
+                .count = 1,
+                .info = stac92xx_mux_enum_info,
+                .get = stac92xx_mux_enum_get,
+                .put = stac92xx_mux_enum_put,
+        },
+        HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
+        HDA_CODEC_MUTE("Capture Switch", 0x1b, 0x0, HDA_OUTPUT),
+        { } /* end */
+};
+
 static snd_kcontrol_new_t stac927x_mixer[] = {
         {
                 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -291,11 +341,17 @@ static unsigned int d945gtp5_pin_configs[10] = {
         0x02a19320, 0x40000100,
 };
 
-static unsigned int *stac922x_brd_tbl[] = {
-        ref922x_pin_configs,
-        d945gtp3_pin_configs,
-        d945gtp5_pin_configs,
-        NULL,           /* STAC_MACMINI */
+static unsigned int d965_2112_pin_configs[10] = {
+        0x0221401f, 0x40000100, 0x40000100, 0x01014011,
+        0x01a19021, 0x01813024, 0x01452130, 0x40000100,
+        0x02a19320, 0x40000100,
+};
+
+static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
+        [STAC_REF] =    ref922x_pin_configs,
+        [STAC_D945GTP3] = d945gtp3_pin_configs,
+        [STAC_D945GTP5] = d945gtp5_pin_configs,
+        [STAC_D965_2112] = d965_2112_pin_configs,
 };
 
 static struct hda_board_config stac922x_cfg_tbl[] = {
@@ -330,6 +386,12 @@ static struct hda_board_config stac922x_cfg_tbl[] = {
         { .pci_subvendor = 0x8384,
           .pci_subdevice = 0x7680,
           .config = STAC_MACMINI },     /* Apple Mac Mini (early 2006) */
+        { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+          .pci_subdevice = 0x2112,
+          .config = STAC_D965_2112 },
+        { .pci_subvendor = PCI_VENDOR_ID_INTEL,
+          .pci_subdevice = 0x284b,
+          .config = STAC_D965_284B },
         {} /* terminator */
 };
 
@@ -713,7 +775,8 @@ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cf
  * A and B is not supported.
  */
 /* fill in the dac_nids table from the parsed pin configuration */
-static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
+static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
+                                       const struct auto_pin_cfg *cfg)
 {
         struct sigmatel_spec *spec = codec->spec;
         hda_nid_t nid;
@@ -732,10 +795,13 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, const struct aut
 }
 
 /* add playback controls from the parsed DAC table */
-static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, const struct auto_pin_cfg *cfg)
+static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
+                                               const struct auto_pin_cfg *cfg)
 {
         char name[32];
-        static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" };
+        static const char *chname[4] = {
+                "Front", "Surround", NULL /*CLFE*/, "Side"
+        };
         hda_nid_t nid;
         int i, err;
 
@@ -893,10 +959,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
                 return err;
         if (! spec->autocfg.line_outs)
                 return 0; /* can't find valid pin config */
+
         if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0)
                 return err;
-        if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0)
-                return err;
+        if (spec->multiout.num_dacs == 0)
+                if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0)
+                        return err;
 
         if ((err = stac92xx_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
             (err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg)) < 0 ||
@@ -1194,7 +1262,8 @@ static int patch_stac922x(struct hda_codec *codec)
         codec->spec = spec;
         spec->board_config = snd_hda_check_board_config(codec, stac922x_cfg_tbl);
         if (spec->board_config < 0)
-                snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, using BIOS defaults\n");
+                snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
+                            "using BIOS defaults\n");
         else if (stac922x_brd_tbl[spec->board_config] != NULL) {
                 spec->num_pins = 10;
                 spec->pin_nids = stac922x_pin_nids;
@@ -1210,6 +1279,25 @@ static int patch_stac922x(struct hda_codec *codec)
         spec->mixer = stac922x_mixer;
 
         spec->multiout.dac_nids = spec->dac_nids;
+        
+        switch (spec->board_config) {
+        case STAC_D965_2112:
+                spec->adc_nids = stac9227_adc_nids;
+                spec->mux_nids = stac9227_mux_nids;
+#if 0
+                spec->multiout.dac_nids = d965_2112_dac_nids;
+                spec->multiout.num_dacs = ARRAY_SIZE(d965_2112_dac_nids);
+#endif
+                spec->init = d965_2112_core_init;
+                spec->mixer = stac9227_mixer;
+                break;
+        case STAC_D965_284B:
+                spec->adc_nids = stac9227_adc_nids;
+                spec->mux_nids = stac9227_mux_nids;
+                spec->init = stac9227_core_init;
+                spec->mixer = stac9227_mixer;
+                break;
+        }
 
         err = stac92xx_parse_auto_config(codec, 0x08, 0x09);
         if (err < 0) {
diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c
index b5754b32b802..fec9440cb310 100644
--- a/sound/pci/ice1712/revo.c
+++ b/sound/pci/ice1712/revo.c
@@ -87,12 +87,25 @@ static void revo_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
  * initialize the chips on M-Audio Revolution cards
  */
 
+static unsigned int revo71_num_stereo_front[] = {2};
+static char *revo71_channel_names_front[] = {"PCM Playback Volume"};
+
+static unsigned int revo71_num_stereo_surround[] = {1, 1, 2, 2};
+static char *revo71_channel_names_surround[] = {"PCM Center Playback Volume", "PCM LFE Playback Volume",
+                                                "PCM Side Playback Volume", "PCM Rear Playback Volume"};
+
+static unsigned int revo51_num_stereo[] = {2, 1, 1, 2};
+static char *revo51_channel_names[] = {"PCM Playback Volume", "PCM Center Playback Volume",
+                                        "PCM LFE Playback Volume", "PCM Rear Playback Volume"};
+
 static struct snd_akm4xxx akm_revo_front __devinitdata = {
         .type = SND_AK4381,
         .num_dacs = 2,
         .ops = {
                 .set_rate_val = revo_set_rate_val
-        }
+        },
+        .num_stereo = revo71_num_stereo_front,
+        .channel_names = revo71_channel_names_front
 };
 
 static struct snd_ak4xxx_private akm_revo_front_priv __devinitdata = {
@@ -113,7 +126,9 @@ static struct snd_akm4xxx akm_revo_surround __devinitdata = {
         .num_dacs = 6,
         .ops = {
                 .set_rate_val = revo_set_rate_val
-        }
+        },
+        .num_stereo = revo71_num_stereo_surround,
+        .channel_names = revo71_channel_names_surround
 };
 
 static struct snd_ak4xxx_private akm_revo_surround_priv __devinitdata = {
@@ -133,7 +148,9 @@ static struct snd_akm4xxx akm_revo51 __devinitdata = {
         .num_dacs = 6,
         .ops = {
                 .set_rate_val = revo_set_rate_val
-        }
+        },
+        .num_stereo = revo51_num_stereo,
+        .channel_names = revo51_channel_names
 };
 
 static struct snd_ak4xxx_private akm_revo51_priv __devinitdata = {
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 627de9525a32..d32d83d970cc 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -3096,6 +3096,32 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev)
 }
 
 /*
+ * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely
+ * documented in the device's data sheet.
+ */
+static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value)
+{
+        u8 buf[4];
+        buf[0] = 0x20;
+        buf[1] = value & 0xff;
+        buf[2] = (value >> 8) & 0xff;
+        buf[3] = reg;
+        return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION,
+                               USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT,
+                               0, 0, &buf, 4, 1000);
+}
+
+static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
+{
+        /*
+         * Enable line-out driver mode, set headphone source to front
+         * channels, enable stereo mic.
+         */
+        return snd_usb_cm106_write_int_reg(dev, 2, 0x8004);
+}
+
+
+/*
  * Setup quirks
  */
 #define AUDIOPHILE_SET                  0x01 /* if set, parse device_setup */
@@ -3365,6 +3391,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
                         goto __err_val;
         }
 
+        /* C-Media CM106 / Turtle Beach Audio Advantage Roadie */
+        if (id == USB_ID(0x10f5, 0x0200)) {
+                if (snd_usb_cm106_boot_quirk(dev) < 0)
+                        goto __err_val;
+        }
+
         /*
          * found a config.  now register to ALSA
          */