diff options
Diffstat (limited to 'sound')
67 files changed, 11331 insertions, 241 deletions
diff --git a/sound/Makefile b/sound/Makefile index a682ea3..1f60797 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 a85194f..2f4334d 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 2c6eb77..bab9754 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(¬if->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 04a7238..cbc8a3b 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 d532d27..7368b7d 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/arm/aaci.c b/sound/arm/aaci.c index 5f22d70..6b18225 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -779,8 +779,9 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev) strlcpy(card->driver, DRIVER_NAME, sizeof(card->driver)); strlcpy(card->shortname, "ARM AC'97 Interface", sizeof(card->shortname)); snprintf(card->longname, sizeof(card->longname), - "%s at 0x%08lx, irq %d", - card->shortname, dev->res.start, dev->irq[0]); + "%s at 0x%016llx, irq %d", + card->shortname, (unsigned long long)dev->res.start, + dev->irq[0]); aaci = card->private_data; mutex_init(&aaci->ac97_sem); diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 4262a1c..b292752 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_memory.h b/sound/core/seq/seq_memory.h index 39c60d9..63e9143 100644 --- a/sound/core/seq/seq_memory.h +++ b/sound/core/seq/seq_memory.h @@ -31,7 +31,7 @@ struct snd_seq_event_cell { struct snd_seq_event_cell *next; /* next cell */ }; -/* design note: the pool is a contigious block of memory, if we dynamicly +/* design note: the pool is a contiguous block of memory, if we dynamicly want to add additional cells to the pool be better store this in another pool as we need to know the base address of the pool when releasing memory. */ diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 334579a9..d467b4f 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/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index d3cbbb0..8b80024 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -160,8 +160,9 @@ static int __devinit snd_mpu401_pnp(int dev, struct pnp_dev *device, return -ENODEV; } if (pnp_port_len(device, 0) < IO_EXTENT) { - snd_printk(KERN_ERR "PnP port length is %ld, expected %d\n", - pnp_port_len(device, 0), IO_EXTENT); + snd_printk(KERN_ERR "PnP port length is %llu, expected %d\n", + (unsigned long long)pnp_port_len(device, 0), + IO_EXTENT); return -ENODEV; } port[dev] = pnp_port_start(device, 0); diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 045e32a..dc7cc20 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/isa/es18xx.c b/sound/isa/es18xx.c index e6945db..af60b0b 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2088,7 +2088,8 @@ static int __devinit snd_audiodrive_pnp(int dev, struct snd_audiodrive *acard, kfree(cfg); return -EAGAIN; } - snd_printdd("pnp: port=0x%lx\n", pnp_port_start(acard->devc, 0)); + snd_printdd("pnp: port=0x%llx\n", + (unsigned long long)pnp_port_start(acard->devc, 0)); /* PnP initialization */ pdev = acard->dev; pnp_init_resource_table(cfg); diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 866300f..c1c86e0 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -611,10 +611,10 @@ static int __devinit snd_interwave_pnp(int dev, struct snd_interwave *iwcard, if (dma2[dev] >= 0) dma2[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - snd_printdd("isapnp IW: sb port=0x%lx, gf1 port=0x%lx, codec port=0x%lx\n", - pnp_port_start(pdev, 0), - pnp_port_start(pdev, 1), - pnp_port_start(pdev, 2)); + snd_printdd("isapnp IW: sb port=0x%llx, gf1 port=0x%llx, codec port=0x%llx\n", + (unsigned long long)pnp_port_start(pdev, 0), + (unsigned long long)pnp_port_start(pdev, 1), + (unsigned long long)pnp_port_start(pdev, 2)); snd_printdd("isapnp IW: dma1=%i, dma2=%i, irq=%i\n", dma1[dev], dma2[dev], irq[dev]); #ifdef SNDRV_STB /* Tone Control initialization */ diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 7f7f05f..d64e67f 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -327,7 +327,8 @@ static int __devinit snd_card_sb16_pnp(int dev, struct snd_card_sb16 *acard, goto __wt_error; } awe_port[dev] = pnp_port_start(pdev, 0); - snd_printdd("pnp SB16: wavetable port=0x%lx\n", pnp_port_start(pdev, 0)); + snd_printdd("pnp SB16: wavetable port=0x%llx\n", + (unsigned long long)pnp_port_start(pdev, 0)); } else { __wt_error: if (pdev) { diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 3b8cdbc..f4980ca 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -493,6 +493,19 @@ config SOUND_CS4232 See <file:Documentation/sound/oss/CS4232> for more information on configuring this card. +config SOUND_SSCAPE + tristate "Ensoniq SoundScape support" + depends on SOUND_OSS + help + Answer Y if you have a sound card based on the Ensoniq SoundScape + chipset. Such cards are being manufactured at least by Ensoniq, Spea + and Reveal (Reveal makes also other cards). + + If you compile the driver into the kernel, you have to add + "sscape=<io>,<irq>,<dma>,<mpuio>,<mpuirq>" to the kernel command + line. + + config SOUND_VMIDI tristate "Loopback MIDI device support" depends on SOUND_OSS diff --git a/sound/oss/cs4232.c b/sound/oss/cs4232.c index c7f86f0..80f6c08 100644 --- a/sound/oss/cs4232.c +++ b/sound/oss/cs4232.c @@ -405,7 +405,7 @@ static const struct pnp_device_id cs4232_pnp_table[] = { MODULE_DEVICE_TABLE(pnp, cs4232_pnp_table); -static int cs4232_pnp_probe(struct pnp_dev *dev, const struct pnp_device_id *dev_id) +static int __init cs4232_pnp_probe(struct pnp_dev *dev, const struct pnp_device_id *dev_id) { struct address_info *isapnpcfg; diff --git a/sound/oss/forte.c b/sound/oss/forte.c index 0294eec..44e5780 100644 --- a/sound/oss/forte.c +++ b/sound/oss/forte.c @@ -2035,8 +2035,9 @@ forte_probe (struct pci_dev *pci_dev, const struct pci_device_id *pci_id) pci_set_drvdata (pci_dev, chip); - printk (KERN_INFO PFX "FM801 chip found at 0x%04lX-0x%04lX IRQ %u\n", - chip->iobase, pci_resource_end (pci_dev, 0), chip->irq); + printk (KERN_INFO PFX "FM801 chip found at 0x%04lX-0x%16llX IRQ %u\n", + chip->iobase, (unsigned long long)pci_resource_end (pci_dev, 0), + chip->irq); /* Power it up */ if ((ret = forte_chip_init (chip)) == 0) diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index fae05fe..180e95c 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -97,19 +97,19 @@ * * The documentation is an adventure: it's close but not fully accurate. I * found out that after a reset some registers are *NOT* reset, though the - * docs say the would be. Interresting ones are 0x7f, 0x7d and 0x7a. They are - * related to the Audio 2 channel. I also was suprised about the consequenses + * docs say the would be. Interesting ones are 0x7f, 0x7d and 0x7a. They are + * related to the Audio 2 channel. I also was surprised about the consequences * of writing 0x00 to 0x7f (which should be done by reset): The ES1887 moves * into ES1888 mode. This means that it claims IRQ 11, which happens to be my * ISDN adapter. Needless to say it no longer worked. I now understand why * after rebooting 0x7f already was 0x05, the value of my choice: the BIOS * did it. * - * Oh, and this is another trap: in ES1887 docs mixer register 0x70 is decribed - * as if it's exactly the same as register 0xa1. This is *NOT* true. The - * description of 0x70 in ES1869 docs is accurate however. + * Oh, and this is another trap: in ES1887 docs mixer register 0x70 is + * described as if it's exactly the same as register 0xa1. This is *NOT* true. + * The description of 0x70 in ES1869 docs is accurate however. * Well, the assumption about ES1869 was wrong: register 0x70 is very much - * like register 0xa1, except that bit 7 is allways 1, whatever you want + * like register 0xa1, except that bit 7 is always 1, whatever you want * it to be. * * When using audio 2 mixer register 0x72 seems te be meaningless. Only 0xa2 @@ -117,10 +117,10 @@ * * Software reset not being able to reset all registers is great! Especially * the fact that register 0x78 isn't reset is great when you wanna change back - * to single dma operation (simplex): audio 2 is still operation, and uses the - * same dma as audio 1: your ess changes into a funny echo machine. + * to single dma operation (simplex): audio 2 is still operational, and uses + * the same dma as audio 1: your ess changes into a funny echo machine. * - * Received the new that ES1688 is detected as a ES1788. Did some thinking: + * Received the news that ES1688 is detected as a ES1788. Did some thinking: * the ES1887 detection scheme suggests in step 2 to try if bit 3 of register * 0x64 can be changed. This is inaccurate, first I inverted the * check: "If * can be modified, it's a 1688", which lead to a correct detection @@ -135,7 +135,7 @@ * About recognition of ESS chips * * The distinction of ES688, ES1688, ES1788, ES1887 and ES1888 is described in - * a (preliminary ??) datasheet on ES1887. It's aim is to identify ES1887, but + * a (preliminary ??) datasheet on ES1887. Its aim is to identify ES1887, but * during detection the text claims that "this chip may be ..." when a step * fails. This scheme is used to distinct between the above chips. * It appears however that some PnP chips like ES1868 are recognized as ES1788 @@ -156,9 +156,9 @@ * * The existing ES1688 support didn't take care of the ES1688+ recording * levels very well. Whenever a device was selected (recmask) for recording - * it's recording level was loud, and it couldn't be changed. The fact that + * its recording level was loud, and it couldn't be changed. The fact that * internal register 0xb4 could take care of RECLEV, didn't work meaning until - * it's value was restored every time the chip was reset; this reset the + * its value was restored every time the chip was reset; this reset the * value of 0xb4 too. I guess that's what 4front also had (have?) trouble with. * * About ES1887 support: @@ -169,9 +169,9 @@ * the latter case the recording volumes are 0. * Now recording levels of inputs can be controlled, by changing the playback * levels. Futhermore several devices can be recorded together (which is not - * possible with the ES1688. + * possible with the ES1688). * Besides the separate recording level control for each input, the common - * recordig level can also be controlled by RECLEV as described above. + * recording level can also be controlled by RECLEV as described above. * * Not only ES1887 have this recording mixer. I know the following from the * documentation: diff --git a/sound/oss/via82cxxx_audio.c b/sound/oss/via82cxxx_audio.c index 1a921ee7..29a6e0c 100644 --- a/sound/oss/via82cxxx_audio.c +++ b/sound/oss/via82cxxx_audio.c @@ -24,6 +24,7 @@ #include <linux/fs.h> #include <linux/mm.h> #include <linux/pci.h> +#include <linux/poison.h> #include <linux/init.h> #include <linux/interrupt.h> #include <linux/proc_fs.h> @@ -308,7 +309,7 @@ struct via_info { unsigned sixchannel: 1; /* 8233/35 with 6 channel support */ unsigned volume: 1; - int locked_rate : 1; + unsigned locked_rate : 1; int mixer_vol; /* 8233/35 volume - not yet implemented */ @@ -3522,7 +3523,7 @@ err_out_have_mixer: err_out_kfree: #ifndef VIA_NDEBUG - memset (card, 0xAB, sizeof (*card)); /* poison memory */ + memset (card, OSS_POISON_FREE, sizeof (*card)); /* poison memory */ #endif kfree (card); @@ -3559,7 +3560,7 @@ static void __devexit via_remove_one (struct pci_dev *pdev) via_ac97_cleanup (card); #ifndef VIA_NDEBUG - memset (card, 0xAB, sizeof (*card)); /* poison memory */ + memset (card, OSS_POISON_FREE, sizeof (*card)); /* poison memory */ #endif kfree (card); diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index d37346b..23e54ce 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 cba5105..e06736d 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 7f197c7..094cfc1 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/bt87x.c b/sound/pci/bt87x.c index c33642d..497ed6b 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -888,8 +888,9 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci, strcpy(card->driver, "Bt87x"); sprintf(card->shortname, "Brooktree Bt%x", pci->device); - sprintf(card->longname, "%s at %#lx, irq %i", - card->shortname, pci_resource_start(pci, 0), chip->irq); + sprintf(card->longname, "%s at %#llx, irq %i", + card->shortname, (unsigned long long)pci_resource_start(pci, 0), + chip->irq); strcpy(card->mixername, "Bt87x"); err = snd_card_register(card); diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index f0a4869..5450a9e 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -143,7 +143,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, if (dma->periods == periods && dma->period_bytes == period_bytes) return 0; - /* the u32 cast is okay because in snd*create we succesfully told + /* the u32 cast is okay because in snd*create we successfully told pci alloc that we're only 32 bit capable so the uppper will be 0 */ addr = (u32) substream->runtime->dma_addr; desc_addr = (u32) dma->desc_buf.addr; diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile new file mode 100644 index 0000000..7b576ae --- /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 0000000..b7108e2 --- /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 0000000..4159e3b --- /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 0000000..e59a982 --- /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 0000000..79938ee --- /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 0000000..12099fe --- /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 0000000..d26a1d1 --- /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 0000000..43b408a --- /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 0000000..7e88c96 --- /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 0000000..9f439ea --- /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 0000000..42afa83 --- /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 0000000..e55ee00 --- /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 0000000..3aa37e7 --- /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 0000000..29d6d12 --- /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 0000000..2757c89 --- /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 0000000..e464d72 --- /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 0000000..144fc56 --- /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 0000000..bfd2467 --- /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 0000000..d6ac773 --- /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 0000000..8ed7ff1 --- /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 0000000..500e150 --- /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 0000000..a8788e9 --- /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 0000000..f3ad13d --- /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 0000000..e503d74 --- /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 0000000..990c9a6 --- /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 0000000..d4581fd --- /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 0000000..7ec5b63 --- /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 0000000..be40c64 --- /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 0000000..891c705 --- /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 0000000..e31f0f1 --- /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 0000000..5dc512a --- /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 0000000..c0b4bf0 --- /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 8c2a817..23201f3 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 dd4e00a..33b7d58 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 98b9f16..18d1052 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 36f1994..fb4bed0 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 b5754b3..fec9440 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/pci/sonicvibes.c b/sound/pci/sonicvibes.c index dcf4029..e551160 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1441,10 +1441,10 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci, strcpy(card->driver, "SonicVibes"); strcpy(card->shortname, "S3 SonicVibes"); - sprintf(card->longname, "%s rev %i at 0x%lx, irq %i", + sprintf(card->longname, "%s rev %i at 0x%llx, irq %i", card->shortname, sonic->revision, - pci_resource_start(pci, 1), + (unsigned long long)pci_resource_start(pci, 1), sonic->irq); if ((err = snd_sonicvibes_pcm(sonic, 0, NULL)) < 0) { diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index b678814..be98f63 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -1170,9 +1170,10 @@ int __init snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) chip->rsrc[i].start + 1, rnames[i]) == NULL) { printk(KERN_ERR "snd: can't request rsrc " - " %d (%s: 0x%08lx:%08lx)\n", - i, rnames[i], chip->rsrc[i].start, - chip->rsrc[i].end); + " %d (%s: 0x%016lx:%016lx)\n", + i, rnames[i], + (unsigned long long)chip->rsrc[i].start, + (unsigned long long)chip->rsrc[i].end); err = -ENODEV; goto __error; } @@ -1201,9 +1202,10 @@ int __init snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) chip->rsrc[i].start + 1, rnames[i]) == NULL) { printk(KERN_ERR "snd: can't request rsrc " - " %d (%s: 0x%08lx:%08lx)\n", - i, rnames[i], chip->rsrc[i].start, - chip->rsrc[i].end); + " %d (%s: 0x%016llx:%016llx)\n", + i, rnames[i], + (unsigned long long)chip->rsrc[i].start, + (unsigned long long)chip->rsrc[i].end); err = -ENODEV; goto __error; } diff --git a/sound/ppc/toonie.c b/sound/ppc/toonie.c deleted file mode 100644 index e69de29..0000000 --- a/sound/ppc/toonie.c +++ /dev/null diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index da54d04..d9d14c2 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -2037,10 +2037,10 @@ static int __init cs4231_sbus_attach(struct sbus_dev *sdev) if (err) return err; - sprintf(card->longname, "%s at 0x%02lx:0x%08lx, irq %d", + sprintf(card->longname, "%s at 0x%02lx:0x%016lx, irq %d", card->shortname, rp->flags & 0xffL, - rp->start, + (unsigned long long)rp->start, sdev->irqs[0]); if ((err = snd_cs4231_sbus_create(card, sdev, dev, &cp)) < 0) { diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 5eecdd0..a7489a3 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -2645,9 +2645,9 @@ static int __init dbri_attach(int prom_node, struct sbus_dev *sdev) strcpy(card->driver, "DBRI"); strcpy(card->shortname, "Sun DBRI"); rp = &sdev->resource[0]; - sprintf(card->longname, "%s at 0x%02lx:0x%08lx, irq %d", + sprintf(card->longname, "%s at 0x%02lx:0x%016lx, irq %d", card->shortname, - rp->flags & 0xffL, rp->start, irq.pri); + rp->flags & 0xffL, (unsigned long long)rp->start, irq.pri); if ((err = snd_dbri_create(card, sdev, &irq, dev)) < 0) { snd_card_free(card); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 627de95..d32d83d 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 */ |
