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authorLinus Torvalds <torvalds@g5.osdl.org>2006-06-29 14:53:31 -0400
committerLinus Torvalds <torvalds@g5.osdl.org>2006-06-29 14:53:31 -0400
commit0950c358ee8e969fce45ba363ca1deaf211e57b0 (patch)
tree4c3b66e8457e1568aa26696d268e0e9c264382cb /sound
parent3aa590c6b7c89d844f81c2e96f295cf2c6967773 (diff)
parent8caf7aa26e0797e5706043f94c491acd1a08636a (diff)
Merge master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa
* master.kernel.org:/pub/scm/linux/kernel/git/perex/alsa: [ALSA] echoaudio - Remove kfree_nocheck() [ALSA] echoaudio - Fix Makefile [ALSA] Add Intel D965 board support [ALSA] Fix/add support of Realtek ALC883 / ALC888 and ALC861 codecs [ALSA] Fix a typo in echoaudio/midi.c [ALSA] snd-aoa: enable dual-edge in GPIOs [ALSA] snd-aoa: support iMac G5 iSight [ALSA] snd-aoa: not experimental [ALSA] Add echoaudio sound drivers [ALSA] ak4xxx-adda - Code clean-up [ALSA] Remove CONFIG_EXPERIMENTAL from intel8x0m driver [ALSA] Stereo controls for M-Audio Revolution cards [ALSA] Fix misuse of __list_add() in seq_ports.c [ALSA] hda-codec - Add model entry for Samsung X60 Chane [ALSA] make CONFIG_SND_DYNAMIC_MINORS non-experimental [ALSA] Fix wrong dependencies of snd-aoa driver [ALSA] fix build failure due to snd-aoa [ALSA] AD1888 mixer controls for DC mode [ALSA] Suppress irq handler mismatch messages in ALSA ISA drivers [ALSA] usb-audio support for Turtle Beach Roadie
Diffstat (limited to 'sound')
-rw-r--r--sound/Makefile3
-rw-r--r--sound/aoa/Kconfig3
-rw-r--r--sound/aoa/core/snd-aoa-gpio-feature.c15
-rw-r--r--sound/aoa/fabrics/snd-aoa-fabric-layout.c14
-rw-r--r--sound/aoa/soundbus/Kconfig3
-rw-r--r--sound/core/Kconfig4
-rw-r--r--sound/core/seq/seq_ports.c6
-rw-r--r--sound/i2c/other/ak4xxx-adda.c284
-rw-r--r--sound/pci/Kconfig141
-rw-r--r--sound/pci/Makefile1
-rw-r--r--sound/pci/ac97/ac97_patch.c2
-rw-r--r--sound/pci/echoaudio/Makefile30
-rw-r--r--sound/pci/echoaudio/darla20.c99
-rw-r--r--sound/pci/echoaudio/darla20_dsp.c125
-rw-r--r--sound/pci/echoaudio/darla24.c106
-rw-r--r--sound/pci/echoaudio/darla24_dsp.c156
-rw-r--r--sound/pci/echoaudio/echo3g.c118
-rw-r--r--sound/pci/echoaudio/echo3g_dsp.c131
-rw-r--r--sound/pci/echoaudio/echoaudio.c2196
-rw-r--r--sound/pci/echoaudio/echoaudio.h590
-rw-r--r--sound/pci/echoaudio/echoaudio_3g.c431
-rw-r--r--sound/pci/echoaudio/echoaudio_dsp.c1125
-rw-r--r--sound/pci/echoaudio/echoaudio_dsp.h694
-rw-r--r--sound/pci/echoaudio/echoaudio_gml.c198
-rw-r--r--sound/pci/echoaudio/gina20.c103
-rw-r--r--sound/pci/echoaudio/gina20_dsp.c215
-rw-r--r--sound/pci/echoaudio/gina24.c123
-rw-r--r--sound/pci/echoaudio/gina24_dsp.c346
-rw-r--r--sound/pci/echoaudio/indigo.c104
-rw-r--r--sound/pci/echoaudio/indigo_dsp.c170
-rw-r--r--sound/pci/echoaudio/indigodj.c104
-rw-r--r--sound/pci/echoaudio/indigodj_dsp.c170
-rw-r--r--sound/pci/echoaudio/indigoio.c105
-rw-r--r--sound/pci/echoaudio/indigoio_dsp.c141
-rw-r--r--sound/pci/echoaudio/layla20.c112
-rw-r--r--sound/pci/echoaudio/layla20_dsp.c290
-rw-r--r--sound/pci/echoaudio/layla24.c121
-rw-r--r--sound/pci/echoaudio/layla24_dsp.c394
-rw-r--r--sound/pci/echoaudio/mia.c117
-rw-r--r--sound/pci/echoaudio/mia_dsp.c229
-rw-r--r--sound/pci/echoaudio/midi.c327
-rw-r--r--sound/pci/echoaudio/mona.c129
-rw-r--r--sound/pci/echoaudio/mona_dsp.c428
-rw-r--r--sound/pci/hda/hda_codec.c4
-rw-r--r--sound/pci/hda/patch_analog.c2
-rw-r--r--sound/pci/hda/patch_realtek.c1084
-rw-r--r--sound/pci/hda/patch_sigmatel.c110
-rw-r--r--sound/pci/ice1712/revo.c23
-rw-r--r--sound/usb/usbaudio.c32
49 files changed, 11263 insertions, 195 deletions
diff --git a/sound/Makefile b/sound/Makefile
index a682ea30f0c9..1f60797afa8a 100644
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -4,7 +4,8 @@
4obj-$(CONFIG_SOUND) += soundcore.o 4obj-$(CONFIG_SOUND) += soundcore.o
5obj-$(CONFIG_SOUND_PRIME) += oss/ 5obj-$(CONFIG_SOUND_PRIME) += oss/
6obj-$(CONFIG_DMASOUND) += oss/ 6obj-$(CONFIG_DMASOUND) += oss/
7obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ aoa/ 7obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/
8obj-$(CONFIG_SND_AOA) += aoa/
8 9
9ifeq ($(CONFIG_SND),y) 10ifeq ($(CONFIG_SND),y)
10 obj-y += last.o 11 obj-y += last.o
diff --git a/sound/aoa/Kconfig b/sound/aoa/Kconfig
index a85194fe0b06..2f4334d19ccd 100644
--- a/sound/aoa/Kconfig
+++ b/sound/aoa/Kconfig
@@ -3,7 +3,8 @@ menu "Apple Onboard Audio driver"
3 3
4config SND_AOA 4config SND_AOA
5 tristate "Apple Onboard Audio driver" 5 tristate "Apple Onboard Audio driver"
6 depends on SOUND && SND_PCM 6 depends on SND
7 select SND_PCM
7 ---help--- 8 ---help---
8 This option enables the new driver for the various 9 This option enables the new driver for the various
9 Apple Onboard Audio components. 10 Apple Onboard Audio components.
diff --git a/sound/aoa/core/snd-aoa-gpio-feature.c b/sound/aoa/core/snd-aoa-gpio-feature.c
index 2c6eb7784cc9..bab97547a052 100644
--- a/sound/aoa/core/snd-aoa-gpio-feature.c
+++ b/sound/aoa/core/snd-aoa-gpio-feature.c
@@ -207,6 +207,17 @@ static void ftr_handle_notify(void *data)
207 mutex_unlock(&notif->mutex); 207 mutex_unlock(&notif->mutex);
208} 208}
209 209
210static void gpio_enable_dual_edge(int gpio)
211{
212 int v;
213
214 if (gpio == -1)
215 return;
216 v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0);
217 v |= 0x80; /* enable dual edge */
218 pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, gpio, v);
219}
220
210static void ftr_gpio_init(struct gpio_runtime *rt) 221static void ftr_gpio_init(struct gpio_runtime *rt)
211{ 222{
212 get_gpio("headphone-mute", NULL, 223 get_gpio("headphone-mute", NULL,
@@ -234,6 +245,10 @@ static void ftr_gpio_init(struct gpio_runtime *rt)
234 &linein_detect_gpio, 245 &linein_detect_gpio,
235 &linein_detect_gpio_activestate); 246 &linein_detect_gpio_activestate);
236 247
248 gpio_enable_dual_edge(headphone_detect_gpio);
249 gpio_enable_dual_edge(lineout_detect_gpio);
250 gpio_enable_dual_edge(linein_detect_gpio);
251
237 get_irq(headphone_detect_node, &headphone_detect_irq); 252 get_irq(headphone_detect_node, &headphone_detect_irq);
238 get_irq(lineout_detect_node, &lineout_detect_irq); 253 get_irq(lineout_detect_node, &lineout_detect_irq);
239 get_irq(linein_detect_node, &linein_detect_irq); 254 get_irq(linein_detect_node, &linein_detect_irq);
diff --git a/sound/aoa/fabrics/snd-aoa-fabric-layout.c b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
index 04a7238e9494..cbc8a3b5cea4 100644
--- a/sound/aoa/fabrics/snd-aoa-fabric-layout.c
+++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
@@ -94,6 +94,7 @@ MODULE_ALIAS("sound-layout-82");
94MODULE_ALIAS("sound-layout-84"); 94MODULE_ALIAS("sound-layout-84");
95MODULE_ALIAS("sound-layout-86"); 95MODULE_ALIAS("sound-layout-86");
96MODULE_ALIAS("sound-layout-92"); 96MODULE_ALIAS("sound-layout-92");
97MODULE_ALIAS("sound-layout-96");
97 98
98/* onyx with all but microphone connected */ 99/* onyx with all but microphone connected */
99static struct codec_connection onyx_connections_nomic[] = { 100static struct codec_connection onyx_connections_nomic[] = {
@@ -381,6 +382,13 @@ static struct layout layouts[] = {
381 .connections = toonie_connections, 382 .connections = toonie_connections,
382 }, 383 },
383 }, 384 },
385 {
386 .layout_id = 96,
387 .codecs[0] = {
388 .name = "onyx",
389 .connections = onyx_connections_noheadphones,
390 },
391 },
384 /* unknown, untested, but this comes from Apple */ 392 /* unknown, untested, but this comes from Apple */
385 { .layout_id = 41, 393 { .layout_id = 41,
386 .codecs[0] = { 394 .codecs[0] = {
@@ -479,12 +487,6 @@ static struct layout layouts[] = {
479 .connections = onyx_connections_noheadphones, 487 .connections = onyx_connections_noheadphones,
480 }, 488 },
481 }, 489 },
482 { .layout_id = 96,
483 .codecs[0] = {
484 .name = "onyx",
485 .connections = onyx_connections_noheadphones,
486 },
487 },
488 { .layout_id = 98, 490 { .layout_id = 98,
489 .codecs[0] = { 491 .codecs[0] = {
490 .name = "toonie", 492 .name = "toonie",
diff --git a/sound/aoa/soundbus/Kconfig b/sound/aoa/soundbus/Kconfig
index d532d27a9f54..7368b7ddfe0d 100644
--- a/sound/aoa/soundbus/Kconfig
+++ b/sound/aoa/soundbus/Kconfig
@@ -1,6 +1,7 @@
1config SND_AOA_SOUNDBUS 1config SND_AOA_SOUNDBUS
2 tristate "Apple Soundbus support" 2 tristate "Apple Soundbus support"
3 depends on SOUND && SND_PCM && EXPERIMENTAL 3 depends on SOUND
4 select SND_PCM
4 ---help--- 5 ---help---
5 This option enables the generic driver for the soundbus 6 This option enables the generic driver for the soundbus
6 support on Apple machines. 7 support on Apple machines.
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 4262a1c87731..b2927523d79d 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -122,8 +122,8 @@ config SND_SEQ_RTCTIMER_DEFAULT
122 If in doubt, say Y. 122 If in doubt, say Y.
123 123
124config SND_DYNAMIC_MINORS 124config SND_DYNAMIC_MINORS
125 bool "Dynamic device file minor numbers (EXPERIMENTAL)" 125 bool "Dynamic device file minor numbers"
126 depends on SND && EXPERIMENTAL 126 depends on SND
127 help 127 help
128 If you say Y here, the minor numbers of ALSA device files in 128 If you say Y here, the minor numbers of ALSA device files in
129 /dev/snd/ are allocated dynamically. This allows you to have 129 /dev/snd/ are allocated dynamically. This allows you to have
diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c
index 334579a9f268..d467b4f0ff2b 100644
--- a/sound/core/seq/seq_ports.c
+++ b/sound/core/seq/seq_ports.c
@@ -322,10 +322,8 @@ int snd_seq_delete_all_ports(struct snd_seq_client *client)
322 mutex_lock(&client->ports_mutex); 322 mutex_lock(&client->ports_mutex);
323 write_lock_irqsave(&client->ports_lock, flags); 323 write_lock_irqsave(&client->ports_lock, flags);
324 if (! list_empty(&client->ports_list_head)) { 324 if (! list_empty(&client->ports_list_head)) {
325 __list_add(&deleted_list, 325 list_add(&deleted_list, &client->ports_list_head);
326 client->ports_list_head.prev, 326 list_del_init(&client->ports_list_head);
327 client->ports_list_head.next);
328 INIT_LIST_HEAD(&client->ports_list_head);
329 } else { 327 } else {
330 INIT_LIST_HEAD(&deleted_list); 328 INIT_LIST_HEAD(&deleted_list);
331 } 329 }
diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c
index 045e32a311e0..dc7cc2001b74 100644
--- a/sound/i2c/other/ak4xxx-adda.c
+++ b/sound/i2c/other/ak4xxx-adda.c
@@ -34,7 +34,8 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@suse.cz>, Takashi Iwai <tiwai@suse.de>");
34MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx AD/DA converters"); 34MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx AD/DA converters");
35MODULE_LICENSE("GPL"); 35MODULE_LICENSE("GPL");
36 36
37void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg, unsigned char val) 37void snd_akm4xxx_write(struct snd_akm4xxx *ak, int chip, unsigned char reg,
38 unsigned char val)
38{ 39{
39 ak->ops.lock(ak, chip); 40 ak->ops.lock(ak, chip);
40 ak->ops.write(ak, chip, reg, val); 41 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
52 ak->ops.unlock(ak, chip); 53 ak->ops.unlock(ak, chip);
53} 54}
54 55
56EXPORT_SYMBOL(snd_akm4xxx_write);
57
58/* reset procedure for AK4524 and AK4528 */
59static void ak4524_reset(struct snd_akm4xxx *ak, int state)
60{
61 unsigned int chip;
62 unsigned char reg, maxreg;
63
64 if (ak->type == SND_AK4528)
65 maxreg = 0x06;
66 else
67 maxreg = 0x08;
68 for (chip = 0; chip < ak->num_dacs/2; chip++) {
69 snd_akm4xxx_write(ak, chip, 0x01, state ? 0x00 : 0x03);
70 if (state)
71 continue;
72 /* DAC volumes */
73 for (reg = 0x04; reg < maxreg; reg++)
74 snd_akm4xxx_write(ak, chip, reg,
75 snd_akm4xxx_get(ak, chip, reg));
76 if (ak->type == SND_AK4528)
77 continue;
78 /* IPGA */
79 for (reg = 0x04; reg < 0x06; reg++)
80 snd_akm4xxx_write(ak, chip, reg,
81 snd_akm4xxx_get_ipga(ak, chip, reg));
82 }
83}
84
85/* reset procedure for AK4355 and AK4358 */
86static void ak4355_reset(struct snd_akm4xxx *ak, int state)
87{
88 unsigned char reg;
89
90 if (state) {
91 snd_akm4xxx_write(ak, 0, 0x01, 0x02); /* reset and soft-mute */
92 return;
93 }
94 for (reg = 0x00; reg < 0x0b; reg++)
95 if (reg != 0x01)
96 snd_akm4xxx_write(ak, 0, reg,
97 snd_akm4xxx_get(ak, 0, reg));
98 snd_akm4xxx_write(ak, 0, 0x01, 0x01); /* un-reset, unmute */
99}
100
101/* reset procedure for AK4381 */
102static void ak4381_reset(struct snd_akm4xxx *ak, int state)
103{
104 unsigned int chip;
105 unsigned char reg;
106
107 for (chip = 0; chip < ak->num_dacs/2; chip++) {
108 snd_akm4xxx_write(ak, chip, 0x00, state ? 0x0c : 0x0f);
109 if (state)
110 continue;
111 for (reg = 0x01; reg < 0x05; reg++)
112 snd_akm4xxx_write(ak, chip, reg,
113 snd_akm4xxx_get(ak, chip, reg));
114 }
115}
116
55/* 117/*
56 * reset the AKM codecs 118 * reset the AKM codecs
57 * @state: 1 = reset codec, 0 = restore the registers 119 * @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
60 */ 122 */
61void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state) 123void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state)
62{ 124{
63 unsigned int chip;
64 unsigned char reg;
65
66 switch (ak->type) { 125 switch (ak->type) {
67 case SND_AK4524: 126 case SND_AK4524:
68 case SND_AK4528: 127 case SND_AK4528:
69 for (chip = 0; chip < ak->num_dacs/2; chip++) { 128 ak4524_reset(ak, state);
70 snd_akm4xxx_write(ak, chip, 0x01, state ? 0x00 : 0x03);
71 if (state)
72 continue;
73 /* DAC volumes */
74 for (reg = 0x04; reg < (ak->type == SND_AK4528 ? 0x06 : 0x08); reg++)
75 snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get(ak, chip, reg));
76 if (ak->type == SND_AK4528)
77 continue;
78 /* IPGA */
79 for (reg = 0x04; reg < 0x06; reg++)
80 snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get_ipga(ak, chip, reg));
81 }
82 break; 129 break;
83 case SND_AK4529: 130 case SND_AK4529:
84 /* FIXME: needed for ak4529? */ 131 /* FIXME: needed for ak4529? */
85 break; 132 break;
86 case SND_AK4355: 133 case SND_AK4355:
87 case SND_AK4358: 134 case SND_AK4358:
88 if (state) { 135 ak4355_reset(ak, state);
89 snd_akm4xxx_write(ak, 0, 0x01, 0x02); /* reset and soft-mute */
90 return;
91 }
92 for (reg = 0x00; reg < 0x0b; reg++)
93 if (reg != 0x01)
94 snd_akm4xxx_write(ak, 0, reg, snd_akm4xxx_get(ak, 0, reg));
95 snd_akm4xxx_write(ak, 0, 0x01, 0x01); /* un-reset, unmute */
96 break; 136 break;
97 case SND_AK4381: 137 case SND_AK4381:
98 for (chip = 0; chip < ak->num_dacs/2; chip++) { 138 ak4381_reset(ak, state);
99 snd_akm4xxx_write(ak, chip, 0x00, state ? 0x0c : 0x0f);
100 if (state)
101 continue;
102 for (reg = 0x01; reg < 0x05; reg++)
103 snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get(ak, chip, reg));
104 }
105 break; 139 break;
106 } 140 }
107} 141}
108 142
143EXPORT_SYMBOL(snd_akm4xxx_reset);
144
109/* 145/*
110 * initialize all the ak4xxx chips 146 * initialize all the ak4xxx chips
111 */ 147 */
@@ -153,7 +189,8 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak)
153 }; 189 };
154 static unsigned char inits_ak4355[] = { 190 static unsigned char inits_ak4355[] = {
155 0x01, 0x02, /* 1: reset and soft-mute */ 191 0x01, 0x02, /* 1: reset and soft-mute */
156 0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect, disable DZF, sharp roll-off, RSTN#=0 */ 192 0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect,
193 * disable DZF, sharp roll-off, RSTN#=0 */
157 0x02, 0x0e, /* 2: DA's power up, normal speed, RSTN#=0 */ 194 0x02, 0x0e, /* 2: DA's power up, normal speed, RSTN#=0 */
158 // 0x02, 0x2e, /* quad speed */ 195 // 0x02, 0x2e, /* quad speed */
159 0x03, 0x01, /* 3: de-emphasis off */ 196 0x03, 0x01, /* 3: de-emphasis off */
@@ -169,7 +206,8 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak)
169 }; 206 };
170 static unsigned char inits_ak4358[] = { 207 static unsigned char inits_ak4358[] = {
171 0x01, 0x02, /* 1: reset and soft-mute */ 208 0x01, 0x02, /* 1: reset and soft-mute */
172 0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect, disable DZF, sharp roll-off, RSTN#=0 */ 209 0x00, 0x06, /* 0: mode3(i2s), disable auto-clock detect,
210 * disable DZF, sharp roll-off, RSTN#=0 */
173 0x02, 0x0e, /* 2: DA's power up, normal speed, RSTN#=0 */ 211 0x02, 0x0e, /* 2: DA's power up, normal speed, RSTN#=0 */
174 // 0x02, 0x2e, /* quad speed */ 212 // 0x02, 0x2e, /* quad speed */
175 0x03, 0x01, /* 3: de-emphasis off */ 213 0x03, 0x01, /* 3: de-emphasis off */
@@ -187,7 +225,8 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak)
187 }; 225 };
188 static unsigned char inits_ak4381[] = { 226 static unsigned char inits_ak4381[] = {
189 0x00, 0x0c, /* 0: mode3(i2s), disable auto-clock detect */ 227 0x00, 0x0c, /* 0: mode3(i2s), disable auto-clock detect */
190 0x01, 0x02, /* 1: de-emphasis off, normal speed, sharp roll-off, DZF off */ 228 0x01, 0x02, /* 1: de-emphasis off, normal speed,
229 * sharp roll-off, DZF off */
191 // 0x01, 0x12, /* quad speed */ 230 // 0x01, 0x12, /* quad speed */
192 0x02, 0x00, /* 2: DZF disabled */ 231 0x02, 0x00, /* 2: DZF disabled */
193 0x03, 0x00, /* 3: LATT 0 */ 232 0x03, 0x00, /* 3: LATT 0 */
@@ -239,12 +278,15 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak)
239 } 278 }
240} 279}
241 280
281EXPORT_SYMBOL(snd_akm4xxx_init);
282
242#define AK_GET_CHIP(val) (((val) >> 8) & 0xff) 283#define AK_GET_CHIP(val) (((val) >> 8) & 0xff)
243#define AK_GET_ADDR(val) ((val) & 0xff) 284#define AK_GET_ADDR(val) ((val) & 0xff)
244#define AK_GET_SHIFT(val) (((val) >> 16) & 0x7f) 285#define AK_GET_SHIFT(val) (((val) >> 16) & 0x7f)
245#define AK_GET_INVERT(val) (((val) >> 23) & 1) 286#define AK_GET_INVERT(val) (((val) >> 23) & 1)
246#define AK_GET_MASK(val) (((val) >> 24) & 0xff) 287#define AK_GET_MASK(val) (((val) >> 24) & 0xff)
247#define AK_COMPOSE(chip,addr,shift,mask) (((chip) << 8) | (addr) | ((shift) << 16) | ((mask) << 24)) 288#define AK_COMPOSE(chip,addr,shift,mask) \
289 (((chip) << 8) | (addr) | ((shift) << 16) | ((mask) << 24))
248#define AK_INVERT (1<<23) 290#define AK_INVERT (1<<23)
249 291
250static int snd_akm4xxx_volume_info(struct snd_kcontrol *kcontrol, 292static int snd_akm4xxx_volume_info(struct snd_kcontrol *kcontrol,
@@ -292,6 +334,64 @@ static int snd_akm4xxx_volume_put(struct snd_kcontrol *kcontrol,
292 return change; 334 return change;
293} 335}
294 336
337static int snd_akm4xxx_stereo_volume_info(struct snd_kcontrol *kcontrol,
338 struct snd_ctl_elem_info *uinfo)
339{
340 unsigned int mask = AK_GET_MASK(kcontrol->private_value);
341
342 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
343 uinfo->count = 2;
344 uinfo->value.integer.min = 0;
345 uinfo->value.integer.max = mask;
346 return 0;
347}
348
349static int snd_akm4xxx_stereo_volume_get(struct snd_kcontrol *kcontrol,
350 struct snd_ctl_elem_value *ucontrol)
351{
352 struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
353 int chip = AK_GET_CHIP(kcontrol->private_value);
354 int addr = AK_GET_ADDR(kcontrol->private_value);
355 int invert = AK_GET_INVERT(kcontrol->private_value);
356 unsigned int mask = AK_GET_MASK(kcontrol->private_value);
357 unsigned char val = snd_akm4xxx_get(ak, chip, addr);
358
359 ucontrol->value.integer.value[0] = invert ? mask - val : val;
360
361 val = snd_akm4xxx_get(ak, chip, addr+1);
362 ucontrol->value.integer.value[1] = invert ? mask - val : val;
363
364 return 0;
365}
366
367static int snd_akm4xxx_stereo_volume_put(struct snd_kcontrol *kcontrol,
368 struct snd_ctl_elem_value *ucontrol)
369{
370 struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
371 int chip = AK_GET_CHIP(kcontrol->private_value);
372 int addr = AK_GET_ADDR(kcontrol->private_value);
373 int invert = AK_GET_INVERT(kcontrol->private_value);
374 unsigned int mask = AK_GET_MASK(kcontrol->private_value);
375 unsigned char nval = ucontrol->value.integer.value[0] % (mask+1);
376 int change0, change1;
377
378 if (invert)
379 nval = mask - nval;
380 change0 = snd_akm4xxx_get(ak, chip, addr) != nval;
381 if (change0)
382 snd_akm4xxx_write(ak, chip, addr, nval);
383
384 nval = ucontrol->value.integer.value[1] % (mask+1);
385 if (invert)
386 nval = mask - nval;
387 change1 = snd_akm4xxx_get(ak, chip, addr+1) != nval;
388 if (change1)
389 snd_akm4xxx_write(ak, chip, addr+1, nval);
390
391
392 return change0 || change1;
393}
394
295static int snd_akm4xxx_ipga_gain_info(struct snd_kcontrol *kcontrol, 395static int snd_akm4xxx_ipga_gain_info(struct snd_kcontrol *kcontrol,
296 struct snd_ctl_elem_info *uinfo) 396 struct snd_ctl_elem_info *uinfo)
297{ 397{
@@ -308,7 +408,8 @@ static int snd_akm4xxx_ipga_gain_get(struct snd_kcontrol *kcontrol,
308 struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol); 408 struct snd_akm4xxx *ak = snd_kcontrol_chip(kcontrol);
309 int chip = AK_GET_CHIP(kcontrol->private_value); 409 int chip = AK_GET_CHIP(kcontrol->private_value);
310 int addr = AK_GET_ADDR(kcontrol->private_value); 410 int addr = AK_GET_ADDR(kcontrol->private_value);
311 ucontrol->value.integer.value[0] = snd_akm4xxx_get_ipga(ak, chip, addr) & 0x7f; 411 ucontrol->value.integer.value[0] =
412 snd_akm4xxx_get_ipga(ak, chip, addr) & 0x7f;
312 return 0; 413 return 0;
313} 414}
314 415
@@ -336,7 +437,8 @@ static int snd_akm4xxx_deemphasis_info(struct snd_kcontrol *kcontrol,
336 uinfo->value.enumerated.items = 4; 437 uinfo->value.enumerated.items = 4;
337 if (uinfo->value.enumerated.item >= 4) 438 if (uinfo->value.enumerated.item >= 4)
338 uinfo->value.enumerated.item = 3; 439 uinfo->value.enumerated.item = 3;
339 strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); 440 strcpy(uinfo->value.enumerated.name,
441 texts[uinfo->value.enumerated.item]);
340 return 0; 442 return 0;
341} 443}
342 444
@@ -347,7 +449,8 @@ static int snd_akm4xxx_deemphasis_get(struct snd_kcontrol *kcontrol,
347 int chip = AK_GET_CHIP(kcontrol->private_value); 449 int chip = AK_GET_CHIP(kcontrol->private_value);
348 int addr = AK_GET_ADDR(kcontrol->private_value); 450 int addr = AK_GET_ADDR(kcontrol->private_value);
349 int shift = AK_GET_SHIFT(kcontrol->private_value); 451 int shift = AK_GET_SHIFT(kcontrol->private_value);
350 ucontrol->value.enumerated.item[0] = (snd_akm4xxx_get(ak, chip, addr) >> shift) & 3; 452 ucontrol->value.enumerated.item[0] =
453 (snd_akm4xxx_get(ak, chip, addr) >> shift) & 3;
351 return 0; 454 return 0;
352} 455}
353 456
@@ -361,7 +464,8 @@ static int snd_akm4xxx_deemphasis_put(struct snd_kcontrol *kcontrol,
361 unsigned char nval = ucontrol->value.enumerated.item[0] & 3; 464 unsigned char nval = ucontrol->value.enumerated.item[0] & 3;
362 int change; 465 int change;
363 466
364 nval = (nval << shift) | (snd_akm4xxx_get(ak, chip, addr) & ~(3 << shift)); 467 nval = (nval << shift) |
468 (snd_akm4xxx_get(ak, chip, addr) & ~(3 << shift));
365 change = snd_akm4xxx_get(ak, chip, addr) != nval; 469 change = snd_akm4xxx_get(ak, chip, addr) != nval;
366 if (change) 470 if (change)
367 snd_akm4xxx_write(ak, chip, addr, nval); 471 snd_akm4xxx_write(ak, chip, addr, nval);
@@ -377,51 +481,86 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
377 unsigned int idx, num_emphs; 481 unsigned int idx, num_emphs;
378 struct snd_kcontrol *ctl; 482 struct snd_kcontrol *ctl;
379 int err; 483 int err;
484 int mixer_ch = 0;
485 int num_stereo;
380 486
381 ctl = kmalloc(sizeof(*ctl), GFP_KERNEL); 487 ctl = kmalloc(sizeof(*ctl), GFP_KERNEL);
382 if (! ctl) 488 if (! ctl)
383 return -ENOMEM; 489 return -ENOMEM;
384 490
385 for (idx = 0; idx < ak->num_dacs; ++idx) { 491 for (idx = 0; idx < ak->num_dacs; ) {
386 memset(ctl, 0, sizeof(*ctl)); 492 memset(ctl, 0, sizeof(*ctl));
387 strcpy(ctl->id.name, "DAC Volume"); 493 if (ak->channel_names == NULL) {
388 ctl->id.index = idx + ak->idx_offset * 2; 494 strcpy(ctl->id.name, "DAC Volume");
495 num_stereo = 1;
496 ctl->id.index = mixer_ch + ak->idx_offset * 2;
497 } else {
498 strcpy(ctl->id.name, ak->channel_names[mixer_ch]);
499 num_stereo = ak->num_stereo[mixer_ch];
500 ctl->id.index = 0;
501 }
389 ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; 502 ctl->id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
390 ctl->count = 1; 503 ctl->count = 1;
391 ctl->info = snd_akm4xxx_volume_info; 504 if (num_stereo == 2) {
392 ctl->get = snd_akm4xxx_volume_get; 505 ctl->info = snd_akm4xxx_stereo_volume_info;
393 ctl->put = snd_akm4xxx_volume_put; 506 ctl->get = snd_akm4xxx_stereo_volume_get;
507 ctl->put = snd_akm4xxx_stereo_volume_put;
508 } else {
509 ctl->info = snd_akm4xxx_volume_info;
510 ctl->get = snd_akm4xxx_volume_get;
511 ctl->put = snd_akm4xxx_volume_put;
512 }
394 switch (ak->type) { 513 switch (ak->type) {
395 case SND_AK4524: 514 case SND_AK4524:
396 ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 6, 0, 127); /* register 6 & 7 */ 515 /* register 6 & 7 */
516 ctl->private_value =
517 AK_COMPOSE(idx/2, (idx%2) + 6, 0, 127);
397 break; 518 break;
398 case SND_AK4528: 519 case SND_AK4528:
399 ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 4, 0, 127); /* register 4 & 5 */ 520 /* register 4 & 5 */
521 ctl->private_value =
522 AK_COMPOSE(idx/2, (idx%2) + 4, 0, 127);
400 break; 523 break;
401 case SND_AK4529: { 524 case SND_AK4529: {
402 int val = idx < 6 ? idx + 2 : (idx - 6) + 0xb; /* registers 2-7 and b,c */ 525 /* registers 2-7 and b,c */
403 ctl->private_value = AK_COMPOSE(0, val, 0, 255) | AK_INVERT; 526 int val = idx < 6 ? idx + 2 : (idx - 6) + 0xb;
527 ctl->private_value =
528 AK_COMPOSE(0, val, 0, 255) | AK_INVERT;
404 break; 529 break;
405 } 530 }
406 case SND_AK4355: 531 case SND_AK4355:
407 ctl->private_value = AK_COMPOSE(0, idx + 4, 0, 255); /* register 4-9, chip #0 only */ 532 /* register 4-9, chip #0 only */
533 ctl->private_value = AK_COMPOSE(0, idx + 4, 0, 255);
408 break; 534 break;
409 case SND_AK4358: 535 case SND_AK4358:
410 if (idx >= 6) 536 if (idx >= 6)
411 ctl->private_value = AK_COMPOSE(0, idx + 5, 0, 255); /* register 4-9, chip #0 only */ 537 /* register 4-9, chip #0 only */
538 ctl->private_value =
539 AK_COMPOSE(0, idx + 5, 0, 255);
412 else 540 else
413 ctl->private_value = AK_COMPOSE(0, idx + 4, 0, 255); /* register 4-9, chip #0 only */ 541 /* register 4-9, chip #0 only */
542 ctl->private_value =
543 AK_COMPOSE(0, idx + 4, 0, 255);
414 break; 544 break;
415 case SND_AK4381: 545 case SND_AK4381:
416 ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 3, 0, 255); /* register 3 & 4 */ 546 /* register 3 & 4 */
547 ctl->private_value =
548 AK_COMPOSE(idx/2, (idx%2) + 3, 0, 255);
417 break; 549 break;
418 default: 550 default:
419 err = -EINVAL; 551 err = -EINVAL;
420 goto __error; 552 goto __error;
421 } 553 }
554
422 ctl->private_data = ak; 555 ctl->private_data = ak;
423 if ((err = snd_ctl_add(ak->card, snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE))) < 0) 556 err = snd_ctl_add(ak->card,
557 snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|
558 SNDRV_CTL_ELEM_ACCESS_WRITE));
559 if (err < 0)
424 goto __error; 560 goto __error;
561
562 idx += num_stereo;
563 mixer_ch++;
425 } 564 }
426 for (idx = 0; idx < ak->num_adcs && ak->type == SND_AK4524; ++idx) { 565 for (idx = 0; idx < ak->num_adcs && ak->type == SND_AK4524; ++idx) {
427 memset(ctl, 0, sizeof(*ctl)); 566 memset(ctl, 0, sizeof(*ctl));
@@ -432,9 +571,14 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
432 ctl->info = snd_akm4xxx_volume_info; 571 ctl->info = snd_akm4xxx_volume_info;
433 ctl->get = snd_akm4xxx_volume_get; 572 ctl->get = snd_akm4xxx_volume_get;
434 ctl->put = snd_akm4xxx_volume_put; 573 ctl->put = snd_akm4xxx_volume_put;
435 ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 4, 0, 127); /* register 4 & 5 */ 574 /* register 4 & 5 */
575 ctl->private_value =
576 AK_COMPOSE(idx/2, (idx%2) + 4, 0, 127);
436 ctl->private_data = ak; 577 ctl->private_data = ak;
437 if ((err = snd_ctl_add(ak->card, snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE))) < 0) 578 err = snd_ctl_add(ak->card,
579 snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|
580 SNDRV_CTL_ELEM_ACCESS_WRITE));
581 if (err < 0)
438 goto __error; 582 goto __error;
439 583
440 memset(ctl, 0, sizeof(*ctl)); 584 memset(ctl, 0, sizeof(*ctl));
@@ -445,9 +589,13 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
445 ctl->info = snd_akm4xxx_ipga_gain_info; 589 ctl->info = snd_akm4xxx_ipga_gain_info;
446 ctl->get = snd_akm4xxx_ipga_gain_get; 590 ctl->get = snd_akm4xxx_ipga_gain_get;
447 ctl->put = snd_akm4xxx_ipga_gain_put; 591 ctl->put = snd_akm4xxx_ipga_gain_put;
448 ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 4, 0, 0); /* register 4 & 5 */ 592 /* register 4 & 5 */
593 ctl->private_value = AK_COMPOSE(idx/2, (idx%2) + 4, 0, 0);
449 ctl->private_data = ak; 594 ctl->private_data = ak;
450 if ((err = snd_ctl_add(ak->card, snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE))) < 0) 595 err = snd_ctl_add(ak->card,
596 snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|
597 SNDRV_CTL_ELEM_ACCESS_WRITE));
598 if (err < 0)
451 goto __error; 599 goto __error;
452 } 600 }
453 if (ak->type == SND_AK4355 || ak->type == SND_AK4358) 601 if (ak->type == SND_AK4355 || ak->type == SND_AK4358)
@@ -466,11 +614,13 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
466 switch (ak->type) { 614 switch (ak->type) {
467 case SND_AK4524: 615 case SND_AK4524:
468 case SND_AK4528: 616 case SND_AK4528:
469 ctl->private_value = AK_COMPOSE(idx, 3, 0, 0); /* register 3 */ 617 /* register 3 */
618 ctl->private_value = AK_COMPOSE(idx, 3, 0, 0);
470 break; 619 break;
471 case SND_AK4529: { 620 case SND_AK4529: {
472 int shift = idx == 3 ? 6 : (2 - idx) * 2; 621 int shift = idx == 3 ? 6 : (2 - idx) * 2;
473 ctl->private_value = AK_COMPOSE(0, 8, shift, 0); /* register 8 with shift */ 622 /* register 8 with shift */
623 ctl->private_value = AK_COMPOSE(0, 8, shift, 0);
474 break; 624 break;
475 } 625 }
476 case SND_AK4355: 626 case SND_AK4355:
@@ -482,7 +632,10 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
482 break; 632 break;
483 } 633 }
484 ctl->private_data = ak; 634 ctl->private_data = ak;
485 if ((err = snd_ctl_add(ak->card, snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|SNDRV_CTL_ELEM_ACCESS_WRITE))) < 0) 635 err = snd_ctl_add(ak->card,
636 snd_ctl_new(ctl, SNDRV_CTL_ELEM_ACCESS_READ|
637 SNDRV_CTL_ELEM_ACCESS_WRITE));
638 if (err < 0)
486 goto __error; 639 goto __error;
487 } 640 }
488 err = 0; 641 err = 0;
@@ -492,6 +645,8 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak)
492 return err; 645 return err;
493} 646}
494 647
648EXPORT_SYMBOL(snd_akm4xxx_build_controls);
649
495static int __init alsa_akm4xxx_module_init(void) 650static int __init alsa_akm4xxx_module_init(void)
496{ 651{
497 return 0; 652 return 0;
@@ -503,8 +658,3 @@ static void __exit alsa_akm4xxx_module_exit(void)
503 658
504module_init(alsa_akm4xxx_module_init) 659module_init(alsa_akm4xxx_module_init)
505module_exit(alsa_akm4xxx_module_exit) 660module_exit(alsa_akm4xxx_module_exit)
506
507EXPORT_SYMBOL(snd_akm4xxx_write);
508EXPORT_SYMBOL(snd_akm4xxx_reset);
509EXPORT_SYMBOL(snd_akm4xxx_init);
510EXPORT_SYMBOL(snd_akm4xxx_build_controls);
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index d37346b12dc0..23e54cedfd4a 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -233,6 +233,143 @@ config SND_CS5535AUDIO
233 To compile this driver as a module, choose M here: the module 233 To compile this driver as a module, choose M here: the module
234 will be called snd-cs5535audio. 234 will be called snd-cs5535audio.
235 235
236config SND_DARLA20
237 tristate "(Echoaudio) Darla20"
238 depends on SND
239 depends on FW_LOADER
240 select SND_PCM
241 help
242 Say 'Y' or 'M' to include support for Echoaudio Darla.
243
244 To compile this driver as a module, choose M here: the module
245 will be called snd-darla20
246
247config SND_GINA20
248 tristate "(Echoaudio) Gina20"
249 depends on SND
250 depends on FW_LOADER
251 select SND_PCM
252 help
253 Say 'Y' or 'M' to include support for Echoaudio Gina.
254
255 To compile this driver as a module, choose M here: the module
256 will be called snd-gina20
257
258config SND_LAYLA20
259 tristate "(Echoaudio) Layla20"
260 depends on SND
261 depends on FW_LOADER
262 select SND_RAWMIDI
263 select SND_PCM
264 help
265 Say 'Y' or 'M' to include support for Echoaudio Layla.
266
267 To compile this driver as a module, choose M here: the module
268 will be called snd-layla20
269
270config SND_DARLA24
271 tristate "(Echoaudio) Darla24"
272 depends on SND
273 depends on FW_LOADER
274 select SND_PCM
275 help
276 Say 'Y' or 'M' to include support for Echoaudio Darla24.
277
278 To compile this driver as a module, choose M here: the module
279 will be called snd-darla24
280
281config SND_GINA24
282 tristate "(Echoaudio) Gina24"
283 depends on SND
284 depends on FW_LOADER
285 select SND_PCM
286 help
287 Say 'Y' or 'M' to include support for Echoaudio Gina24.
288
289 To compile this driver as a module, choose M here: the module
290 will be called snd-gina24
291
292config SND_LAYLA24
293 tristate "(Echoaudio) Layla24"
294 depends on SND
295 depends on FW_LOADER
296 select SND_RAWMIDI
297 select SND_PCM
298 help
299 Say 'Y' or 'M' to include support for Echoaudio Layla24.
300
301 To compile this driver as a module, choose M here: the module
302 will be called snd-layla24
303
304config SND_MONA
305 tristate "(Echoaudio) Mona"
306 depends on SND
307 depends on FW_LOADER
308 select SND_RAWMIDI
309 select SND_PCM
310 help
311 Say 'Y' or 'M' to include support for Echoaudio Mona.
312
313 To compile this driver as a module, choose M here: the module
314 will be called snd-mona
315
316config SND_MIA
317 tristate "(Echoaudio) Mia"
318 depends on SND
319 depends on FW_LOADER
320 select SND_RAWMIDI
321 select SND_PCM
322 help
323 Say 'Y' or 'M' to include support for Echoaudio Mia and Mia-midi.
324
325 To compile this driver as a module, choose M here: the module
326 will be called snd-mia
327
328config SND_ECHO3G
329 tristate "(Echoaudio) 3G cards"
330 depends on SND
331 depends on FW_LOADER
332 select SND_RAWMIDI
333 select SND_PCM
334 help
335 Say 'Y' or 'M' to include support for Echoaudio Gina3G and Layla3G.
336
337 To compile this driver as a module, choose M here: the module
338 will be called snd-echo3g
339
340config SND_INDIGO
341 tristate "(Echoaudio) Indigo"
342 depends on SND
343 depends on FW_LOADER
344 select SND_PCM
345 help
346 Say 'Y' or 'M' to include support for Echoaudio Indigo.
347
348 To compile this driver as a module, choose M here: the module
349 will be called snd-indigo
350
351config SND_INDIGOIO
352 tristate "(Echoaudio) Indigo IO"
353 depends on SND
354 depends on FW_LOADER
355 select SND_PCM
356 help
357 Say 'Y' or 'M' to include support for Echoaudio Indigo IO.
358
359 To compile this driver as a module, choose M here: the module
360 will be called snd-indigoio
361
362config SND_INDIGODJ
363 tristate "(Echoaudio) Indigo DJ"
364 depends on SND
365 depends on FW_LOADER
366 select SND_PCM
367 help
368 Say 'Y' or 'M' to include support for Echoaudio Indigo DJ.
369
370 To compile this driver as a module, choose M here: the module
371 will be called snd-indigodj
372
236config SND_EMU10K1 373config SND_EMU10K1
237 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" 374 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)"
238 depends on SND 375 depends on SND
@@ -420,8 +557,8 @@ config SND_INTEL8X0
420 will be called snd-intel8x0. 557 will be called snd-intel8x0.
421 558
422config SND_INTEL8X0M 559config SND_INTEL8X0M
423 tristate "Intel/SiS/nVidia/AMD MC97 Modem (EXPERIMENTAL)" 560 tristate "Intel/SiS/nVidia/AMD MC97 Modem"
424 depends on SND && EXPERIMENTAL 561 depends on SND
425 select SND_AC97_CODEC 562 select SND_AC97_CODEC
426 help 563 help
427 Say Y here to include support for the integrated MC97 modem on 564 Say Y here to include support for the integrated MC97 modem on
diff --git a/sound/pci/Makefile b/sound/pci/Makefile
index cba5105aafea..e06736da9ef1 100644
--- a/sound/pci/Makefile
+++ b/sound/pci/Makefile
@@ -57,6 +57,7 @@ obj-$(CONFIG_SND) += \
57 ca0106/ \ 57 ca0106/ \
58 cs46xx/ \ 58 cs46xx/ \
59 cs5535audio/ \ 59 cs5535audio/ \
60 echoaudio/ \
60 emu10k1/ \ 61 emu10k1/ \
61 hda/ \ 62 hda/ \
62 ice1712/ \ 63 ice1712/ \
diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c
index 7f197c780816..094cfc1f3a19 100644
--- a/sound/pci/ac97/ac97_patch.c
+++ b/sound/pci/ac97/ac97_patch.c
@@ -1824,6 +1824,8 @@ static const struct snd_kcontrol_new snd_ac97_ad1888_controls[] = {
1824 .get = snd_ac97_ad1888_lohpsel_get, 1824 .get = snd_ac97_ad1888_lohpsel_get,
1825 .put = snd_ac97_ad1888_lohpsel_put 1825 .put = snd_ac97_ad1888_lohpsel_put
1826 }, 1826 },
1827 AC97_SINGLE("V_REFOUT Enable", AC97_AD_MISC, 2, 1, 1),
1828 AC97_SINGLE("High Pass Filter Enable", AC97_AD_TEST2, 12, 1, 1),
1827 AC97_SINGLE("Spread Front to Surround and Center/LFE", AC97_AD_MISC, 7, 1, 0), 1829 AC97_SINGLE("Spread Front to Surround and Center/LFE", AC97_AD_MISC, 7, 1, 0),
1828 { 1830 {
1829 .iface = SNDRV_CTL_ELEM_IFACE_MIXER, 1831 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile
new file mode 100644
index 000000000000..7b576aeb3f8d
--- /dev/null
+++ b/sound/pci/echoaudio/Makefile
@@ -0,0 +1,30 @@
1#
2# Makefile for ALSA Echoaudio soundcard drivers
3# Copyright (c) 2003 by Giuliano Pochini <pochini@shiny.it>
4#
5
6snd-darla20-objs := darla20.o
7snd-gina20-objs := gina20.o
8snd-layla20-objs := layla20.o
9snd-darla24-objs := darla24.o
10snd-gina24-objs := gina24.o
11snd-layla24-objs := layla24.o
12snd-mona-objs := mona.o
13snd-mia-objs := mia.o
14snd-echo3g-objs := echo3g.o
15snd-indigo-objs := indigo.o
16snd-indigoio-objs := indigoio.o
17snd-indigodj-objs := indigodj.o
18
19obj-$(CONFIG_SND_DARLA20) += snd-darla20.o
20obj-$(CONFIG_SND_GINA20) += snd-gina20.o
21obj-$(CONFIG_SND_LAYLA20) += snd-layla20.o
22obj-$(CONFIG_SND_DARLA24) += snd-darla24.o
23obj-$(CONFIG_SND_GINA24) += snd-gina24.o
24obj-$(CONFIG_SND_LAYLA24) += snd-layla24.o
25obj-$(CONFIG_SND_MONA) += snd-mona.o
26obj-$(CONFIG_SND_MIA) += snd-mia.o
27obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o
28obj-$(CONFIG_SND_INDIGO) += snd-indigo.o
29obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o
30obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o
diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c
new file mode 100644
index 000000000000..b7108e29a668
--- /dev/null
+++ b/sound/pci/echoaudio/darla20.c
@@ -0,0 +1,99 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define ECHOGALS_FAMILY
20#define ECHOCARD_DARLA20
21#define ECHOCARD_NAME "Darla20"
22#define ECHOCARD_HAS_MONITOR
23
24/* Pipe indexes */
25#define PX_ANALOG_OUT 0 /* 8 */
26#define PX_DIGITAL_OUT 8 /* 0 */
27#define PX_ANALOG_IN 8 /* 2 */
28#define PX_DIGITAL_IN 10 /* 0 */
29#define PX_NUM 10
30
31/* Bus indexes */
32#define BX_ANALOG_OUT 0 /* 8 */
33#define BX_DIGITAL_OUT 8 /* 0 */
34#define BX_ANALOG_IN 8 /* 2 */
35#define BX_DIGITAL_IN 10 /* 0 */
36#define BX_NUM 10
37
38
39#include <sound/driver.h>
40#include <linux/delay.h>
41#include <linux/init.h>
42#include <linux/interrupt.h>
43#include <linux/pci.h>
44#include <linux/slab.h>
45#include <linux/moduleparam.h>
46#include <linux/firmware.h>
47#include <sound/core.h>
48#include <sound/info.h>
49#include <sound/control.h>
50#include <sound/pcm.h>
51#include <sound/pcm_params.h>
52#include <sound/asoundef.h>
53#include <sound/initval.h>
54#include <asm/io.h>
55#include <asm/atomic.h>
56#include "echoaudio.h"
57
58#define FW_DARLA20_DSP 0
59
60static const struct firmware card_fw[] = {
61 {0, "darla20_dsp.fw"}
62};
63
64static struct pci_device_id snd_echo_ids[] = {
65 {0x1057, 0x1801, 0xECC0, 0x0010, 0, 0, 0}, /* DSP 56301 Darla20 rev.0 */
66 {0,}
67};
68
69static struct snd_pcm_hardware pcm_hardware_skel = {
70 .info = SNDRV_PCM_INFO_MMAP |
71 SNDRV_PCM_INFO_INTERLEAVED |
72 SNDRV_PCM_INFO_BLOCK_TRANSFER |
73 SNDRV_PCM_INFO_MMAP_VALID |
74 SNDRV_PCM_INFO_PAUSE |
75 SNDRV_PCM_INFO_SYNC_START,
76 .formats = SNDRV_PCM_FMTBIT_U8 |
77 SNDRV_PCM_FMTBIT_S16_LE |
78 SNDRV_PCM_FMTBIT_S24_3LE |
79 SNDRV_PCM_FMTBIT_S32_LE |
80 SNDRV_PCM_FMTBIT_S32_BE,
81 .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
82 .rate_min = 44100,
83 .rate_max = 48000,
84 .channels_min = 1,
85 .channels_max = 2,
86 .buffer_bytes_max = 262144,
87 .period_bytes_min = 32,
88 .period_bytes_max = 131072,
89 .periods_min = 2,
90 .periods_max = 220,
91 /* One page (4k) contains 512 instructions. I don't know if the hw
92 supports lists longer than this. In this case periods_max=220 is a
93 safe limit to make sure the list never exceeds 512 instructions. */
94};
95
96
97#include "darla20_dsp.c"
98#include "echoaudio_dsp.c"
99#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c
new file mode 100644
index 000000000000..4159e3bc186f
--- /dev/null
+++ b/sound/pci/echoaudio/darla20_dsp.c
@@ -0,0 +1,125 @@
1/***************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
33{
34 int err;
35
36 DE_INIT(("init_hw() - Darla20\n"));
37 snd_assert((subdevice_id & 0xfff0) == DARLA20, return -ENODEV);
38
39 if ((err = init_dsp_comm_page(chip))) {
40 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
41 return err;
42 }
43
44 chip->device_id = device_id;
45 chip->subdevice_id = subdevice_id;
46 chip->bad_board = TRUE;
47 chip->dsp_code_to_load = &card_fw[FW_DARLA20_DSP];
48 chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
49 chip->clock_state = GD_CLOCK_UNDEF;
50 /* Since this card has no ASIC, mark it as loaded so everything
51 works OK */
52 chip->asic_loaded = TRUE;
53 chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
54
55 if ((err = load_firmware(chip)) < 0)
56 return err;
57 chip->bad_board = FALSE;
58
59 if ((err = init_line_levels(chip)) < 0)
60 return err;
61
62 DE_INIT(("init_hw done\n"));
63 return err;
64}
65
66
67
68/* The Darla20 has no external clock sources */
69static u32 detect_input_clocks(const struct echoaudio *chip)
70{
71 return ECHO_CLOCK_BIT_INTERNAL;
72}
73
74
75
76/* The Darla20 has no ASIC. Just do nothing */
77static int load_asic(struct echoaudio *chip)
78{
79 return 0;
80}
81
82
83
84static int set_sample_rate(struct echoaudio *chip, u32 rate)
85{
86 u8 clock_state, spdif_status;
87
88 if (wait_handshake(chip))
89 return -EIO;
90
91 switch (rate) {
92 case 44100:
93 clock_state = GD_CLOCK_44;
94 spdif_status = GD_SPDIF_STATUS_44;
95 break;
96 case 48000:
97 clock_state = GD_CLOCK_48;
98 spdif_status = GD_SPDIF_STATUS_48;
99 break;
100 default:
101 clock_state = GD_CLOCK_NOCHANGE;
102 spdif_status = GD_SPDIF_STATUS_NOCHANGE;
103 break;
104 }
105
106 if (chip->clock_state == clock_state)
107 clock_state = GD_CLOCK_NOCHANGE;
108 if (spdif_status == chip->spdif_status)
109 spdif_status = GD_SPDIF_STATUS_NOCHANGE;
110
111 chip->comm_page->sample_rate = cpu_to_le32(rate);
112 chip->comm_page->gd_clock_state = clock_state;
113 chip->comm_page->gd_spdif_status = spdif_status;
114 chip->comm_page->gd_resampler_state = 3; /* magic number - should always be 3 */
115
116 /* Save the new audio state if it changed */
117 if (clock_state != GD_CLOCK_NOCHANGE)
118 chip->clock_state = clock_state;
119 if (spdif_status != GD_SPDIF_STATUS_NOCHANGE)
120 chip->spdif_status = spdif_status;
121 chip->sample_rate = rate;
122
123 clear_handshake(chip);
124 return send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
125}
diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c
new file mode 100644
index 000000000000..e59a982ee361
--- /dev/null
+++ b/sound/pci/echoaudio/darla24.c
@@ -0,0 +1,106 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define ECHOGALS_FAMILY
20#define ECHOCARD_DARLA24
21#define ECHOCARD_NAME "Darla24"
22#define ECHOCARD_HAS_MONITOR
23#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
24#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
25#define ECHOCARD_HAS_EXTERNAL_CLOCK
26#define ECHOCARD_HAS_SUPER_INTERLEAVE
27
28/* Pipe indexes */
29#define PX_ANALOG_OUT 0 /* 8 */
30#define PX_DIGITAL_OUT 8 /* 0 */
31#define PX_ANALOG_IN 8 /* 2 */
32#define PX_DIGITAL_IN 10 /* 0 */
33#define PX_NUM 10
34
35/* Bus indexes */
36#define BX_ANALOG_OUT 0 /* 8 */
37#define BX_DIGITAL_OUT 8 /* 0 */
38#define BX_ANALOG_IN 8 /* 2 */
39#define BX_DIGITAL_IN 10 /* 0 */
40#define BX_NUM 10
41
42
43#include <sound/driver.h>
44#include <linux/delay.h>
45#include <linux/init.h>
46#include <linux/interrupt.h>
47#include <linux/pci.h>
48#include <linux/slab.h>
49#include <linux/moduleparam.h>
50#include <linux/firmware.h>
51#include <sound/core.h>
52#include <sound/info.h>
53#include <sound/control.h>
54#include <sound/pcm.h>
55#include <sound/pcm_params.h>
56#include <sound/asoundef.h>
57#include <sound/initval.h>
58#include <asm/io.h>
59#include <asm/atomic.h>
60#include "echoaudio.h"
61
62#define FW_DARLA24_DSP 0
63
64static const struct firmware card_fw[] = {
65 {0, "darla24_dsp.fw"}
66};
67
68static struct pci_device_id snd_echo_ids[] = {
69 {0x1057, 0x1801, 0xECC0, 0x0040, 0, 0, 0}, /* DSP 56301 Darla24 rev.0 */
70 {0x1057, 0x1801, 0xECC0, 0x0041, 0, 0, 0}, /* DSP 56301 Darla24 rev.1 */
71 {0,}
72};
73
74static struct snd_pcm_hardware pcm_hardware_skel = {
75 .info = SNDRV_PCM_INFO_MMAP |
76 SNDRV_PCM_INFO_INTERLEAVED |
77 SNDRV_PCM_INFO_BLOCK_TRANSFER |
78 SNDRV_PCM_INFO_MMAP_VALID |
79 SNDRV_PCM_INFO_PAUSE |
80 SNDRV_PCM_INFO_SYNC_START,
81 .formats = SNDRV_PCM_FMTBIT_U8 |
82 SNDRV_PCM_FMTBIT_S16_LE |
83 SNDRV_PCM_FMTBIT_S24_3LE |
84 SNDRV_PCM_FMTBIT_S32_LE |
85 SNDRV_PCM_FMTBIT_S32_BE,
86 .rates = SNDRV_PCM_RATE_8000_48000 |
87 SNDRV_PCM_RATE_88200 |
88 SNDRV_PCM_RATE_96000,
89 .rate_min = 8000,
90 .rate_max = 96000,
91 .channels_min = 1,
92 .channels_max = 8,
93 .buffer_bytes_max = 262144,
94 .period_bytes_min = 32,
95 .period_bytes_max = 131072,
96 .periods_min = 2,
97 .periods_max = 220,
98 /* One page (4k) contains 512 instructions. I don't know if the hw
99 supports lists longer than this. In this case periods_max=220 is a
100 safe limit to make sure the list never exceeds 512 instructions. */
101};
102
103
104#include "darla24_dsp.c"
105#include "echoaudio_dsp.c"
106#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c
new file mode 100644
index 000000000000..79938eed7e9c
--- /dev/null
+++ b/sound/pci/echoaudio/darla24_dsp.c
@@ -0,0 +1,156 @@
1/***************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
33{
34 int err;
35
36 DE_INIT(("init_hw() - Darla24\n"));
37 snd_assert((subdevice_id & 0xfff0) == DARLA24, return -ENODEV);
38
39 if ((err = init_dsp_comm_page(chip))) {
40 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
41 return err;
42 }
43
44 chip->device_id = device_id;
45 chip->subdevice_id = subdevice_id;
46 chip->bad_board = TRUE;
47 chip->dsp_code_to_load = &card_fw[FW_DARLA24_DSP];
48 /* Since this card has no ASIC, mark it as loaded so everything
49 works OK */
50 chip->asic_loaded = TRUE;
51 chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
52 ECHO_CLOCK_BIT_ESYNC;
53
54 if ((err = load_firmware(chip)) < 0)
55 return err;
56 chip->bad_board = FALSE;
57
58 if ((err = init_line_levels(chip)) < 0)
59 return err;
60
61 DE_INIT(("init_hw done\n"));
62 return err;
63}
64
65
66
67static u32 detect_input_clocks(const struct echoaudio *chip)
68{
69 u32 clocks_from_dsp, clock_bits;
70
71 /* Map the DSP clock detect bits to the generic driver clock
72 detect bits */
73 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
74
75 clock_bits = ECHO_CLOCK_BIT_INTERNAL;
76
77 if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_ESYNC)
78 clock_bits |= ECHO_CLOCK_BIT_ESYNC;
79
80 return clock_bits;
81}
82
83
84
85/* The Darla24 has no ASIC. Just do nothing */
86static int load_asic(struct echoaudio *chip)
87{
88 return 0;
89}
90
91
92
93static int set_sample_rate(struct echoaudio *chip, u32 rate)
94{
95 u8 clock;
96
97 switch (rate) {
98 case 96000:
99 clock = GD24_96000;
100 break;
101 case 88200:
102 clock = GD24_88200;
103 break;
104 case 48000:
105 clock = GD24_48000;
106 break;
107 case 44100:
108 clock = GD24_44100;
109 break;
110 case 32000:
111 clock = GD24_32000;
112 break;
113 case 22050:
114 clock = GD24_22050;
115 break;
116 case 16000:
117 clock = GD24_16000;
118 break;
119 case 11025:
120 clock = GD24_11025;
121 break;
122 case 8000:
123 clock = GD24_8000;
124 break;
125 default:
126 DE_ACT(("set_sample_rate: Error, invalid sample rate %d\n",
127 rate));
128 return -EINVAL;
129 }
130
131 if (wait_handshake(chip))
132 return -EIO;
133
134 DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
135 chip->sample_rate = rate;
136
137 /* Override the sample rate if this card is set to Echo sync. */
138 if (chip->input_clock == ECHO_CLOCK_ESYNC)
139 clock = GD24_EXT_SYNC;
140
141 chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP ? */
142 chip->comm_page->gd_clock_state = clock;
143 clear_handshake(chip);
144 return send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
145}
146
147
148
149static int set_input_clock(struct echoaudio *chip, u16 clock)
150{
151 snd_assert(clock == ECHO_CLOCK_INTERNAL ||
152 clock == ECHO_CLOCK_ESYNC, return -EINVAL);
153 chip->input_clock = clock;
154 return set_sample_rate(chip, chip->sample_rate);
155}
156
diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c
new file mode 100644
index 000000000000..12099fe1547d
--- /dev/null
+++ b/sound/pci/echoaudio/echo3g.c
@@ -0,0 +1,118 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define ECHO3G_FAMILY
20#define ECHOCARD_ECHO3G
21#define ECHOCARD_NAME "Echo3G"
22#define ECHOCARD_HAS_MONITOR
23#define ECHOCARD_HAS_ASIC
24#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
25#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
26#define ECHOCARD_HAS_SUPER_INTERLEAVE
27#define ECHOCARD_HAS_DIGITAL_IO
28#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
29#define ECHOCARD_HAS_ADAT 6
30#define ECHOCARD_HAS_EXTERNAL_CLOCK
31#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
32#define ECHOCARD_HAS_MIDI
33#define ECHOCARD_HAS_PHANTOM_POWER
34
35/* Pipe indexes */
36#define PX_ANALOG_OUT 0
37#define PX_DIGITAL_OUT chip->px_digital_out
38#define PX_ANALOG_IN chip->px_analog_in
39#define PX_DIGITAL_IN chip->px_digital_in
40#define PX_NUM chip->px_num
41
42/* Bus indexes */
43#define BX_ANALOG_OUT 0
44#define BX_DIGITAL_OUT chip->bx_digital_out
45#define BX_ANALOG_IN chip->bx_analog_in
46#define BX_DIGITAL_IN chip->bx_digital_in
47#define BX_NUM chip->bx_num
48
49
50#include <sound/driver.h>
51#include <linux/delay.h>
52#include <linux/init.h>
53#include <linux/interrupt.h>
54#include <linux/pci.h>
55#include <linux/slab.h>
56#include <linux/moduleparam.h>
57#include <linux/firmware.h>
58#include <sound/core.h>
59#include <sound/info.h>
60#include <sound/control.h>
61#include <sound/pcm.h>
62#include <sound/pcm_params.h>
63#include <sound/asoundef.h>
64#include <sound/initval.h>
65#include <sound/rawmidi.h>
66#include <asm/io.h>
67#include <asm/atomic.h>
68#include "echoaudio.h"
69
70#define FW_361_LOADER 0
71#define FW_ECHO3G_DSP 1
72#define FW_3G_ASIC 2
73
74static const struct firmware card_fw[] = {
75 {0, "loader_dsp.fw"},
76 {0, "echo3g_dsp.fw"},
77 {0, "3g_asic.fw"}
78};
79
80static struct pci_device_id snd_echo_ids[] = {
81 {0x1057, 0x3410, 0xECC0, 0x0100, 0, 0, 0}, /* Echo 3G */
82 {0,}
83};
84
85static struct snd_pcm_hardware pcm_hardware_skel = {
86 .info = SNDRV_PCM_INFO_MMAP |
87 SNDRV_PCM_INFO_INTERLEAVED |
88 SNDRV_PCM_INFO_BLOCK_TRANSFER |
89 SNDRV_PCM_INFO_MMAP_VALID |
90 SNDRV_PCM_INFO_PAUSE |
91 SNDRV_PCM_INFO_SYNC_START,
92 .formats = SNDRV_PCM_FMTBIT_U8 |
93 SNDRV_PCM_FMTBIT_S16_LE |
94 SNDRV_PCM_FMTBIT_S24_3LE |
95 SNDRV_PCM_FMTBIT_S32_LE |
96 SNDRV_PCM_FMTBIT_S32_BE,
97 .rates = SNDRV_PCM_RATE_32000 |
98 SNDRV_PCM_RATE_44100 |
99 SNDRV_PCM_RATE_48000 |
100 SNDRV_PCM_RATE_88200 |
101 SNDRV_PCM_RATE_96000 |
102 SNDRV_PCM_RATE_CONTINUOUS,
103 .rate_min = 32000,
104 .rate_max = 100000,
105 .channels_min = 1,
106 .channels_max = 8,
107 .buffer_bytes_max = 262144,
108 .period_bytes_min = 32,
109 .period_bytes_max = 131072,
110 .periods_min = 2,
111 .periods_max = 220,
112};
113
114#include "echo3g_dsp.c"
115#include "echoaudio_dsp.c"
116#include "echoaudio_3g.c"
117#include "echoaudio.c"
118#include "midi.c"
diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c
new file mode 100644
index 000000000000..d26a1d1f3ed1
--- /dev/null
+++ b/sound/pci/echoaudio/echo3g_dsp.c
@@ -0,0 +1,131 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31static int load_asic(struct echoaudio *chip);
32static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode);
33static int set_digital_mode(struct echoaudio *chip, u8 mode);
34static int check_asic_status(struct echoaudio *chip);
35static int set_sample_rate(struct echoaudio *chip, u32 rate);
36static int set_input_clock(struct echoaudio *chip, u16 clock);
37static int set_professional_spdif(struct echoaudio *chip, char prof);
38static int set_phantom_power(struct echoaudio *chip, char on);
39static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq,
40 char force);
41
42#include <linux/irq.h>
43
44static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
45{
46 int err;
47
48 local_irq_enable();
49 DE_INIT(("init_hw() - Echo3G\n"));
50 snd_assert((subdevice_id & 0xfff0) == ECHO3G, return -ENODEV);
51
52 if ((err = init_dsp_comm_page(chip))) {
53 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
54 return err;
55 }
56
57 chip->comm_page->e3g_frq_register =
58 __constant_cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2);
59 chip->device_id = device_id;
60 chip->subdevice_id = subdevice_id;
61 chip->bad_board = TRUE;
62 chip->has_midi = TRUE;
63 chip->dsp_code_to_load = &card_fw[FW_ECHO3G_DSP];
64
65 /* Load the DSP code and the ASIC on the PCI card and get
66 what type of external box is attached */
67 err = load_firmware(chip);
68
69 if (err < 0) {
70 return err;
71 } else if (err == E3G_GINA3G_BOX_TYPE) {
72 chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
73 ECHO_CLOCK_BIT_SPDIF |
74 ECHO_CLOCK_BIT_ADAT;
75 chip->card_name = "Gina3G";
76 chip->px_digital_out = chip->bx_digital_out = 6;
77 chip->px_analog_in = chip->bx_analog_in = 14;
78 chip->px_digital_in = chip->bx_digital_in = 16;
79 chip->px_num = chip->bx_num = 24;
80 chip->has_phantom_power = TRUE;
81 chip->hasnt_input_nominal_level = TRUE;
82 } else if (err == E3G_LAYLA3G_BOX_TYPE) {
83 chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
84 ECHO_CLOCK_BIT_SPDIF |
85 ECHO_CLOCK_BIT_ADAT |
86 ECHO_CLOCK_BIT_WORD;
87 chip->card_name = "Layla3G";
88 chip->px_digital_out = chip->bx_digital_out = 8;
89 chip->px_analog_in = chip->bx_analog_in = 16;
90 chip->px_digital_in = chip->bx_digital_in = 24;
91 chip->px_num = chip->bx_num = 32;
92 } else {
93 return -ENODEV;
94 }
95
96 chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
97 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
98 ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
99 chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
100 chip->professional_spdif = FALSE;
101 chip->non_audio_spdif = FALSE;
102 chip->bad_board = FALSE;
103
104 if ((err = init_line_levels(chip)) < 0)
105 return err;
106 err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
107 snd_assert(err >= 0, return err);
108 err = set_phantom_power(chip, 0);
109 snd_assert(err >= 0, return err);
110 err = set_professional_spdif(chip, TRUE);
111
112 DE_INIT(("init_hw done\n"));
113 return err;
114}
115
116
117
118static int set_phantom_power(struct echoaudio *chip, char on)
119{
120 u32 control_reg = le32_to_cpu(chip->comm_page->control_register);
121
122 if (on)
123 control_reg |= E3G_PHANTOM_POWER;
124 else
125 control_reg &= ~E3G_PHANTOM_POWER;
126
127 chip->phantom_power = on;
128 return write_control_reg(chip, control_reg,
129 le32_to_cpu(chip->comm_page->e3g_frq_register),
130 0);
131}
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
new file mode 100644
index 000000000000..43b408ada1da
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -0,0 +1,2196 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19MODULE_AUTHOR("Giuliano Pochini <pochini@shiny.it>");
20MODULE_LICENSE("GPL v2");
21MODULE_DESCRIPTION("Echoaudio " ECHOCARD_NAME " soundcards driver");
22MODULE_SUPPORTED_DEVICE("{{Echoaudio," ECHOCARD_NAME "}}");
23MODULE_DEVICE_TABLE(pci, snd_echo_ids);
24
25static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
26static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
27static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
28
29module_param_array(index, int, NULL, 0444);
30MODULE_PARM_DESC(index, "Index value for " ECHOCARD_NAME " soundcard.");
31module_param_array(id, charp, NULL, 0444);
32MODULE_PARM_DESC(id, "ID string for " ECHOCARD_NAME " soundcard.");
33module_param_array(enable, bool, NULL, 0444);
34MODULE_PARM_DESC(enable, "Enable " ECHOCARD_NAME " soundcard.");
35
36static unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999};
37
38static int get_firmware(const struct firmware **fw_entry,
39 const struct firmware *frm, struct echoaudio *chip)
40{
41 int err;
42 char name[30];
43 DE_ACT(("firmware requested: %s\n", frm->data));
44 snprintf(name, sizeof(name), "ea/%s", frm->data);
45 if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0)
46 snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err);
47 return err;
48}
49
50static void free_firmware(const struct firmware *fw_entry)
51{
52 release_firmware(fw_entry);
53 DE_ACT(("firmware released\n"));
54}
55
56
57
58/******************************************************************************
59 PCM interface
60******************************************************************************/
61
62static void audiopipe_free(struct snd_pcm_runtime *runtime)
63{
64 struct audiopipe *pipe = runtime->private_data;
65
66 if (pipe->sgpage.area)
67 snd_dma_free_pages(&pipe->sgpage);
68 kfree(pipe);
69}
70
71
72
73static int hw_rule_capture_format_by_channels(struct snd_pcm_hw_params *params,
74 struct snd_pcm_hw_rule *rule)
75{
76 struct snd_interval *c = hw_param_interval(params,
77 SNDRV_PCM_HW_PARAM_CHANNELS);
78 struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
79 struct snd_mask fmt;
80
81 snd_mask_any(&fmt);
82
83#ifndef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
84 /* >=2 channels cannot be S32_BE */
85 if (c->min == 2) {
86 fmt.bits[0] &= ~SNDRV_PCM_FMTBIT_S32_BE;
87 return snd_mask_refine(f, &fmt);
88 }
89#endif
90 /* > 2 channels cannot be U8 and S32_BE */
91 if (c->min > 2) {
92 fmt.bits[0] &= ~(SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_BE);
93 return snd_mask_refine(f, &fmt);
94 }
95 /* Mono is ok with any format */
96 return 0;
97}
98
99
100
101static int hw_rule_capture_channels_by_format(struct snd_pcm_hw_params *params,
102 struct snd_pcm_hw_rule *rule)
103{
104 struct snd_interval *c = hw_param_interval(params,
105 SNDRV_PCM_HW_PARAM_CHANNELS);
106 struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
107 struct snd_interval ch;
108
109 snd_interval_any(&ch);
110
111 /* S32_BE is mono (and stereo) only */
112 if (f->bits[0] == SNDRV_PCM_FMTBIT_S32_BE) {
113 ch.min = 1;
114#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
115 ch.max = 2;
116#else
117 ch.max = 1;
118#endif
119 ch.integer = 1;
120 return snd_interval_refine(c, &ch);
121 }
122 /* U8 can be only mono or stereo */
123 if (f->bits[0] == SNDRV_PCM_FMTBIT_U8) {
124 ch.min = 1;
125 ch.max = 2;
126 ch.integer = 1;
127 return snd_interval_refine(c, &ch);
128 }
129 /* S16_LE, S24_3LE and S32_LE support any number of channels. */
130 return 0;
131}
132
133
134
135static int hw_rule_playback_format_by_channels(struct snd_pcm_hw_params *params,
136 struct snd_pcm_hw_rule *rule)
137{
138 struct snd_interval *c = hw_param_interval(params,
139 SNDRV_PCM_HW_PARAM_CHANNELS);
140 struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
141 struct snd_mask fmt;
142 u64 fmask;
143 snd_mask_any(&fmt);
144
145 fmask = fmt.bits[0] + ((u64)fmt.bits[1] << 32);
146
147 /* >2 channels must be S16_LE, S24_3LE or S32_LE */
148 if (c->min > 2) {
149 fmask &= SNDRV_PCM_FMTBIT_S16_LE |
150 SNDRV_PCM_FMTBIT_S24_3LE |
151 SNDRV_PCM_FMTBIT_S32_LE;
152 /* 1 channel must be S32_BE or S32_LE */
153 } else if (c->max == 1)
154 fmask &= SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE;
155#ifndef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
156 /* 2 channels cannot be S32_BE */
157 else if (c->min == 2 && c->max == 2)
158 fmask &= ~SNDRV_PCM_FMTBIT_S32_BE;
159#endif
160 else
161 return 0;
162
163 fmt.bits[0] &= (u32)fmask;
164 fmt.bits[1] &= (u32)(fmask >> 32);
165 return snd_mask_refine(f, &fmt);
166}
167
168
169
170static int hw_rule_playback_channels_by_format(struct snd_pcm_hw_params *params,
171 struct snd_pcm_hw_rule *rule)
172{
173 struct snd_interval *c = hw_param_interval(params,
174 SNDRV_PCM_HW_PARAM_CHANNELS);
175 struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
176 struct snd_interval ch;
177 u64 fmask;
178
179 snd_interval_any(&ch);
180 ch.integer = 1;
181 fmask = f->bits[0] + ((u64)f->bits[1] << 32);
182
183 /* S32_BE is mono (and stereo) only */
184 if (fmask == SNDRV_PCM_FMTBIT_S32_BE) {
185 ch.min = 1;
186#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
187 ch.max = 2;
188#else
189 ch.max = 1;
190#endif
191 /* U8 is stereo only */
192 } else if (fmask == SNDRV_PCM_FMTBIT_U8)
193 ch.min = ch.max = 2;
194 /* S16_LE and S24_3LE must be at least stereo */
195 else if (!(fmask & ~(SNDRV_PCM_FMTBIT_S16_LE |
196 SNDRV_PCM_FMTBIT_S24_3LE)))
197 ch.min = 2;
198 else
199 return 0;
200
201 return snd_interval_refine(c, &ch);
202}
203
204
205
206/* Since the sample rate is a global setting, do allow the user to change the
207sample rate only if there is only one pcm device open. */
208static int hw_rule_sample_rate(struct snd_pcm_hw_params *params,
209 struct snd_pcm_hw_rule *rule)
210{
211 struct snd_interval *rate = hw_param_interval(params,
212 SNDRV_PCM_HW_PARAM_RATE);
213 struct echoaudio *chip = rule->private;
214 struct snd_interval fixed;
215
216 if (!chip->can_set_rate) {
217 snd_interval_any(&fixed);
218 fixed.min = fixed.max = chip->sample_rate;
219 return snd_interval_refine(rate, &fixed);
220 }
221 return 0;
222}
223
224
225static int pcm_open(struct snd_pcm_substream *substream,
226 signed char max_channels)
227{
228 struct echoaudio *chip;
229 struct snd_pcm_runtime *runtime;
230 struct audiopipe *pipe;
231 int err, i;
232
233 if (max_channels <= 0)
234 return -EAGAIN;
235
236 chip = snd_pcm_substream_chip(substream);
237 runtime = substream->runtime;
238
239 if (!(pipe = kmalloc(sizeof(struct audiopipe), GFP_KERNEL)))
240 return -ENOMEM;
241 memset(pipe, 0, sizeof(struct audiopipe));
242 pipe->index = -1; /* Not configured yet */
243
244 /* Set up hw capabilities and contraints */
245 memcpy(&pipe->hw, &pcm_hardware_skel, sizeof(struct snd_pcm_hardware));
246 DE_HWP(("max_channels=%d\n", max_channels));
247 pipe->constr.list = channels_list;
248 pipe->constr.mask = 0;
249 for (i = 0; channels_list[i] <= max_channels; i++);
250 pipe->constr.count = i;
251 if (pipe->hw.channels_max > max_channels)
252 pipe->hw.channels_max = max_channels;
253 if (chip->digital_mode == DIGITAL_MODE_ADAT) {
254 pipe->hw.rate_max = 48000;
255 pipe->hw.rates &= SNDRV_PCM_RATE_8000_48000;
256 }
257
258 runtime->hw = pipe->hw;
259 runtime->private_data = pipe;
260 runtime->private_free = audiopipe_free;
261 snd_pcm_set_sync(substream);
262
263 /* Only mono and any even number of channels are allowed */
264 if ((err = snd_pcm_hw_constraint_list(runtime, 0,
265 SNDRV_PCM_HW_PARAM_CHANNELS,
266 &pipe->constr)) < 0)
267 return err;
268
269 /* All periods should have the same size */
270 if ((err = snd_pcm_hw_constraint_integer(runtime,
271 SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
272 return err;
273
274 /* The hw accesses memory in chunks 32 frames long and they should be
275 32-bytes-aligned. It's not a requirement, but it seems that IRQs are
276 generated with a resolution of 32 frames. Thus we need the following */
277 if ((err = snd_pcm_hw_constraint_step(runtime, 0,
278 SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
279 32)) < 0)
280 return err;
281 if ((err = snd_pcm_hw_constraint_step(runtime, 0,
282 SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
283 32)) < 0)
284 return err;
285
286 if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
287 SNDRV_PCM_HW_PARAM_RATE,
288 hw_rule_sample_rate, chip,
289 SNDRV_PCM_HW_PARAM_RATE, -1)) < 0)
290 return err;
291
292 /* Finally allocate a page for the scatter-gather list */
293 if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV,
294 snd_dma_pci_data(chip->pci),
295 PAGE_SIZE, &pipe->sgpage)) < 0) {
296 DE_HWP(("s-g list allocation failed\n"));
297 return err;
298 }
299
300 return 0;
301}
302
303
304
305static int pcm_analog_in_open(struct snd_pcm_substream *substream)
306{
307 struct echoaudio *chip = snd_pcm_substream_chip(substream);
308 int err;
309
310 DE_ACT(("pcm_analog_in_open\n"));
311 if ((err = pcm_open(substream, num_analog_busses_in(chip) -
312 substream->number)) < 0)
313 return err;
314 if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
315 SNDRV_PCM_HW_PARAM_CHANNELS,
316 hw_rule_capture_channels_by_format, NULL,
317 SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0)
318 return err;
319 if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
320 SNDRV_PCM_HW_PARAM_FORMAT,
321 hw_rule_capture_format_by_channels, NULL,
322 SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
323 return err;
324 atomic_inc(&chip->opencount);
325 if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
326 chip->can_set_rate=0;
327 DE_HWP(("pcm_analog_in_open cs=%d oc=%d r=%d\n",
328 chip->can_set_rate, atomic_read(&chip->opencount),
329 chip->sample_rate));
330 return 0;
331}
332
333
334
335static int pcm_analog_out_open(struct snd_pcm_substream *substream)
336{
337 struct echoaudio *chip = snd_pcm_substream_chip(substream);
338 int max_channels, err;
339
340#ifdef ECHOCARD_HAS_VMIXER
341 max_channels = num_pipes_out(chip);
342#else
343 max_channels = num_analog_busses_out(chip);
344#endif
345 DE_ACT(("pcm_analog_out_open\n"));
346 if ((err = pcm_open(substream, max_channels - substream->number)) < 0)
347 return err;
348 if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
349 SNDRV_PCM_HW_PARAM_CHANNELS,
350 hw_rule_playback_channels_by_format,
351 NULL,
352 SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0)
353 return err;
354 if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
355 SNDRV_PCM_HW_PARAM_FORMAT,
356 hw_rule_playback_format_by_channels,
357 NULL,
358 SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
359 return err;
360 atomic_inc(&chip->opencount);
361 if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
362 chip->can_set_rate=0;
363 DE_HWP(("pcm_analog_out_open cs=%d oc=%d r=%d\n",
364 chip->can_set_rate, atomic_read(&chip->opencount),
365 chip->sample_rate));
366 return 0;
367}
368
369
370
371#ifdef ECHOCARD_HAS_DIGITAL_IO
372
373static int pcm_digital_in_open(struct snd_pcm_substream *substream)
374{
375 struct echoaudio *chip = snd_pcm_substream_chip(substream);
376 int err, max_channels;
377
378 DE_ACT(("pcm_digital_in_open\n"));
379 max_channels = num_digital_busses_in(chip) - substream->number;
380 down(&chip->mode_mutex);
381 if (chip->digital_mode == DIGITAL_MODE_ADAT)
382 err = pcm_open(substream, max_channels);
383 else /* If the card has ADAT, subtract the 6 channels
384 * that S/PDIF doesn't have
385 */
386 err = pcm_open(substream, max_channels - ECHOCARD_HAS_ADAT);
387
388 if (err < 0)
389 goto din_exit;
390
391 if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
392 SNDRV_PCM_HW_PARAM_CHANNELS,
393 hw_rule_capture_channels_by_format, NULL,
394 SNDRV_PCM_HW_PARAM_FORMAT, -1)) < 0)
395 goto din_exit;
396 if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
397 SNDRV_PCM_HW_PARAM_FORMAT,
398 hw_rule_capture_format_by_channels, NULL,
399 SNDRV_PCM_HW_PARAM_CHANNELS, -1)) < 0)
400 goto din_exit;
401
402 atomic_inc(&chip->opencount);
403 if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
404 chip->can_set_rate=0;
405
406din_exit:
407 up(&chip->mode_mutex);
408 return err;
409}
410
411
412
413#ifndef ECHOCARD_HAS_VMIXER /* See the note in snd_echo_new_pcm() */
414
415static int pcm_digital_out_open(struct snd_pcm_substream *substream)
416{
417 struct echoaudio *chip = snd_pcm_substream_chip(substream);
418 int err, max_channels;
419
420 DE_ACT(("pcm_digital_out_open\n"));
421 max_channels = num_digital_busses_out(chip) - substream->number;
422 down(&chip->mode_mutex);
423 if (chip->digital_mode == DIGITAL_MODE_ADAT)
424 err = pcm_open(substream, max_channels);
425 else /* If the card has ADAT, subtract the 6 channels
426 * that S/PDIF doesn't have
427 */
428 err = pcm_open(substream, max_channels - ECHOCARD_HAS_ADAT);
429
430 if (err < 0)
431 goto dout_exit;
432
433 if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
434 SNDRV_PCM_HW_PARAM_CHANNELS,
435 hw_rule_playback_channels_by_format,
436 NULL, SNDRV_PCM_HW_PARAM_FORMAT,
437 -1)) < 0)
438 goto dout_exit;
439 if ((err = snd_pcm_hw_rule_add(substream->runtime, 0,
440 SNDRV_PCM_HW_PARAM_FORMAT,
441 hw_rule_playback_format_by_channels,
442 NULL, SNDRV_PCM_HW_PARAM_CHANNELS,
443 -1)) < 0)
444 goto dout_exit;
445 atomic_inc(&chip->opencount);
446 if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
447 chip->can_set_rate=0;
448dout_exit:
449 up(&chip->mode_mutex);
450 return err;
451}
452
453#endif /* !ECHOCARD_HAS_VMIXER */
454
455#endif /* ECHOCARD_HAS_DIGITAL_IO */
456
457
458
459static int pcm_close(struct snd_pcm_substream *substream)
460{
461 struct echoaudio *chip = snd_pcm_substream_chip(substream);
462 int oc;
463
464 /* Nothing to do here. Audio is already off and pipe will be
465 * freed by its callback
466 */
467 DE_ACT(("pcm_close\n"));
468
469 atomic_dec(&chip->opencount);
470 oc = atomic_read(&chip->opencount);
471 DE_ACT(("pcm_close oc=%d cs=%d rs=%d\n", oc,
472 chip->can_set_rate, chip->rate_set));
473 if (oc < 2)
474 chip->can_set_rate = 1;
475 if (oc == 0)
476 chip->rate_set = 0;
477 DE_ACT(("pcm_close2 oc=%d cs=%d rs=%d\n", oc,
478 chip->can_set_rate,chip->rate_set));
479
480 return 0;
481}
482
483
484
485/* Channel allocation and scatter-gather list setup */
486static int init_engine(struct snd_pcm_substream *substream,
487 struct snd_pcm_hw_params *hw_params,
488 int pipe_index, int interleave)
489{
490 struct echoaudio *chip;
491 int err, per, rest, page, edge, offs;
492 struct snd_sg_buf *sgbuf;
493 struct audiopipe *pipe;
494
495 chip = snd_pcm_substream_chip(substream);
496 pipe = (struct audiopipe *) substream->runtime->private_data;
497
498 /* Sets up che hardware. If it's already initialized, reset and
499 * redo with the new parameters
500 */
501 spin_lock_irq(&chip->lock);
502 if (pipe->index >= 0) {
503 DE_HWP(("hwp_ie free(%d)\n", pipe->index));
504 err = free_pipes(chip, pipe);
505 snd_assert(!err);
506 chip->substream[pipe->index] = NULL;
507 }
508
509 err = allocate_pipes(chip, pipe, pipe_index, interleave);
510 if (err < 0) {
511 spin_unlock_irq(&chip->lock);
512 DE_ACT((KERN_NOTICE "allocate_pipes(%d) err=%d\n",
513 pipe_index, err));
514 return err;
515 }
516 spin_unlock_irq(&chip->lock);
517 DE_ACT((KERN_NOTICE "allocate_pipes()=%d\n", pipe_index));
518
519 DE_HWP(("pcm_hw_params (bufsize=%dB periods=%d persize=%dB)\n",
520 params_buffer_bytes(hw_params), params_periods(hw_params),
521 params_period_bytes(hw_params)));
522 err = snd_pcm_lib_malloc_pages(substream,
523 params_buffer_bytes(hw_params));
524 if (err < 0) {
525 snd_printk(KERN_ERR "malloc_pages err=%d\n", err);
526 spin_lock_irq(&chip->lock);
527 free_pipes(chip, pipe);
528 spin_unlock_irq(&chip->lock);
529 pipe->index = -1;
530 return err;
531 }
532
533 sgbuf = snd_pcm_substream_sgbuf(substream);
534
535 DE_HWP(("pcm_hw_params table size=%d pages=%d\n",
536 sgbuf->size, sgbuf->pages));
537 sglist_init(chip, pipe);
538 edge = PAGE_SIZE;
539 for (offs = page = per = 0; offs < params_buffer_bytes(hw_params);
540 per++) {
541 rest = params_period_bytes(hw_params);
542 if (offs + rest > params_buffer_bytes(hw_params))
543 rest = params_buffer_bytes(hw_params) - offs;
544 while (rest) {
545 if (rest <= edge - offs) {
546 sglist_add_mapping(chip, pipe,
547 snd_sgbuf_get_addr(sgbuf, offs),
548 rest);
549 sglist_add_irq(chip, pipe);
550 offs += rest;
551 rest = 0;
552 } else {
553 sglist_add_mapping(chip, pipe,
554 snd_sgbuf_get_addr(sgbuf, offs),
555 edge - offs);
556 rest -= edge - offs;
557 offs = edge;
558 }
559 if (offs == edge) {
560 edge += PAGE_SIZE;
561 page++;
562 }
563 }
564 }
565
566 /* Close the ring buffer */
567 sglist_wrap(chip, pipe);
568
569 /* This stuff is used by the irq handler, so it must be
570 * initialized before chip->substream
571 */
572 chip->last_period[pipe_index] = 0;
573 pipe->last_counter = 0;
574 pipe->position = 0;
575 smp_wmb();
576 chip->substream[pipe_index] = substream;
577 chip->rate_set = 1;
578 spin_lock_irq(&chip->lock);
579 set_sample_rate(chip, hw_params->rate_num / hw_params->rate_den);
580 spin_unlock_irq(&chip->lock);
581 DE_HWP(("pcm_hw_params ok\n"));
582 return 0;
583}
584
585
586
587static int pcm_analog_in_hw_params(struct snd_pcm_substream *substream,
588 struct snd_pcm_hw_params *hw_params)
589{
590 struct echoaudio *chip = snd_pcm_substream_chip(substream);
591
592 return init_engine(substream, hw_params, px_analog_in(chip) +
593 substream->number, params_channels(hw_params));
594}
595
596
597
598static int pcm_analog_out_hw_params(struct snd_pcm_substream *substream,
599 struct snd_pcm_hw_params *hw_params)
600{
601 return init_engine(substream, hw_params, substream->number,
602 params_channels(hw_params));
603}
604
605
606
607#ifdef ECHOCARD_HAS_DIGITAL_IO
608
609static int pcm_digital_in_hw_params(struct snd_pcm_substream *substream,
610 struct snd_pcm_hw_params *hw_params)
611{
612 struct echoaudio *chip = snd_pcm_substream_chip(substream);
613
614 return init_engine(substream, hw_params, px_digital_in(chip) +
615 substream->number, params_channels(hw_params));
616}
617
618
619
620#ifndef ECHOCARD_HAS_VMIXER /* See the note in snd_echo_new_pcm() */
621static int pcm_digital_out_hw_params(struct snd_pcm_substream *substream,
622 struct snd_pcm_hw_params *hw_params)
623{
624 struct echoaudio *chip = snd_pcm_substream_chip(substream);
625
626 return init_engine(substream, hw_params, px_digital_out(chip) +
627 substream->number, params_channels(hw_params));
628}
629#endif /* !ECHOCARD_HAS_VMIXER */
630
631#endif /* ECHOCARD_HAS_DIGITAL_IO */
632
633
634
635static int pcm_hw_free(struct snd_pcm_substream *substream)
636{
637 struct echoaudio *chip;
638 struct audiopipe *pipe;
639
640 chip = snd_pcm_substream_chip(substream);
641 pipe = (struct audiopipe *) substream->runtime->private_data;
642
643 spin_lock_irq(&chip->lock);
644 if (pipe->index >= 0) {
645 DE_HWP(("pcm_hw_free(%d)\n", pipe->index));
646 free_pipes(chip, pipe);
647 chip->substream[pipe->index] = NULL;
648 pipe->index = -1;
649 }
650 spin_unlock_irq(&chip->lock);
651
652 DE_HWP(("pcm_hw_freed\n"));
653 snd_pcm_lib_free_pages(substream);
654 return 0;
655}
656
657
658
659static int pcm_prepare(struct snd_pcm_substream *substream)
660{
661 struct echoaudio *chip = snd_pcm_substream_chip(substream);
662 struct snd_pcm_runtime *runtime = substream->runtime;
663 struct audioformat format;
664 int pipe_index = ((struct audiopipe *)runtime->private_data)->index;
665
666 DE_HWP(("Prepare rate=%d format=%d channels=%d\n",
667 runtime->rate, runtime->format, runtime->channels));
668 format.interleave = runtime->channels;
669 format.data_are_bigendian = 0;
670 format.mono_to_stereo = 0;
671 switch (runtime->format) {
672 case SNDRV_PCM_FORMAT_U8:
673 format.bits_per_sample = 8;
674 break;
675 case SNDRV_PCM_FORMAT_S16_LE:
676 format.bits_per_sample = 16;
677 break;
678 case SNDRV_PCM_FORMAT_S24_3LE:
679 format.bits_per_sample = 24;
680 break;
681 case SNDRV_PCM_FORMAT_S32_BE:
682 format.data_are_bigendian = 1;
683 case SNDRV_PCM_FORMAT_S32_LE:
684 format.bits_per_sample = 32;
685 break;
686 default:
687 DE_HWP(("Prepare error: unsupported format %d\n",
688 runtime->format));
689 return -EINVAL;
690 }
691
692 snd_assert(pipe_index < px_num(chip), return -EINVAL);
693 snd_assert(is_pipe_allocated(chip, pipe_index), return -EINVAL);
694 set_audio_format(chip, pipe_index, &format);
695 return 0;
696}
697
698
699
700static int pcm_trigger(struct snd_pcm_substream *substream, int cmd)
701{
702 struct echoaudio *chip = snd_pcm_substream_chip(substream);
703 struct snd_pcm_runtime *runtime = substream->runtime;
704 struct audiopipe *pipe = runtime->private_data;
705 int i, err;
706 u32 channelmask = 0;
707 struct list_head *pos;
708 struct snd_pcm_substream *s;
709
710 snd_pcm_group_for_each(pos, substream) {
711 s = snd_pcm_group_substream_entry(pos);
712 for (i = 0; i < DSP_MAXPIPES; i++) {
713 if (s == chip->substream[i]) {
714 channelmask |= 1 << i;
715 snd_pcm_trigger_done(s, substream);
716 }
717 }
718 }
719
720 spin_lock(&chip->lock);
721 switch (cmd) {
722 case SNDRV_PCM_TRIGGER_START:
723 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
724 DE_ACT(("pcm_trigger start\n"));
725 for (i = 0; i < DSP_MAXPIPES; i++) {
726 if (channelmask & (1 << i)) {
727 pipe = chip->substream[i]->runtime->private_data;
728 switch (pipe->state) {
729 case PIPE_STATE_STOPPED:
730 chip->last_period[i] = 0;
731 pipe->last_counter = 0;
732 pipe->position = 0;
733 *pipe->dma_counter = 0;
734 case PIPE_STATE_PAUSED:
735 pipe->state = PIPE_STATE_STARTED;
736 break;
737 case PIPE_STATE_STARTED:
738 break;
739 }
740 }
741 }
742 err = start_transport(chip, channelmask,
743 chip->pipe_cyclic_mask);
744 break;
745 case SNDRV_PCM_TRIGGER_STOP:
746 DE_ACT(("pcm_trigger stop\n"));
747 for (i = 0; i < DSP_MAXPIPES; i++) {
748 if (channelmask & (1 << i)) {
749 pipe = chip->substream[i]->runtime->private_data;
750 pipe->state = PIPE_STATE_STOPPED;
751 }
752 }
753 err = stop_transport(chip, channelmask);
754 break;
755 case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
756 DE_ACT(("pcm_trigger pause\n"));
757 for (i = 0; i < DSP_MAXPIPES; i++) {
758 if (channelmask & (1 << i)) {
759 pipe = chip->substream[i]->runtime->private_data;
760 pipe->state = PIPE_STATE_PAUSED;
761 }
762 }
763 err = pause_transport(chip, channelmask);
764 break;
765 default:
766 err = -EINVAL;
767 }
768 spin_unlock(&chip->lock);
769 return err;
770}
771
772
773
774static snd_pcm_uframes_t pcm_pointer(struct snd_pcm_substream *substream)
775{
776 struct snd_pcm_runtime *runtime = substream->runtime;
777 struct audiopipe *pipe = runtime->private_data;
778 size_t cnt, bufsize, pos;
779
780 cnt = le32_to_cpu(*pipe->dma_counter);
781 pipe->position += cnt - pipe->last_counter;
782 pipe->last_counter = cnt;
783 bufsize = substream->runtime->buffer_size;
784 pos = bytes_to_frames(substream->runtime, pipe->position);
785
786 while (pos >= bufsize) {
787 pipe->position -= frames_to_bytes(substream->runtime, bufsize);
788 pos -= bufsize;
789 }
790 return pos;
791}
792
793
794
795/* pcm *_ops structures */
796static struct snd_pcm_ops analog_playback_ops = {
797 .open = pcm_analog_out_open,
798 .close = pcm_close,
799 .ioctl = snd_pcm_lib_ioctl,
800 .hw_params = pcm_analog_out_hw_params,
801 .hw_free = pcm_hw_free,
802 .prepare = pcm_prepare,
803 .trigger = pcm_trigger,
804 .pointer = pcm_pointer,
805 .page = snd_pcm_sgbuf_ops_page,
806};
807static struct snd_pcm_ops analog_capture_ops = {
808 .open = pcm_analog_in_open,
809 .close = pcm_close,
810 .ioctl = snd_pcm_lib_ioctl,
811 .hw_params = pcm_analog_in_hw_params,
812 .hw_free = pcm_hw_free,
813 .prepare = pcm_prepare,
814 .trigger = pcm_trigger,
815 .pointer = pcm_pointer,
816 .page = snd_pcm_sgbuf_ops_page,
817};
818#ifdef ECHOCARD_HAS_DIGITAL_IO
819#ifndef ECHOCARD_HAS_VMIXER
820static struct snd_pcm_ops digital_playback_ops = {
821 .open = pcm_digital_out_open,
822 .close = pcm_close,
823 .ioctl = snd_pcm_lib_ioctl,
824 .hw_params = pcm_digital_out_hw_params,
825 .hw_free = pcm_hw_free,
826 .prepare = pcm_prepare,
827 .trigger = pcm_trigger,
828 .pointer = pcm_pointer,
829 .page = snd_pcm_sgbuf_ops_page,
830};
831#endif /* !ECHOCARD_HAS_VMIXER */
832static struct snd_pcm_ops digital_capture_ops = {
833 .open = pcm_digital_in_open,
834 .close = pcm_close,
835 .ioctl = snd_pcm_lib_ioctl,
836 .hw_params = pcm_digital_in_hw_params,
837 .hw_free = pcm_hw_free,
838 .prepare = pcm_prepare,
839 .trigger = pcm_trigger,
840 .pointer = pcm_pointer,
841 .page = snd_pcm_sgbuf_ops_page,
842};
843#endif /* ECHOCARD_HAS_DIGITAL_IO */
844
845
846
847/* Preallocate memory only for the first substream because it's the most
848 * used one
849 */
850static int snd_echo_preallocate_pages(struct snd_pcm *pcm, struct device *dev)
851{
852 struct snd_pcm_substream *ss;
853 int stream, err;
854
855 for (stream = 0; stream < 2; stream++)
856 for (ss = pcm->streams[stream].substream; ss; ss = ss->next) {
857 err = snd_pcm_lib_preallocate_pages(ss, SNDRV_DMA_TYPE_DEV_SG,
858 dev,
859 ss->number ? 0 : 128<<10,
860 256<<10);
861 if (err < 0)
862 return err;
863 }
864 return 0;
865}
866
867
868
869/*<--snd_echo_probe() */
870static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
871{
872 struct snd_pcm *pcm;
873 int err;
874
875#ifdef ECHOCARD_HAS_VMIXER
876 /* This card has a Vmixer, that is there is no direct mapping from PCM
877 streams to physical outputs. The user can mix the streams as he wishes
878 via control interface and it's possible to send any stream to any
879 output, thus it makes no sense to keep analog and digital outputs
880 separated */
881
882 /* PCM#0 Virtual outputs and analog inputs */
883 if ((err = snd_pcm_new(chip->card, "PCM", 0, num_pipes_out(chip),
884 num_analog_busses_in(chip), &pcm)) < 0)
885 return err;
886 pcm->private_data = chip;
887 chip->analog_pcm = pcm;
888 strcpy(pcm->name, chip->card->shortname);
889 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops);
890 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
891 if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
892 return err;
893 DE_INIT(("Analog PCM ok\n"));
894
895#ifdef ECHOCARD_HAS_DIGITAL_IO
896 /* PCM#1 Digital inputs, no outputs */
897 if ((err = snd_pcm_new(chip->card, "Digital PCM", 1, 0,
898 num_digital_busses_in(chip), &pcm)) < 0)
899 return err;
900 pcm->private_data = chip;
901 chip->digital_pcm = pcm;
902 strcpy(pcm->name, chip->card->shortname);
903 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
904 if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
905 return err;
906 DE_INIT(("Digital PCM ok\n"));
907#endif /* ECHOCARD_HAS_DIGITAL_IO */
908
909#else /* ECHOCARD_HAS_VMIXER */
910
911 /* The card can manage substreams formed by analog and digital channels
912 at the same time, but I prefer to keep analog and digital channels
913 separated, because that mixed thing is confusing and useless. So we
914 register two PCM devices: */
915
916 /* PCM#0 Analog i/o */
917 if ((err = snd_pcm_new(chip->card, "Analog PCM", 0,
918 num_analog_busses_out(chip),
919 num_analog_busses_in(chip), &pcm)) < 0)
920 return err;
921 pcm->private_data = chip;
922 chip->analog_pcm = pcm;
923 strcpy(pcm->name, chip->card->shortname);
924 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &analog_playback_ops);
925 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &analog_capture_ops);
926 if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
927 return err;
928 DE_INIT(("Analog PCM ok\n"));
929
930#ifdef ECHOCARD_HAS_DIGITAL_IO
931 /* PCM#1 Digital i/o */
932 if ((err = snd_pcm_new(chip->card, "Digital PCM", 1,
933 num_digital_busses_out(chip),
934 num_digital_busses_in(chip), &pcm)) < 0)
935 return err;
936 pcm->private_data = chip;
937 chip->digital_pcm = pcm;
938 strcpy(pcm->name, chip->card->shortname);
939 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &digital_playback_ops);
940 snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &digital_capture_ops);
941 if ((err = snd_echo_preallocate_pages(pcm, snd_dma_pci_data(chip->pci))) < 0)
942 return err;
943 DE_INIT(("Digital PCM ok\n"));
944#endif /* ECHOCARD_HAS_DIGITAL_IO */
945
946#endif /* ECHOCARD_HAS_VMIXER */
947
948 return 0;
949}
950
951
952
953
954/******************************************************************************
955 Control interface
956******************************************************************************/
957
958/******************* PCM output volume *******************/
959static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
960 struct snd_ctl_elem_info *uinfo)
961{
962 struct echoaudio *chip;
963
964 chip = snd_kcontrol_chip(kcontrol);
965 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
966 uinfo->count = num_busses_out(chip);
967 uinfo->value.integer.min = ECHOGAIN_MINOUT;
968 uinfo->value.integer.max = ECHOGAIN_MAXOUT;
969 return 0;
970}
971
972static int snd_echo_output_gain_get(struct snd_kcontrol *kcontrol,
973 struct snd_ctl_elem_value *ucontrol)
974{
975 struct echoaudio *chip;
976 int c;
977
978 chip = snd_kcontrol_chip(kcontrol);
979 for (c = 0; c < num_busses_out(chip); c++)
980 ucontrol->value.integer.value[c] = chip->output_gain[c];
981 return 0;
982}
983
984static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
985 struct snd_ctl_elem_value *ucontrol)
986{
987 struct echoaudio *chip;
988 int c, changed, gain;
989
990 changed = 0;
991 chip = snd_kcontrol_chip(kcontrol);
992 spin_lock_irq(&chip->lock);
993 for (c = 0; c < num_busses_out(chip); c++) {
994 gain = ucontrol->value.integer.value[c];
995 /* Ignore out of range values */
996 if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
997 continue;
998 if (chip->output_gain[c] != gain) {
999 set_output_gain(chip, c, gain);
1000 changed = 1;
1001 }
1002 }
1003 if (changed)
1004 update_output_line_level(chip);
1005 spin_unlock_irq(&chip->lock);
1006 return changed;
1007}
1008
1009#ifdef ECHOCARD_HAS_VMIXER
1010/* On Vmixer cards this one controls the line-out volume */
1011static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
1012 .name = "Line Playback Volume",
1013 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
1014 .info = snd_echo_output_gain_info,
1015 .get = snd_echo_output_gain_get,
1016 .put = snd_echo_output_gain_put,
1017};
1018#else
1019static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
1020 .name = "PCM Playback Volume",
1021 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
1022 .info = snd_echo_output_gain_info,
1023 .get = snd_echo_output_gain_get,
1024 .put = snd_echo_output_gain_put,
1025};
1026#endif
1027
1028
1029
1030#ifdef ECHOCARD_HAS_INPUT_GAIN
1031
1032/******************* Analog input volume *******************/
1033static int snd_echo_input_gain_info(struct snd_kcontrol *kcontrol,
1034 struct snd_ctl_elem_info *uinfo)
1035{
1036 struct echoaudio *chip;
1037
1038 chip = snd_kcontrol_chip(kcontrol);
1039 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
1040 uinfo->count = num_analog_busses_in(chip);
1041 uinfo->value.integer.min = ECHOGAIN_MININP;
1042 uinfo->value.integer.max = ECHOGAIN_MAXINP;
1043 return 0;
1044}
1045
1046static int snd_echo_input_gain_get(struct snd_kcontrol *kcontrol,
1047 struct snd_ctl_elem_value *ucontrol)
1048{
1049 struct echoaudio *chip;
1050 int c;
1051
1052 chip = snd_kcontrol_chip(kcontrol);
1053 for (c = 0; c < num_analog_busses_in(chip); c++)
1054 ucontrol->value.integer.value[c] = chip->input_gain[c];
1055 return 0;
1056}
1057
1058static int snd_echo_input_gain_put(struct snd_kcontrol *kcontrol,
1059 struct snd_ctl_elem_value *ucontrol)
1060{
1061 struct echoaudio *chip;
1062 int c, gain, changed;
1063
1064 changed = 0;
1065 chip = snd_kcontrol_chip(kcontrol);
1066 spin_lock_irq(&chip->lock);
1067 for (c = 0; c < num_analog_busses_in(chip); c++) {
1068 gain = ucontrol->value.integer.value[c];
1069 /* Ignore out of range values */
1070 if (gain < ECHOGAIN_MININP || gain > ECHOGAIN_MAXINP)
1071 continue;
1072 if (chip->input_gain[c] != gain) {
1073 set_input_gain(chip, c, gain);
1074 changed = 1;
1075 }
1076 }
1077 if (changed)
1078 update_input_line_level(chip);
1079 spin_unlock_irq(&chip->lock);
1080 return changed;
1081}
1082
1083static struct snd_kcontrol_new snd_echo_line_input_gain __devinitdata = {
1084 .name = "Line Capture Volume",
1085 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
1086 .info = snd_echo_input_gain_info,
1087 .get = snd_echo_input_gain_get,
1088 .put = snd_echo_input_gain_put,
1089};
1090
1091#endif /* ECHOCARD_HAS_INPUT_GAIN */
1092
1093
1094
1095#ifdef ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
1096
1097/************ Analog output nominal level (+4dBu / -10dBV) ***************/
1098static int snd_echo_output_nominal_info (struct snd_kcontrol *kcontrol,
1099 struct snd_ctl_elem_info *uinfo)
1100{
1101 struct echoaudio *chip;
1102
1103 chip = snd_kcontrol_chip(kcontrol);
1104 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
1105 uinfo->count = num_analog_busses_out(chip);
1106 uinfo->value.integer.min = 0;
1107 uinfo->value.integer.max = 1;
1108 return 0;
1109}
1110
1111static int snd_echo_output_nominal_get(struct snd_kcontrol *kcontrol,
1112 struct snd_ctl_elem_value *ucontrol)
1113{
1114 struct echoaudio *chip;
1115 int c;
1116
1117 chip = snd_kcontrol_chip(kcontrol);
1118 for (c = 0; c < num_analog_busses_out(chip); c++)
1119 ucontrol->value.integer.value[c] = chip->nominal_level[c];
1120 return 0;
1121}
1122
1123static int snd_echo_output_nominal_put(struct snd_kcontrol *kcontrol,
1124 struct snd_ctl_elem_value *ucontrol)
1125{
1126 struct echoaudio *chip;
1127 int c, changed;
1128
1129 changed = 0;
1130 chip = snd_kcontrol_chip(kcontrol);
1131 spin_lock_irq(&chip->lock);
1132 for (c = 0; c < num_analog_busses_out(chip); c++) {
1133 if (chip->nominal_level[c] != ucontrol->value.integer.value[c]) {
1134 set_nominal_level(chip, c,
1135 ucontrol->value.integer.value[c]);
1136 changed = 1;
1137 }
1138 }
1139 if (changed)
1140 update_output_line_level(chip);
1141 spin_unlock_irq(&chip->lock);
1142 return changed;
1143}
1144
1145static struct snd_kcontrol_new snd_echo_output_nominal_level __devinitdata = {
1146 .name = "Line Playback Switch (-10dBV)",
1147 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
1148 .info = snd_echo_output_nominal_info,
1149 .get = snd_echo_output_nominal_get,
1150 .put = snd_echo_output_nominal_put,
1151};
1152
1153#endif /* ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL */
1154
1155
1156
1157#ifdef ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
1158
1159/*************** Analog input nominal level (+4dBu / -10dBV) ***************/
1160static int snd_echo_input_nominal_info(struct snd_kcontrol *kcontrol,
1161 struct snd_ctl_elem_info *uinfo)
1162{
1163 struct echoaudio *chip;
1164
1165 chip = snd_kcontrol_chip(kcontrol);
1166 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
1167 uinfo->count = num_analog_busses_in(chip);
1168 uinfo->value.integer.min = 0;
1169 uinfo->value.integer.max = 1;
1170 return 0;
1171}
1172
1173static int snd_echo_input_nominal_get(struct snd_kcontrol *kcontrol,
1174 struct snd_ctl_elem_value *ucontrol)
1175{
1176 struct echoaudio *chip;
1177 int c;
1178
1179 chip = snd_kcontrol_chip(kcontrol);
1180 for (c = 0; c < num_analog_busses_in(chip); c++)
1181 ucontrol->value.integer.value[c] =
1182 chip->nominal_level[bx_analog_in(chip) + c];
1183 return 0;
1184}
1185
1186static int snd_echo_input_nominal_put(struct snd_kcontrol *kcontrol,
1187 struct snd_ctl_elem_value *ucontrol)
1188{
1189 struct echoaudio *chip;
1190 int c, changed;
1191
1192 changed = 0;
1193 chip = snd_kcontrol_chip(kcontrol);
1194 spin_lock_irq(&chip->lock);
1195 for (c = 0; c < num_analog_busses_in(chip); c++) {
1196 if (chip->nominal_level[bx_analog_in(chip) + c] !=
1197 ucontrol->value.integer.value[c]) {
1198 set_nominal_level(chip, bx_analog_in(chip) + c,
1199 ucontrol->value.integer.value[c]);
1200 changed = 1;
1201 }
1202 }
1203 if (changed)
1204 update_output_line_level(chip); /* "Output" is not a mistake
1205 * here.
1206 */
1207 spin_unlock_irq(&chip->lock);
1208 return changed;
1209}
1210
1211static struct snd_kcontrol_new snd_echo_intput_nominal_level __devinitdata = {
1212 .name = "Line Capture Switch (-10dBV)",
1213 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
1214 .info = snd_echo_input_nominal_info,
1215 .get = snd_echo_input_nominal_get,
1216 .put = snd_echo_input_nominal_put,
1217};
1218
1219#endif /* ECHOCARD_HAS_INPUT_NOMINAL_LEVEL */
1220
1221
1222
1223#ifdef ECHOCARD_HAS_MONITOR
1224
1225/******************* Monitor mixer *******************/
1226static int snd_echo_mixer_info(struct snd_kcontrol *kcontrol,
1227 struct snd_ctl_elem_info *uinfo)
1228{
1229 struct echoaudio *chip;
1230
1231 chip = snd_kcontrol_chip(kcontrol);
1232 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
1233 uinfo->count = 1;
1234 uinfo->value.integer.min = ECHOGAIN_MINOUT;
1235 uinfo->value.integer.max = ECHOGAIN_MAXOUT;
1236 uinfo->dimen.d[0] = num_busses_out(chip);
1237 uinfo->dimen.d[1] = num_busses_in(chip);
1238 return 0;
1239}
1240
1241static int snd_echo_mixer_get(struct snd_kcontrol *kcontrol,
1242 struct snd_ctl_elem_value *ucontrol)
1243{
1244 struct echoaudio *chip;
1245
1246 chip = snd_kcontrol_chip(kcontrol);
1247 ucontrol->value.integer.value[0] =
1248 chip->monitor_gain[ucontrol->id.index / num_busses_in(chip)]
1249 [ucontrol->id.index % num_busses_in(chip)];
1250 return 0;
1251}
1252
1253static int snd_echo_mixer_put(struct snd_kcontrol *kcontrol,
1254 struct snd_ctl_elem_value *ucontrol)
1255{
1256 struct echoaudio *chip;
1257 int changed, gain;
1258 short out, in;
1259
1260 changed = 0;
1261 chip = snd_kcontrol_chip(kcontrol);
1262 out = ucontrol->id.index / num_busses_in(chip);
1263 in = ucontrol->id.index % num_busses_in(chip);
1264 gain = ucontrol->value.integer.value[0];
1265 if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
1266 return -EINVAL;
1267 if (chip->monitor_gain[out][in] != gain) {
1268 spin_lock_irq(&chip->lock);
1269 set_monitor_gain(chip, out, in, gain);
1270 update_output_line_level(chip);
1271 spin_unlock_irq(&chip->lock);
1272 changed = 1;
1273 }
1274 return changed;
1275}
1276
1277static struct snd_kcontrol_new snd_echo_monitor_mixer __devinitdata = {
1278 .name = "Monitor Mixer Volume",
1279 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
1280 .info = snd_echo_mixer_info,
1281 .get = snd_echo_mixer_get,
1282 .put = snd_echo_mixer_put,
1283};
1284
1285#endif /* ECHOCARD_HAS_MONITOR */
1286
1287
1288
1289#ifdef ECHOCARD_HAS_VMIXER
1290
1291/******************* Vmixer *******************/
1292static int snd_echo_vmixer_info(struct snd_kcontrol *kcontrol,
1293 struct snd_ctl_elem_info *uinfo)
1294{
1295 struct echoaudio *chip;
1296
1297 chip = snd_kcontrol_chip(kcontrol);
1298 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
1299 uinfo->count = 1;
1300 uinfo->value.integer.min = ECHOGAIN_MINOUT;
1301 uinfo->value.integer.max = ECHOGAIN_MAXOUT;
1302 uinfo->dimen.d[0] = num_busses_out(chip);
1303 uinfo->dimen.d[1] = num_pipes_out(chip);
1304 return 0;
1305}
1306
1307static int snd_echo_vmixer_get(struct snd_kcontrol *kcontrol,
1308 struct snd_ctl_elem_value *ucontrol)
1309{
1310 struct echoaudio *chip;
1311
1312 chip = snd_kcontrol_chip(kcontrol);
1313 ucontrol->value.integer.value[0] =
1314 chip->vmixer_gain[ucontrol->id.index / num_pipes_out(chip)]
1315 [ucontrol->id.index % num_pipes_out(chip)];
1316 return 0;
1317}
1318
1319static int snd_echo_vmixer_put(struct snd_kcontrol *kcontrol,
1320 struct snd_ctl_elem_value *ucontrol)
1321{
1322 struct echoaudio *chip;
1323 int gain, changed;
1324 short vch, out;
1325
1326 changed = 0;
1327 chip = snd_kcontrol_chip(kcontrol);
1328 out = ucontrol->id.index / num_pipes_out(chip);
1329 vch = ucontrol->id.index % num_pipes_out(chip);
1330 gain = ucontrol->value.integer.value[0];
1331 if (gain < ECHOGAIN_MINOUT || gain > ECHOGAIN_MAXOUT)
1332 return -EINVAL;
1333 if (chip->vmixer_gain[out][vch] != ucontrol->value.integer.value[0]) {
1334 spin_lock_irq(&chip->lock);
1335 set_vmixer_gain(chip, out, vch, ucontrol->value.integer.value[0]);
1336 update_vmixer_level(chip);
1337 spin_unlock_irq(&chip->lock);
1338 changed = 1;
1339 }
1340 return changed;
1341}
1342
1343static struct snd_kcontrol_new snd_echo_vmixer __devinitdata = {
1344 .name = "VMixer Volume",
1345 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
1346 .info = snd_echo_vmixer_info,
1347 .get = snd_echo_vmixer_get,
1348 .put = snd_echo_vmixer_put,
1349};
1350
1351#endif /* ECHOCARD_HAS_VMIXER */
1352
1353
1354
1355#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH
1356
1357/******************* Digital mode switch *******************/
1358static int snd_echo_digital_mode_info(struct snd_kcontrol *kcontrol,
1359 struct snd_ctl_elem_info *uinfo)
1360{
1361 static char *names[4] = {
1362 "S/PDIF Coaxial", "S/PDIF Optical", "ADAT Optical",
1363 "S/PDIF Cdrom"
1364 };
1365 struct echoaudio *chip;
1366
1367 chip = snd_kcontrol_chip(kcontrol);
1368 uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
1369 uinfo->value.enumerated.items = chip->num_digital_modes;
1370 uinfo->count = 1;
1371 if (uinfo->value.enumerated.item >= chip->num_digital_modes)
1372 uinfo->value.enumerated.item = chip->num_digital_modes - 1;
1373 strcpy(uinfo->value.enumerated.name, names[
1374 chip->digital_mode_list[uinfo->value.enumerated.item]]);
1375 return 0;
1376}
1377
1378static int snd_echo_digital_mode_get(struct snd_kcontrol *kcontrol,
1379 struct snd_ctl_elem_value *ucontrol)
1380{
1381 struct echoaudio *chip;
1382 int i, mode;
1383
1384 chip = snd_kcontrol_chip(kcontrol);
1385 mode = chip->digital_mode;
1386 for (i = chip->num_digital_modes - 1; i >= 0; i--)
1387 if (mode == chip->digital_mode_list[i]) {
1388 ucontrol->value.enumerated.item[0] = i;
1389 break;
1390 }
1391 return 0;
1392}
1393
1394static int snd_echo_digital_mode_put(struct snd_kcontrol *kcontrol,
1395 struct snd_ctl_elem_value *ucontrol)
1396{
1397 struct echoaudio *chip;
1398 int changed;
1399 unsigned short emode, dmode;
1400
1401 changed = 0;
1402 chip = snd_kcontrol_chip(kcontrol);
1403
1404 emode = ucontrol->value.enumerated.item[0];
1405 if (emode >= chip->num_digital_modes)
1406 return -EINVAL;
1407 dmode = chip->digital_mode_list[emode];
1408
1409 if (dmode != chip->digital_mode) {
1410 /* mode_mutex is required to make this operation atomic wrt
1411 pcm_digital_*_open() and set_input_clock() functions. */
1412 down(&chip->mode_mutex);
1413
1414 /* Do not allow the user to change the digital mode when a pcm
1415 device is open because it also changes the number of channels
1416 and the allowed sample rates */
1417 if (atomic_read(&chip->opencount)) {
1418 changed = -EAGAIN;
1419 } else {
1420 changed = set_digital_mode(chip, dmode);
1421 /* If we had to change the clock source, report it */
1422 if (changed > 0 && chip->clock_src_ctl) {
1423 snd_ctl_notify(chip->card,
1424 SNDRV_CTL_EVENT_MASK_VALUE,
1425 &chip->clock_src_ctl->id);
1426 DE_ACT(("SDM() =%d\n", changed));
1427 }
1428 if (changed >= 0)
1429 changed = 1; /* No errors */
1430 }
1431 up(&chip->mode_mutex);
1432 }
1433 return changed;
1434}
1435
1436static struct snd_kcontrol_new snd_echo_digital_mode_switch __devinitdata = {
1437 .name = "Digital mode Switch",
1438 .iface = SNDRV_CTL_ELEM_IFACE_CARD,
1439 .info = snd_echo_digital_mode_info,
1440 .get = snd_echo_digital_mode_get,
1441 .put = snd_echo_digital_mode_put,
1442};
1443
1444#endif /* ECHOCARD_HAS_DIGITAL_MODE_SWITCH */
1445
1446
1447
1448#ifdef ECHOCARD_HAS_DIGITAL_IO
1449
1450/******************* S/PDIF mode switch *******************/
1451static int snd_echo_spdif_mode_info(struct snd_kcontrol *kcontrol,
1452 struct snd_ctl_elem_info *uinfo)
1453{
1454 static char *names[2] = {"Consumer", "Professional"};
1455
1456 uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
1457 uinfo->value.enumerated.items = 2;
1458 uinfo->count = 1;
1459 if (uinfo->value.enumerated.item)
1460 uinfo->value.enumerated.item = 1;
1461 strcpy(uinfo->value.enumerated.name,
1462 names[uinfo->value.enumerated.item]);
1463 return 0;
1464}
1465
1466static int snd_echo_spdif_mode_get(struct snd_kcontrol *kcontrol,
1467 struct snd_ctl_elem_value *ucontrol)
1468{
1469 struct echoaudio *chip;
1470
1471 chip = snd_kcontrol_chip(kcontrol);
1472 ucontrol->value.enumerated.item[0] = !!chip->professional_spdif;
1473 return 0;
1474}
1475
1476static int snd_echo_spdif_mode_put(struct snd_kcontrol *kcontrol,
1477 struct snd_ctl_elem_value *ucontrol)
1478{
1479 struct echoaudio *chip;
1480 int mode;
1481
1482 chip = snd_kcontrol_chip(kcontrol);
1483 mode = !!ucontrol->value.enumerated.item[0];
1484 if (mode != chip->professional_spdif) {
1485 spin_lock_irq(&chip->lock);
1486 set_professional_spdif(chip, mode);
1487 spin_unlock_irq(&chip->lock);
1488 return 1;
1489 }
1490 return 0;
1491}
1492
1493static struct snd_kcontrol_new snd_echo_spdif_mode_switch __devinitdata = {
1494 .name = "S/PDIF mode Switch",
1495 .iface = SNDRV_CTL_ELEM_IFACE_CARD,
1496 .info = snd_echo_spdif_mode_info,
1497 .get = snd_echo_spdif_mode_get,
1498 .put = snd_echo_spdif_mode_put,
1499};
1500
1501#endif /* ECHOCARD_HAS_DIGITAL_IO */
1502
1503
1504
1505#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
1506
1507/******************* Select input clock source *******************/
1508static int snd_echo_clock_source_info(struct snd_kcontrol *kcontrol,
1509 struct snd_ctl_elem_info *uinfo)
1510{
1511 static char *names[8] = {
1512 "Internal", "Word", "Super", "S/PDIF", "ADAT", "ESync",
1513 "ESync96", "MTC"
1514 };
1515 struct echoaudio *chip;
1516
1517 chip = snd_kcontrol_chip(kcontrol);
1518 uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
1519 uinfo->value.enumerated.items = chip->num_clock_sources;
1520 uinfo->count = 1;
1521 if (uinfo->value.enumerated.item >= chip->num_clock_sources)
1522 uinfo->value.enumerated.item = chip->num_clock_sources - 1;
1523 strcpy(uinfo->value.enumerated.name, names[
1524 chip->clock_source_list[uinfo->value.enumerated.item]]);
1525 return 0;
1526}
1527
1528static int snd_echo_clock_source_get(struct snd_kcontrol *kcontrol,
1529 struct snd_ctl_elem_value *ucontrol)
1530{
1531 struct echoaudio *chip;
1532 int i, clock;
1533
1534 chip = snd_kcontrol_chip(kcontrol);
1535 clock = chip->input_clock;
1536
1537 for (i = 0; i < chip->num_clock_sources; i++)
1538 if (clock == chip->clock_source_list[i])
1539 ucontrol->value.enumerated.item[0] = i;
1540
1541 return 0;
1542}
1543
1544static int snd_echo_clock_source_put(struct snd_kcontrol *kcontrol,
1545 struct snd_ctl_elem_value *ucontrol)
1546{
1547 struct echoaudio *chip;
1548 int changed;
1549 unsigned int eclock, dclock;
1550
1551 changed = 0;
1552 chip = snd_kcontrol_chip(kcontrol);
1553 eclock = ucontrol->value.enumerated.item[0];
1554 if (eclock >= chip->input_clock_types)
1555 return -EINVAL;
1556 dclock = chip->clock_source_list[eclock];
1557 if (chip->input_clock != dclock) {
1558 down(&chip->mode_mutex);
1559 spin_lock_irq(&chip->lock);
1560 if ((changed = set_input_clock(chip, dclock)) == 0)
1561 changed = 1; /* no errors */
1562 spin_unlock_irq(&chip->lock);
1563 up(&chip->mode_mutex);
1564 }
1565
1566 if (changed < 0)
1567 DE_ACT(("seticlk val%d err 0x%x\n", dclock, changed));
1568
1569 return changed;
1570}
1571
1572static struct snd_kcontrol_new snd_echo_clock_source_switch __devinitdata = {
1573 .name = "Sample Clock Source",
1574 .iface = SNDRV_CTL_ELEM_IFACE_PCM,
1575 .info = snd_echo_clock_source_info,
1576 .get = snd_echo_clock_source_get,
1577 .put = snd_echo_clock_source_put,
1578};
1579
1580#endif /* ECHOCARD_HAS_EXTERNAL_CLOCK */
1581
1582
1583
1584#ifdef ECHOCARD_HAS_PHANTOM_POWER
1585
1586/******************* Phantom power switch *******************/
1587static int snd_echo_phantom_power_info(struct snd_kcontrol *kcontrol,
1588 struct snd_ctl_elem_info *uinfo)
1589{
1590 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
1591 uinfo->count = 1;
1592 uinfo->value.integer.min = 0;
1593 uinfo->value.integer.max = 1;
1594 return 0;
1595}
1596
1597static int snd_echo_phantom_power_get(struct snd_kcontrol *kcontrol,
1598 struct snd_ctl_elem_value *ucontrol)
1599{
1600 struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
1601
1602 ucontrol->value.integer.value[0] = chip->phantom_power;
1603 return 0;
1604}
1605
1606static int snd_echo_phantom_power_put(struct snd_kcontrol *kcontrol,
1607 struct snd_ctl_elem_value *ucontrol)
1608{
1609 struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
1610 int power, changed = 0;
1611
1612 power = !!ucontrol->value.integer.value[0];
1613 if (chip->phantom_power != power) {
1614 spin_lock_irq(&chip->lock);
1615 changed = set_phantom_power(chip, power);
1616 spin_unlock_irq(&chip->lock);
1617 if (changed == 0)
1618 changed = 1; /* no errors */
1619 }
1620 return changed;
1621}
1622
1623static struct snd_kcontrol_new snd_echo_phantom_power_switch __devinitdata = {
1624 .name = "Phantom power Switch",
1625 .iface = SNDRV_CTL_ELEM_IFACE_CARD,
1626 .info = snd_echo_phantom_power_info,
1627 .get = snd_echo_phantom_power_get,
1628 .put = snd_echo_phantom_power_put,
1629};
1630
1631#endif /* ECHOCARD_HAS_PHANTOM_POWER */
1632
1633
1634
1635#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
1636
1637/******************* Digital input automute switch *******************/
1638static int snd_echo_automute_info(struct snd_kcontrol *kcontrol,
1639 struct snd_ctl_elem_info *uinfo)
1640{
1641 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
1642 uinfo->count = 1;
1643 uinfo->value.integer.min = 0;
1644 uinfo->value.integer.max = 1;
1645 return 0;
1646}
1647
1648static int snd_echo_automute_get(struct snd_kcontrol *kcontrol,
1649 struct snd_ctl_elem_value *ucontrol)
1650{
1651 struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
1652
1653 ucontrol->value.integer.value[0] = chip->digital_in_automute;
1654 return 0;
1655}
1656
1657static int snd_echo_automute_put(struct snd_kcontrol *kcontrol,
1658 struct snd_ctl_elem_value *ucontrol)
1659{
1660 struct echoaudio *chip = snd_kcontrol_chip(kcontrol);
1661 int automute, changed = 0;
1662
1663 automute = !!ucontrol->value.integer.value[0];
1664 if (chip->digital_in_automute != automute) {
1665 spin_lock_irq(&chip->lock);
1666 changed = set_input_auto_mute(chip, automute);
1667 spin_unlock_irq(&chip->lock);
1668 if (changed == 0)
1669 changed = 1; /* no errors */
1670 }
1671 return changed;
1672}
1673
1674static struct snd_kcontrol_new snd_echo_automute_switch __devinitdata = {
1675 .name = "Digital Capture Switch (automute)",
1676 .iface = SNDRV_CTL_ELEM_IFACE_CARD,
1677 .info = snd_echo_automute_info,
1678 .get = snd_echo_automute_get,
1679 .put = snd_echo_automute_put,
1680};
1681
1682#endif /* ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE */
1683
1684
1685
1686/******************* VU-meters switch *******************/
1687static int snd_echo_vumeters_switch_info(struct snd_kcontrol *kcontrol,
1688 struct snd_ctl_elem_info *uinfo)
1689{
1690 struct echoaudio *chip;
1691
1692 chip = snd_kcontrol_chip(kcontrol);
1693 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
1694 uinfo->count = 1;
1695 uinfo->value.integer.min = 0;
1696 uinfo->value.integer.max = 1;
1697 return 0;
1698}
1699
1700static int snd_echo_vumeters_switch_put(struct snd_kcontrol *kcontrol,
1701 struct snd_ctl_elem_value *ucontrol)
1702{
1703 struct echoaudio *chip;
1704
1705 chip = snd_kcontrol_chip(kcontrol);
1706 spin_lock_irq(&chip->lock);
1707 set_meters_on(chip, ucontrol->value.integer.value[0]);
1708 spin_unlock_irq(&chip->lock);
1709 return 1;
1710}
1711
1712static struct snd_kcontrol_new snd_echo_vumeters_switch __devinitdata = {
1713 .name = "VU-meters Switch",
1714 .iface = SNDRV_CTL_ELEM_IFACE_CARD,
1715 .access = SNDRV_CTL_ELEM_ACCESS_WRITE,
1716 .info = snd_echo_vumeters_switch_info,
1717 .put = snd_echo_vumeters_switch_put,
1718};
1719
1720
1721
1722/***** Read VU-meters (input, output, analog and digital together) *****/
1723static int snd_echo_vumeters_info(struct snd_kcontrol *kcontrol,
1724 struct snd_ctl_elem_info *uinfo)
1725{
1726 struct echoaudio *chip;
1727
1728 chip = snd_kcontrol_chip(kcontrol);
1729 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
1730 uinfo->count = 96;
1731 uinfo->value.integer.min = ECHOGAIN_MINOUT;
1732 uinfo->value.integer.max = 0;
1733#ifdef ECHOCARD_HAS_VMIXER
1734 uinfo->dimen.d[0] = 3; /* Out, In, Virt */
1735#else
1736 uinfo->dimen.d[0] = 2; /* Out, In */
1737#endif
1738 uinfo->dimen.d[1] = 16; /* 16 channels */
1739 uinfo->dimen.d[2] = 2; /* 0=level, 1=peak */
1740 return 0;
1741}
1742
1743static int snd_echo_vumeters_get(struct snd_kcontrol *kcontrol,
1744 struct snd_ctl_elem_value *ucontrol)
1745{
1746 struct echoaudio *chip;
1747
1748 chip = snd_kcontrol_chip(kcontrol);
1749 get_audio_meters(chip, ucontrol->value.integer.value);
1750 return 0;
1751}
1752
1753static struct snd_kcontrol_new snd_echo_vumeters __devinitdata = {
1754 .name = "VU-meters",
1755 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
1756 .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
1757 .info = snd_echo_vumeters_info,
1758 .get = snd_echo_vumeters_get,
1759};
1760
1761
1762
1763/*** Channels info - it exports informations about the number of channels ***/
1764static int snd_echo_channels_info_info(struct snd_kcontrol *kcontrol,
1765 struct snd_ctl_elem_info *uinfo)
1766{
1767 struct echoaudio *chip;
1768
1769 chip = snd_kcontrol_chip(kcontrol);
1770 uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
1771 uinfo->count = 6;
1772 uinfo->value.integer.min = 0;
1773 uinfo->value.integer.max = 1 << ECHO_CLOCK_NUMBER;
1774 return 0;
1775}
1776
1777static int snd_echo_channels_info_get(struct snd_kcontrol *kcontrol,
1778 struct snd_ctl_elem_value *ucontrol)
1779{
1780 struct echoaudio *chip;
1781 int detected, clocks, bit, src;
1782
1783 chip = snd_kcontrol_chip(kcontrol);
1784 ucontrol->value.integer.value[0] = num_busses_in(chip);
1785 ucontrol->value.integer.value[1] = num_analog_busses_in(chip);
1786 ucontrol->value.integer.value[2] = num_busses_out(chip);
1787 ucontrol->value.integer.value[3] = num_analog_busses_out(chip);
1788 ucontrol->value.integer.value[4] = num_pipes_out(chip);
1789
1790 /* Compute the bitmask of the currently valid input clocks */
1791 detected = detect_input_clocks(chip);
1792 clocks = 0;
1793 src = chip->num_clock_sources - 1;
1794 for (bit = ECHO_CLOCK_NUMBER - 1; bit >= 0; bit--)
1795 if (detected & (1 << bit))
1796 for (; src >= 0; src--)
1797 if (bit == chip->clock_source_list[src]) {
1798 clocks |= 1 << src;
1799 break;
1800 }
1801 ucontrol->value.integer.value[5] = clocks;
1802
1803 return 0;
1804}
1805
1806static struct snd_kcontrol_new snd_echo_channels_info __devinitdata = {
1807 .name = "Channels info",
1808 .iface = SNDRV_CTL_ELEM_IFACE_HWDEP,
1809 .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE,
1810 .info = snd_echo_channels_info_info,
1811 .get = snd_echo_channels_info_get,
1812};
1813
1814
1815
1816
1817/******************************************************************************
1818 IRQ Handler
1819******************************************************************************/
1820
1821static irqreturn_t snd_echo_interrupt(int irq, void *dev_id,
1822 struct pt_regs *regs)
1823{
1824 struct echoaudio *chip = dev_id;
1825 struct snd_pcm_substream *substream;
1826 int period, ss, st;
1827
1828 spin_lock(&chip->lock);
1829 st = service_irq(chip);
1830 if (st < 0) {
1831 spin_unlock(&chip->lock);
1832 return IRQ_NONE;
1833 }
1834 /* The hardware doesn't tell us which substream caused the irq,
1835 thus we have to check all running substreams. */
1836 for (ss = 0; ss < DSP_MAXPIPES; ss++) {
1837 if ((substream = chip->substream[ss])) {
1838 period = pcm_pointer(substream) /
1839 substream->runtime->period_size;
1840 if (period != chip->last_period[ss]) {
1841 chip->last_period[ss] = period;
1842 spin_unlock(&chip->lock);
1843 snd_pcm_period_elapsed(substream);
1844 spin_lock(&chip->lock);
1845 }
1846 }
1847 }
1848 spin_unlock(&chip->lock);
1849
1850#ifdef ECHOCARD_HAS_MIDI
1851 if (st > 0 && chip->midi_in) {
1852 snd_rawmidi_receive(chip->midi_in, chip->midi_buffer, st);
1853 DE_MID(("rawmidi_iread=%d\n", st));
1854 }
1855#endif
1856 return IRQ_HANDLED;
1857}
1858
1859
1860
1861
1862/******************************************************************************
1863 Module construction / destruction
1864******************************************************************************/
1865
1866static int snd_echo_free(struct echoaudio *chip)
1867{
1868 DE_INIT(("Stop DSP...\n"));
1869 if (chip->comm_page) {
1870 rest_in_peace(chip);
1871 snd_dma_free_pages(&chip->commpage_dma_buf);
1872 }
1873 DE_INIT(("Stopped.\n"));
1874
1875 if (chip->irq >= 0)
1876 free_irq(chip->irq, (void *)chip);
1877
1878 if (chip->dsp_registers)
1879 iounmap(chip->dsp_registers);
1880
1881 if (chip->iores)
1882 release_and_free_resource(chip->iores);
1883
1884 DE_INIT(("MMIO freed.\n"));
1885
1886 pci_disable_device(chip->pci);
1887
1888 /* release chip data */
1889 kfree(chip);
1890 DE_INIT(("Chip freed.\n"));
1891 return 0;
1892}
1893
1894
1895
1896static int snd_echo_dev_free(struct snd_device *device)
1897{
1898 struct echoaudio *chip = device->device_data;
1899
1900 DE_INIT(("snd_echo_dev_free()...\n"));
1901 return snd_echo_free(chip);
1902}
1903
1904
1905
1906/* <--snd_echo_probe() */
1907static __devinit int snd_echo_create(struct snd_card *card,
1908 struct pci_dev *pci,
1909 struct echoaudio **rchip)
1910{
1911 struct echoaudio *chip;
1912 int err;
1913 size_t sz;
1914 static struct snd_device_ops ops = {
1915 .dev_free = snd_echo_dev_free,
1916 };
1917
1918 *rchip = NULL;
1919
1920 pci_write_config_byte(pci, PCI_LATENCY_TIMER, 0xC0);
1921
1922 if ((err = pci_enable_device(pci)) < 0)
1923 return err;
1924 pci_set_master(pci);
1925
1926 /* allocate a chip-specific data */
1927 chip = kzalloc(sizeof(*chip), GFP_KERNEL);
1928 if (!chip) {
1929 pci_disable_device(pci);
1930 return -ENOMEM;
1931 }
1932 DE_INIT(("chip=%p\n", chip));
1933
1934 spin_lock_init(&chip->lock);
1935 chip->card = card;
1936 chip->pci = pci;
1937 chip->irq = -1;
1938
1939 /* PCI resource allocation */
1940 chip->dsp_registers_phys = pci_resource_start(pci, 0);
1941 sz = pci_resource_len(pci, 0);
1942 if (sz > PAGE_SIZE)
1943 sz = PAGE_SIZE; /* We map only the required part */
1944
1945 if ((chip->iores = request_mem_region(chip->dsp_registers_phys, sz,
1946 ECHOCARD_NAME)) == NULL) {
1947 snd_echo_free(chip);
1948 snd_printk(KERN_ERR "cannot get memory region\n");
1949 return -EBUSY;
1950 }
1951 chip->dsp_registers = (volatile u32 __iomem *)
1952 ioremap_nocache(chip->dsp_registers_phys, sz);
1953
1954 if (request_irq(pci->irq, snd_echo_interrupt, SA_INTERRUPT | SA_SHIRQ,
1955 ECHOCARD_NAME, (void *)chip)) {
1956 snd_echo_free(chip);
1957 snd_printk(KERN_ERR "cannot grab irq\n");
1958 return -EBUSY;
1959 }
1960 chip->irq = pci->irq;
1961 DE_INIT(("pci=%p irq=%d subdev=%04x Init hardware...\n",
1962 chip->pci, chip->irq, chip->pci->subsystem_device));
1963
1964 /* Create the DSP comm page - this is the area of memory used for most
1965 of the communication with the DSP, which accesses it via bus mastering */
1966 if (snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
1967 sizeof(struct comm_page),
1968 &chip->commpage_dma_buf) < 0) {
1969 snd_echo_free(chip);
1970 snd_printk(KERN_ERR "cannot allocate the comm page\n");
1971 return -ENOMEM;
1972 }
1973 chip->comm_page_phys = chip->commpage_dma_buf.addr;
1974 chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area;
1975
1976 err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device);
1977 if (err) {
1978 DE_INIT(("init_hw err=%d\n", err));
1979 snd_echo_free(chip);
1980 return err;
1981 }
1982 DE_INIT(("Card init OK\n"));
1983
1984 if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) {
1985 snd_echo_free(chip);
1986 return err;
1987 }
1988 atomic_set(&chip->opencount, 0);
1989 init_MUTEX(&chip->mode_mutex);
1990 chip->can_set_rate = 1;
1991 *rchip = chip;
1992 /* Init done ! */
1993 return 0;
1994}
1995
1996
1997
1998/* constructor */
1999static int __devinit snd_echo_probe(struct pci_dev *pci,
2000 const struct pci_device_id *pci_id)
2001{
2002 static int dev;
2003 struct snd_card *card;
2004 struct echoaudio *chip;
2005 char *dsp;
2006 int i, err;
2007
2008 if (dev >= SNDRV_CARDS)
2009 return -ENODEV;
2010 if (!enable[dev]) {
2011 dev++;
2012 return -ENOENT;
2013 }
2014
2015 DE_INIT(("Echoaudio driver starting...\n"));
2016 i = 0;
2017 card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
2018 if (card == NULL)
2019 return -ENOMEM;
2020
2021 if ((err = snd_echo_create(card, pci, &chip)) < 0) {
2022 snd_card_free(card);
2023 return err;
2024 }
2025
2026 strcpy(card->driver, "Echo_" ECHOCARD_NAME);
2027 strcpy(card->shortname, chip->card_name);
2028
2029 dsp = "56301";
2030 if (pci_id->device == 0x3410)
2031 dsp = "56361";
2032
2033 sprintf(card->longname, "%s rev.%d (DSP%s) at 0x%lx irq %i",
2034 card->shortname, pci_id->subdevice & 0x000f, dsp,
2035 chip->dsp_registers_phys, chip->irq);
2036
2037 if ((err = snd_echo_new_pcm(chip)) < 0) {
2038 snd_printk(KERN_ERR "new pcm error %d\n", err);
2039 snd_card_free(card);
2040 return err;
2041 }
2042
2043#ifdef ECHOCARD_HAS_MIDI
2044 if (chip->has_midi) { /* Some Mia's do not have midi */
2045 if ((err = snd_echo_midi_create(card, chip)) < 0) {
2046 snd_printk(KERN_ERR "new midi error %d\n", err);
2047 snd_card_free(card);
2048 return err;
2049 }
2050 }
2051#endif
2052
2053#ifdef ECHOCARD_HAS_VMIXER
2054 snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
2055 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0)
2056 goto ctl_error;
2057 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
2058 goto ctl_error;
2059#else
2060 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_pcm_output_gain, chip))) < 0)
2061 goto ctl_error;
2062#endif
2063
2064#ifdef ECHOCARD_HAS_INPUT_GAIN
2065 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_input_gain, chip))) < 0)
2066 goto ctl_error;
2067#endif
2068
2069#ifdef ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
2070 if (!chip->hasnt_input_nominal_level)
2071 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_intput_nominal_level, chip))) < 0)
2072 goto ctl_error;
2073#endif
2074
2075#ifdef ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
2076 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_output_nominal_level, chip))) < 0)
2077 goto ctl_error;
2078#endif
2079
2080 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vumeters_switch, chip))) < 0)
2081 goto ctl_error;
2082
2083 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vumeters, chip))) < 0)
2084 goto ctl_error;
2085
2086#ifdef ECHOCARD_HAS_MONITOR
2087 snd_echo_monitor_mixer.count = num_busses_in(chip) * num_busses_out(chip);
2088 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_monitor_mixer, chip))) < 0)
2089 goto ctl_error;
2090#endif
2091
2092#ifdef ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
2093 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_automute_switch, chip))) < 0)
2094 goto ctl_error;
2095#endif
2096
2097 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_channels_info, chip))) < 0)
2098 goto ctl_error;
2099
2100#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH
2101 /* Creates a list of available digital modes */
2102 chip->num_digital_modes = 0;
2103 for (i = 0; i < 6; i++)
2104 if (chip->digital_modes & (1 << i))
2105 chip->digital_mode_list[chip->num_digital_modes++] = i;
2106
2107 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_digital_mode_switch, chip))) < 0)
2108 goto ctl_error;
2109#endif /* ECHOCARD_HAS_DIGITAL_MODE_SWITCH */
2110
2111#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
2112 /* Creates a list of available clock sources */
2113 chip->num_clock_sources = 0;
2114 for (i = 0; i < 10; i++)
2115 if (chip->input_clock_types & (1 << i))
2116 chip->clock_source_list[chip->num_clock_sources++] = i;
2117
2118 if (chip->num_clock_sources > 1) {
2119 chip->clock_src_ctl = snd_ctl_new1(&snd_echo_clock_source_switch, chip);
2120 if ((err = snd_ctl_add(chip->card, chip->clock_src_ctl)) < 0)
2121 goto ctl_error;
2122 }
2123#endif /* ECHOCARD_HAS_EXTERNAL_CLOCK */
2124
2125#ifdef ECHOCARD_HAS_DIGITAL_IO
2126 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_spdif_mode_switch, chip))) < 0)
2127 goto ctl_error;
2128#endif
2129
2130#ifdef ECHOCARD_HAS_PHANTOM_POWER
2131 if (chip->has_phantom_power)
2132 if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_phantom_power_switch, chip))) < 0)
2133 goto ctl_error;
2134#endif
2135
2136 if ((err = snd_card_register(card)) < 0) {
2137 snd_card_free(card);
2138 goto ctl_error;
2139 }
2140 snd_printk(KERN_INFO "Card registered: %s\n", card->longname);
2141
2142 pci_set_drvdata(pci, chip);
2143 dev++;
2144 return 0;
2145
2146ctl_error:
2147 snd_printk(KERN_ERR "new control error %d\n", err);
2148 snd_card_free(card);
2149 return err;
2150}
2151
2152
2153
2154static void __devexit snd_echo_remove(struct pci_dev *pci)
2155{
2156 struct echoaudio *chip;
2157
2158 chip = pci_get_drvdata(pci);
2159 if (chip)
2160 snd_card_free(chip->card);
2161 pci_set_drvdata(pci, NULL);
2162}
2163
2164
2165
2166/******************************************************************************
2167 Everything starts and ends here
2168******************************************************************************/
2169
2170/* pci_driver definition */
2171static struct pci_driver driver = {
2172 .name = "Echoaudio " ECHOCARD_NAME,
2173 .id_table = snd_echo_ids,
2174 .probe = snd_echo_probe,
2175 .remove = __devexit_p(snd_echo_remove),
2176};
2177
2178
2179
2180/* initialization of the module */
2181static int __init alsa_card_echo_init(void)
2182{
2183 return pci_register_driver(&driver);
2184}
2185
2186
2187
2188/* clean up the module */
2189static void __exit alsa_card_echo_exit(void)
2190{
2191 pci_unregister_driver(&driver);
2192}
2193
2194
2195module_init(alsa_card_echo_init)
2196module_exit(alsa_card_echo_exit)
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
new file mode 100644
index 000000000000..7e88c968e22f
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -0,0 +1,590 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 ****************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29 ****************************************************************************
30
31
32 Here's a block diagram of how most of the cards work:
33
34 +-----------+
35 record | |<-------------------- Inputs
36 <-------| | |
37 PCI | Transport | |
38 bus | engine | \|/
39 ------->| | +-------+
40 play | |--->|monitor|-------> Outputs
41 +-----------+ | mixer |
42 +-------+
43
44 The lines going to and from the PCI bus represent "pipes". A pipe performs
45 audio transport - moving audio data to and from buffers on the host via
46 bus mastering.
47
48 The inputs and outputs on the right represent input and output "busses."
49 A bus is a physical, real connection to the outside world. An example
50 of a bus would be the 1/4" analog connectors on the back of Layla or
51 an RCA S/PDIF connector.
52
53 For most cards, there is a one-to-one correspondence between outputs
54 and busses; that is, each individual pipe is hard-wired to a single bus.
55
56 Cards that work this way are Darla20, Gina20, Layla20, Darla24, Gina24,
57 Layla24, Mona, and Indigo.
58
59
60 Mia has a feature called "virtual outputs."
61
62
63 +-----------+
64 record | |<----------------------------- Inputs
65 <-------| | |
66 PCI | Transport | |
67 bus | engine | \|/
68 ------->| | +------+ +-------+
69 play | |-->|vmixer|-->|monitor|-------> Outputs
70 +-----------+ +------+ | mixer |
71 +-------+
72
73
74 Obviously, the difference here is the box labeled "vmixer." Vmixer is
75 short for "virtual output mixer." For Mia, pipes are *not* hard-wired
76 to a single bus; the vmixer lets you mix any pipe to any bus in any
77 combination.
78
79 Note, however, that the left-hand side of the diagram is unchanged.
80 Transport works exactly the same way - the difference is in the mixer stage.
81
82
83 Pipes and busses are numbered starting at zero.
84
85
86
87 Pipe index
88 ==========
89
90 A number of calls in CEchoGals refer to a "pipe index". A pipe index is
91 a unique number for a pipe that unambiguously refers to a playback or record
92 pipe. Pipe indices are numbered starting with analog outputs, followed by
93 digital outputs, then analog inputs, then digital inputs.
94
95 Take Gina24 as an example:
96
97 Pipe index
98
99 0-7 Analog outputs (0 .. FirstDigitalBusOut-1)
100 8-15 Digital outputs (FirstDigitalBusOut .. NumBussesOut-1)
101 16-17 Analog inputs
102 18-25 Digital inputs
103
104
105 You get the pipe index by calling CEchoGals::OpenAudio; the other transport
106 functions take the pipe index as a parameter. If you need a pipe index for
107 some other reason, use the handy Makepipe_index method.
108
109
110 Some calls take a CChannelMask parameter; CChannelMask is a handy way to
111 group pipe indices.
112
113
114
115 Digital mode switch
116 ===================
117
118 Some cards (right now, Gina24, Layla24, and Mona) have a Digital Mode Switch
119 or DMS. Cards with a DMS can be set to one of three mutually exclusive
120 digital modes: S/PDIF RCA, S/PDIF optical, or ADAT optical.
121
122 This may create some confusion since ADAT optical is 8 channels wide and
123 S/PDIF is only two channels wide. Gina24, Layla24, and Mona handle this
124 by acting as if they always have 8 digital outs and ins. If you are in
125 either S/PDIF mode, the last 6 channels don't do anything - data sent
126 out these channels is thrown away and you will always record zeros.
127
128 Note that with Gina24, Layla24, and Mona, sample rates above 50 kHz are
129 only available if you have the card configured for S/PDIF optical or S/PDIF
130 RCA.
131
132
133
134 Double speed mode
135 =================
136
137 Some of the cards support 88.2 kHz and 96 kHz sampling (Darla24, Gina24,
138 Layla24, Mona, Mia, and Indigo). For these cards, the driver sometimes has
139 to worry about "double speed mode"; double speed mode applies whenever the
140 sampling rate is above 50 kHz.
141
142 For instance, Mona and Layla24 support word clock sync. However, they
143 actually support two different word clock modes - single speed (below
144 50 kHz) and double speed (above 50 kHz). The hardware detects if a single
145 or double speed word clock signal is present; the generic code uses that
146 information to determine which mode to use.
147
148 The generic code takes care of all this for you.
149*/
150
151
152#ifndef _ECHOAUDIO_H_
153#define _ECHOAUDIO_H_
154
155
156#define TRUE 1
157#define FALSE 0
158
159#include "echoaudio_dsp.h"
160
161
162
163/***********************************************************************
164
165 PCI configuration space
166
167***********************************************************************/
168
169/*
170 * PCI vendor ID and device IDs for the hardware
171 */
172#define VENDOR_ID 0x1057
173#define DEVICE_ID_56301 0x1801
174#define DEVICE_ID_56361 0x3410
175#define SUBVENDOR_ID 0xECC0
176
177
178/*
179 * Valid Echo PCI subsystem card IDs
180 */
181#define DARLA20 0x0010
182#define GINA20 0x0020
183#define LAYLA20 0x0030
184#define DARLA24 0x0040
185#define GINA24 0x0050
186#define LAYLA24 0x0060
187#define MONA 0x0070
188#define MIA 0x0080
189#define INDIGO 0x0090
190#define INDIGO_IO 0x00a0
191#define INDIGO_DJ 0x00b0
192#define ECHO3G 0x0100
193
194
195/************************************************************************
196
197 Array sizes and so forth
198
199***********************************************************************/
200
201/*
202 * Sizes
203 */
204#define ECHO_MAXAUDIOINPUTS 32 /* Max audio input channels */
205#define ECHO_MAXAUDIOOUTPUTS 32 /* Max audio output channels */
206#define ECHO_MAXAUDIOPIPES 32 /* Max number of input and output
207 * pipes */
208#define E3G_MAX_OUTPUTS 16
209#define ECHO_MAXMIDIJACKS 1 /* Max MIDI ports */
210#define ECHO_MIDI_QUEUE_SZ 512 /* Max MIDI input queue entries */
211#define ECHO_MTC_QUEUE_SZ 32 /* Max MIDI time code input queue
212 * entries */
213
214/*
215 * MIDI activity indicator timeout
216 */
217#define MIDI_ACTIVITY_TIMEOUT_USEC 200000
218
219
220/****************************************************************************
221
222 Clocks
223
224*****************************************************************************/
225
226/*
227 * Clock numbers
228 */
229#define ECHO_CLOCK_INTERNAL 0
230#define ECHO_CLOCK_WORD 1
231#define ECHO_CLOCK_SUPER 2
232#define ECHO_CLOCK_SPDIF 3
233#define ECHO_CLOCK_ADAT 4
234#define ECHO_CLOCK_ESYNC 5
235#define ECHO_CLOCK_ESYNC96 6
236#define ECHO_CLOCK_MTC 7
237#define ECHO_CLOCK_NUMBER 8
238#define ECHO_CLOCKS 0xffff
239
240/*
241 * Clock bit numbers - used to report capabilities and whatever clocks
242 * are being detected dynamically.
243 */
244#define ECHO_CLOCK_BIT_INTERNAL (1 << ECHO_CLOCK_INTERNAL)
245#define ECHO_CLOCK_BIT_WORD (1 << ECHO_CLOCK_WORD)
246#define ECHO_CLOCK_BIT_SUPER (1 << ECHO_CLOCK_SUPER)
247#define ECHO_CLOCK_BIT_SPDIF (1 << ECHO_CLOCK_SPDIF)
248#define ECHO_CLOCK_BIT_ADAT (1 << ECHO_CLOCK_ADAT)
249#define ECHO_CLOCK_BIT_ESYNC (1 << ECHO_CLOCK_ESYNC)
250#define ECHO_CLOCK_BIT_ESYNC96 (1 << ECHO_CLOCK_ESYNC96)
251#define ECHO_CLOCK_BIT_MTC (1<<ECHO_CLOCK_MTC)
252
253
254/***************************************************************************
255
256 Digital modes
257
258****************************************************************************/
259
260/*
261 * Digital modes for Mona, Layla24, and Gina24
262 */
263#define DIGITAL_MODE_NONE 0xFF
264#define DIGITAL_MODE_SPDIF_RCA 0
265#define DIGITAL_MODE_SPDIF_OPTICAL 1
266#define DIGITAL_MODE_ADAT 2
267#define DIGITAL_MODE_SPDIF_CDROM 3
268#define DIGITAL_MODES 4
269
270/*
271 * Digital mode capability masks
272 */
273#define ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA (1 << DIGITAL_MODE_SPDIF_RCA)
274#define ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL (1 << DIGITAL_MODE_SPDIF_OPTICAL)
275#define ECHOCAPS_HAS_DIGITAL_MODE_ADAT (1 << DIGITAL_MODE_ADAT)
276#define ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_CDROM (1 << DIGITAL_MODE_SPDIF_CDROM)
277
278
279#define EXT_3GBOX_NC 0x01 /* 3G box not connected */
280#define EXT_3GBOX_NOT_SET 0x02 /* 3G box not detected yet */
281
282
283#define ECHOGAIN_MUTED (-128) /* Minimum possible gain */
284#define ECHOGAIN_MINOUT (-128) /* Min output gain (dB) */
285#define ECHOGAIN_MAXOUT (6) /* Max output gain (dB) */
286#define ECHOGAIN_MININP (-50) /* Min input gain (0.5 dB) */
287#define ECHOGAIN_MAXINP (50) /* Max input gain (0.5 dB) */
288
289#define PIPE_STATE_STOPPED 0 /* Pipe has been reset */
290#define PIPE_STATE_PAUSED 1 /* Pipe has been stopped */
291#define PIPE_STATE_STARTED 2 /* Pipe has been started */
292#define PIPE_STATE_PENDING 3 /* Pipe has pending start */
293
294
295/* Debug initialization */
296#ifdef CONFIG_SND_DEBUG
297#define DE_INIT(x) snd_printk x
298#else
299#define DE_INIT(x)
300#endif
301
302/* Debug hw_params callbacks */
303#ifdef CONFIG_SND_DEBUG
304#define DE_HWP(x) snd_printk x
305#else
306#define DE_HWP(x)
307#endif
308
309/* Debug normal activity (open, start, stop...) */
310#ifdef CONFIG_SND_DEBUG
311#define DE_ACT(x) snd_printk x
312#else
313#define DE_ACT(x)
314#endif
315
316/* Debug midi activity */
317#ifdef CONFIG_SND_DEBUG
318#define DE_MID(x) snd_printk x
319#else
320#define DE_MID(x)
321#endif
322
323
324struct audiopipe {
325 volatile u32 *dma_counter; /* Commpage register that contains
326 * the current dma position
327 * (lower 32 bits only)
328 */
329 u32 last_counter; /* The last position, which is used
330 * to compute...
331 */
332 u32 position; /* ...the number of bytes tranferred
333 * by the DMA engine, modulo the
334 * buffer size
335 */
336 short index; /* Index of the first channel or <0
337 * if hw is not configured yet
338 */
339 short interleave;
340 struct snd_dma_buffer sgpage; /* Room for the scatter-gather list */
341 struct snd_pcm_hardware hw;
342 struct snd_pcm_hw_constraint_list constr;
343 short sglist_head;
344 char state; /* pipe state */
345};
346
347
348struct audioformat {
349 u8 interleave; /* How the data is arranged in memory:
350 * mono = 1, stereo = 2, ...
351 */
352 u8 bits_per_sample; /* 8, 16, 24, 32 (24 bits left aligned) */
353 char mono_to_stereo; /* Only used if interleave is 1 and
354 * if this is an output pipe.
355 */
356 char data_are_bigendian; /* 1 = big endian, 0 = little endian */
357};
358
359
360struct echoaudio {
361 spinlock_t lock;
362 struct snd_pcm_substream *substream[DSP_MAXPIPES];
363 int last_period[DSP_MAXPIPES];
364 struct semaphore mode_mutex;
365 u16 num_digital_modes, digital_mode_list[6];
366 u16 num_clock_sources, clock_source_list[10];
367 atomic_t opencount;
368 struct snd_kcontrol *clock_src_ctl;
369 struct snd_pcm *analog_pcm, *digital_pcm;
370 struct snd_card *card;
371 const char *card_name;
372 struct pci_dev *pci;
373 unsigned long dsp_registers_phys;
374 struct resource *iores;
375 struct snd_dma_buffer commpage_dma_buf;
376 int irq;
377#ifdef ECHOCARD_HAS_MIDI
378 struct snd_rawmidi *rmidi;
379 struct snd_rawmidi_substream *midi_in, *midi_out;
380#endif
381 struct timer_list timer;
382 char tinuse; /* Timer in use */
383 char midi_full; /* MIDI output buffer is full */
384 char can_set_rate;
385 char rate_set;
386
387 /* This stuff is used mainly by the lowlevel code */
388 struct comm_page *comm_page; /* Virtual address of the memory
389 * seen by DSP
390 */
391 u32 pipe_alloc_mask; /* Bitmask of allocated pipes */
392 u32 pipe_cyclic_mask; /* Bitmask of pipes with cyclic
393 * buffers
394 */
395 u32 sample_rate; /* Card sample rate in Hz */
396 u8 digital_mode; /* Current digital mode
397 * (see DIGITAL_MODE_*)
398 */
399 u8 spdif_status; /* Gina20, Darla20, Darla24 - only */
400 u8 clock_state; /* Gina20, Darla20, Darla24 - only */
401 u8 input_clock; /* Currently selected sample clock
402 * source
403 */
404 u8 output_clock; /* Layla20 only */
405 char meters_enabled; /* VU-meters status */
406 char asic_loaded; /* Set TRUE when ASIC loaded */
407 char bad_board; /* Set TRUE if DSP won't load */
408 char professional_spdif; /* 0 = consumer; 1 = professional */
409 char non_audio_spdif; /* 3G - only */
410 char digital_in_automute; /* Gina24, Layla24, Mona - only */
411 char has_phantom_power;
412 char hasnt_input_nominal_level; /* Gina3G */
413 char phantom_power; /* Gina3G - only */
414 char has_midi;
415 char midi_input_enabled;
416
417#ifdef ECHOCARD_ECHO3G
418 /* External module -dependent pipe and bus indexes */
419 char px_digital_out, px_analog_in, px_digital_in, px_num;
420 char bx_digital_out, bx_analog_in, bx_digital_in, bx_num;
421#endif
422
423 char nominal_level[ECHO_MAXAUDIOPIPES]; /* True == -10dBV
424 * False == +4dBu */
425 s8 input_gain[ECHO_MAXAUDIOINPUTS]; /* Input level -50..+50
426 * unit is 0.5dB */
427 s8 output_gain[ECHO_MAXAUDIOOUTPUTS]; /* Output level -128..+6 dB
428 * (-128=muted) */
429 s8 monitor_gain[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOINPUTS];
430 /* -128..+6 dB */
431 s8 vmixer_gain[ECHO_MAXAUDIOOUTPUTS][ECHO_MAXAUDIOOUTPUTS];
432 /* -128..+6 dB */
433
434 u16 digital_modes; /* Bitmask of supported modes
435 * (see ECHOCAPS_HAS_DIGITAL_MODE_*) */
436 u16 input_clock_types; /* Suppoted input clock types */
437 u16 output_clock_types; /* Suppoted output clock types -
438 * Layla20 only */
439 u16 device_id, subdevice_id;
440 u16 *dsp_code; /* Current DSP code loaded,
441 * NULL if nothing loaded */
442 const struct firmware *dsp_code_to_load;/* DSP code to load */
443 const struct firmware *asic_code; /* Current ASIC code */
444 u32 comm_page_phys; /* Physical address of the
445 * memory seen by DSP */
446 volatile u32 __iomem *dsp_registers; /* DSP's register base */
447 u32 active_mask; /* Chs. active mask or
448 * punks out */
449
450#ifdef ECHOCARD_HAS_MIDI
451 u16 mtc_state; /* State for MIDI input parsing state machine */
452 u8 midi_buffer[MIDI_IN_BUFFER_SIZE];
453#endif
454};
455
456
457static int init_dsp_comm_page(struct echoaudio *chip);
458static int init_line_levels(struct echoaudio *chip);
459static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe);
460static int load_firmware(struct echoaudio *chip);
461static int wait_handshake(struct echoaudio *chip);
462static int send_vector(struct echoaudio *chip, u32 command);
463static int get_firmware(const struct firmware **fw_entry,
464 const struct firmware *frm, struct echoaudio *chip);
465static void free_firmware(const struct firmware *fw_entry);
466
467#ifdef ECHOCARD_HAS_MIDI
468static int enable_midi_input(struct echoaudio *chip, char enable);
469static int midi_service_irq(struct echoaudio *chip);
470static int __devinit snd_echo_midi_create(struct snd_card *card,
471 struct echoaudio *chip);
472#endif
473
474
475static inline void clear_handshake(struct echoaudio *chip)
476{
477 chip->comm_page->handshake = 0;
478}
479
480static inline u32 get_dsp_register(struct echoaudio *chip, u32 index)
481{
482 return readl(&chip->dsp_registers[index]);
483}
484
485static inline void set_dsp_register(struct echoaudio *chip, u32 index,
486 u32 value)
487{
488 writel(value, &chip->dsp_registers[index]);
489}
490
491
492/* Pipe and bus indexes. PX_* and BX_* are defined as chip->px_* and chip->bx_*
493for 3G cards because they depend on the external box. They are integer
494constants for all other cards.
495Never use those defines directly, use the following functions instead. */
496
497static inline int px_digital_out(const struct echoaudio *chip)
498{
499 return PX_DIGITAL_OUT;
500}
501
502static inline int px_analog_in(const struct echoaudio *chip)
503{
504 return PX_ANALOG_IN;
505}
506
507static inline int px_digital_in(const struct echoaudio *chip)
508{
509 return PX_DIGITAL_IN;
510}
511
512static inline int px_num(const struct echoaudio *chip)
513{
514 return PX_NUM;
515}
516
517static inline int bx_digital_out(const struct echoaudio *chip)
518{
519 return BX_DIGITAL_OUT;
520}
521
522static inline int bx_analog_in(const struct echoaudio *chip)
523{
524 return BX_ANALOG_IN;
525}
526
527static inline int bx_digital_in(const struct echoaudio *chip)
528{
529 return BX_DIGITAL_IN;
530}
531
532static inline int bx_num(const struct echoaudio *chip)
533{
534 return BX_NUM;
535}
536
537static inline int num_pipes_out(const struct echoaudio *chip)
538{
539 return px_analog_in(chip);
540}
541
542static inline int num_pipes_in(const struct echoaudio *chip)
543{
544 return px_num(chip) - px_analog_in(chip);
545}
546
547static inline int num_busses_out(const struct echoaudio *chip)
548{
549 return bx_analog_in(chip);
550}
551
552static inline int num_busses_in(const struct echoaudio *chip)
553{
554 return bx_num(chip) - bx_analog_in(chip);
555}
556
557static inline int num_analog_busses_out(const struct echoaudio *chip)
558{
559 return bx_digital_out(chip);
560}
561
562static inline int num_analog_busses_in(const struct echoaudio *chip)
563{
564 return bx_digital_in(chip) - bx_analog_in(chip);
565}
566
567static inline int num_digital_busses_out(const struct echoaudio *chip)
568{
569 return num_busses_out(chip) - num_analog_busses_out(chip);
570}
571
572static inline int num_digital_busses_in(const struct echoaudio *chip)
573{
574 return num_busses_in(chip) - num_analog_busses_in(chip);
575}
576
577/* The monitor array is a one-dimensional array; compute the offset
578 * into the array */
579static inline int monitor_index(const struct echoaudio *chip, int out, int in)
580{
581 return out * num_busses_in(chip) + in;
582}
583
584
585#ifndef pci_device
586#define pci_device(chip) (&chip->pci->dev)
587#endif
588
589
590#endif /* _ECHOAUDIO_H_ */
diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c
new file mode 100644
index 000000000000..9f439ea459f4
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_3g.c
@@ -0,0 +1,431 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32
33/* These functions are common for all "3G" cards */
34
35
36static int check_asic_status(struct echoaudio *chip)
37{
38 u32 box_status;
39
40 if (wait_handshake(chip))
41 return -EIO;
42
43 chip->comm_page->ext_box_status =
44 __constant_cpu_to_le32(E3G_ASIC_NOT_LOADED);
45 chip->asic_loaded = FALSE;
46 clear_handshake(chip);
47 send_vector(chip, DSP_VC_TEST_ASIC);
48
49 if (wait_handshake(chip)) {
50 chip->dsp_code = NULL;
51 return -EIO;
52 }
53
54 box_status = le32_to_cpu(chip->comm_page->ext_box_status);
55 DE_INIT(("box_status=%x\n", box_status));
56 if (box_status == E3G_ASIC_NOT_LOADED)
57 return -ENODEV;
58
59 chip->asic_loaded = TRUE;
60 return box_status & E3G_BOX_TYPE_MASK;
61}
62
63
64
65static inline u32 get_frq_reg(struct echoaudio *chip)
66{
67 return le32_to_cpu(chip->comm_page->e3g_frq_register);
68}
69
70
71
72/* Most configuration of 3G cards is accomplished by writing the control
73register. write_control_reg sends the new control register value to the DSP. */
74static int write_control_reg(struct echoaudio *chip, u32 ctl, u32 frq,
75 char force)
76{
77 if (wait_handshake(chip))
78 return -EIO;
79
80 DE_ACT(("WriteControlReg: Setting 0x%x, 0x%x\n", ctl, frq));
81
82 ctl = cpu_to_le32(ctl);
83 frq = cpu_to_le32(frq);
84
85 if (ctl != chip->comm_page->control_register ||
86 frq != chip->comm_page->e3g_frq_register || force) {
87 chip->comm_page->e3g_frq_register = frq;
88 chip->comm_page->control_register = ctl;
89 clear_handshake(chip);
90 return send_vector(chip, DSP_VC_WRITE_CONTROL_REG);
91 }
92
93 DE_ACT(("WriteControlReg: not written, no change\n"));
94 return 0;
95}
96
97
98
99/* Set the digital mode - currently for Gina24, Layla24, Mona, 3G */
100static int set_digital_mode(struct echoaudio *chip, u8 mode)
101{
102 u8 previous_mode;
103 int err, i, o;
104
105 /* All audio channels must be closed before changing the digital mode */
106 snd_assert(!chip->pipe_alloc_mask, return -EAGAIN);
107
108 snd_assert(chip->digital_modes & (1 << mode), return -EINVAL);
109
110 previous_mode = chip->digital_mode;
111 err = dsp_set_digital_mode(chip, mode);
112
113 /* If we successfully changed the digital mode from or to ADAT,
114 * then make sure all output, input and monitor levels are
115 * updated by the DSP comm object. */
116 if (err >= 0 && previous_mode != mode &&
117 (previous_mode == DIGITAL_MODE_ADAT || mode == DIGITAL_MODE_ADAT)) {
118 spin_lock_irq(&chip->lock);
119 for (o = 0; o < num_busses_out(chip); o++)
120 for (i = 0; i < num_busses_in(chip); i++)
121 set_monitor_gain(chip, o, i,
122 chip->monitor_gain[o][i]);
123
124#ifdef ECHOCARD_HAS_INPUT_GAIN
125 for (i = 0; i < num_busses_in(chip); i++)
126 set_input_gain(chip, i, chip->input_gain[i]);
127 update_input_line_level(chip);
128#endif
129
130 for (o = 0; o < num_busses_out(chip); o++)
131 set_output_gain(chip, o, chip->output_gain[o]);
132 update_output_line_level(chip);
133 spin_unlock_irq(&chip->lock);
134 }
135
136 return err;
137}
138
139
140
141static u32 set_spdif_bits(struct echoaudio *chip, u32 control_reg, u32 rate)
142{
143 control_reg &= E3G_SPDIF_FORMAT_CLEAR_MASK;
144
145 switch (rate) {
146 case 32000 :
147 control_reg |= E3G_SPDIF_SAMPLE_RATE0 | E3G_SPDIF_SAMPLE_RATE1;
148 break;
149 case 44100 :
150 if (chip->professional_spdif)
151 control_reg |= E3G_SPDIF_SAMPLE_RATE0;
152 break;
153 case 48000 :
154 control_reg |= E3G_SPDIF_SAMPLE_RATE1;
155 break;
156 }
157
158 if (chip->professional_spdif)
159 control_reg |= E3G_SPDIF_PRO_MODE;
160
161 if (chip->non_audio_spdif)
162 control_reg |= E3G_SPDIF_NOT_AUDIO;
163
164 control_reg |= E3G_SPDIF_24_BIT | E3G_SPDIF_TWO_CHANNEL |
165 E3G_SPDIF_COPY_PERMIT;
166
167 return control_reg;
168}
169
170
171
172/* Set the S/PDIF output format */
173static int set_professional_spdif(struct echoaudio *chip, char prof)
174{
175 u32 control_reg;
176
177 control_reg = le32_to_cpu(chip->comm_page->control_register);
178 chip->professional_spdif = prof;
179 control_reg = set_spdif_bits(chip, control_reg, chip->sample_rate);
180 return write_control_reg(chip, control_reg, get_frq_reg(chip), 0);
181}
182
183
184
185/* detect_input_clocks() returns a bitmask consisting of all the input clocks
186currently connected to the hardware; this changes as the user connects and
187disconnects clock inputs. You should use this information to determine which
188clocks the user is allowed to select. */
189static u32 detect_input_clocks(const struct echoaudio *chip)
190{
191 u32 clocks_from_dsp, clock_bits;
192
193 /* Map the DSP clock detect bits to the generic driver clock
194 * detect bits */
195 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
196
197 clock_bits = ECHO_CLOCK_BIT_INTERNAL;
198
199 if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD)
200 clock_bits |= ECHO_CLOCK_BIT_WORD;
201
202 switch(chip->digital_mode) {
203 case DIGITAL_MODE_SPDIF_RCA:
204 case DIGITAL_MODE_SPDIF_OPTICAL:
205 if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF)
206 clock_bits |= ECHO_CLOCK_BIT_SPDIF;
207 break;
208 case DIGITAL_MODE_ADAT:
209 if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_ADAT)
210 clock_bits |= ECHO_CLOCK_BIT_ADAT;
211 break;
212 }
213
214 return clock_bits;
215}
216
217
218
219static int load_asic(struct echoaudio *chip)
220{
221 int box_type, err;
222
223 if (chip->asic_loaded)
224 return 0;
225
226 /* Give the DSP a few milliseconds to settle down */
227 mdelay(2);
228
229 err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC,
230 &card_fw[FW_3G_ASIC]);
231 if (err < 0)
232 return err;
233
234 chip->asic_code = &card_fw[FW_3G_ASIC];
235
236 /* Now give the new ASIC a little time to set up */
237 mdelay(2);
238 /* See if it worked */
239 box_type = check_asic_status(chip);
240
241 /* Set up the control register if the load succeeded -
242 * 48 kHz, internal clock, S/PDIF RCA mode */
243 if (box_type >= 0) {
244 err = write_control_reg(chip, E3G_48KHZ,
245 E3G_FREQ_REG_DEFAULT, TRUE);
246 if (err < 0)
247 return err;
248 }
249
250 return box_type;
251}
252
253
254
255static int set_sample_rate(struct echoaudio *chip, u32 rate)
256{
257 u32 control_reg, clock, base_rate, frq_reg;
258
259 /* Only set the clock for internal mode. */
260 if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
261 DE_ACT(("set_sample_rate: Cannot set sample rate - "
262 "clock not set to CLK_CLOCKININTERNAL\n"));
263 /* Save the rate anyhow */
264 chip->comm_page->sample_rate = cpu_to_le32(rate);
265 chip->sample_rate = rate;
266 set_input_clock(chip, chip->input_clock);
267 return 0;
268 }
269
270 snd_assert(rate < 50000 || chip->digital_mode != DIGITAL_MODE_ADAT,
271 return -EINVAL);
272
273 clock = 0;
274 control_reg = le32_to_cpu(chip->comm_page->control_register);
275 control_reg &= E3G_CLOCK_CLEAR_MASK;
276
277 switch (rate) {
278 case 96000:
279 clock = E3G_96KHZ;
280 break;
281 case 88200:
282 clock = E3G_88KHZ;
283 break;
284 case 48000:
285 clock = E3G_48KHZ;
286 break;
287 case 44100:
288 clock = E3G_44KHZ;
289 break;
290 case 32000:
291 clock = E3G_32KHZ;
292 break;
293 default:
294 clock = E3G_CONTINUOUS_CLOCK;
295 if (rate > 50000)
296 clock |= E3G_DOUBLE_SPEED_MODE;
297 break;
298 }
299
300 control_reg |= clock;
301 control_reg = set_spdif_bits(chip, control_reg, rate);
302
303 base_rate = rate;
304 if (base_rate > 50000)
305 base_rate /= 2;
306 if (base_rate < 32000)
307 base_rate = 32000;
308
309 frq_reg = E3G_MAGIC_NUMBER / base_rate - 2;
310 if (frq_reg > E3G_FREQ_REG_MAX)
311 frq_reg = E3G_FREQ_REG_MAX;
312
313 chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
314 chip->sample_rate = rate;
315 DE_ACT(("SetSampleRate: %d clock %x\n", rate, control_reg));
316
317 /* Tell the DSP about it - DSP reads both control reg & freq reg */
318 return write_control_reg(chip, control_reg, frq_reg, 0);
319}
320
321
322
323/* Set the sample clock source to internal, S/PDIF, ADAT */
324static int set_input_clock(struct echoaudio *chip, u16 clock)
325{
326 u32 control_reg, clocks_from_dsp;
327
328 DE_ACT(("set_input_clock:\n"));
329
330 /* Mask off the clock select bits */
331 control_reg = le32_to_cpu(chip->comm_page->control_register) &
332 E3G_CLOCK_CLEAR_MASK;
333 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
334
335 switch (clock) {
336 case ECHO_CLOCK_INTERNAL:
337 DE_ACT(("Set Echo3G clock to INTERNAL\n"));
338 chip->input_clock = ECHO_CLOCK_INTERNAL;
339 return set_sample_rate(chip, chip->sample_rate);
340 case ECHO_CLOCK_SPDIF:
341 if (chip->digital_mode == DIGITAL_MODE_ADAT)
342 return -EAGAIN;
343 DE_ACT(("Set Echo3G clock to SPDIF\n"));
344 control_reg |= E3G_SPDIF_CLOCK;
345 if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_SPDIF96)
346 control_reg |= E3G_DOUBLE_SPEED_MODE;
347 else
348 control_reg &= ~E3G_DOUBLE_SPEED_MODE;
349 break;
350 case ECHO_CLOCK_ADAT:
351 if (chip->digital_mode != DIGITAL_MODE_ADAT)
352 return -EAGAIN;
353 DE_ACT(("Set Echo3G clock to ADAT\n"));
354 control_reg |= E3G_ADAT_CLOCK;
355 control_reg &= ~E3G_DOUBLE_SPEED_MODE;
356 break;
357 case ECHO_CLOCK_WORD:
358 DE_ACT(("Set Echo3G clock to WORD\n"));
359 control_reg |= E3G_WORD_CLOCK;
360 if (clocks_from_dsp & E3G_CLOCK_DETECT_BIT_WORD96)
361 control_reg |= E3G_DOUBLE_SPEED_MODE;
362 else
363 control_reg &= ~E3G_DOUBLE_SPEED_MODE;
364 break;
365 default:
366 DE_ACT(("Input clock 0x%x not supported for Echo3G\n", clock));
367 return -EINVAL;
368 }
369
370 chip->input_clock = clock;
371 return write_control_reg(chip, control_reg, get_frq_reg(chip), 1);
372}
373
374
375
376static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
377{
378 u32 control_reg;
379 int err, incompatible_clock;
380
381 /* Set clock to "internal" if it's not compatible with the new mode */
382 incompatible_clock = FALSE;
383 switch (mode) {
384 case DIGITAL_MODE_SPDIF_OPTICAL:
385 case DIGITAL_MODE_SPDIF_RCA:
386 if (chip->input_clock == ECHO_CLOCK_ADAT)
387 incompatible_clock = TRUE;
388 break;
389 case DIGITAL_MODE_ADAT:
390 if (chip->input_clock == ECHO_CLOCK_SPDIF)
391 incompatible_clock = TRUE;
392 break;
393 default:
394 DE_ACT(("Digital mode not supported: %d\n", mode));
395 return -EINVAL;
396 }
397
398 spin_lock_irq(&chip->lock);
399
400 if (incompatible_clock) {
401 chip->sample_rate = 48000;
402 set_input_clock(chip, ECHO_CLOCK_INTERNAL);
403 }
404
405 /* Clear the current digital mode */
406 control_reg = le32_to_cpu(chip->comm_page->control_register);
407 control_reg &= E3G_DIGITAL_MODE_CLEAR_MASK;
408
409 /* Tweak the control reg */
410 switch (mode) {
411 case DIGITAL_MODE_SPDIF_OPTICAL:
412 control_reg |= E3G_SPDIF_OPTICAL_MODE;
413 break;
414 case DIGITAL_MODE_SPDIF_RCA:
415 /* E3G_SPDIF_OPTICAL_MODE bit cleared */
416 break;
417 case DIGITAL_MODE_ADAT:
418 control_reg |= E3G_ADAT_MODE;
419 control_reg &= ~E3G_DOUBLE_SPEED_MODE; /* @@ useless */
420 break;
421 }
422
423 err = write_control_reg(chip, control_reg, get_frq_reg(chip), 1);
424 spin_unlock_irq(&chip->lock);
425 if (err < 0)
426 return err;
427 chip->digital_mode = mode;
428
429 DE_ACT(("set_digital_mode(%d)\n", chip->digital_mode));
430 return incompatible_clock;
431}
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
new file mode 100644
index 000000000000..42afa837d9b4
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -0,0 +1,1125 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31#if PAGE_SIZE < 4096
32#error PAGE_SIZE is < 4k
33#endif
34
35static int restore_dsp_rettings(struct echoaudio *chip);
36
37
38/* Some vector commands involve the DSP reading or writing data to and from the
39comm page; if you send one of these commands to the DSP, it will complete the
40command and then write a non-zero value to the Handshake field in the
41comm page. This function waits for the handshake to show up. */
42static int wait_handshake(struct echoaudio *chip)
43{
44 int i;
45
46 /* Wait up to 10ms for the handshake from the DSP */
47 for (i = 0; i < HANDSHAKE_TIMEOUT; i++) {
48 /* Look for the handshake value */
49 if (chip->comm_page->handshake) {
50 /*if (i) DE_ACT(("Handshake time: %d\n", i));*/
51 return 0;
52 }
53 udelay(1);
54 }
55
56 snd_printk(KERN_ERR "wait_handshake(): Timeout waiting for DSP\n");
57 return -EBUSY;
58}
59
60
61
62/* Much of the interaction between the DSP and the driver is done via vector
63commands; send_vector writes a vector command to the DSP. Typically, this
64causes the DSP to read or write fields in the comm page.
65PCI posting is not required thanks to the handshake logic. */
66static int send_vector(struct echoaudio *chip, u32 command)
67{
68 int i;
69
70 wmb(); /* Flush all pending writes before sending the command */
71
72 /* Wait up to 100ms for the "vector busy" bit to be off */
73 for (i = 0; i < VECTOR_BUSY_TIMEOUT; i++) {
74 if (!(get_dsp_register(chip, CHI32_VECTOR_REG) &
75 CHI32_VECTOR_BUSY)) {
76 set_dsp_register(chip, CHI32_VECTOR_REG, command);
77 /*if (i) DE_ACT(("send_vector time: %d\n", i));*/
78 return 0;
79 }
80 udelay(1);
81 }
82
83 DE_ACT((KERN_ERR "timeout on send_vector\n"));
84 return -EBUSY;
85}
86
87
88
89/* write_dsp writes a 32-bit value to the DSP; this is used almost
90exclusively for loading the DSP. */
91static int write_dsp(struct echoaudio *chip, u32 data)
92{
93 u32 status, i;
94
95 for (i = 0; i < 10000000; i++) { /* timeout = 10s */
96 status = get_dsp_register(chip, CHI32_STATUS_REG);
97 if ((status & CHI32_STATUS_HOST_WRITE_EMPTY) != 0) {
98 set_dsp_register(chip, CHI32_DATA_REG, data);
99 wmb(); /* write it immediately */
100 return 0;
101 }
102 udelay(1);
103 cond_resched();
104 }
105
106 chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */
107 DE_ACT((KERN_ERR "write_dsp: Set bad_board to TRUE\n"));
108 return -EIO;
109}
110
111
112
113/* read_dsp reads a 32-bit value from the DSP; this is used almost
114exclusively for loading the DSP and checking the status of the ASIC. */
115static int read_dsp(struct echoaudio *chip, u32 *data)
116{
117 u32 status, i;
118
119 for (i = 0; i < READ_DSP_TIMEOUT; i++) {
120 status = get_dsp_register(chip, CHI32_STATUS_REG);
121 if ((status & CHI32_STATUS_HOST_READ_FULL) != 0) {
122 *data = get_dsp_register(chip, CHI32_DATA_REG);
123 return 0;
124 }
125 udelay(1);
126 cond_resched();
127 }
128
129 chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */
130 DE_INIT((KERN_ERR "read_dsp: Set bad_board to TRUE\n"));
131 return -EIO;
132}
133
134
135
136/****************************************************************************
137 Firmware loading functions
138 ****************************************************************************/
139
140/* This function is used to read back the serial number from the DSP;
141this is triggered by the SET_COMMPAGE_ADDR command.
142Only some early Echogals products have serial numbers in the ROM;
143the serial number is not used, but you still need to do this as
144part of the DSP load process. */
145static int read_sn(struct echoaudio *chip)
146{
147 int i;
148 u32 sn[6];
149
150 for (i = 0; i < 5; i++) {
151 if (read_dsp(chip, &sn[i])) {
152 snd_printk(KERN_ERR "Failed to read serial number\n");
153 return -EIO;
154 }
155 }
156 DE_INIT(("Read serial number %08x %08x %08x %08x %08x\n",
157 sn[0], sn[1], sn[2], sn[3], sn[4]));
158 return 0;
159}
160
161
162
163#ifndef ECHOCARD_HAS_ASIC
164/* This card has no ASIC, just return ok */
165static inline int check_asic_status(struct echoaudio *chip)
166{
167 chip->asic_loaded = TRUE;
168 return 0;
169}
170
171#endif /* !ECHOCARD_HAS_ASIC */
172
173
174
175#ifdef ECHOCARD_HAS_ASIC
176
177/* Load ASIC code - done after the DSP is loaded */
178static int load_asic_generic(struct echoaudio *chip, u32 cmd,
179 const struct firmware *asic)
180{
181 const struct firmware *fw;
182 int err;
183 u32 i, size;
184 u8 *code;
185
186 if ((err = get_firmware(&fw, asic, chip)) < 0) {
187 snd_printk(KERN_WARNING "Firmware not found !\n");
188 return err;
189 }
190
191 code = (u8 *)fw->data;
192 size = fw->size;
193
194 /* Send the "Here comes the ASIC" command */
195 if (write_dsp(chip, cmd) < 0)
196 goto la_error;
197
198 /* Write length of ASIC file in bytes */
199 if (write_dsp(chip, size) < 0)
200 goto la_error;
201
202 for (i = 0; i < size; i++) {
203 if (write_dsp(chip, code[i]) < 0)
204 goto la_error;
205 }
206
207 DE_INIT(("ASIC loaded\n"));
208 free_firmware(fw);
209 return 0;
210
211la_error:
212 DE_INIT(("failed on write_dsp\n"));
213 free_firmware(fw);
214 return -EIO;
215}
216
217#endif /* ECHOCARD_HAS_ASIC */
218
219
220
221#ifdef DSP_56361
222
223/* Install the resident loader for 56361 DSPs; The resident loader is on
224the EPROM on the board for 56301 DSP. The resident loader is a tiny little
225program that is used to load the real DSP code. */
226static int install_resident_loader(struct echoaudio *chip)
227{
228 u32 address;
229 int index, words, i;
230 u16 *code;
231 u32 status;
232 const struct firmware *fw;
233
234 /* 56361 cards only! This check is required by the old 56301-based
235 Mona and Gina24 */
236 if (chip->device_id != DEVICE_ID_56361)
237 return 0;
238
239 /* Look to see if the resident loader is present. If the resident
240 loader is already installed, host flag 5 will be on. */
241 status = get_dsp_register(chip, CHI32_STATUS_REG);
242 if (status & CHI32_STATUS_REG_HF5) {
243 DE_INIT(("Resident loader already installed; status is 0x%x\n",
244 status));
245 return 0;
246 }
247
248 if ((i = get_firmware(&fw, &card_fw[FW_361_LOADER], chip)) < 0) {
249 snd_printk(KERN_WARNING "Firmware not found !\n");
250 return i;
251 }
252
253 /* The DSP code is an array of 16 bit words. The array is divided up
254 into sections. The first word of each section is the size in words,
255 followed by the section type.
256 Since DSP addresses and data are 24 bits wide, they each take up two
257 16 bit words in the array.
258 This is a lot like the other loader loop, but it's not a loop, you
259 don't write the memory type, and you don't write a zero at the end. */
260
261 /* Set DSP format bits for 24 bit mode */
262 set_dsp_register(chip, CHI32_CONTROL_REG,
263 get_dsp_register(chip, CHI32_CONTROL_REG) | 0x900);
264
265 code = (u16 *)fw->data;
266
267 /* Skip the header section; the first word in the array is the size
268 of the first section, so the first real section of code is pointed
269 to by Code[0]. */
270 index = code[0];
271
272 /* Skip the section size, LRS block type, and DSP memory type */
273 index += 3;
274
275 /* Get the number of DSP words to write */
276 words = code[index++];
277
278 /* Get the DSP address for this block; 24 bits, so build from two words */
279 address = ((u32)code[index] << 16) + code[index + 1];
280 index += 2;
281
282 /* Write the count to the DSP */
283 if (write_dsp(chip, words)) {
284 DE_INIT(("install_resident_loader: Failed to write word count!\n"));
285 goto irl_error;
286 }
287 /* Write the DSP address */
288 if (write_dsp(chip, address)) {
289 DE_INIT(("install_resident_loader: Failed to write DSP address!\n"));
290 goto irl_error;
291 }
292 /* Write out this block of code to the DSP */
293 for (i = 0; i < words; i++) {
294 u32 data;
295
296 data = ((u32)code[index] << 16) + code[index + 1];
297 if (write_dsp(chip, data)) {
298 DE_INIT(("install_resident_loader: Failed to write DSP code\n"));
299 goto irl_error;
300 }
301 index += 2;
302 }
303
304 /* Wait for flag 5 to come up */
305 for (i = 0; i < 200; i++) { /* Timeout is 50us * 200 = 10ms */
306 udelay(50);
307 status = get_dsp_register(chip, CHI32_STATUS_REG);
308 if (status & CHI32_STATUS_REG_HF5)
309 break;
310 }
311
312 if (i == 200) {
313 DE_INIT(("Resident loader failed to set HF5\n"));
314 goto irl_error;
315 }
316
317 DE_INIT(("Resident loader successfully installed\n"));
318 free_firmware(fw);
319 return 0;
320
321irl_error:
322 free_firmware(fw);
323 return -EIO;
324}
325
326#endif /* DSP_56361 */
327
328
329static int load_dsp(struct echoaudio *chip, u16 *code)
330{
331 u32 address, data;
332 int index, words, i;
333
334 if (chip->dsp_code == code) {
335 DE_INIT(("DSP is already loaded!\n"));
336 return 0;
337 }
338 chip->bad_board = TRUE; /* Set TRUE until DSP loaded */
339 chip->dsp_code = NULL; /* Current DSP code not loaded */
340 chip->asic_loaded = FALSE; /* Loading the DSP code will reset the ASIC */
341
342 DE_INIT(("load_dsp: Set bad_board to TRUE\n"));
343
344 /* If this board requires a resident loader, install it. */
345#ifdef DSP_56361
346 if ((i = install_resident_loader(chip)) < 0)
347 return i;
348#endif
349
350 /* Send software reset command */
351 if (send_vector(chip, DSP_VC_RESET) < 0) {
352 DE_INIT(("LoadDsp: send_vector DSP_VC_RESET failed, Critical Failure\n"));
353 return -EIO;
354 }
355 /* Delay 10us */
356 udelay(10);
357
358 /* Wait 10ms for HF3 to indicate that software reset is complete */
359 for (i = 0; i < 1000; i++) { /* Timeout is 10us * 1000 = 10ms */
360 if (get_dsp_register(chip, CHI32_STATUS_REG) &
361 CHI32_STATUS_REG_HF3)
362 break;
363 udelay(10);
364 }
365
366 if (i == 1000) {
367 DE_INIT(("load_dsp: Timeout waiting for CHI32_STATUS_REG_HF3\n"));
368 return -EIO;
369 }
370
371 /* Set DSP format bits for 24 bit mode now that soft reset is done */
372 set_dsp_register(chip, CHI32_CONTROL_REG,
373 get_dsp_register(chip, CHI32_CONTROL_REG) | 0x900);
374
375 /* Main loader loop */
376
377 index = code[0];
378 for (;;) {
379 int block_type, mem_type;
380
381 /* Total Block Size */
382 index++;
383
384 /* Block Type */
385 block_type = code[index];
386 if (block_type == 4) /* We're finished */
387 break;
388
389 index++;
390
391 /* Memory Type P=0,X=1,Y=2 */
392 mem_type = code[index++];
393
394 /* Block Code Size */
395 words = code[index++];
396 if (words == 0) /* We're finished */
397 break;
398
399 /* Start Address */
400 address = ((u32)code[index] << 16) + code[index + 1];
401 index += 2;
402
403 if (write_dsp(chip, words) < 0) {
404 DE_INIT(("load_dsp: failed to write number of DSP words\n"));
405 return -EIO;
406 }
407 if (write_dsp(chip, address) < 0) {
408 DE_INIT(("load_dsp: failed to write DSP address\n"));
409 return -EIO;
410 }
411 if (write_dsp(chip, mem_type) < 0) {
412 DE_INIT(("load_dsp: failed to write DSP memory type\n"));
413 return -EIO;
414 }
415 /* Code */
416 for (i = 0; i < words; i++, index+=2) {
417 data = ((u32)code[index] << 16) + code[index + 1];
418 if (write_dsp(chip, data) < 0) {
419 DE_INIT(("load_dsp: failed to write DSP data\n"));
420 return -EIO;
421 }
422 }
423 }
424
425 if (write_dsp(chip, 0) < 0) { /* We're done!!! */
426 DE_INIT(("load_dsp: Failed to write final zero\n"));
427 return -EIO;
428 }
429 udelay(10);
430
431 for (i = 0; i < 5000; i++) { /* Timeout is 100us * 5000 = 500ms */
432 /* Wait for flag 4 - indicates that the DSP loaded OK */
433 if (get_dsp_register(chip, CHI32_STATUS_REG) &
434 CHI32_STATUS_REG_HF4) {
435 set_dsp_register(chip, CHI32_CONTROL_REG,
436 get_dsp_register(chip, CHI32_CONTROL_REG) & ~0x1b00);
437
438 if (write_dsp(chip, DSP_FNC_SET_COMMPAGE_ADDR) < 0) {
439 DE_INIT(("load_dsp: Failed to write DSP_FNC_SET_COMMPAGE_ADDR\n"));
440 return -EIO;
441 }
442
443 if (write_dsp(chip, chip->comm_page_phys) < 0) {
444 DE_INIT(("load_dsp: Failed to write comm page address\n"));
445 return -EIO;
446 }
447
448 /* Get the serial number via slave mode.
449 This is triggered by the SET_COMMPAGE_ADDR command.
450 We don't actually use the serial number but we have to
451 get it as part of the DSP init voodoo. */
452 if (read_sn(chip) < 0) {
453 DE_INIT(("load_dsp: Failed to read serial number\n"));
454 return -EIO;
455 }
456
457 chip->dsp_code = code; /* Show which DSP code loaded */
458 chip->bad_board = FALSE; /* DSP OK */
459 DE_INIT(("load_dsp: OK!\n"));
460 return 0;
461 }
462 udelay(100);
463 }
464
465 DE_INIT(("load_dsp: DSP load timed out waiting for HF4\n"));
466 return -EIO;
467}
468
469
470
471/* load_firmware takes care of loading the DSP and any ASIC code. */
472static int load_firmware(struct echoaudio *chip)
473{
474 const struct firmware *fw;
475 int box_type, err;
476
477 snd_assert(chip->dsp_code_to_load && chip->comm_page, return -EPERM);
478
479 /* See if the ASIC is present and working - only if the DSP is already loaded */
480 if (chip->dsp_code) {
481 if ((box_type = check_asic_status(chip)) >= 0)
482 return box_type;
483 /* ASIC check failed; force the DSP to reload */
484 chip->dsp_code = NULL;
485 }
486
487 if ((err = get_firmware(&fw, chip->dsp_code_to_load, chip)) < 0)
488 return err;
489 err = load_dsp(chip, (u16 *)fw->data);
490 free_firmware(fw);
491 if (err < 0)
492 return err;
493
494 if ((box_type = load_asic(chip)) < 0)
495 return box_type; /* error */
496
497 if ((err = restore_dsp_rettings(chip)) < 0)
498 return err;
499
500 return box_type;
501}
502
503
504
505/****************************************************************************
506 Mixer functions
507 ****************************************************************************/
508
509#if defined(ECHOCARD_HAS_INPUT_NOMINAL_LEVEL) || \
510 defined(ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL)
511
512/* Set the nominal level for an input or output bus (true = -10dBV, false = +4dBu) */
513static int set_nominal_level(struct echoaudio *chip, u16 index, char consumer)
514{
515 snd_assert(index < num_busses_out(chip) + num_busses_in(chip),
516 return -EINVAL);
517
518 /* Wait for the handshake (OK even if ASIC is not loaded) */
519 if (wait_handshake(chip))
520 return -EIO;
521
522 chip->nominal_level[index] = consumer;
523
524 if (consumer)
525 chip->comm_page->nominal_level_mask |= cpu_to_le32(1 << index);
526 else
527 chip->comm_page->nominal_level_mask &= ~cpu_to_le32(1 << index);
528
529 return 0;
530}
531
532#endif /* ECHOCARD_HAS_*_NOMINAL_LEVEL */
533
534
535
536/* Set the gain for a single physical output channel (dB). */
537static int set_output_gain(struct echoaudio *chip, u16 channel, s8 gain)
538{
539 snd_assert(channel < num_busses_out(chip), return -EINVAL);
540
541 if (wait_handshake(chip))
542 return -EIO;
543
544 /* Save the new value */
545 chip->output_gain[channel] = gain;
546 chip->comm_page->line_out_level[channel] = gain;
547 return 0;
548}
549
550
551
552#ifdef ECHOCARD_HAS_MONITOR
553/* Set the monitor level from an input bus to an output bus. */
554static int set_monitor_gain(struct echoaudio *chip, u16 output, u16 input,
555 s8 gain)
556{
557 snd_assert(output < num_busses_out(chip) &&
558 input < num_busses_in(chip), return -EINVAL);
559
560 if (wait_handshake(chip))
561 return -EIO;
562
563 chip->monitor_gain[output][input] = gain;
564 chip->comm_page->monitors[monitor_index(chip, output, input)] = gain;
565 return 0;
566}
567#endif /* ECHOCARD_HAS_MONITOR */
568
569
570/* Tell the DSP to read and update output, nominal & monitor levels in comm page. */
571static int update_output_line_level(struct echoaudio *chip)
572{
573 if (wait_handshake(chip))
574 return -EIO;
575 clear_handshake(chip);
576 return send_vector(chip, DSP_VC_UPDATE_OUTVOL);
577}
578
579
580
581/* Tell the DSP to read and update input levels in comm page */
582static int update_input_line_level(struct echoaudio *chip)
583{
584 if (wait_handshake(chip))
585 return -EIO;
586 clear_handshake(chip);
587 return send_vector(chip, DSP_VC_UPDATE_INGAIN);
588}
589
590
591
592/* set_meters_on turns the meters on or off. If meters are turned on, the DSP
593will write the meter and clock detect values to the comm page at about 30Hz */
594static void set_meters_on(struct echoaudio *chip, char on)
595{
596 if (on && !chip->meters_enabled) {
597 send_vector(chip, DSP_VC_METERS_ON);
598 chip->meters_enabled = 1;
599 } else if (!on && chip->meters_enabled) {
600 send_vector(chip, DSP_VC_METERS_OFF);
601 chip->meters_enabled = 0;
602 memset((s8 *)chip->comm_page->vu_meter, ECHOGAIN_MUTED,
603 DSP_MAXPIPES);
604 memset((s8 *)chip->comm_page->peak_meter, ECHOGAIN_MUTED,
605 DSP_MAXPIPES);
606 }
607}
608
609
610
611/* Fill out an the given array using the current values in the comm page.
612Meters are written in the comm page by the DSP in this order:
613 Output busses
614 Input busses
615 Output pipes (vmixer cards only)
616
617This function assumes there are no more than 16 in/out busses or pipes
618Meters is an array [3][16][2] of long. */
619static void get_audio_meters(struct echoaudio *chip, long *meters)
620{
621 int i, m, n;
622
623 m = 0;
624 n = 0;
625 for (i = 0; i < num_busses_out(chip); i++, m++) {
626 meters[n++] = chip->comm_page->vu_meter[m];
627 meters[n++] = chip->comm_page->peak_meter[m];
628 }
629 for (; n < 32; n++)
630 meters[n] = 0;
631
632#ifdef ECHOCARD_ECHO3G
633 m = E3G_MAX_OUTPUTS; /* Skip unused meters */
634#endif
635
636 for (i = 0; i < num_busses_in(chip); i++, m++) {
637 meters[n++] = chip->comm_page->vu_meter[m];
638 meters[n++] = chip->comm_page->peak_meter[m];
639 }
640 for (; n < 64; n++)
641 meters[n] = 0;
642
643#ifdef ECHOCARD_HAS_VMIXER
644 for (i = 0; i < num_pipes_out(chip); i++, m++) {
645 meters[n++] = chip->comm_page->vu_meter[m];
646 meters[n++] = chip->comm_page->peak_meter[m];
647 }
648#endif
649 for (; n < 96; n++)
650 meters[n] = 0;
651}
652
653
654
655static int restore_dsp_rettings(struct echoaudio *chip)
656{
657 int err;
658 DE_INIT(("restore_dsp_settings\n"));
659
660 if ((err = check_asic_status(chip)) < 0)
661 return err;
662
663 /* @ Gina20/Darla20 only. Should be harmless for other cards. */
664 chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF;
665 chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF;
666 chip->comm_page->handshake = 0xffffffff;
667
668 if ((err = set_sample_rate(chip, chip->sample_rate)) < 0)
669 return err;
670
671 if (chip->meters_enabled)
672 if (send_vector(chip, DSP_VC_METERS_ON) < 0)
673 return -EIO;
674
675#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK
676 if (set_input_clock(chip, chip->input_clock) < 0)
677 return -EIO;
678#endif
679
680#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH
681 if (set_output_clock(chip, chip->output_clock) < 0)
682 return -EIO;
683#endif
684
685 if (update_output_line_level(chip) < 0)
686 return -EIO;
687
688 if (update_input_line_level(chip) < 0)
689 return -EIO;
690
691#ifdef ECHOCARD_HAS_VMIXER
692 if (update_vmixer_level(chip) < 0)
693 return -EIO;
694#endif
695
696 if (wait_handshake(chip) < 0)
697 return -EIO;
698 clear_handshake(chip);
699
700 DE_INIT(("restore_dsp_rettings done\n"));
701 return send_vector(chip, DSP_VC_UPDATE_FLAGS);
702}
703
704
705
706/****************************************************************************
707 Transport functions
708 ****************************************************************************/
709
710/* set_audio_format() sets the format of the audio data in host memory for
711this pipe. Note that _MS_ (mono-to-stereo) playback modes are not used by ALSA
712but they are here because they are just mono while capturing */
713static void set_audio_format(struct echoaudio *chip, u16 pipe_index,
714 const struct audioformat *format)
715{
716 u16 dsp_format;
717
718 dsp_format = DSP_AUDIOFORM_SS_16LE;
719
720 /* Look for super-interleave (no big-endian and 8 bits) */
721 if (format->interleave > 2) {
722 switch (format->bits_per_sample) {
723 case 16:
724 dsp_format = DSP_AUDIOFORM_SUPER_INTERLEAVE_16LE;
725 break;
726 case 24:
727 dsp_format = DSP_AUDIOFORM_SUPER_INTERLEAVE_24LE;
728 break;
729 case 32:
730 dsp_format = DSP_AUDIOFORM_SUPER_INTERLEAVE_32LE;
731 break;
732 }
733 dsp_format |= format->interleave;
734 } else if (format->data_are_bigendian) {
735 /* For big-endian data, only 32 bit samples are supported */
736 switch (format->interleave) {
737 case 1:
738 dsp_format = DSP_AUDIOFORM_MM_32BE;
739 break;
740#ifdef ECHOCARD_HAS_STEREO_BIG_ENDIAN32
741 case 2:
742 dsp_format = DSP_AUDIOFORM_SS_32BE;
743 break;
744#endif
745 }
746 } else if (format->interleave == 1 &&
747 format->bits_per_sample == 32 && !format->mono_to_stereo) {
748 /* 32 bit little-endian mono->mono case */
749 dsp_format = DSP_AUDIOFORM_MM_32LE;
750 } else {
751 /* Handle the other little-endian formats */
752 switch (format->bits_per_sample) {
753 case 8:
754 if (format->interleave == 2)
755 dsp_format = DSP_AUDIOFORM_SS_8;
756 else
757 dsp_format = DSP_AUDIOFORM_MS_8;
758 break;
759 default:
760 case 16:
761 if (format->interleave == 2)
762 dsp_format = DSP_AUDIOFORM_SS_16LE;
763 else
764 dsp_format = DSP_AUDIOFORM_MS_16LE;
765 break;
766 case 24:
767 if (format->interleave == 2)
768 dsp_format = DSP_AUDIOFORM_SS_24LE;
769 else
770 dsp_format = DSP_AUDIOFORM_MS_24LE;
771 break;
772 case 32:
773 if (format->interleave == 2)
774 dsp_format = DSP_AUDIOFORM_SS_32LE;
775 else
776 dsp_format = DSP_AUDIOFORM_MS_32LE;
777 break;
778 }
779 }
780 DE_ACT(("set_audio_format[%d] = %x\n", pipe_index, dsp_format));
781 chip->comm_page->audio_format[pipe_index] = cpu_to_le16(dsp_format);
782}
783
784
785
786/* start_transport starts transport for a set of pipes.
787The bits 1 in channel_mask specify what pipes to start. Only the bit of the
788first channel must be set, regardless its interleave.
789Same thing for pause_ and stop_ -trasport below. */
790static int start_transport(struct echoaudio *chip, u32 channel_mask,
791 u32 cyclic_mask)
792{
793 DE_ACT(("start_transport %x\n", channel_mask));
794
795 if (wait_handshake(chip))
796 return -EIO;
797
798 chip->comm_page->cmd_start |= cpu_to_le32(channel_mask);
799
800 if (chip->comm_page->cmd_start) {
801 clear_handshake(chip);
802 send_vector(chip, DSP_VC_START_TRANSFER);
803 if (wait_handshake(chip))
804 return -EIO;
805 /* Keep track of which pipes are transporting */
806 chip->active_mask |= channel_mask;
807 chip->comm_page->cmd_start = 0;
808 return 0;
809 }
810
811 DE_ACT(("start_transport: No pipes to start!\n"));
812 return -EINVAL;
813}
814
815
816
817static int pause_transport(struct echoaudio *chip, u32 channel_mask)
818{
819 DE_ACT(("pause_transport %x\n", channel_mask));
820
821 if (wait_handshake(chip))
822 return -EIO;
823
824 chip->comm_page->cmd_stop |= cpu_to_le32(channel_mask);
825 chip->comm_page->cmd_reset = 0;
826 if (chip->comm_page->cmd_stop) {
827 clear_handshake(chip);
828 send_vector(chip, DSP_VC_STOP_TRANSFER);
829 if (wait_handshake(chip))
830 return -EIO;
831 /* Keep track of which pipes are transporting */
832 chip->active_mask &= ~channel_mask;
833 chip->comm_page->cmd_stop = 0;
834 chip->comm_page->cmd_reset = 0;
835 return 0;
836 }
837
838 DE_ACT(("pause_transport: No pipes to stop!\n"));
839 return 0;
840}
841
842
843
844static int stop_transport(struct echoaudio *chip, u32 channel_mask)
845{
846 DE_ACT(("stop_transport %x\n", channel_mask));
847
848 if (wait_handshake(chip))
849 return -EIO;
850
851 chip->comm_page->cmd_stop |= cpu_to_le32(channel_mask);
852 chip->comm_page->cmd_reset |= cpu_to_le32(channel_mask);
853 if (chip->comm_page->cmd_reset) {
854 clear_handshake(chip);
855 send_vector(chip, DSP_VC_STOP_TRANSFER);
856 if (wait_handshake(chip))
857 return -EIO;
858 /* Keep track of which pipes are transporting */
859 chip->active_mask &= ~channel_mask;
860 chip->comm_page->cmd_stop = 0;
861 chip->comm_page->cmd_reset = 0;
862 return 0;
863 }
864
865 DE_ACT(("stop_transport: No pipes to stop!\n"));
866 return 0;
867}
868
869
870
871static inline int is_pipe_allocated(struct echoaudio *chip, u16 pipe_index)
872{
873 return (chip->pipe_alloc_mask & (1 << pipe_index));
874}
875
876
877
878/* Stops everything and turns off the DSP. All pipes should be already
879stopped and unallocated. */
880static int rest_in_peace(struct echoaudio *chip)
881{
882 DE_ACT(("rest_in_peace() open=%x\n", chip->pipe_alloc_mask));
883
884 /* Stops all active pipes (just to be sure) */
885 stop_transport(chip, chip->active_mask);
886
887 set_meters_on(chip, FALSE);
888
889#ifdef ECHOCARD_HAS_MIDI
890 enable_midi_input(chip, FALSE);
891#endif
892
893 /* Go to sleep */
894 if (chip->dsp_code) {
895 /* Make load_firmware do a complete reload */
896 chip->dsp_code = NULL;
897 /* Put the DSP to sleep */
898 return send_vector(chip, DSP_VC_GO_COMATOSE);
899 }
900 return 0;
901}
902
903
904
905/* Fills the comm page with default values */
906static int init_dsp_comm_page(struct echoaudio *chip)
907{
908 /* Check if the compiler added extra padding inside the structure */
909 if (offsetof(struct comm_page, midi_output) != 0xbe0) {
910 DE_INIT(("init_dsp_comm_page() - Invalid struct comm_page structure\n"));
911 return -EPERM;
912 }
913
914 /* Init all the basic stuff */
915 chip->card_name = ECHOCARD_NAME;
916 chip->bad_board = TRUE; /* Set TRUE until DSP loaded */
917 chip->dsp_code = NULL; /* Current DSP code not loaded */
918 chip->digital_mode = DIGITAL_MODE_NONE;
919 chip->input_clock = ECHO_CLOCK_INTERNAL;
920 chip->output_clock = ECHO_CLOCK_WORD;
921 chip->asic_loaded = FALSE;
922 memset(chip->comm_page, 0, sizeof(struct comm_page));
923
924 /* Init the comm page */
925 chip->comm_page->comm_size =
926 __constant_cpu_to_le32(sizeof(struct comm_page));
927 chip->comm_page->handshake = 0xffffffff;
928 chip->comm_page->midi_out_free_count =
929 __constant_cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE);
930 chip->comm_page->sample_rate = __constant_cpu_to_le32(44100);
931 chip->sample_rate = 44100;
932
933 /* Set line levels so we don't blast any inputs on startup */
934 memset(chip->comm_page->monitors, ECHOGAIN_MUTED, MONITOR_ARRAY_SIZE);
935 memset(chip->comm_page->vmixer, ECHOGAIN_MUTED, VMIXER_ARRAY_SIZE);
936
937 return 0;
938}
939
940
941
942/* This function initializes the several volume controls for busses and pipes.
943This MUST be called after the DSP is up and running ! */
944static int init_line_levels(struct echoaudio *chip)
945{
946 int st, i, o;
947
948 DE_INIT(("init_line_levels\n"));
949
950 /* Mute output busses */
951 for (i = 0; i < num_busses_out(chip); i++)
952 if ((st = set_output_gain(chip, i, ECHOGAIN_MUTED)))
953 return st;
954 if ((st = update_output_line_level(chip)))
955 return st;
956
957#ifdef ECHOCARD_HAS_VMIXER
958 /* Mute the Vmixer */
959 for (i = 0; i < num_pipes_out(chip); i++)
960 for (o = 0; o < num_busses_out(chip); o++)
961 if ((st = set_vmixer_gain(chip, o, i, ECHOGAIN_MUTED)))
962 return st;
963 if ((st = update_vmixer_level(chip)))
964 return st;
965#endif /* ECHOCARD_HAS_VMIXER */
966
967#ifdef ECHOCARD_HAS_MONITOR
968 /* Mute the monitor mixer */
969 for (o = 0; o < num_busses_out(chip); o++)
970 for (i = 0; i < num_busses_in(chip); i++)
971 if ((st = set_monitor_gain(chip, o, i, ECHOGAIN_MUTED)))
972 return st;
973 if ((st = update_output_line_level(chip)))
974 return st;
975#endif /* ECHOCARD_HAS_MONITOR */
976
977#ifdef ECHOCARD_HAS_INPUT_GAIN
978 for (i = 0; i < num_busses_in(chip); i++)
979 if ((st = set_input_gain(chip, i, ECHOGAIN_MUTED)))
980 return st;
981 if ((st = update_input_line_level(chip)))
982 return st;
983#endif /* ECHOCARD_HAS_INPUT_GAIN */
984
985 return 0;
986}
987
988
989
990/* This is low level part of the interrupt handler.
991It returns -1 if the IRQ is not ours, or N>=0 if it is, where N is the number
992of midi data in the input queue. */
993static int service_irq(struct echoaudio *chip)
994{
995 int st;
996
997 /* Read the DSP status register and see if this DSP generated this interrupt */
998 if (get_dsp_register(chip, CHI32_STATUS_REG) & CHI32_STATUS_IRQ) {
999 st = 0;
1000#ifdef ECHOCARD_HAS_MIDI
1001 /* Get and parse midi data if present */
1002 if (chip->comm_page->midi_input[0]) /* The count is at index 0 */
1003 st = midi_service_irq(chip); /* Returns how many midi bytes we received */
1004#endif
1005 /* Clear the hardware interrupt */
1006 chip->comm_page->midi_input[0] = 0;
1007 send_vector(chip, DSP_VC_ACK_INT);
1008 return st;
1009 }
1010 return -1;
1011}
1012
1013
1014
1015
1016/******************************************************************************
1017 Functions for opening and closing pipes
1018 ******************************************************************************/
1019
1020/* allocate_pipes is used to reserve audio pipes for your exclusive use.
1021The call will fail if some pipes are already allocated. */
1022static int allocate_pipes(struct echoaudio *chip, struct audiopipe *pipe,
1023 int pipe_index, int interleave)
1024{
1025 int i;
1026 u32 channel_mask;
1027 char is_cyclic;
1028
1029 DE_ACT(("allocate_pipes: ch=%d int=%d\n", pipe_index, interleave));
1030
1031 if (chip->bad_board)
1032 return -EIO;
1033
1034 is_cyclic = 1; /* This driver uses cyclic buffers only */
1035
1036 for (channel_mask = i = 0; i < interleave; i++)
1037 channel_mask |= 1 << (pipe_index + i);
1038 if (chip->pipe_alloc_mask & channel_mask) {
1039 DE_ACT(("allocate_pipes: channel already open\n"));
1040 return -EAGAIN;
1041 }
1042
1043 chip->comm_page->position[pipe_index] = 0;
1044 chip->pipe_alloc_mask |= channel_mask;
1045 if (is_cyclic)
1046 chip->pipe_cyclic_mask |= channel_mask;
1047 pipe->index = pipe_index;
1048 pipe->interleave = interleave;
1049 pipe->state = PIPE_STATE_STOPPED;
1050
1051 /* The counter register is where the DSP writes the 32 bit DMA
1052 position for a pipe. The DSP is constantly updating this value as
1053 it moves data. The DMA counter is in units of bytes, not samples. */
1054 pipe->dma_counter = &chip->comm_page->position[pipe_index];
1055 *pipe->dma_counter = 0;
1056 DE_ACT(("allocate_pipes: ok\n"));
1057 return pipe_index;
1058}
1059
1060
1061
1062static int free_pipes(struct echoaudio *chip, struct audiopipe *pipe)
1063{
1064 u32 channel_mask;
1065 int i;
1066
1067 DE_ACT(("free_pipes: Pipe %d\n", pipe->index));
1068 snd_assert(is_pipe_allocated(chip, pipe->index), return -EINVAL);
1069 snd_assert(pipe->state == PIPE_STATE_STOPPED, return -EINVAL);
1070
1071 for (channel_mask = i = 0; i < pipe->interleave; i++)
1072 channel_mask |= 1 << (pipe->index + i);
1073
1074 chip->pipe_alloc_mask &= ~channel_mask;
1075 chip->pipe_cyclic_mask &= ~channel_mask;
1076 return 0;
1077}
1078
1079
1080
1081/******************************************************************************
1082 Functions for managing the scatter-gather list
1083******************************************************************************/
1084
1085static int sglist_init(struct echoaudio *chip, struct audiopipe *pipe)
1086{
1087 pipe->sglist_head = 0;
1088 memset(pipe->sgpage.area, 0, PAGE_SIZE);
1089 chip->comm_page->sglist_addr[pipe->index].addr =
1090 cpu_to_le32(pipe->sgpage.addr);
1091 return 0;
1092}
1093
1094
1095
1096static int sglist_add_mapping(struct echoaudio *chip, struct audiopipe *pipe,
1097 dma_addr_t address, size_t length)
1098{
1099 int head = pipe->sglist_head;
1100 struct sg_entry *list = (struct sg_entry *)pipe->sgpage.area;
1101
1102 if (head < MAX_SGLIST_ENTRIES - 1) {
1103 list[head].addr = cpu_to_le32(address);
1104 list[head].size = cpu_to_le32(length);
1105 pipe->sglist_head++;
1106 } else {
1107 DE_ACT(("SGlist: too many fragments\n"));
1108 return -ENOMEM;
1109 }
1110 return 0;
1111}
1112
1113
1114
1115static inline int sglist_add_irq(struct echoaudio *chip, struct audiopipe *pipe)
1116{
1117 return sglist_add_mapping(chip, pipe, 0, 0);
1118}
1119
1120
1121
1122static inline int sglist_wrap(struct echoaudio *chip, struct audiopipe *pipe)
1123{
1124 return sglist_add_mapping(chip, pipe, pipe->sgpage.addr, 0);
1125}
diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h
new file mode 100644
index 000000000000..e55ee00991ac
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_dsp.h
@@ -0,0 +1,694 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31#ifndef _ECHO_DSP_
32#define _ECHO_DSP_
33
34
35/**** Echogals: Darla20, Gina20, Layla20, and Darla24 ****/
36#if defined(ECHOGALS_FAMILY)
37
38#define NUM_ASIC_TESTS 5
39#define READ_DSP_TIMEOUT 1000000L /* one second */
40
41/**** Echo24: Gina24, Layla24, Mona, Mia, Mia-midi ****/
42#elif defined(ECHO24_FAMILY)
43
44#define DSP_56361 /* Some Echo24 cards use the 56361 DSP */
45#define READ_DSP_TIMEOUT 100000L /* .1 second */
46
47/**** 3G: Gina3G, Layla3G ****/
48#elif defined(ECHO3G_FAMILY)
49
50#define DSP_56361
51#define READ_DSP_TIMEOUT 100000L /* .1 second */
52#define MIN_MTC_1X_RATE 32000
53
54/**** Indigo: Indigo, Indigo IO, Indigo DJ ****/
55#elif defined(INDIGO_FAMILY)
56
57#define DSP_56361
58#define READ_DSP_TIMEOUT 100000L /* .1 second */
59
60#else
61
62#error No family is defined
63
64#endif
65
66
67
68/*
69 *
70 * Max inputs and outputs
71 *
72 */
73
74#define DSP_MAXAUDIOINPUTS 16 /* Max audio input channels */
75#define DSP_MAXAUDIOOUTPUTS 16 /* Max audio output channels */
76#define DSP_MAXPIPES 32 /* Max total pipes (input + output) */
77
78
79/*
80 *
81 * These are the offsets for the memory-mapped DSP registers; the DSP base
82 * address is treated as the start of a u32 array.
83 */
84
85#define CHI32_CONTROL_REG 4
86#define CHI32_STATUS_REG 5
87#define CHI32_VECTOR_REG 6
88#define CHI32_DATA_REG 7
89
90
91/*
92 *
93 * Interesting bits within the DSP registers
94 *
95 */
96
97#define CHI32_VECTOR_BUSY 0x00000001
98#define CHI32_STATUS_REG_HF3 0x00000008
99#define CHI32_STATUS_REG_HF4 0x00000010
100#define CHI32_STATUS_REG_HF5 0x00000020
101#define CHI32_STATUS_HOST_READ_FULL 0x00000004
102#define CHI32_STATUS_HOST_WRITE_EMPTY 0x00000002
103#define CHI32_STATUS_IRQ 0x00000040
104
105
106/*
107 *
108 * DSP commands sent via slave mode; these are sent to the DSP by write_dsp()
109 *
110 */
111
112#define DSP_FNC_SET_COMMPAGE_ADDR 0x02
113#define DSP_FNC_LOAD_LAYLA_ASIC 0xa0
114#define DSP_FNC_LOAD_GINA24_ASIC 0xa0
115#define DSP_FNC_LOAD_MONA_PCI_CARD_ASIC 0xa0
116#define DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC 0xa0
117#define DSP_FNC_LOAD_MONA_EXTERNAL_ASIC 0xa1
118#define DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC 0xa1
119#define DSP_FNC_LOAD_3G_ASIC 0xa0
120
121
122/*
123 *
124 * Defines to handle the MIDI input state engine; these are used to properly
125 * extract MIDI time code bytes and their timestamps from the MIDI input stream.
126 *
127 */
128
129#define MIDI_IN_STATE_NORMAL 0
130#define MIDI_IN_STATE_TS_HIGH 1
131#define MIDI_IN_STATE_TS_LOW 2
132#define MIDI_IN_STATE_F1_DATA 3
133#define MIDI_IN_SKIP_DATA (-1)
134
135
136/*----------------------------------------------------------------------------
137
138Setting the sample rates on Layla24 is somewhat schizophrenic.
139
140For standard rates, it works exactly like Mona and Gina24. That is, for
1418, 11.025, 16, 22.05, 32, 44.1, 48, 88.2, and 96 kHz, you just set the
142appropriate bits in the control register and write the control register.
143
144In order to support MIDI time code sync (and possibly SMPTE LTC sync in
145the future), Layla24 also has "continuous sample rate mode". In this mode,
146Layla24 can generate any sample rate between 25 and 50 kHz inclusive, or
14750 to 100 kHz inclusive for double speed mode.
148
149To use continuous mode:
150
151-Set the clock select bits in the control register to 0xe (see the #define
152 below)
153
154-Set double-speed mode if you want to use sample rates above 50 kHz
155
156-Write the control register as you would normally
157
158-Now, you need to set the frequency register. First, you need to determine the
159 value for the frequency register. This is given by the following formula:
160
161frequency_reg = (LAYLA24_MAGIC_NUMBER / sample_rate) - 2
162
163Note the #define below for the magic number
164
165-Wait for the DSP handshake
166-Write the frequency_reg value to the .SampleRate field of the comm page
167-Send the vector command SET_LAYLA24_FREQUENCY_REG (see vmonkey.h)
168
169Once you have set the control register up for continuous mode, you can just
170write the frequency register to change the sample rate. This could be
171used for MIDI time code sync. For MTC sync, the control register is set for
172continuous mode. The driver then just keeps writing the
173SET_LAYLA24_FREQUENCY_REG command.
174
175-----------------------------------------------------------------------------*/
176
177#define LAYLA24_MAGIC_NUMBER 677376000
178#define LAYLA24_CONTINUOUS_CLOCK 0x000e
179
180
181/*
182 *
183 * DSP vector commands
184 *
185 */
186
187#define DSP_VC_RESET 0x80ff
188
189#ifndef DSP_56361
190
191#define DSP_VC_ACK_INT 0x8073
192#define DSP_VC_SET_VMIXER_GAIN 0x0000 /* Not used, only for compile */
193#define DSP_VC_START_TRANSFER 0x0075 /* Handshke rqd. */
194#define DSP_VC_METERS_ON 0x0079
195#define DSP_VC_METERS_OFF 0x007b
196#define DSP_VC_UPDATE_OUTVOL 0x007d /* Handshke rqd. */
197#define DSP_VC_UPDATE_INGAIN 0x007f /* Handshke rqd. */
198#define DSP_VC_ADD_AUDIO_BUFFER 0x0081 /* Handshke rqd. */
199#define DSP_VC_TEST_ASIC 0x00eb
200#define DSP_VC_UPDATE_CLOCKS 0x00ef /* Handshke rqd. */
201#define DSP_VC_SET_LAYLA_SAMPLE_RATE 0x00f1 /* Handshke rqd. */
202#define DSP_VC_SET_GD_AUDIO_STATE 0x00f1 /* Handshke rqd. */
203#define DSP_VC_WRITE_CONTROL_REG 0x00f1 /* Handshke rqd. */
204#define DSP_VC_MIDI_WRITE 0x00f5 /* Handshke rqd. */
205#define DSP_VC_STOP_TRANSFER 0x00f7 /* Handshke rqd. */
206#define DSP_VC_UPDATE_FLAGS 0x00fd /* Handshke rqd. */
207#define DSP_VC_GO_COMATOSE 0x00f9
208
209#else /* !DSP_56361 */
210
211/* Vector commands for families that use either the 56301 or 56361 */
212#define DSP_VC_ACK_INT 0x80F5
213#define DSP_VC_SET_VMIXER_GAIN 0x00DB /* Handshke rqd. */
214#define DSP_VC_START_TRANSFER 0x00DD /* Handshke rqd. */
215#define DSP_VC_METERS_ON 0x00EF
216#define DSP_VC_METERS_OFF 0x00F1
217#define DSP_VC_UPDATE_OUTVOL 0x00E3 /* Handshke rqd. */
218#define DSP_VC_UPDATE_INGAIN 0x00E5 /* Handshke rqd. */
219#define DSP_VC_ADD_AUDIO_BUFFER 0x00E1 /* Handshke rqd. */
220#define DSP_VC_TEST_ASIC 0x00ED
221#define DSP_VC_UPDATE_CLOCKS 0x00E9 /* Handshke rqd. */
222#define DSP_VC_SET_LAYLA24_FREQUENCY_REG 0x00E9 /* Handshke rqd. */
223#define DSP_VC_SET_LAYLA_SAMPLE_RATE 0x00EB /* Handshke rqd. */
224#define DSP_VC_SET_GD_AUDIO_STATE 0x00EB /* Handshke rqd. */
225#define DSP_VC_WRITE_CONTROL_REG 0x00EB /* Handshke rqd. */
226#define DSP_VC_MIDI_WRITE 0x00E7 /* Handshke rqd. */
227#define DSP_VC_STOP_TRANSFER 0x00DF /* Handshke rqd. */
228#define DSP_VC_UPDATE_FLAGS 0x00FB /* Handshke rqd. */
229#define DSP_VC_GO_COMATOSE 0x00d9
230
231#endif /* !DSP_56361 */
232
233
234/*
235 *
236 * Timeouts
237 *
238 */
239
240#define HANDSHAKE_TIMEOUT 20000 /* send_vector command timeout (20ms) */
241#define VECTOR_BUSY_TIMEOUT 100000 /* 100ms */
242#define MIDI_OUT_DELAY_USEC 2000 /* How long to wait after MIDI fills up */
243
244
245/*
246 *
247 * Flags for .Flags field in the comm page
248 *
249 */
250
251#define DSP_FLAG_MIDI_INPUT 0x0001 /* Enable MIDI input */
252#define DSP_FLAG_SPDIF_NONAUDIO 0x0002 /* Sets the "non-audio" bit
253 * in the S/PDIF out status
254 * bits. Clear this flag for
255 * audio data;
256 * set it for AC3 or WMA or
257 * some such */
258#define DSP_FLAG_PROFESSIONAL_SPDIF 0x0008 /* 1 Professional, 0 Consumer */
259
260
261/*
262 *
263 * Clock detect bits reported by the DSP for Gina20, Layla20, Darla24, and Mia
264 *
265 */
266
267#define GLDM_CLOCK_DETECT_BIT_WORD 0x0002
268#define GLDM_CLOCK_DETECT_BIT_SUPER 0x0004
269#define GLDM_CLOCK_DETECT_BIT_SPDIF 0x0008
270#define GLDM_CLOCK_DETECT_BIT_ESYNC 0x0010
271
272
273/*
274 *
275 * Clock detect bits reported by the DSP for Gina24, Mona, and Layla24
276 *
277 */
278
279#define GML_CLOCK_DETECT_BIT_WORD96 0x0002
280#define GML_CLOCK_DETECT_BIT_WORD48 0x0004
281#define GML_CLOCK_DETECT_BIT_SPDIF48 0x0008
282#define GML_CLOCK_DETECT_BIT_SPDIF96 0x0010
283#define GML_CLOCK_DETECT_BIT_WORD (GML_CLOCK_DETECT_BIT_WORD96 | GML_CLOCK_DETECT_BIT_WORD48)
284#define GML_CLOCK_DETECT_BIT_SPDIF (GML_CLOCK_DETECT_BIT_SPDIF48 | GML_CLOCK_DETECT_BIT_SPDIF96)
285#define GML_CLOCK_DETECT_BIT_ESYNC 0x0020
286#define GML_CLOCK_DETECT_BIT_ADAT 0x0040
287
288
289/*
290 *
291 * Layla clock numbers to send to DSP
292 *
293 */
294
295#define LAYLA20_CLOCK_INTERNAL 0
296#define LAYLA20_CLOCK_SPDIF 1
297#define LAYLA20_CLOCK_WORD 2
298#define LAYLA20_CLOCK_SUPER 3
299
300
301/*
302 *
303 * Gina/Darla clock states
304 *
305 */
306
307#define GD_CLOCK_NOCHANGE 0
308#define GD_CLOCK_44 1
309#define GD_CLOCK_48 2
310#define GD_CLOCK_SPDIFIN 3
311#define GD_CLOCK_UNDEF 0xff
312
313
314/*
315 *
316 * Gina/Darla S/PDIF status bits
317 *
318 */
319
320#define GD_SPDIF_STATUS_NOCHANGE 0
321#define GD_SPDIF_STATUS_44 1
322#define GD_SPDIF_STATUS_48 2
323#define GD_SPDIF_STATUS_UNDEF 0xff
324
325
326/*
327 *
328 * Layla20 output clocks
329 *
330 */
331
332#define LAYLA20_OUTPUT_CLOCK_SUPER 0
333#define LAYLA20_OUTPUT_CLOCK_WORD 1
334
335
336/****************************************************************************
337
338 Magic constants for the Darla24 hardware
339
340 ****************************************************************************/
341
342#define GD24_96000 0x0
343#define GD24_48000 0x1
344#define GD24_44100 0x2
345#define GD24_32000 0x3
346#define GD24_22050 0x4
347#define GD24_16000 0x5
348#define GD24_11025 0x6
349#define GD24_8000 0x7
350#define GD24_88200 0x8
351#define GD24_EXT_SYNC 0x9
352
353
354/*
355 *
356 * Return values from the DSP when ASIC is loaded
357 *
358 */
359
360#define ASIC_ALREADY_LOADED 0x1
361#define ASIC_NOT_LOADED 0x0
362
363
364/*
365 *
366 * DSP Audio formats
367 *
368 * These are the audio formats that the DSP can transfer
369 * via input and output pipes. LE means little-endian,
370 * BE means big-endian.
371 *
372 * DSP_AUDIOFORM_MS_8
373 *
374 * 8-bit mono unsigned samples. For playback,
375 * mono data is duplicated out the left and right channels
376 * of the output bus. The "MS" part of the name
377 * means mono->stereo.
378 *
379 * DSP_AUDIOFORM_MS_16LE
380 *
381 * 16-bit signed little-endian mono samples. Playback works
382 * like the previous code.
383 *
384 * DSP_AUDIOFORM_MS_24LE
385 *
386 * 24-bit signed little-endian mono samples. Data is packed
387 * three bytes per sample; if you had two samples 0x112233 and 0x445566
388 * they would be stored in memory like this: 33 22 11 66 55 44.
389 *
390 * DSP_AUDIOFORM_MS_32LE
391 *
392 * 24-bit signed little-endian mono samples in a 32-bit
393 * container. In other words, each sample is a 32-bit signed
394 * integer, where the actual audio data is left-justified
395 * in the 32 bits and only the 24 most significant bits are valid.
396 *
397 * DSP_AUDIOFORM_SS_8
398 * DSP_AUDIOFORM_SS_16LE
399 * DSP_AUDIOFORM_SS_24LE
400 * DSP_AUDIOFORM_SS_32LE
401 *
402 * Like the previous ones, except now with stereo interleaved
403 * data. "SS" means stereo->stereo.
404 *
405 * DSP_AUDIOFORM_MM_32LE
406 *
407 * Similar to DSP_AUDIOFORM_MS_32LE, except that the mono
408 * data is not duplicated out both the left and right outputs.
409 * This mode is used by the ASIO driver. Here, "MM" means
410 * mono->mono.
411 *
412 * DSP_AUDIOFORM_MM_32BE
413 *
414 * Just like DSP_AUDIOFORM_MM_32LE, but now the data is
415 * in big-endian format.
416 *
417 */
418
419#define DSP_AUDIOFORM_MS_8 0 /* 8 bit mono */
420#define DSP_AUDIOFORM_MS_16LE 1 /* 16 bit mono */
421#define DSP_AUDIOFORM_MS_24LE 2 /* 24 bit mono */
422#define DSP_AUDIOFORM_MS_32LE 3 /* 32 bit mono */
423#define DSP_AUDIOFORM_SS_8 4 /* 8 bit stereo */
424#define DSP_AUDIOFORM_SS_16LE 5 /* 16 bit stereo */
425#define DSP_AUDIOFORM_SS_24LE 6 /* 24 bit stereo */
426#define DSP_AUDIOFORM_SS_32LE 7 /* 32 bit stereo */
427#define DSP_AUDIOFORM_MM_32LE 8 /* 32 bit mono->mono little-endian */
428#define DSP_AUDIOFORM_MM_32BE 9 /* 32 bit mono->mono big-endian */
429#define DSP_AUDIOFORM_SS_32BE 10 /* 32 bit stereo big endian */
430#define DSP_AUDIOFORM_INVALID 0xFF /* Invalid audio format */
431
432
433/*
434 *
435 * Super-interleave is defined as interleaving by 4 or more. Darla20 and Gina20
436 * do not support super interleave.
437 *
438 * 16 bit, 24 bit, and 32 bit little endian samples are supported for super
439 * interleave. The interleave factor must be even. 16 - way interleave is the
440 * current maximum, so you can interleave by 4, 6, 8, 10, 12, 14, and 16.
441 *
442 * The actual format code is derived by taking the define below and or-ing with
443 * the interleave factor. So, 32 bit interleave by 6 is 0x86 and
444 * 16 bit interleave by 16 is (0x40 | 0x10) = 0x50.
445 *
446 */
447
448#define DSP_AUDIOFORM_SUPER_INTERLEAVE_16LE 0x40
449#define DSP_AUDIOFORM_SUPER_INTERLEAVE_24LE 0xc0
450#define DSP_AUDIOFORM_SUPER_INTERLEAVE_32LE 0x80
451
452
453/*
454 *
455 * Gina24, Mona, and Layla24 control register defines
456 *
457 */
458
459#define GML_CONVERTER_ENABLE 0x0010
460#define GML_SPDIF_PRO_MODE 0x0020 /* Professional S/PDIF == 1,
461 consumer == 0 */
462#define GML_SPDIF_SAMPLE_RATE0 0x0040
463#define GML_SPDIF_SAMPLE_RATE1 0x0080
464#define GML_SPDIF_TWO_CHANNEL 0x0100 /* 1 == two channels,
465 0 == one channel */
466#define GML_SPDIF_NOT_AUDIO 0x0200
467#define GML_SPDIF_COPY_PERMIT 0x0400
468#define GML_SPDIF_24_BIT 0x0800 /* 1 == 24 bit, 0 == 20 bit */
469#define GML_ADAT_MODE 0x1000 /* 1 == ADAT mode, 0 == S/PDIF mode */
470#define GML_SPDIF_OPTICAL_MODE 0x2000 /* 1 == optical mode, 0 == RCA mode */
471#define GML_SPDIF_CDROM_MODE 0x3000 /* 1 == CDROM mode,
472 * 0 == RCA or optical mode */
473#define GML_DOUBLE_SPEED_MODE 0x4000 /* 1 == double speed,
474 0 == single speed */
475
476#define GML_DIGITAL_IN_AUTO_MUTE 0x800000
477
478#define GML_96KHZ (0x0 | GML_DOUBLE_SPEED_MODE)
479#define GML_88KHZ (0x1 | GML_DOUBLE_SPEED_MODE)
480#define GML_48KHZ 0x2
481#define GML_44KHZ 0x3
482#define GML_32KHZ 0x4
483#define GML_22KHZ 0x5
484#define GML_16KHZ 0x6
485#define GML_11KHZ 0x7
486#define GML_8KHZ 0x8
487#define GML_SPDIF_CLOCK 0x9
488#define GML_ADAT_CLOCK 0xA
489#define GML_WORD_CLOCK 0xB
490#define GML_ESYNC_CLOCK 0xC
491#define GML_ESYNCx2_CLOCK 0xD
492
493#define GML_CLOCK_CLEAR_MASK 0xffffbff0
494#define GML_SPDIF_RATE_CLEAR_MASK (~(GML_SPDIF_SAMPLE_RATE0|GML_SPDIF_SAMPLE_RATE1))
495#define GML_DIGITAL_MODE_CLEAR_MASK 0xffffcfff
496#define GML_SPDIF_FORMAT_CLEAR_MASK 0xfffff01f
497
498
499/*
500 *
501 * Mia sample rate and clock setting constants
502 *
503 */
504
505#define MIA_32000 0x0040
506#define MIA_44100 0x0042
507#define MIA_48000 0x0041
508#define MIA_88200 0x0142
509#define MIA_96000 0x0141
510
511#define MIA_SPDIF 0x00000044
512#define MIA_SPDIF96 0x00000144
513
514#define MIA_MIDI_REV 1 /* Must be Mia rev 1 for MIDI support */
515
516
517/*
518 *
519 * 3G register bits
520 *
521 */
522
523#define E3G_CONVERTER_ENABLE 0x0010
524#define E3G_SPDIF_PRO_MODE 0x0020 /* Professional S/PDIF == 1,
525 consumer == 0 */
526#define E3G_SPDIF_SAMPLE_RATE0 0x0040
527#define E3G_SPDIF_SAMPLE_RATE1 0x0080
528#define E3G_SPDIF_TWO_CHANNEL 0x0100 /* 1 == two channels,
529 0 == one channel */
530#define E3G_SPDIF_NOT_AUDIO 0x0200
531#define E3G_SPDIF_COPY_PERMIT 0x0400
532#define E3G_SPDIF_24_BIT 0x0800 /* 1 == 24 bit, 0 == 20 bit */
533#define E3G_DOUBLE_SPEED_MODE 0x4000 /* 1 == double speed,
534 0 == single speed */
535#define E3G_PHANTOM_POWER 0x8000 /* 1 == phantom power on,
536 0 == phantom power off */
537
538#define E3G_96KHZ (0x0 | E3G_DOUBLE_SPEED_MODE)
539#define E3G_88KHZ (0x1 | E3G_DOUBLE_SPEED_MODE)
540#define E3G_48KHZ 0x2
541#define E3G_44KHZ 0x3
542#define E3G_32KHZ 0x4
543#define E3G_22KHZ 0x5
544#define E3G_16KHZ 0x6
545#define E3G_11KHZ 0x7
546#define E3G_8KHZ 0x8
547#define E3G_SPDIF_CLOCK 0x9
548#define E3G_ADAT_CLOCK 0xA
549#define E3G_WORD_CLOCK 0xB
550#define E3G_CONTINUOUS_CLOCK 0xE
551
552#define E3G_ADAT_MODE 0x1000
553#define E3G_SPDIF_OPTICAL_MODE 0x2000
554
555#define E3G_CLOCK_CLEAR_MASK 0xbfffbff0
556#define E3G_DIGITAL_MODE_CLEAR_MASK 0xffffcfff
557#define E3G_SPDIF_FORMAT_CLEAR_MASK 0xfffff01f
558
559/* Clock detect bits reported by the DSP */
560#define E3G_CLOCK_DETECT_BIT_WORD96 0x0001
561#define E3G_CLOCK_DETECT_BIT_WORD48 0x0002
562#define E3G_CLOCK_DETECT_BIT_SPDIF48 0x0004
563#define E3G_CLOCK_DETECT_BIT_ADAT 0x0004
564#define E3G_CLOCK_DETECT_BIT_SPDIF96 0x0008
565#define E3G_CLOCK_DETECT_BIT_WORD (E3G_CLOCK_DETECT_BIT_WORD96|E3G_CLOCK_DETECT_BIT_WORD48)
566#define E3G_CLOCK_DETECT_BIT_SPDIF (E3G_CLOCK_DETECT_BIT_SPDIF48|E3G_CLOCK_DETECT_BIT_SPDIF96)
567
568/* Frequency control register */
569#define E3G_MAGIC_NUMBER 677376000
570#define E3G_FREQ_REG_DEFAULT (E3G_MAGIC_NUMBER / 48000 - 2)
571#define E3G_FREQ_REG_MAX 0xffff
572
573/* 3G external box types */
574#define E3G_GINA3G_BOX_TYPE 0x00
575#define E3G_LAYLA3G_BOX_TYPE 0x10
576#define E3G_ASIC_NOT_LOADED 0xffff
577#define E3G_BOX_TYPE_MASK 0xf0
578
579#define EXT_3GBOX_NC 0x01
580#define EXT_3GBOX_NOT_SET 0x02
581
582
583/*
584 *
585 * Gina20 & Layla20 have input gain controls for the analog inputs;
586 * this is the magic number for the hardware that gives you 0 dB at -10.
587 *
588 */
589
590#define GL20_INPUT_GAIN_MAGIC_NUMBER 0xC8
591
592
593/*
594 *
595 * Defines how much time must pass between DSP load attempts
596 *
597 */
598
599#define DSP_LOAD_ATTEMPT_PERIOD 1000000L /* One second */
600
601
602/*
603 *
604 * Size of arrays for the comm page. MAX_PLAY_TAPS and MAX_REC_TAPS are
605 * no longer used, but the sizes must still be right for the DSP to see
606 * the comm page correctly.
607 *
608 */
609
610#define MONITOR_ARRAY_SIZE 0x180
611#define VMIXER_ARRAY_SIZE 0x40
612#define MIDI_OUT_BUFFER_SIZE 32
613#define MIDI_IN_BUFFER_SIZE 256
614#define MAX_PLAY_TAPS 168
615#define MAX_REC_TAPS 192
616#define DSP_MIDI_OUT_FIFO_SIZE 64
617
618
619/* sg_entry is a single entry for the scatter-gather list. The array of struct
620sg_entry struct is read by the DSP, so all values must be little-endian. */
621
622#define MAX_SGLIST_ENTRIES 512
623
624struct sg_entry {
625 u32 addr;
626 u32 size;
627};
628
629
630/****************************************************************************
631
632 The comm page. This structure is read and written by the DSP; the
633 DSP code is a firm believer in the byte offsets written in the comments
634 at the end of each line. This structure should not be changed.
635
636 Any reads from or writes to this structure should be in little-endian format.
637
638 ****************************************************************************/
639
640struct comm_page { /* Base Length*/
641 u32 comm_size; /* size of this object 0x000 4 */
642 u32 flags; /* See Appendix A below 0x004 4 */
643 u32 unused; /* Unused entry 0x008 4 */
644 u32 sample_rate; /* Card sample rate in Hz 0x00c 4 */
645 volatile u32 handshake; /* DSP command handshake 0x010 4 */
646 u32 cmd_start; /* Chs. to start mask 0x014 4 */
647 u32 cmd_stop; /* Chs. to stop mask 0x018 4 */
648 u32 cmd_reset; /* Chs. to reset mask 0x01c 4 */
649 u16 audio_format[DSP_MAXPIPES]; /* Chs. audio format 0x020 32*2 */
650 struct sg_entry sglist_addr[DSP_MAXPIPES];
651 /* Chs. Physical sglist addrs 0x060 32*8 */
652 volatile u32 position[DSP_MAXPIPES];
653 /* Positions for ea. ch. 0x160 32*4 */
654 volatile s8 vu_meter[DSP_MAXPIPES];
655 /* VU meters 0x1e0 32*1 */
656 volatile s8 peak_meter[DSP_MAXPIPES];
657 /* Peak meters 0x200 32*1 */
658 s8 line_out_level[DSP_MAXAUDIOOUTPUTS];
659 /* Output gain 0x220 16*1 */
660 s8 line_in_level[DSP_MAXAUDIOINPUTS];
661 /* Input gain 0x230 16*1 */
662 s8 monitors[MONITOR_ARRAY_SIZE];
663 /* Monitor map 0x240 0x180 */
664 u32 play_coeff[MAX_PLAY_TAPS];
665 /* Gina/Darla play filters - obsolete 0x3c0 168*4 */
666 u32 rec_coeff[MAX_REC_TAPS];
667 /* Gina/Darla record filters - obsolete 0x660 192*4 */
668 volatile u16 midi_input[MIDI_IN_BUFFER_SIZE];
669 /* MIDI input data transfer buffer 0x960 256*2 */
670 u8 gd_clock_state; /* Chg Gina/Darla clock state 0xb60 1 */
671 u8 gd_spdif_status; /* Chg. Gina/Darla S/PDIF state 0xb61 1 */
672 u8 gd_resampler_state; /* Should always be 3 0xb62 1 */
673 u8 filler2; /* 0xb63 1 */
674 u32 nominal_level_mask; /* -10 level enable mask 0xb64 4 */
675 u16 input_clock; /* Chg. Input clock state 0xb68 2 */
676 u16 output_clock; /* Chg. Output clock state 0xb6a 2 */
677 volatile u32 status_clocks;
678 /* Current Input clock state 0xb6c 4 */
679 u32 ext_box_status; /* External box status 0xb70 4 */
680 u32 cmd_add_buffer; /* Pipes to add (obsolete) 0xb74 4 */
681 volatile u32 midi_out_free_count;
682 /* # of bytes free in MIDI output FIFO 0xb78 4 */
683 u32 unused2; /* Cyclic pipes 0xb7c 4 */
684 u32 control_register;
685 /* Mona, Gina24, Layla24, 3G ctrl reg 0xb80 4 */
686 u32 e3g_frq_register; /* 3G frequency register 0xb84 4 */
687 u8 filler[24]; /* filler 0xb88 24*1 */
688 s8 vmixer[VMIXER_ARRAY_SIZE];
689 /* Vmixer levels 0xba0 64*1 */
690 u8 midi_output[MIDI_OUT_BUFFER_SIZE];
691 /* MIDI output data 0xbe0 32*1 */
692};
693
694#endif /* _ECHO_DSP_ */
diff --git a/sound/pci/echoaudio/echoaudio_gml.c b/sound/pci/echoaudio/echoaudio_gml.c
new file mode 100644
index 000000000000..3aa37e76ebab
--- /dev/null
+++ b/sound/pci/echoaudio/echoaudio_gml.c
@@ -0,0 +1,198 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32/* These functions are common for Gina24, Layla24 and Mona cards */
33
34
35/* ASIC status check - some cards have one or two ASICs that need to be
36loaded. Once that load is complete, this function is called to see if
37the load was successful.
38If this load fails, it does not necessarily mean that the hardware is
39defective - the external box may be disconnected or turned off. */
40static int check_asic_status(struct echoaudio *chip)
41{
42 u32 asic_status;
43
44 send_vector(chip, DSP_VC_TEST_ASIC);
45
46 /* The DSP will return a value to indicate whether or not the
47 ASIC is currently loaded */
48 if (read_dsp(chip, &asic_status) < 0) {
49 DE_INIT(("check_asic_status: failed on read_dsp\n"));
50 chip->asic_loaded = FALSE;
51 return -EIO;
52 }
53
54 chip->asic_loaded = (asic_status == ASIC_ALREADY_LOADED);
55 return chip->asic_loaded ? 0 : -EIO;
56}
57
58
59
60/* Most configuration of Gina24, Layla24, or Mona is accomplished by writing
61the control register. write_control_reg sends the new control register
62value to the DSP. */
63static int write_control_reg(struct echoaudio *chip, u32 value, char force)
64{
65 /* Handle the digital input auto-mute */
66 if (chip->digital_in_automute)
67 value |= GML_DIGITAL_IN_AUTO_MUTE;
68 else
69 value &= ~GML_DIGITAL_IN_AUTO_MUTE;
70
71 DE_ACT(("write_control_reg: 0x%x\n", value));
72
73 /* Write the control register */
74 value = cpu_to_le32(value);
75 if (value != chip->comm_page->control_register || force) {
76 if (wait_handshake(chip))
77 return -EIO;
78 chip->comm_page->control_register = value;
79 clear_handshake(chip);
80 return send_vector(chip, DSP_VC_WRITE_CONTROL_REG);
81 }
82 return 0;
83}
84
85
86
87/* Gina24, Layla24, and Mona support digital input auto-mute. If the digital
88input auto-mute is enabled, the DSP will only enable the digital inputs if
89the card is syncing to a valid clock on the ADAT or S/PDIF inputs.
90If the auto-mute is disabled, the digital inputs are enabled regardless of
91what the input clock is set or what is connected. */
92static int set_input_auto_mute(struct echoaudio *chip, int automute)
93{
94 DE_ACT(("set_input_auto_mute %d\n", automute));
95
96 chip->digital_in_automute = automute;
97
98 /* Re-set the input clock to the current value - indirectly causes
99 the auto-mute flag to be sent to the DSP */
100 return set_input_clock(chip, chip->input_clock);
101}
102
103
104
105/* S/PDIF coax / S/PDIF optical / ADAT - switch */
106static int set_digital_mode(struct echoaudio *chip, u8 mode)
107{
108 u8 previous_mode;
109 int err, i, o;
110
111 if (chip->bad_board)
112 return -EIO;
113
114 /* All audio channels must be closed before changing the digital mode */
115 snd_assert(!chip->pipe_alloc_mask, return -EAGAIN);
116
117 snd_assert(chip->digital_modes & (1 << mode), return -EINVAL);
118
119 previous_mode = chip->digital_mode;
120 err = dsp_set_digital_mode(chip, mode);
121
122 /* If we successfully changed the digital mode from or to ADAT,
123 then make sure all output, input and monitor levels are
124 updated by the DSP comm object. */
125 if (err >= 0 && previous_mode != mode &&
126 (previous_mode == DIGITAL_MODE_ADAT || mode == DIGITAL_MODE_ADAT)) {
127 spin_lock_irq(&chip->lock);
128 for (o = 0; o < num_busses_out(chip); o++)
129 for (i = 0; i < num_busses_in(chip); i++)
130 set_monitor_gain(chip, o, i,
131 chip->monitor_gain[o][i]);
132
133#ifdef ECHOCARD_HAS_INPUT_GAIN
134 for (i = 0; i < num_busses_in(chip); i++)
135 set_input_gain(chip, i, chip->input_gain[i]);
136 update_input_line_level(chip);
137#endif
138
139 for (o = 0; o < num_busses_out(chip); o++)
140 set_output_gain(chip, o, chip->output_gain[o]);
141 update_output_line_level(chip);
142 spin_unlock_irq(&chip->lock);
143 }
144
145 return err;
146}
147
148
149
150/* Set the S/PDIF output format */
151static int set_professional_spdif(struct echoaudio *chip, char prof)
152{
153 u32 control_reg;
154 int err;
155
156 /* Clear the current S/PDIF flags */
157 control_reg = le32_to_cpu(chip->comm_page->control_register);
158 control_reg &= GML_SPDIF_FORMAT_CLEAR_MASK;
159
160 /* Set the new S/PDIF flags depending on the mode */
161 control_reg |= GML_SPDIF_TWO_CHANNEL | GML_SPDIF_24_BIT |
162 GML_SPDIF_COPY_PERMIT;
163 if (prof) {
164 /* Professional mode */
165 control_reg |= GML_SPDIF_PRO_MODE;
166
167 switch (chip->sample_rate) {
168 case 32000:
169 control_reg |= GML_SPDIF_SAMPLE_RATE0 |
170 GML_SPDIF_SAMPLE_RATE1;
171 break;
172 case 44100:
173 control_reg |= GML_SPDIF_SAMPLE_RATE0;
174 break;
175 case 48000:
176 control_reg |= GML_SPDIF_SAMPLE_RATE1;
177 break;
178 }
179 } else {
180 /* Consumer mode */
181 switch (chip->sample_rate) {
182 case 32000:
183 control_reg |= GML_SPDIF_SAMPLE_RATE0 |
184 GML_SPDIF_SAMPLE_RATE1;
185 break;
186 case 48000:
187 control_reg |= GML_SPDIF_SAMPLE_RATE1;
188 break;
189 }
190 }
191
192 if ((err = write_control_reg(chip, control_reg, FALSE)))
193 return err;
194 chip->professional_spdif = prof;
195 DE_ACT(("set_professional_spdif to %s\n",
196 prof ? "Professional" : "Consumer"));
197 return 0;
198}
diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c
new file mode 100644
index 000000000000..29d6d12f80ca
--- /dev/null
+++ b/sound/pci/echoaudio/gina20.c
@@ -0,0 +1,103 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define ECHOGALS_FAMILY
20#define ECHOCARD_GINA20
21#define ECHOCARD_NAME "Gina20"
22#define ECHOCARD_HAS_MONITOR
23#define ECHOCARD_HAS_INPUT_GAIN
24#define ECHOCARD_HAS_DIGITAL_IO
25#define ECHOCARD_HAS_EXTERNAL_CLOCK
26#define ECHOCARD_HAS_ADAT FALSE
27
28/* Pipe indexes */
29#define PX_ANALOG_OUT 0 /* 8 */
30#define PX_DIGITAL_OUT 8 /* 2 */
31#define PX_ANALOG_IN 10 /* 2 */
32#define PX_DIGITAL_IN 12 /* 2 */
33#define PX_NUM 14
34
35/* Bus indexes */
36#define BX_ANALOG_OUT 0 /* 8 */
37#define BX_DIGITAL_OUT 8 /* 2 */
38#define BX_ANALOG_IN 10 /* 2 */
39#define BX_DIGITAL_IN 12 /* 2 */
40#define BX_NUM 14
41
42
43#include <sound/driver.h>
44#include <linux/delay.h>
45#include <linux/init.h>
46#include <linux/interrupt.h>
47#include <linux/pci.h>
48#include <linux/slab.h>
49#include <linux/moduleparam.h>
50#include <linux/firmware.h>
51#include <sound/core.h>
52#include <sound/info.h>
53#include <sound/control.h>
54#include <sound/pcm.h>
55#include <sound/pcm_params.h>
56#include <sound/asoundef.h>
57#include <sound/initval.h>
58#include <asm/io.h>
59#include <asm/atomic.h>
60#include "echoaudio.h"
61
62#define FW_GINA20_DSP 0
63
64static const struct firmware card_fw[] = {
65 {0, "gina20_dsp.fw"}
66};
67
68static struct pci_device_id snd_echo_ids[] = {
69 {0x1057, 0x1801, 0xECC0, 0x0020, 0, 0, 0}, /* DSP 56301 Gina20 rev.0 */
70 {0,}
71};
72
73static struct snd_pcm_hardware pcm_hardware_skel = {
74 .info = SNDRV_PCM_INFO_MMAP |
75 SNDRV_PCM_INFO_INTERLEAVED |
76 SNDRV_PCM_INFO_BLOCK_TRANSFER |
77 SNDRV_PCM_INFO_MMAP_VALID |
78 SNDRV_PCM_INFO_PAUSE |
79 SNDRV_PCM_INFO_SYNC_START,
80 .formats = SNDRV_PCM_FMTBIT_U8 |
81 SNDRV_PCM_FMTBIT_S16_LE |
82 SNDRV_PCM_FMTBIT_S24_3LE |
83 SNDRV_PCM_FMTBIT_S32_LE |
84 SNDRV_PCM_FMTBIT_S32_BE,
85 .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
86 .rate_min = 44100,
87 .rate_max = 48000,
88 .channels_min = 1,
89 .channels_max = 2,
90 .buffer_bytes_max = 262144,
91 .period_bytes_min = 32,
92 .period_bytes_max = 131072,
93 .periods_min = 2,
94 .periods_max = 220,
95 /* One page (4k) contains 512 instructions. I don't know if the hw
96 supports lists longer than this. In this case periods_max=220 is a
97 safe limit to make sure the list never exceeds 512 instructions. */
98};
99
100
101#include "gina20_dsp.c"
102#include "echoaudio_dsp.c"
103#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c
new file mode 100644
index 000000000000..2757c8960843
--- /dev/null
+++ b/sound/pci/echoaudio/gina20_dsp.c
@@ -0,0 +1,215 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int set_professional_spdif(struct echoaudio *chip, char prof);
33static int update_flags(struct echoaudio *chip);
34
35
36static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
37{
38 int err;
39
40 DE_INIT(("init_hw() - Gina20\n"));
41 snd_assert((subdevice_id & 0xfff0) == GINA20, return -ENODEV);
42
43 if ((err = init_dsp_comm_page(chip))) {
44 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
45 return err;
46 }
47
48 chip->device_id = device_id;
49 chip->subdevice_id = subdevice_id;
50 chip->bad_board = TRUE;
51 chip->dsp_code_to_load = &card_fw[FW_GINA20_DSP];
52 chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
53 chip->clock_state = GD_CLOCK_UNDEF;
54 /* Since this card has no ASIC, mark it as loaded so everything
55 works OK */
56 chip->asic_loaded = TRUE;
57 chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
58 ECHO_CLOCK_BIT_SPDIF;
59
60 if ((err = load_firmware(chip)) < 0)
61 return err;
62 chip->bad_board = FALSE;
63
64 if ((err = init_line_levels(chip)) < 0)
65 return err;
66
67 err = set_professional_spdif(chip, TRUE);
68
69 DE_INIT(("init_hw done\n"));
70 return err;
71}
72
73
74
75static u32 detect_input_clocks(const struct echoaudio *chip)
76{
77 u32 clocks_from_dsp, clock_bits;
78
79 /* Map the DSP clock detect bits to the generic driver clock
80 detect bits */
81 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
82
83 clock_bits = ECHO_CLOCK_BIT_INTERNAL;
84
85 if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
86 clock_bits |= ECHO_CLOCK_BIT_SPDIF;
87
88 return clock_bits;
89}
90
91
92
93/* The Gina20 has no ASIC. Just do nothing */
94static int load_asic(struct echoaudio *chip)
95{
96 return 0;
97}
98
99
100
101static int set_sample_rate(struct echoaudio *chip, u32 rate)
102{
103 u8 clock_state, spdif_status;
104
105 if (wait_handshake(chip))
106 return -EIO;
107
108 switch (rate) {
109 case 44100:
110 clock_state = GD_CLOCK_44;
111 spdif_status = GD_SPDIF_STATUS_44;
112 break;
113 case 48000:
114 clock_state = GD_CLOCK_48;
115 spdif_status = GD_SPDIF_STATUS_48;
116 break;
117 default:
118 clock_state = GD_CLOCK_NOCHANGE;
119 spdif_status = GD_SPDIF_STATUS_NOCHANGE;
120 break;
121 }
122
123 if (chip->clock_state == clock_state)
124 clock_state = GD_CLOCK_NOCHANGE;
125 if (spdif_status == chip->spdif_status)
126 spdif_status = GD_SPDIF_STATUS_NOCHANGE;
127
128 chip->comm_page->sample_rate = cpu_to_le32(rate);
129 chip->comm_page->gd_clock_state = clock_state;
130 chip->comm_page->gd_spdif_status = spdif_status;
131 chip->comm_page->gd_resampler_state = 3; /* magic number - should always be 3 */
132
133 /* Save the new audio state if it changed */
134 if (clock_state != GD_CLOCK_NOCHANGE)
135 chip->clock_state = clock_state;
136 if (spdif_status != GD_SPDIF_STATUS_NOCHANGE)
137 chip->spdif_status = spdif_status;
138 chip->sample_rate = rate;
139
140 clear_handshake(chip);
141 return send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
142}
143
144
145
146static int set_input_clock(struct echoaudio *chip, u16 clock)
147{
148 DE_ACT(("set_input_clock:\n"));
149
150 switch (clock) {
151 case ECHO_CLOCK_INTERNAL:
152 /* Reset the audio state to unknown (just in case) */
153 chip->clock_state = GD_CLOCK_UNDEF;
154 chip->spdif_status = GD_SPDIF_STATUS_UNDEF;
155 set_sample_rate(chip, chip->sample_rate);
156 chip->input_clock = clock;
157 DE_ACT(("Set Gina clock to INTERNAL\n"));
158 break;
159 case ECHO_CLOCK_SPDIF:
160 chip->comm_page->gd_clock_state = GD_CLOCK_SPDIFIN;
161 chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_NOCHANGE;
162 clear_handshake(chip);
163 send_vector(chip, DSP_VC_SET_GD_AUDIO_STATE);
164 chip->clock_state = GD_CLOCK_SPDIFIN;
165 DE_ACT(("Set Gina20 clock to SPDIF\n"));
166 chip->input_clock = clock;
167 break;
168 default:
169 return -EINVAL;
170 }
171
172 return 0;
173}
174
175
176
177/* Set input bus gain (one unit is 0.5dB !) */
178static int set_input_gain(struct echoaudio *chip, u16 input, int gain)
179{
180 snd_assert(input < num_busses_in(chip), return -EINVAL);
181
182 if (wait_handshake(chip))
183 return -EIO;
184
185 chip->input_gain[input] = gain;
186 gain += GL20_INPUT_GAIN_MAGIC_NUMBER;
187 chip->comm_page->line_in_level[input] = gain;
188 return 0;
189}
190
191
192
193/* Tell the DSP to reread the flags from the comm page */
194static int update_flags(struct echoaudio *chip)
195{
196 if (wait_handshake(chip))
197 return -EIO;
198 clear_handshake(chip);
199 return send_vector(chip, DSP_VC_UPDATE_FLAGS);
200}
201
202
203
204static int set_professional_spdif(struct echoaudio *chip, char prof)
205{
206 DE_ACT(("set_professional_spdif %d\n", prof));
207 if (prof)
208 chip->comm_page->flags |=
209 __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
210 else
211 chip->comm_page->flags &=
212 ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
213 chip->professional_spdif = prof;
214 return update_flags(chip);
215}
diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c
new file mode 100644
index 000000000000..e464d720d0bd
--- /dev/null
+++ b/sound/pci/echoaudio/gina24.c
@@ -0,0 +1,123 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define ECHO24_FAMILY
20#define ECHOCARD_GINA24
21#define ECHOCARD_NAME "Gina24"
22#define ECHOCARD_HAS_MONITOR
23#define ECHOCARD_HAS_ASIC
24#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
25#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
26#define ECHOCARD_HAS_SUPER_INTERLEAVE
27#define ECHOCARD_HAS_DIGITAL_IO
28#define ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
29#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
30#define ECHOCARD_HAS_EXTERNAL_CLOCK
31#define ECHOCARD_HAS_ADAT 6
32#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
33
34/* Pipe indexes */
35#define PX_ANALOG_OUT 0 /* 8 */
36#define PX_DIGITAL_OUT 8 /* 8 */
37#define PX_ANALOG_IN 16 /* 2 */
38#define PX_DIGITAL_IN 18 /* 8 */
39#define PX_NUM 26
40
41/* Bus indexes */
42#define BX_ANALOG_OUT 0 /* 8 */
43#define BX_DIGITAL_OUT 8 /* 8 */
44#define BX_ANALOG_IN 16 /* 2 */
45#define BX_DIGITAL_IN 18 /* 8 */
46#define BX_NUM 26
47
48
49#include <sound/driver.h>
50#include <linux/delay.h>
51#include <linux/init.h>
52#include <linux/interrupt.h>
53#include <linux/pci.h>
54#include <linux/slab.h>
55#include <linux/moduleparam.h>
56#include <linux/firmware.h>
57#include <sound/core.h>
58#include <sound/info.h>
59#include <sound/control.h>
60#include <sound/pcm.h>
61#include <sound/pcm_params.h>
62#include <sound/asoundef.h>
63#include <sound/initval.h>
64#include <asm/io.h>
65#include <asm/atomic.h>
66#include "echoaudio.h"
67
68#define FW_361_LOADER 0
69#define FW_GINA24_301_DSP 1
70#define FW_GINA24_361_DSP 2
71#define FW_GINA24_301_ASIC 3
72#define FW_GINA24_361_ASIC 4
73
74static const struct firmware card_fw[] = {
75 {0, "loader_dsp.fw"},
76 {0, "gina24_301_dsp.fw"},
77 {0, "gina24_361_dsp.fw"},
78 {0, "gina24_301_asic.fw"},
79 {0, "gina24_361_asic.fw"}
80};
81
82static struct pci_device_id snd_echo_ids[] = {
83 {0x1057, 0x1801, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56301 Gina24 rev.0 */
84 {0x1057, 0x1801, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56301 Gina24 rev.1 */
85 {0x1057, 0x3410, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56361 Gina24 rev.0 */
86 {0x1057, 0x3410, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56361 Gina24 rev.1 */
87 {0,}
88};
89
90static struct snd_pcm_hardware pcm_hardware_skel = {
91 .info = SNDRV_PCM_INFO_MMAP |
92 SNDRV_PCM_INFO_INTERLEAVED |
93 SNDRV_PCM_INFO_BLOCK_TRANSFER |
94 SNDRV_PCM_INFO_MMAP_VALID |
95 SNDRV_PCM_INFO_PAUSE |
96 SNDRV_PCM_INFO_SYNC_START,
97 .formats = SNDRV_PCM_FMTBIT_U8 |
98 SNDRV_PCM_FMTBIT_S16_LE |
99 SNDRV_PCM_FMTBIT_S24_3LE |
100 SNDRV_PCM_FMTBIT_S32_LE |
101 SNDRV_PCM_FMTBIT_S32_BE,
102 .rates = SNDRV_PCM_RATE_8000_48000 |
103 SNDRV_PCM_RATE_88200 |
104 SNDRV_PCM_RATE_96000,
105 .rate_min = 8000,
106 .rate_max = 96000,
107 .channels_min = 1,
108 .channels_max = 8,
109 .buffer_bytes_max = 262144,
110 .period_bytes_min = 32,
111 .period_bytes_max = 131072,
112 .periods_min = 2,
113 .periods_max = 220,
114 /* One page (4k) contains 512 instructions. I don't know if the hw
115 supports lists longer than this. In this case periods_max=220 is a
116 safe limit to make sure the list never exceeds 512 instructions.
117 220 ~= (512 - 1 - (BUFFER_BYTES_MAX / PAGE_SIZE)) / 2 */
118};
119
120#include "gina24_dsp.c"
121#include "echoaudio_dsp.c"
122#include "echoaudio_gml.c"
123#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c
new file mode 100644
index 000000000000..144fc567becf
--- /dev/null
+++ b/sound/pci/echoaudio/gina24_dsp.c
@@ -0,0 +1,346 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int write_control_reg(struct echoaudio *chip, u32 value, char force);
33static int set_input_clock(struct echoaudio *chip, u16 clock);
34static int set_professional_spdif(struct echoaudio *chip, char prof);
35static int set_digital_mode(struct echoaudio *chip, u8 mode);
36static int load_asic_generic(struct echoaudio *chip, u32 cmd,
37 const struct firmware *asic);
38static int check_asic_status(struct echoaudio *chip);
39
40
41static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
42{
43 int err;
44
45 DE_INIT(("init_hw() - Gina24\n"));
46 snd_assert((subdevice_id & 0xfff0) == GINA24, return -ENODEV);
47
48 if ((err = init_dsp_comm_page(chip))) {
49 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
50 return err;
51 }
52
53 chip->device_id = device_id;
54 chip->subdevice_id = subdevice_id;
55 chip->bad_board = TRUE;
56 chip->input_clock_types =
57 ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
58 ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 |
59 ECHO_CLOCK_BIT_ADAT;
60 chip->professional_spdif = FALSE;
61 chip->digital_in_automute = TRUE;
62 chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
63
64 /* Gina24 comes in both '301 and '361 flavors */
65 if (chip->device_id == DEVICE_ID_56361) {
66 chip->dsp_code_to_load = &card_fw[FW_GINA24_361_DSP];
67 chip->digital_modes =
68 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
69 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
70 ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
71 } else {
72 chip->dsp_code_to_load = &card_fw[FW_GINA24_301_DSP];
73 chip->digital_modes =
74 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
75 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
76 ECHOCAPS_HAS_DIGITAL_MODE_ADAT |
77 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_CDROM;
78 }
79
80 if ((err = load_firmware(chip)) < 0)
81 return err;
82 chip->bad_board = FALSE;
83
84 if ((err = init_line_levels(chip)) < 0)
85 return err;
86 err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
87 snd_assert(err >= 0, return err);
88 err = set_professional_spdif(chip, TRUE);
89
90 DE_INIT(("init_hw done\n"));
91 return err;
92}
93
94
95
96static u32 detect_input_clocks(const struct echoaudio *chip)
97{
98 u32 clocks_from_dsp, clock_bits;
99
100 /* Map the DSP clock detect bits to the generic driver clock
101 detect bits */
102 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
103
104 clock_bits = ECHO_CLOCK_BIT_INTERNAL;
105
106 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF)
107 clock_bits |= ECHO_CLOCK_BIT_SPDIF;
108
109 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ADAT)
110 clock_bits |= ECHO_CLOCK_BIT_ADAT;
111
112 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ESYNC)
113 clock_bits |= ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96;
114
115 return clock_bits;
116}
117
118
119
120/* Gina24 has an ASIC on the PCI card which must be loaded for anything
121interesting to happen. */
122static int load_asic(struct echoaudio *chip)
123{
124 u32 control_reg;
125 int err;
126 const struct firmware *fw;
127
128 if (chip->asic_loaded)
129 return 1;
130
131 /* Give the DSP a few milliseconds to settle down */
132 mdelay(10);
133
134 /* Pick the correct ASIC for '301 or '361 Gina24 */
135 if (chip->device_id == DEVICE_ID_56361)
136 fw = &card_fw[FW_GINA24_361_ASIC];
137 else
138 fw = &card_fw[FW_GINA24_301_ASIC];
139
140 if ((err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, fw)) < 0)
141 return err;
142
143 chip->asic_code = fw;
144
145 /* Now give the new ASIC a little time to set up */
146 mdelay(10);
147 /* See if it worked */
148 err = check_asic_status(chip);
149
150 /* Set up the control register if the load succeeded -
151 48 kHz, internal clock, S/PDIF RCA mode */
152 if (!err) {
153 control_reg = GML_CONVERTER_ENABLE | GML_48KHZ;
154 err = write_control_reg(chip, control_reg, TRUE);
155 }
156 DE_INIT(("load_asic() done\n"));
157 return err;
158}
159
160
161
162static int set_sample_rate(struct echoaudio *chip, u32 rate)
163{
164 u32 control_reg, clock;
165
166 snd_assert(rate < 50000 || chip->digital_mode != DIGITAL_MODE_ADAT,
167 return -EINVAL);
168
169 /* Only set the clock for internal mode. */
170 if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
171 DE_ACT(("set_sample_rate: Cannot set sample rate - "
172 "clock not set to CLK_CLOCKININTERNAL\n"));
173 /* Save the rate anyhow */
174 chip->comm_page->sample_rate = cpu_to_le32(rate);
175 chip->sample_rate = rate;
176 return 0;
177 }
178
179 clock = 0;
180
181 control_reg = le32_to_cpu(chip->comm_page->control_register);
182 control_reg &= GML_CLOCK_CLEAR_MASK & GML_SPDIF_RATE_CLEAR_MASK;
183
184 switch (rate) {
185 case 96000:
186 clock = GML_96KHZ;
187 break;
188 case 88200:
189 clock = GML_88KHZ;
190 break;
191 case 48000:
192 clock = GML_48KHZ | GML_SPDIF_SAMPLE_RATE1;
193 break;
194 case 44100:
195 clock = GML_44KHZ;
196 /* Professional mode ? */
197 if (control_reg & GML_SPDIF_PRO_MODE)
198 clock |= GML_SPDIF_SAMPLE_RATE0;
199 break;
200 case 32000:
201 clock = GML_32KHZ | GML_SPDIF_SAMPLE_RATE0 |
202 GML_SPDIF_SAMPLE_RATE1;
203 break;
204 case 22050:
205 clock = GML_22KHZ;
206 break;
207 case 16000:
208 clock = GML_16KHZ;
209 break;
210 case 11025:
211 clock = GML_11KHZ;
212 break;
213 case 8000:
214 clock = GML_8KHZ;
215 break;
216 default:
217 DE_ACT(("set_sample_rate: %d invalid!\n", rate));
218 return -EINVAL;
219 }
220
221 control_reg |= clock;
222
223 chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
224 chip->sample_rate = rate;
225 DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
226
227 return write_control_reg(chip, control_reg, FALSE);
228}
229
230
231
232static int set_input_clock(struct echoaudio *chip, u16 clock)
233{
234 u32 control_reg, clocks_from_dsp;
235
236 DE_ACT(("set_input_clock:\n"));
237
238 /* Mask off the clock select bits */
239 control_reg = le32_to_cpu(chip->comm_page->control_register) &
240 GML_CLOCK_CLEAR_MASK;
241 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
242
243 switch (clock) {
244 case ECHO_CLOCK_INTERNAL:
245 DE_ACT(("Set Gina24 clock to INTERNAL\n"));
246 chip->input_clock = ECHO_CLOCK_INTERNAL;
247 return set_sample_rate(chip, chip->sample_rate);
248 case ECHO_CLOCK_SPDIF:
249 if (chip->digital_mode == DIGITAL_MODE_ADAT)
250 return -EAGAIN;
251 DE_ACT(("Set Gina24 clock to SPDIF\n"));
252 control_reg |= GML_SPDIF_CLOCK;
253 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
254 control_reg |= GML_DOUBLE_SPEED_MODE;
255 else
256 control_reg &= ~GML_DOUBLE_SPEED_MODE;
257 break;
258 case ECHO_CLOCK_ADAT:
259 if (chip->digital_mode != DIGITAL_MODE_ADAT)
260 return -EAGAIN;
261 DE_ACT(("Set Gina24 clock to ADAT\n"));
262 control_reg |= GML_ADAT_CLOCK;
263 control_reg &= ~GML_DOUBLE_SPEED_MODE;
264 break;
265 case ECHO_CLOCK_ESYNC:
266 DE_ACT(("Set Gina24 clock to ESYNC\n"));
267 control_reg |= GML_ESYNC_CLOCK;
268 control_reg &= ~GML_DOUBLE_SPEED_MODE;
269 break;
270 case ECHO_CLOCK_ESYNC96:
271 DE_ACT(("Set Gina24 clock to ESYNC96\n"));
272 control_reg |= GML_ESYNC_CLOCK | GML_DOUBLE_SPEED_MODE;
273 break;
274 default:
275 DE_ACT(("Input clock 0x%x not supported for Gina24\n", clock));
276 return -EINVAL;
277 }
278
279 chip->input_clock = clock;
280 return write_control_reg(chip, control_reg, TRUE);
281}
282
283
284
285static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
286{
287 u32 control_reg;
288 int err, incompatible_clock;
289
290 /* Set clock to "internal" if it's not compatible with the new mode */
291 incompatible_clock = FALSE;
292 switch (mode) {
293 case DIGITAL_MODE_SPDIF_OPTICAL:
294 case DIGITAL_MODE_SPDIF_CDROM:
295 case DIGITAL_MODE_SPDIF_RCA:
296 if (chip->input_clock == ECHO_CLOCK_ADAT)
297 incompatible_clock = TRUE;
298 break;
299 case DIGITAL_MODE_ADAT:
300 if (chip->input_clock == ECHO_CLOCK_SPDIF)
301 incompatible_clock = TRUE;
302 break;
303 default:
304 DE_ACT(("Digital mode not supported: %d\n", mode));
305 return -EINVAL;
306 }
307
308 spin_lock_irq(&chip->lock);
309
310 if (incompatible_clock) { /* Switch to 48KHz, internal */
311 chip->sample_rate = 48000;
312 set_input_clock(chip, ECHO_CLOCK_INTERNAL);
313 }
314
315 /* Clear the current digital mode */
316 control_reg = le32_to_cpu(chip->comm_page->control_register);
317 control_reg &= GML_DIGITAL_MODE_CLEAR_MASK;
318
319 /* Tweak the control reg */
320 switch (mode) {
321 case DIGITAL_MODE_SPDIF_OPTICAL:
322 control_reg |= GML_SPDIF_OPTICAL_MODE;
323 break;
324 case DIGITAL_MODE_SPDIF_CDROM:
325 /* '361 Gina24 cards do not have the S/PDIF CD-ROM mode */
326 if (chip->device_id == DEVICE_ID_56301)
327 control_reg |= GML_SPDIF_CDROM_MODE;
328 break;
329 case DIGITAL_MODE_SPDIF_RCA:
330 /* GML_SPDIF_OPTICAL_MODE bit cleared */
331 break;
332 case DIGITAL_MODE_ADAT:
333 control_reg |= GML_ADAT_MODE;
334 control_reg &= ~GML_DOUBLE_SPEED_MODE;
335 break;
336 }
337
338 err = write_control_reg(chip, control_reg, TRUE);
339 spin_unlock_irq(&chip->lock);
340 if (err < 0)
341 return err;
342 chip->digital_mode = mode;
343
344 DE_ACT(("set_digital_mode to %d\n", chip->digital_mode));
345 return incompatible_clock;
346}
diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c
new file mode 100644
index 000000000000..bfd2467099ac
--- /dev/null
+++ b/sound/pci/echoaudio/indigo.c
@@ -0,0 +1,104 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define INDIGO_FAMILY
20#define ECHOCARD_INDIGO
21#define ECHOCARD_NAME "Indigo"
22#define ECHOCARD_HAS_SUPER_INTERLEAVE
23#define ECHOCARD_HAS_VMIXER
24#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
25
26/* Pipe indexes */
27#define PX_ANALOG_OUT 0 /* 8 */
28#define PX_DIGITAL_OUT 8 /* 0 */
29#define PX_ANALOG_IN 8 /* 0 */
30#define PX_DIGITAL_IN 8 /* 0 */
31#define PX_NUM 8
32
33/* Bus indexes */
34#define BX_ANALOG_OUT 0 /* 2 */
35#define BX_DIGITAL_OUT 2 /* 0 */
36#define BX_ANALOG_IN 2 /* 0 */
37#define BX_DIGITAL_IN 2 /* 0 */
38#define BX_NUM 2
39
40
41#include <sound/driver.h>
42#include <linux/delay.h>
43#include <linux/init.h>
44#include <linux/interrupt.h>
45#include <linux/pci.h>
46#include <linux/slab.h>
47#include <linux/moduleparam.h>
48#include <linux/firmware.h>
49#include <sound/core.h>
50#include <sound/info.h>
51#include <sound/control.h>
52#include <sound/pcm.h>
53#include <sound/pcm_params.h>
54#include <sound/asoundef.h>
55#include <sound/initval.h>
56#include <asm/io.h>
57#include <asm/atomic.h>
58#include "echoaudio.h"
59
60#define FW_361_LOADER 0
61#define FW_INDIGO_DSP 1
62
63static const struct firmware card_fw[] = {
64 {0, "loader_dsp.fw"},
65 {0, "indigo_dsp.fw"}
66};
67
68static struct pci_device_id snd_echo_ids[] = {
69 {0x1057, 0x3410, 0xECC0, 0x0090, 0, 0, 0}, /* Indigo */
70 {0,}
71};
72
73static struct snd_pcm_hardware pcm_hardware_skel = {
74 .info = SNDRV_PCM_INFO_MMAP |
75 SNDRV_PCM_INFO_INTERLEAVED |
76 SNDRV_PCM_INFO_BLOCK_TRANSFER |
77 SNDRV_PCM_INFO_MMAP_VALID |
78 SNDRV_PCM_INFO_PAUSE |
79 SNDRV_PCM_INFO_SYNC_START,
80 .formats = SNDRV_PCM_FMTBIT_U8 |
81 SNDRV_PCM_FMTBIT_S16_LE |
82 SNDRV_PCM_FMTBIT_S24_3LE |
83 SNDRV_PCM_FMTBIT_S32_LE |
84 SNDRV_PCM_FMTBIT_S32_BE,
85 .rates = SNDRV_PCM_RATE_32000 |
86 SNDRV_PCM_RATE_44100 |
87 SNDRV_PCM_RATE_48000 |
88 SNDRV_PCM_RATE_88200 |
89 SNDRV_PCM_RATE_96000,
90 .rate_min = 32000,
91 .rate_max = 96000,
92 .channels_min = 1,
93 .channels_max = 8,
94 .buffer_bytes_max = 262144,
95 .period_bytes_min = 32,
96 .period_bytes_max = 131072,
97 .periods_min = 2,
98 .periods_max = 220,
99};
100
101#include "indigo_dsp.c"
102#include "echoaudio_dsp.c"
103#include "echoaudio.c"
104
diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c
new file mode 100644
index 000000000000..d6ac7734609e
--- /dev/null
+++ b/sound/pci/echoaudio/indigo_dsp.c
@@ -0,0 +1,170 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
33 int gain);
34static int update_vmixer_level(struct echoaudio *chip);
35
36
37static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
38{
39 int err;
40
41 DE_INIT(("init_hw() - Indigo\n"));
42 snd_assert((subdevice_id & 0xfff0) == INDIGO, return -ENODEV);
43
44 if ((err = init_dsp_comm_page(chip))) {
45 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
46 return err;
47 }
48
49 chip->device_id = device_id;
50 chip->subdevice_id = subdevice_id;
51 chip->bad_board = TRUE;
52 chip->dsp_code_to_load = &card_fw[FW_INDIGO_DSP];
53 /* Since this card has no ASIC, mark it as loaded so everything
54 works OK */
55 chip->asic_loaded = TRUE;
56 chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
57
58 if ((err = load_firmware(chip)) < 0)
59 return err;
60 chip->bad_board = FALSE;
61
62 if ((err = init_line_levels(chip)) < 0)
63 return err;
64
65 /* Default routing of the virtual channels: all vchannels are routed
66 to the stereo output */
67 set_vmixer_gain(chip, 0, 0, 0);
68 set_vmixer_gain(chip, 1, 1, 0);
69 set_vmixer_gain(chip, 0, 2, 0);
70 set_vmixer_gain(chip, 1, 3, 0);
71 set_vmixer_gain(chip, 0, 4, 0);
72 set_vmixer_gain(chip, 1, 5, 0);
73 set_vmixer_gain(chip, 0, 6, 0);
74 set_vmixer_gain(chip, 1, 7, 0);
75 err = update_vmixer_level(chip);
76
77 DE_INIT(("init_hw done\n"));
78 return err;
79}
80
81
82
83static u32 detect_input_clocks(const struct echoaudio *chip)
84{
85 return ECHO_CLOCK_BIT_INTERNAL;
86}
87
88
89
90/* The Indigo has no ASIC. Just do nothing */
91static int load_asic(struct echoaudio *chip)
92{
93 return 0;
94}
95
96
97
98static int set_sample_rate(struct echoaudio *chip, u32 rate)
99{
100 u32 control_reg;
101
102 switch (rate) {
103 case 96000:
104 control_reg = MIA_96000;
105 break;
106 case 88200:
107 control_reg = MIA_88200;
108 break;
109 case 48000:
110 control_reg = MIA_48000;
111 break;
112 case 44100:
113 control_reg = MIA_44100;
114 break;
115 case 32000:
116 control_reg = MIA_32000;
117 break;
118 default:
119 DE_ACT(("set_sample_rate: %d invalid!\n", rate));
120 return -EINVAL;
121 }
122
123 /* Set the control register if it has changed */
124 if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
125 if (wait_handshake(chip))
126 return -EIO;
127
128 chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
129 chip->comm_page->control_register = cpu_to_le32(control_reg);
130 chip->sample_rate = rate;
131
132 clear_handshake(chip);
133 return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
134 }
135 return 0;
136}
137
138
139
140/* This function routes the sound from a virtual channel to a real output */
141static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
142 int gain)
143{
144 int index;
145
146 snd_assert(pipe < num_pipes_out(chip) &&
147 output < num_busses_out(chip), return -EINVAL);
148
149 if (wait_handshake(chip))
150 return -EIO;
151
152 chip->vmixer_gain[output][pipe] = gain;
153 index = output * num_pipes_out(chip) + pipe;
154 chip->comm_page->vmixer[index] = gain;
155
156 DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
157 return 0;
158}
159
160
161
162/* Tell the DSP to read and update virtual mixer levels in comm page. */
163static int update_vmixer_level(struct echoaudio *chip)
164{
165 if (wait_handshake(chip))
166 return -EIO;
167 clear_handshake(chip);
168 return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
169}
170
diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c
new file mode 100644
index 000000000000..8ed7ff1fd875
--- /dev/null
+++ b/sound/pci/echoaudio/indigodj.c
@@ -0,0 +1,104 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define INDIGO_FAMILY
20#define ECHOCARD_INDIGO_DJ
21#define ECHOCARD_NAME "Indigo DJ"
22#define ECHOCARD_HAS_SUPER_INTERLEAVE
23#define ECHOCARD_HAS_VMIXER
24#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
25
26/* Pipe indexes */
27#define PX_ANALOG_OUT 0 /* 8 */
28#define PX_DIGITAL_OUT 8 /* 0 */
29#define PX_ANALOG_IN 8 /* 0 */
30#define PX_DIGITAL_IN 8 /* 0 */
31#define PX_NUM 8
32
33/* Bus indexes */
34#define BX_ANALOG_OUT 0 /* 4 */
35#define BX_DIGITAL_OUT 4 /* 0 */
36#define BX_ANALOG_IN 4 /* 0 */
37#define BX_DIGITAL_IN 4 /* 0 */
38#define BX_NUM 4
39
40
41#include <sound/driver.h>
42#include <linux/delay.h>
43#include <linux/init.h>
44#include <linux/interrupt.h>
45#include <linux/pci.h>
46#include <linux/slab.h>
47#include <linux/moduleparam.h>
48#include <linux/firmware.h>
49#include <sound/core.h>
50#include <sound/info.h>
51#include <sound/control.h>
52#include <sound/pcm.h>
53#include <sound/pcm_params.h>
54#include <sound/asoundef.h>
55#include <sound/initval.h>
56#include <asm/io.h>
57#include <asm/atomic.h>
58#include "echoaudio.h"
59
60#define FW_361_LOADER 0
61#define FW_INDIGO_DJ_DSP 1
62
63static const struct firmware card_fw[] = {
64 {0, "loader_dsp.fw"},
65 {0, "indigo_dj_dsp.fw"}
66};
67
68static struct pci_device_id snd_echo_ids[] = {
69 {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/
70 {0,}
71};
72
73static struct snd_pcm_hardware pcm_hardware_skel = {
74 .info = SNDRV_PCM_INFO_MMAP |
75 SNDRV_PCM_INFO_INTERLEAVED |
76 SNDRV_PCM_INFO_BLOCK_TRANSFER |
77 SNDRV_PCM_INFO_MMAP_VALID |
78 SNDRV_PCM_INFO_PAUSE |
79 SNDRV_PCM_INFO_SYNC_START,
80 .formats = SNDRV_PCM_FMTBIT_U8 |
81 SNDRV_PCM_FMTBIT_S16_LE |
82 SNDRV_PCM_FMTBIT_S24_3LE |
83 SNDRV_PCM_FMTBIT_S32_LE |
84 SNDRV_PCM_FMTBIT_S32_BE,
85 .rates = SNDRV_PCM_RATE_32000 |
86 SNDRV_PCM_RATE_44100 |
87 SNDRV_PCM_RATE_48000 |
88 SNDRV_PCM_RATE_88200 |
89 SNDRV_PCM_RATE_96000,
90 .rate_min = 32000,
91 .rate_max = 96000,
92 .channels_min = 1,
93 .channels_max = 4,
94 .buffer_bytes_max = 262144,
95 .period_bytes_min = 32,
96 .period_bytes_max = 131072,
97 .periods_min = 2,
98 .periods_max = 220,
99};
100
101#include "indigodj_dsp.c"
102#include "echoaudio_dsp.c"
103#include "echoaudio.c"
104
diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c
new file mode 100644
index 000000000000..500e150b49fc
--- /dev/null
+++ b/sound/pci/echoaudio/indigodj_dsp.c
@@ -0,0 +1,170 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
33 int gain);
34static int update_vmixer_level(struct echoaudio *chip);
35
36
37static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
38{
39 int err;
40
41 DE_INIT(("init_hw() - Indigo DJ\n"));
42 snd_assert((subdevice_id & 0xfff0) == INDIGO_DJ, return -ENODEV);
43
44 if ((err = init_dsp_comm_page(chip))) {
45 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
46 return err;
47 }
48
49 chip->device_id = device_id;
50 chip->subdevice_id = subdevice_id;
51 chip->bad_board = TRUE;
52 chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP];
53 /* Since this card has no ASIC, mark it as loaded so everything
54 works OK */
55 chip->asic_loaded = TRUE;
56 chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
57
58 if ((err = load_firmware(chip)) < 0)
59 return err;
60 chip->bad_board = FALSE;
61
62 if ((err = init_line_levels(chip)) < 0)
63 return err;
64
65 /* Default routing of the virtual channels: vchannels 0-3 and
66 vchannels 4-7 are routed to real channels 0-4 */
67 set_vmixer_gain(chip, 0, 0, 0);
68 set_vmixer_gain(chip, 1, 1, 0);
69 set_vmixer_gain(chip, 2, 2, 0);
70 set_vmixer_gain(chip, 3, 3, 0);
71 set_vmixer_gain(chip, 0, 4, 0);
72 set_vmixer_gain(chip, 1, 5, 0);
73 set_vmixer_gain(chip, 2, 6, 0);
74 set_vmixer_gain(chip, 3, 7, 0);
75 err = update_vmixer_level(chip);
76
77 DE_INIT(("init_hw done\n"));
78 return err;
79}
80
81
82
83static u32 detect_input_clocks(const struct echoaudio *chip)
84{
85 return ECHO_CLOCK_BIT_INTERNAL;
86}
87
88
89
90/* The IndigoDJ has no ASIC. Just do nothing */
91static int load_asic(struct echoaudio *chip)
92{
93 return 0;
94}
95
96
97
98static int set_sample_rate(struct echoaudio *chip, u32 rate)
99{
100 u32 control_reg;
101
102 switch (rate) {
103 case 96000:
104 control_reg = MIA_96000;
105 break;
106 case 88200:
107 control_reg = MIA_88200;
108 break;
109 case 48000:
110 control_reg = MIA_48000;
111 break;
112 case 44100:
113 control_reg = MIA_44100;
114 break;
115 case 32000:
116 control_reg = MIA_32000;
117 break;
118 default:
119 DE_ACT(("set_sample_rate: %d invalid!\n", rate));
120 return -EINVAL;
121 }
122
123 /* Set the control register if it has changed */
124 if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
125 if (wait_handshake(chip))
126 return -EIO;
127
128 chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
129 chip->comm_page->control_register = cpu_to_le32(control_reg);
130 chip->sample_rate = rate;
131
132 clear_handshake(chip);
133 return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
134 }
135 return 0;
136}
137
138
139
140/* This function routes the sound from a virtual channel to a real output */
141static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
142 int gain)
143{
144 int index;
145
146 snd_assert(pipe < num_pipes_out(chip) &&
147 output < num_busses_out(chip), return -EINVAL);
148
149 if (wait_handshake(chip))
150 return -EIO;
151
152 chip->vmixer_gain[output][pipe] = gain;
153 index = output * num_pipes_out(chip) + pipe;
154 chip->comm_page->vmixer[index] = gain;
155
156 DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
157 return 0;
158}
159
160
161
162/* Tell the DSP to read and update virtual mixer levels in comm page. */
163static int update_vmixer_level(struct echoaudio *chip)
164{
165 if (wait_handshake(chip))
166 return -EIO;
167 clear_handshake(chip);
168 return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
169}
170
diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c
new file mode 100644
index 000000000000..a8788e959171
--- /dev/null
+++ b/sound/pci/echoaudio/indigoio.c
@@ -0,0 +1,105 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define INDIGO_FAMILY
20#define ECHOCARD_INDIGO_IO
21#define ECHOCARD_NAME "Indigo IO"
22#define ECHOCARD_HAS_MONITOR
23#define ECHOCARD_HAS_SUPER_INTERLEAVE
24#define ECHOCARD_HAS_VMIXER
25#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
26
27/* Pipe indexes */
28#define PX_ANALOG_OUT 0 /* 8 */
29#define PX_DIGITAL_OUT 8 /* 0 */
30#define PX_ANALOG_IN 8 /* 2 */
31#define PX_DIGITAL_IN 10 /* 0 */
32#define PX_NUM 10
33
34/* Bus indexes */
35#define BX_ANALOG_OUT 0 /* 2 */
36#define BX_DIGITAL_OUT 2 /* 0 */
37#define BX_ANALOG_IN 2 /* 2 */
38#define BX_DIGITAL_IN 4 /* 0 */
39#define BX_NUM 4
40
41
42#include <sound/driver.h>
43#include <linux/delay.h>
44#include <linux/init.h>
45#include <linux/interrupt.h>
46#include <linux/pci.h>
47#include <linux/slab.h>
48#include <linux/moduleparam.h>
49#include <linux/firmware.h>
50#include <sound/core.h>
51#include <sound/info.h>
52#include <sound/control.h>
53#include <sound/pcm.h>
54#include <sound/pcm_params.h>
55#include <sound/asoundef.h>
56#include <sound/initval.h>
57#include <asm/io.h>
58#include <asm/atomic.h>
59#include "echoaudio.h"
60
61#define FW_361_LOADER 0
62#define FW_INDIGO_IO_DSP 1
63
64static const struct firmware card_fw[] = {
65 {0, "loader_dsp.fw"},
66 {0, "indigo_io_dsp.fw"}
67};
68
69static struct pci_device_id snd_echo_ids[] = {
70 {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/
71 {0,}
72};
73
74static struct snd_pcm_hardware pcm_hardware_skel = {
75 .info = SNDRV_PCM_INFO_MMAP |
76 SNDRV_PCM_INFO_INTERLEAVED |
77 SNDRV_PCM_INFO_BLOCK_TRANSFER |
78 SNDRV_PCM_INFO_MMAP_VALID |
79 SNDRV_PCM_INFO_PAUSE |
80 SNDRV_PCM_INFO_SYNC_START,
81 .formats = SNDRV_PCM_FMTBIT_U8 |
82 SNDRV_PCM_FMTBIT_S16_LE |
83 SNDRV_PCM_FMTBIT_S24_3LE |
84 SNDRV_PCM_FMTBIT_S32_LE |
85 SNDRV_PCM_FMTBIT_S32_BE,
86 .rates = SNDRV_PCM_RATE_32000 |
87 SNDRV_PCM_RATE_44100 |
88 SNDRV_PCM_RATE_48000 |
89 SNDRV_PCM_RATE_88200 |
90 SNDRV_PCM_RATE_96000,
91 .rate_min = 32000,
92 .rate_max = 96000,
93 .channels_min = 1,
94 .channels_max = 8,
95 .buffer_bytes_max = 262144,
96 .period_bytes_min = 32,
97 .period_bytes_max = 131072,
98 .periods_min = 2,
99 .periods_max = 220,
100};
101
102#include "indigoio_dsp.c"
103#include "echoaudio_dsp.c"
104#include "echoaudio.c"
105
diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c
new file mode 100644
index 000000000000..f3ad13d06be0
--- /dev/null
+++ b/sound/pci/echoaudio/indigoio_dsp.c
@@ -0,0 +1,141 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
33 int gain);
34static int update_vmixer_level(struct echoaudio *chip);
35
36
37static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
38{
39 int err;
40
41 DE_INIT(("init_hw() - Indigo IO\n"));
42 snd_assert((subdevice_id & 0xfff0) == INDIGO_IO, return -ENODEV);
43
44 if ((err = init_dsp_comm_page(chip))) {
45 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
46 return err;
47 }
48
49 chip->device_id = device_id;
50 chip->subdevice_id = subdevice_id;
51 chip->bad_board = TRUE;
52 chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP];
53 /* Since this card has no ASIC, mark it as loaded so everything
54 works OK */
55 chip->asic_loaded = TRUE;
56 chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
57
58 if ((err = load_firmware(chip)) < 0)
59 return err;
60 chip->bad_board = FALSE;
61
62 if ((err = init_line_levels(chip)) < 0)
63 return err;
64
65 /* Default routing of the virtual channels: all vchannels are routed
66 to the stereo output */
67 set_vmixer_gain(chip, 0, 0, 0);
68 set_vmixer_gain(chip, 1, 1, 0);
69 set_vmixer_gain(chip, 0, 2, 0);
70 set_vmixer_gain(chip, 1, 3, 0);
71 set_vmixer_gain(chip, 0, 4, 0);
72 set_vmixer_gain(chip, 1, 5, 0);
73 set_vmixer_gain(chip, 0, 6, 0);
74 set_vmixer_gain(chip, 1, 7, 0);
75 err = update_vmixer_level(chip);
76
77 DE_INIT(("init_hw done\n"));
78 return err;
79}
80
81
82
83static u32 detect_input_clocks(const struct echoaudio *chip)
84{
85 return ECHO_CLOCK_BIT_INTERNAL;
86}
87
88
89
90/* The IndigoIO has no ASIC. Just do nothing */
91static int load_asic(struct echoaudio *chip)
92{
93 return 0;
94}
95
96
97
98static int set_sample_rate(struct echoaudio *chip, u32 rate)
99{
100 if (wait_handshake(chip))
101 return -EIO;
102
103 chip->sample_rate = rate;
104 chip->comm_page->sample_rate = cpu_to_le32(rate);
105 clear_handshake(chip);
106 return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
107}
108
109
110
111/* This function routes the sound from a virtual channel to a real output */
112static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
113 int gain)
114{
115 int index;
116
117 snd_assert(pipe < num_pipes_out(chip) &&
118 output < num_busses_out(chip), return -EINVAL);
119
120 if (wait_handshake(chip))
121 return -EIO;
122
123 chip->vmixer_gain[output][pipe] = gain;
124 index = output * num_pipes_out(chip) + pipe;
125 chip->comm_page->vmixer[index] = gain;
126
127 DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
128 return 0;
129}
130
131
132
133/* Tell the DSP to read and update virtual mixer levels in comm page. */
134static int update_vmixer_level(struct echoaudio *chip)
135{
136 if (wait_handshake(chip))
137 return -EIO;
138 clear_handshake(chip);
139 return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
140}
141
diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c
new file mode 100644
index 000000000000..e503d74b3ba9
--- /dev/null
+++ b/sound/pci/echoaudio/layla20.c
@@ -0,0 +1,112 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define ECHOGALS_FAMILY
20#define ECHOCARD_LAYLA20
21#define ECHOCARD_NAME "Layla20"
22#define ECHOCARD_HAS_MONITOR
23#define ECHOCARD_HAS_ASIC
24#define ECHOCARD_HAS_INPUT_GAIN
25#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
26#define ECHOCARD_HAS_SUPER_INTERLEAVE
27#define ECHOCARD_HAS_DIGITAL_IO
28#define ECHOCARD_HAS_EXTERNAL_CLOCK
29#define ECHOCARD_HAS_ADAT FALSE
30#define ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH
31#define ECHOCARD_HAS_MIDI
32
33/* Pipe indexes */
34#define PX_ANALOG_OUT 0 /* 10 */
35#define PX_DIGITAL_OUT 10 /* 2 */
36#define PX_ANALOG_IN 12 /* 8 */
37#define PX_DIGITAL_IN 20 /* 2 */
38#define PX_NUM 22
39
40/* Bus indexes */
41#define BX_ANALOG_OUT 0 /* 10 */
42#define BX_DIGITAL_OUT 10 /* 2 */
43#define BX_ANALOG_IN 12 /* 8 */
44#define BX_DIGITAL_IN 20 /* 2 */
45#define BX_NUM 22
46
47
48#include <sound/driver.h>
49#include <linux/delay.h>
50#include <linux/init.h>
51#include <linux/interrupt.h>
52#include <linux/pci.h>
53#include <linux/slab.h>
54#include <linux/moduleparam.h>
55#include <linux/firmware.h>
56#include <sound/core.h>
57#include <sound/info.h>
58#include <sound/control.h>
59#include <sound/pcm.h>
60#include <sound/pcm_params.h>
61#include <sound/asoundef.h>
62#include <sound/initval.h>
63#include <sound/rawmidi.h>
64#include <asm/io.h>
65#include <asm/atomic.h>
66#include "echoaudio.h"
67
68#define FW_LAYLA20_DSP 0
69#define FW_LAYLA20_ASIC 1
70
71static const struct firmware card_fw[] = {
72 {0, "layla20_dsp.fw"},
73 {0, "layla20_asic.fw"}
74};
75
76static struct pci_device_id snd_echo_ids[] = {
77 {0x1057, 0x1801, 0xECC0, 0x0030, 0, 0, 0}, /* DSP 56301 Layla20 rev.0 */
78 {0x1057, 0x1801, 0xECC0, 0x0031, 0, 0, 0}, /* DSP 56301 Layla20 rev.1 */
79 {0,}
80};
81
82static struct snd_pcm_hardware pcm_hardware_skel = {
83 .info = SNDRV_PCM_INFO_MMAP |
84 SNDRV_PCM_INFO_INTERLEAVED |
85 SNDRV_PCM_INFO_BLOCK_TRANSFER |
86 SNDRV_PCM_INFO_MMAP_VALID |
87 SNDRV_PCM_INFO_PAUSE |
88 SNDRV_PCM_INFO_SYNC_START,
89 .formats = SNDRV_PCM_FMTBIT_U8 |
90 SNDRV_PCM_FMTBIT_S16_LE |
91 SNDRV_PCM_FMTBIT_S24_3LE |
92 SNDRV_PCM_FMTBIT_S32_LE |
93 SNDRV_PCM_FMTBIT_S32_BE,
94 .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS,
95 .rate_min = 8000,
96 .rate_max = 50000,
97 .channels_min = 1,
98 .channels_max = 10,
99 .buffer_bytes_max = 262144,
100 .period_bytes_min = 32,
101 .period_bytes_max = 131072,
102 .periods_min = 2,
103 .periods_max = 220,
104 /* One page (4k) contains 512 instructions. I don't know if the hw
105 supports lists longer than this. In this case periods_max=220 is a
106 safe limit to make sure the list never exceeds 512 instructions. */
107};
108
109#include "layla20_dsp.c"
110#include "echoaudio_dsp.c"
111#include "echoaudio.c"
112#include "midi.c"
diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c
new file mode 100644
index 000000000000..990c9a60a0a8
--- /dev/null
+++ b/sound/pci/echoaudio/layla20_dsp.c
@@ -0,0 +1,290 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int read_dsp(struct echoaudio *chip, u32 *data);
33static int set_professional_spdif(struct echoaudio *chip, char prof);
34static int load_asic_generic(struct echoaudio *chip, u32 cmd,
35 const struct firmware *asic);
36static int check_asic_status(struct echoaudio *chip);
37static int update_flags(struct echoaudio *chip);
38
39
40static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
41{
42 int err;
43
44 DE_INIT(("init_hw() - Layla20\n"));
45 snd_assert((subdevice_id & 0xfff0) == LAYLA20, return -ENODEV);
46
47 if ((err = init_dsp_comm_page(chip))) {
48 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
49 return err;
50 }
51
52 chip->device_id = device_id;
53 chip->subdevice_id = subdevice_id;
54 chip->bad_board = TRUE;
55 chip->has_midi = TRUE;
56 chip->dsp_code_to_load = &card_fw[FW_LAYLA20_DSP];
57 chip->input_clock_types =
58 ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
59 ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER;
60 chip->output_clock_types =
61 ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER;
62
63 if ((err = load_firmware(chip)) < 0)
64 return err;
65 chip->bad_board = FALSE;
66
67 if ((err = init_line_levels(chip)) < 0)
68 return err;
69
70 err = set_professional_spdif(chip, TRUE);
71
72 DE_INIT(("init_hw done\n"));
73 return err;
74}
75
76
77
78static u32 detect_input_clocks(const struct echoaudio *chip)
79{
80 u32 clocks_from_dsp, clock_bits;
81
82 /* Map the DSP clock detect bits to the generic driver clock detect bits */
83 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
84
85 clock_bits = ECHO_CLOCK_BIT_INTERNAL;
86
87 if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
88 clock_bits |= ECHO_CLOCK_BIT_SPDIF;
89
90 if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_WORD) {
91 if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SUPER)
92 clock_bits |= ECHO_CLOCK_BIT_SUPER;
93 else
94 clock_bits |= ECHO_CLOCK_BIT_WORD;
95 }
96
97 return clock_bits;
98}
99
100
101
102/* ASIC status check - some cards have one or two ASICs that need to be
103loaded. Once that load is complete, this function is called to see if
104the load was successful.
105If this load fails, it does not necessarily mean that the hardware is
106defective - the external box may be disconnected or turned off.
107This routine sometimes fails for Layla20; for Layla20, the loop runs
1085 times and succeeds if it wins on three of the loops. */
109static int check_asic_status(struct echoaudio *chip)
110{
111 u32 asic_status;
112 int goodcnt, i;
113
114 chip->asic_loaded = FALSE;
115 for (i = goodcnt = 0; i < 5; i++) {
116 send_vector(chip, DSP_VC_TEST_ASIC);
117
118 /* The DSP will return a value to indicate whether or not
119 the ASIC is currently loaded */
120 if (read_dsp(chip, &asic_status) < 0) {
121 DE_ACT(("check_asic_status: failed on read_dsp\n"));
122 return -EIO;
123 }
124
125 if (asic_status == ASIC_ALREADY_LOADED) {
126 if (++goodcnt == 3) {
127 chip->asic_loaded = TRUE;
128 return 0;
129 }
130 }
131 }
132 return -EIO;
133}
134
135
136
137/* Layla20 has an ASIC in the external box */
138static int load_asic(struct echoaudio *chip)
139{
140 int err;
141
142 if (chip->asic_loaded)
143 return 0;
144
145 err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA_ASIC,
146 &card_fw[FW_LAYLA20_ASIC]);
147 if (err < 0)
148 return err;
149
150 /* Check if ASIC is alive and well. */
151 return check_asic_status(chip);
152}
153
154
155
156static int set_sample_rate(struct echoaudio *chip, u32 rate)
157{
158 snd_assert(rate >= 8000 && rate <= 50000, return -EINVAL);
159
160 /* Only set the clock for internal mode. Do not return failure,
161 simply treat it as a non-event. */
162 if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
163 DE_ACT(("set_sample_rate: Cannot set sample rate - "
164 "clock not set to CLK_CLOCKININTERNAL\n"));
165 chip->comm_page->sample_rate = cpu_to_le32(rate);
166 chip->sample_rate = rate;
167 return 0;
168 }
169
170 if (wait_handshake(chip))
171 return -EIO;
172
173 DE_ACT(("set_sample_rate(%d)\n", rate));
174 chip->sample_rate = rate;
175 chip->comm_page->sample_rate = cpu_to_le32(rate);
176 clear_handshake(chip);
177 return send_vector(chip, DSP_VC_SET_LAYLA_SAMPLE_RATE);
178}
179
180
181
182static int set_input_clock(struct echoaudio *chip, u16 clock_source)
183{
184 u16 clock;
185 u32 rate;
186
187 DE_ACT(("set_input_clock:\n"));
188 rate = 0;
189 switch (clock_source) {
190 case ECHO_CLOCK_INTERNAL:
191 DE_ACT(("Set Layla20 clock to INTERNAL\n"));
192 rate = chip->sample_rate;
193 clock = LAYLA20_CLOCK_INTERNAL;
194 break;
195 case ECHO_CLOCK_SPDIF:
196 DE_ACT(("Set Layla20 clock to SPDIF\n"));
197 clock = LAYLA20_CLOCK_SPDIF;
198 break;
199 case ECHO_CLOCK_WORD:
200 DE_ACT(("Set Layla20 clock to WORD\n"));
201 clock = LAYLA20_CLOCK_WORD;
202 break;
203 case ECHO_CLOCK_SUPER:
204 DE_ACT(("Set Layla20 clock to SUPER\n"));
205 clock = LAYLA20_CLOCK_SUPER;
206 break;
207 default:
208 DE_ACT(("Input clock 0x%x not supported for Layla24\n",
209 clock_source));
210 return -EINVAL;
211 }
212 chip->input_clock = clock_source;
213
214 chip->comm_page->input_clock = cpu_to_le16(clock);
215 clear_handshake(chip);
216 send_vector(chip, DSP_VC_UPDATE_CLOCKS);
217
218 if (rate)
219 set_sample_rate(chip, rate);
220
221 return 0;
222}
223
224
225
226static int set_output_clock(struct echoaudio *chip, u16 clock)
227{
228 DE_ACT(("set_output_clock: %d\n", clock));
229 switch (clock) {
230 case ECHO_CLOCK_SUPER:
231 clock = LAYLA20_OUTPUT_CLOCK_SUPER;
232 break;
233 case ECHO_CLOCK_WORD:
234 clock = LAYLA20_OUTPUT_CLOCK_WORD;
235 break;
236 default:
237 DE_ACT(("set_output_clock wrong clock\n"));
238 return -EINVAL;
239 }
240
241 if (wait_handshake(chip))
242 return -EIO;
243
244 chip->comm_page->output_clock = cpu_to_le16(clock);
245 chip->output_clock = clock;
246 clear_handshake(chip);
247 return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
248}
249
250
251
252/* Set input bus gain (one unit is 0.5dB !) */
253static int set_input_gain(struct echoaudio *chip, u16 input, int gain)
254{
255 snd_assert(input < num_busses_in(chip), return -EINVAL);
256
257 if (wait_handshake(chip))
258 return -EIO;
259
260 chip->input_gain[input] = gain;
261 gain += GL20_INPUT_GAIN_MAGIC_NUMBER;
262 chip->comm_page->line_in_level[input] = gain;
263 return 0;
264}
265
266
267
268/* Tell the DSP to reread the flags from the comm page */
269static int update_flags(struct echoaudio *chip)
270{
271 if (wait_handshake(chip))
272 return -EIO;
273 clear_handshake(chip);
274 return send_vector(chip, DSP_VC_UPDATE_FLAGS);
275}
276
277
278
279static int set_professional_spdif(struct echoaudio *chip, char prof)
280{
281 DE_ACT(("set_professional_spdif %d\n", prof));
282 if (prof)
283 chip->comm_page->flags |=
284 __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
285 else
286 chip->comm_page->flags &=
287 ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
288 chip->professional_spdif = prof;
289 return update_flags(chip);
290}
diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c
new file mode 100644
index 000000000000..d4581fdc841c
--- /dev/null
+++ b/sound/pci/echoaudio/layla24.c
@@ -0,0 +1,121 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define ECHO24_FAMILY
20#define ECHOCARD_LAYLA24
21#define ECHOCARD_NAME "Layla24"
22#define ECHOCARD_HAS_MONITOR
23#define ECHOCARD_HAS_ASIC
24#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
25#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
26#define ECHOCARD_HAS_SUPER_INTERLEAVE
27#define ECHOCARD_HAS_DIGITAL_IO
28#define ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
29#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
30#define ECHOCARD_HAS_EXTERNAL_CLOCK
31#define ECHOCARD_HAS_ADAT 6
32#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
33#define ECHOCARD_HAS_MIDI
34
35/* Pipe indexes */
36#define PX_ANALOG_OUT 0 /* 8 */
37#define PX_DIGITAL_OUT 8 /* 8 */
38#define PX_ANALOG_IN 16 /* 8 */
39#define PX_DIGITAL_IN 24 /* 8 */
40#define PX_NUM 32
41
42/* Bus indexes */
43#define BX_ANALOG_OUT 0 /* 8 */
44#define BX_DIGITAL_OUT 8 /* 8 */
45#define BX_ANALOG_IN 16 /* 8 */
46#define BX_DIGITAL_IN 24 /* 8 */
47#define BX_NUM 32
48
49
50#include <sound/driver.h>
51#include <linux/delay.h>
52#include <linux/init.h>
53#include <linux/interrupt.h>
54#include <linux/pci.h>
55#include <linux/slab.h>
56#include <linux/moduleparam.h>
57#include <linux/firmware.h>
58#include <sound/core.h>
59#include <sound/info.h>
60#include <sound/control.h>
61#include <sound/pcm.h>
62#include <sound/pcm_params.h>
63#include <sound/asoundef.h>
64#include <sound/initval.h>
65#include <sound/rawmidi.h>
66#include <asm/io.h>
67#include <asm/atomic.h>
68#include "echoaudio.h"
69
70#define FW_361_LOADER 0
71#define FW_LAYLA24_DSP 1
72#define FW_LAYLA24_1_ASIC 2
73#define FW_LAYLA24_2A_ASIC 3
74#define FW_LAYLA24_2S_ASIC 4
75
76static const struct firmware card_fw[] = {
77 {0, "loader_dsp.fw"},
78 {0, "layla24_dsp.fw"},
79 {0, "layla24_1_asic.fw"},
80 {0, "layla24_2A_asic.fw"},
81 {0, "layla24_2S_asic.fw"}
82};
83
84static struct pci_device_id snd_echo_ids[] = {
85 {0x1057, 0x3410, 0xECC0, 0x0060, 0, 0, 0}, /* DSP 56361 Layla24 rev.0 */
86 {0,}
87};
88
89static struct snd_pcm_hardware pcm_hardware_skel = {
90 .info = SNDRV_PCM_INFO_MMAP |
91 SNDRV_PCM_INFO_INTERLEAVED |
92 SNDRV_PCM_INFO_BLOCK_TRANSFER |
93 SNDRV_PCM_INFO_MMAP_VALID |
94 SNDRV_PCM_INFO_PAUSE |
95 SNDRV_PCM_INFO_SYNC_START,
96 .formats = SNDRV_PCM_FMTBIT_U8 |
97 SNDRV_PCM_FMTBIT_S16_LE |
98 SNDRV_PCM_FMTBIT_S24_3LE |
99 SNDRV_PCM_FMTBIT_S32_LE |
100 SNDRV_PCM_FMTBIT_S32_BE,
101 .rates = SNDRV_PCM_RATE_8000_96000,
102 .rate_min = 8000,
103 .rate_max = 100000,
104 .channels_min = 1,
105 .channels_max = 8,
106 .buffer_bytes_max = 262144,
107 .period_bytes_min = 32,
108 .period_bytes_max = 131072,
109 .periods_min = 2,
110 .periods_max = 220,
111 /* One page (4k) contains 512 instructions. I don't know if the hw
112 supports lists longer than this. In this case periods_max=220 is a
113 safe limit to make sure the list never exceeds 512 instructions. */
114};
115
116
117#include "layla24_dsp.c"
118#include "echoaudio_dsp.c"
119#include "echoaudio_gml.c"
120#include "echoaudio.c"
121#include "midi.c"
diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c
new file mode 100644
index 000000000000..7ec5b63d0dce
--- /dev/null
+++ b/sound/pci/echoaudio/layla24_dsp.c
@@ -0,0 +1,394 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software Foundation.
12
13 This program is distributed in the hope that it will be useful,
14 but WITHOUT ANY WARRANTY; without even the implied warranty of
15 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
16 GNU General Public License for more details.
17
18 You should have received a copy of the GNU General Public License
19 along with this program; if not, write to the Free Software
20 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
21 MA 02111-1307, USA.
22
23 *************************************************************************
24
25 Translation from C++ and adaptation for use in ALSA-Driver
26 were made by Giuliano Pochini <pochini@shiny.it>
27
28****************************************************************************/
29
30
31static int write_control_reg(struct echoaudio *chip, u32 value, char force);
32static int set_input_clock(struct echoaudio *chip, u16 clock);
33static int set_professional_spdif(struct echoaudio *chip, char prof);
34static int set_digital_mode(struct echoaudio *chip, u8 mode);
35static int load_asic_generic(struct echoaudio *chip, u32 cmd,
36 const struct firmware *asic);
37static int check_asic_status(struct echoaudio *chip);
38
39
40static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
41{
42 int err;
43
44 DE_INIT(("init_hw() - Layla24\n"));
45 snd_assert((subdevice_id & 0xfff0) == LAYLA24, return -ENODEV);
46
47 if ((err = init_dsp_comm_page(chip))) {
48 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
49 return err;
50 }
51
52 chip->device_id = device_id;
53 chip->subdevice_id = subdevice_id;
54 chip->bad_board = TRUE;
55 chip->has_midi = TRUE;
56 chip->dsp_code_to_load = &card_fw[FW_LAYLA24_DSP];
57 chip->input_clock_types =
58 ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
59 ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT;
60 chip->digital_modes =
61 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
62 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
63 ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
64 chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
65 chip->professional_spdif = FALSE;
66 chip->digital_in_automute = TRUE;
67
68 if ((err = load_firmware(chip)) < 0)
69 return err;
70 chip->bad_board = FALSE;
71
72 if ((err = init_line_levels(chip)) < 0)
73 return err;
74
75 err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
76 snd_assert(err >= 0, return err);
77 err = set_professional_spdif(chip, TRUE);
78
79 DE_INIT(("init_hw done\n"));
80 return err;
81}
82
83
84
85static u32 detect_input_clocks(const struct echoaudio *chip)
86{
87 u32 clocks_from_dsp, clock_bits;
88
89 /* Map the DSP clock detect bits to the generic driver clock detect bits */
90 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
91
92 clock_bits = ECHO_CLOCK_BIT_INTERNAL;
93
94 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF)
95 clock_bits |= ECHO_CLOCK_BIT_SPDIF;
96
97 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ADAT)
98 clock_bits |= ECHO_CLOCK_BIT_ADAT;
99
100 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD)
101 clock_bits |= ECHO_CLOCK_BIT_WORD;
102
103 return clock_bits;
104}
105
106
107
108/* Layla24 has an ASIC on the PCI card and another ASIC in the external box;
109both need to be loaded. */
110static int load_asic(struct echoaudio *chip)
111{
112 int err;
113
114 if (chip->asic_loaded)
115 return 1;
116
117 DE_INIT(("load_asic\n"));
118
119 /* Give the DSP a few milliseconds to settle down */
120 mdelay(10);
121
122 /* Load the ASIC for the PCI card */
123 err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC,
124 &card_fw[FW_LAYLA24_1_ASIC]);
125 if (err < 0)
126 return err;
127
128 chip->asic_code = &card_fw[FW_LAYLA24_2S_ASIC];
129
130 /* Now give the new ASIC a little time to set up */
131 mdelay(10);
132
133 /* Do the external one */
134 err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC,
135 &card_fw[FW_LAYLA24_2S_ASIC]);
136 if (err < 0)
137 return FALSE;
138
139 /* Now give the external ASIC a little time to set up */
140 mdelay(10);
141
142 /* See if it worked */
143 err = check_asic_status(chip);
144
145 /* Set up the control register if the load succeeded -
146 48 kHz, internal clock, S/PDIF RCA mode */
147 if (!err)
148 err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ,
149 TRUE);
150
151 DE_INIT(("load_asic() done\n"));
152 return err;
153}
154
155
156
157static int set_sample_rate(struct echoaudio *chip, u32 rate)
158{
159 u32 control_reg, clock, base_rate;
160
161 snd_assert(rate < 50000 || chip->digital_mode != DIGITAL_MODE_ADAT,
162 return -EINVAL);
163
164 /* Only set the clock for internal mode. */
165 if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
166 DE_ACT(("set_sample_rate: Cannot set sample rate - "
167 "clock not set to CLK_CLOCKININTERNAL\n"));
168 /* Save the rate anyhow */
169 chip->comm_page->sample_rate = cpu_to_le32(rate);
170 chip->sample_rate = rate;
171 return 0;
172 }
173
174 /* Get the control register & clear the appropriate bits */
175 control_reg = le32_to_cpu(chip->comm_page->control_register);
176 control_reg &= GML_CLOCK_CLEAR_MASK & GML_SPDIF_RATE_CLEAR_MASK;
177
178 clock = 0;
179
180 switch (rate) {
181 case 96000:
182 clock = GML_96KHZ;
183 break;
184 case 88200:
185 clock = GML_88KHZ;
186 break;
187 case 48000:
188 clock = GML_48KHZ | GML_SPDIF_SAMPLE_RATE1;
189 break;
190 case 44100:
191 clock = GML_44KHZ;
192 /* Professional mode */
193 if (control_reg & GML_SPDIF_PRO_MODE)
194 clock |= GML_SPDIF_SAMPLE_RATE0;
195 break;
196 case 32000:
197 clock = GML_32KHZ | GML_SPDIF_SAMPLE_RATE0 |
198 GML_SPDIF_SAMPLE_RATE1;
199 break;
200 case 22050:
201 clock = GML_22KHZ;
202 break;
203 case 16000:
204 clock = GML_16KHZ;
205 break;
206 case 11025:
207 clock = GML_11KHZ;
208 break;
209 case 8000:
210 clock = GML_8KHZ;
211 break;
212 default:
213 /* If this is a non-standard rate, then the driver needs to
214 use Layla24's special "continuous frequency" mode */
215 clock = LAYLA24_CONTINUOUS_CLOCK;
216 if (rate > 50000) {
217 base_rate = rate >> 1;
218 control_reg |= GML_DOUBLE_SPEED_MODE;
219 } else {
220 base_rate = rate;
221 }
222
223 if (base_rate < 25000)
224 base_rate = 25000;
225
226 if (wait_handshake(chip))
227 return -EIO;
228
229 chip->comm_page->sample_rate =
230 cpu_to_le32(LAYLA24_MAGIC_NUMBER / base_rate - 2);
231
232 clear_handshake(chip);
233 send_vector(chip, DSP_VC_SET_LAYLA24_FREQUENCY_REG);
234 }
235
236 control_reg |= clock;
237
238 chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP ? */
239 chip->sample_rate = rate;
240 DE_ACT(("set_sample_rate: %d clock %d\n", rate, control_reg));
241
242 return write_control_reg(chip, control_reg, FALSE);
243}
244
245
246
247static int set_input_clock(struct echoaudio *chip, u16 clock)
248{
249 u32 control_reg, clocks_from_dsp;
250
251 /* Mask off the clock select bits */
252 control_reg = le32_to_cpu(chip->comm_page->control_register) &
253 GML_CLOCK_CLEAR_MASK;
254 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
255
256 /* Pick the new clock */
257 switch (clock) {
258 case ECHO_CLOCK_INTERNAL:
259 DE_ACT(("Set Layla24 clock to INTERNAL\n"));
260 chip->input_clock = ECHO_CLOCK_INTERNAL;
261 return set_sample_rate(chip, chip->sample_rate);
262 case ECHO_CLOCK_SPDIF:
263 if (chip->digital_mode == DIGITAL_MODE_ADAT)
264 return -EAGAIN;
265 control_reg |= GML_SPDIF_CLOCK;
266 /* Layla24 doesn't support 96KHz S/PDIF */
267 control_reg &= ~GML_DOUBLE_SPEED_MODE;
268 DE_ACT(("Set Layla24 clock to SPDIF\n"));
269 break;
270 case ECHO_CLOCK_WORD:
271 control_reg |= GML_WORD_CLOCK;
272 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96)
273 control_reg |= GML_DOUBLE_SPEED_MODE;
274 else
275 control_reg &= ~GML_DOUBLE_SPEED_MODE;
276 DE_ACT(("Set Layla24 clock to WORD\n"));
277 break;
278 case ECHO_CLOCK_ADAT:
279 if (chip->digital_mode != DIGITAL_MODE_ADAT)
280 return -EAGAIN;
281 control_reg |= GML_ADAT_CLOCK;
282 control_reg &= ~GML_DOUBLE_SPEED_MODE;
283 DE_ACT(("Set Layla24 clock to ADAT\n"));
284 break;
285 default:
286 DE_ACT(("Input clock 0x%x not supported for Layla24\n", clock));
287 return -EINVAL;
288 }
289
290 chip->input_clock = clock;
291 return write_control_reg(chip, control_reg, TRUE);
292}
293
294
295
296/* Depending on what digital mode you want, Layla24 needs different ASICs
297loaded. This function checks the ASIC needed for the new mode and sees
298if it matches the one already loaded. */
299static int switch_asic(struct echoaudio *chip, const struct firmware *asic)
300{
301 s8 *monitors;
302
303 /* Check to see if this is already loaded */
304 if (asic != chip->asic_code) {
305 monitors = kmalloc(MONITOR_ARRAY_SIZE, GFP_KERNEL);
306 if (! monitors)
307 return -ENOMEM;
308
309 memcpy(monitors, chip->comm_page->monitors, MONITOR_ARRAY_SIZE);
310 memset(chip->comm_page->monitors, ECHOGAIN_MUTED,
311 MONITOR_ARRAY_SIZE);
312
313 /* Load the desired ASIC */
314 if (load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC,
315 asic) < 0) {
316 memcpy(chip->comm_page->monitors, monitors,
317 MONITOR_ARRAY_SIZE);
318 kfree(monitors);
319 return -EIO;
320 }
321 chip->asic_code = asic;
322 memcpy(chip->comm_page->monitors, monitors, MONITOR_ARRAY_SIZE);
323 kfree(monitors);
324 }
325
326 return 0;
327}
328
329
330
331static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
332{
333 u32 control_reg;
334 int err, incompatible_clock;
335 const struct firmware *asic;
336
337 /* Set clock to "internal" if it's not compatible with the new mode */
338 incompatible_clock = FALSE;
339 switch (mode) {
340 case DIGITAL_MODE_SPDIF_OPTICAL:
341 case DIGITAL_MODE_SPDIF_RCA:
342 if (chip->input_clock == ECHO_CLOCK_ADAT)
343 incompatible_clock = TRUE;
344 asic = &card_fw[FW_LAYLA24_2S_ASIC];
345 break;
346 case DIGITAL_MODE_ADAT:
347 if (chip->input_clock == ECHO_CLOCK_SPDIF)
348 incompatible_clock = TRUE;
349 asic = &card_fw[FW_LAYLA24_2A_ASIC];
350 break;
351 default:
352 DE_ACT(("Digital mode not supported: %d\n", mode));
353 return -EINVAL;
354 }
355
356 if (incompatible_clock) { /* Switch to 48KHz, internal */
357 chip->sample_rate = 48000;
358 spin_lock_irq(&chip->lock);
359 set_input_clock(chip, ECHO_CLOCK_INTERNAL);
360 spin_unlock_irq(&chip->lock);
361 }
362
363 /* switch_asic() can sleep */
364 if (switch_asic(chip, asic) < 0)
365 return -EIO;
366
367 spin_lock_irq(&chip->lock);
368
369 /* Tweak the control register */
370 control_reg = le32_to_cpu(chip->comm_page->control_register);
371 control_reg &= GML_DIGITAL_MODE_CLEAR_MASK;
372
373 switch (mode) {
374 case DIGITAL_MODE_SPDIF_OPTICAL:
375 control_reg |= GML_SPDIF_OPTICAL_MODE;
376 break;
377 case DIGITAL_MODE_SPDIF_RCA:
378 /* GML_SPDIF_OPTICAL_MODE bit cleared */
379 break;
380 case DIGITAL_MODE_ADAT:
381 control_reg |= GML_ADAT_MODE;
382 control_reg &= ~GML_DOUBLE_SPEED_MODE;
383 break;
384 }
385
386 err = write_control_reg(chip, control_reg, TRUE);
387 spin_unlock_irq(&chip->lock);
388 if (err < 0)
389 return err;
390 chip->digital_mode = mode;
391
392 DE_ACT(("set_digital_mode to %d\n", mode));
393 return incompatible_clock;
394}
diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c
new file mode 100644
index 000000000000..be40c64263d2
--- /dev/null
+++ b/sound/pci/echoaudio/mia.c
@@ -0,0 +1,117 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define ECHO24_FAMILY
20#define ECHOCARD_MIA
21#define ECHOCARD_NAME "Mia"
22#define ECHOCARD_HAS_MONITOR
23#define ECHOCARD_HAS_INPUT_NOMINAL_LEVEL
24#define ECHOCARD_HAS_OUTPUT_NOMINAL_LEVEL
25#define ECHOCARD_HAS_SUPER_INTERLEAVE
26#define ECHOCARD_HAS_VMIXER
27#define ECHOCARD_HAS_DIGITAL_IO
28#define ECHOCARD_HAS_EXTERNAL_CLOCK
29#define ECHOCARD_HAS_ADAT FALSE
30#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
31#define ECHOCARD_HAS_MIDI
32
33/* Pipe indexes */
34#define PX_ANALOG_OUT 0 /* 8 */
35#define PX_DIGITAL_OUT 8 /* 0 */
36#define PX_ANALOG_IN 8 /* 2 */
37#define PX_DIGITAL_IN 10 /* 2 */
38#define PX_NUM 12
39
40/* Bus indexes */
41#define BX_ANALOG_OUT 0 /* 2 */
42#define BX_DIGITAL_OUT 2 /* 2 */
43#define BX_ANALOG_IN 4 /* 2 */
44#define BX_DIGITAL_IN 6 /* 2 */
45#define BX_NUM 8
46
47
48#include <sound/driver.h>
49#include <linux/delay.h>
50#include <linux/init.h>
51#include <linux/interrupt.h>
52#include <linux/pci.h>
53#include <linux/slab.h>
54#include <linux/moduleparam.h>
55#include <linux/firmware.h>
56#include <sound/core.h>
57#include <sound/info.h>
58#include <sound/control.h>
59#include <sound/pcm.h>
60#include <sound/pcm_params.h>
61#include <sound/asoundef.h>
62#include <sound/initval.h>
63#include <sound/rawmidi.h>
64#include <asm/io.h>
65#include <asm/atomic.h>
66#include "echoaudio.h"
67
68#define FW_361_LOADER 0
69#define FW_MIA_DSP 1
70
71static const struct firmware card_fw[] = {
72 {0, "loader_dsp.fw"},
73 {0, "mia_dsp.fw"}
74};
75
76static struct pci_device_id snd_echo_ids[] = {
77 {0x1057, 0x3410, 0xECC0, 0x0080, 0, 0, 0}, /* DSP 56361 Mia rev.0 */
78 {0x1057, 0x3410, 0xECC0, 0x0081, 0, 0, 0}, /* DSP 56361 Mia rev.1 */
79 {0,}
80};
81
82static struct snd_pcm_hardware pcm_hardware_skel = {
83 .info = SNDRV_PCM_INFO_MMAP |
84 SNDRV_PCM_INFO_INTERLEAVED |
85 SNDRV_PCM_INFO_BLOCK_TRANSFER |
86 SNDRV_PCM_INFO_MMAP_VALID |
87 SNDRV_PCM_INFO_PAUSE |
88 SNDRV_PCM_INFO_SYNC_START,
89 .formats = SNDRV_PCM_FMTBIT_U8 |
90 SNDRV_PCM_FMTBIT_S16_LE |
91 SNDRV_PCM_FMTBIT_S24_3LE |
92 SNDRV_PCM_FMTBIT_S32_LE |
93 SNDRV_PCM_FMTBIT_S32_BE,
94 .rates = SNDRV_PCM_RATE_32000 |
95 SNDRV_PCM_RATE_44100 |
96 SNDRV_PCM_RATE_48000 |
97 SNDRV_PCM_RATE_88200 |
98 SNDRV_PCM_RATE_96000,
99 .rate_min = 8000,
100 .rate_max = 96000,
101 .channels_min = 1,
102 .channels_max = 8,
103 .buffer_bytes_max = 262144,
104 .period_bytes_min = 32,
105 .period_bytes_max = 131072,
106 .periods_min = 2,
107 .periods_max = 220,
108 /* One page (4k) contains 512 instructions. I don't know if the hw
109 supports lists longer than this. In this case periods_max=220 is a
110 safe limit to make sure the list never exceeds 512 instructions. */
111};
112
113
114#include "mia_dsp.c"
115#include "echoaudio_dsp.c"
116#include "echoaudio.c"
117#include "midi.c"
diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c
new file mode 100644
index 000000000000..891c70519096
--- /dev/null
+++ b/sound/pci/echoaudio/mia_dsp.c
@@ -0,0 +1,229 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int set_input_clock(struct echoaudio *chip, u16 clock);
33static int set_professional_spdif(struct echoaudio *chip, char prof);
34static int update_flags(struct echoaudio *chip);
35static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
36 int gain);
37static int update_vmixer_level(struct echoaudio *chip);
38
39
40static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
41{
42 int err;
43
44 DE_INIT(("init_hw() - Mia\n"));
45 snd_assert((subdevice_id & 0xfff0) == MIA, return -ENODEV);
46
47 if ((err = init_dsp_comm_page(chip))) {
48 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
49 return err;
50 }
51
52 chip->device_id = device_id;
53 chip->subdevice_id = subdevice_id;
54 chip->bad_board = TRUE;
55 chip->dsp_code_to_load = &card_fw[FW_MIA_DSP];
56 /* Since this card has no ASIC, mark it as loaded so everything
57 works OK */
58 chip->asic_loaded = TRUE;
59 if ((subdevice_id & 0x0000f) == MIA_MIDI_REV)
60 chip->has_midi = TRUE;
61 chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL |
62 ECHO_CLOCK_BIT_SPDIF;
63
64 if ((err = load_firmware(chip)) < 0)
65 return err;
66 chip->bad_board = FALSE;
67
68 if ((err = init_line_levels(chip)))
69 return err;
70
71 /* Default routing of the virtual channels: vchannels 0-3 go to analog
72 outputs and vchannels 4-7 go to S/PDIF outputs */
73 set_vmixer_gain(chip, 0, 0, 0);
74 set_vmixer_gain(chip, 1, 1, 0);
75 set_vmixer_gain(chip, 0, 2, 0);
76 set_vmixer_gain(chip, 1, 3, 0);
77 set_vmixer_gain(chip, 2, 4, 0);
78 set_vmixer_gain(chip, 3, 5, 0);
79 set_vmixer_gain(chip, 2, 6, 0);
80 set_vmixer_gain(chip, 3, 7, 0);
81 err = update_vmixer_level(chip);
82
83 DE_INIT(("init_hw done\n"));
84 return err;
85}
86
87
88
89static u32 detect_input_clocks(const struct echoaudio *chip)
90{
91 u32 clocks_from_dsp, clock_bits;
92
93 /* Map the DSP clock detect bits to the generic driver clock
94 detect bits */
95 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
96
97 clock_bits = ECHO_CLOCK_BIT_INTERNAL;
98
99 if (clocks_from_dsp & GLDM_CLOCK_DETECT_BIT_SPDIF)
100 clock_bits |= ECHO_CLOCK_BIT_SPDIF;
101
102 return clock_bits;
103}
104
105
106
107/* The Mia has no ASIC. Just do nothing */
108static int load_asic(struct echoaudio *chip)
109{
110 return 0;
111}
112
113
114
115static int set_sample_rate(struct echoaudio *chip, u32 rate)
116{
117 u32 control_reg;
118
119 switch (rate) {
120 case 96000:
121 control_reg = MIA_96000;
122 break;
123 case 88200:
124 control_reg = MIA_88200;
125 break;
126 case 48000:
127 control_reg = MIA_48000;
128 break;
129 case 44100:
130 control_reg = MIA_44100;
131 break;
132 case 32000:
133 control_reg = MIA_32000;
134 break;
135 default:
136 DE_ACT(("set_sample_rate: %d invalid!\n", rate));
137 return -EINVAL;
138 }
139
140 /* Override the clock setting if this Mia is set to S/PDIF clock */
141 if (chip->input_clock == ECHO_CLOCK_SPDIF)
142 control_reg |= MIA_SPDIF;
143
144 /* Set the control register if it has changed */
145 if (control_reg != le32_to_cpu(chip->comm_page->control_register)) {
146 if (wait_handshake(chip))
147 return -EIO;
148
149 chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
150 chip->comm_page->control_register = cpu_to_le32(control_reg);
151 chip->sample_rate = rate;
152
153 clear_handshake(chip);
154 return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
155 }
156 return 0;
157}
158
159
160
161static int set_input_clock(struct echoaudio *chip, u16 clock)
162{
163 DE_ACT(("set_input_clock(%d)\n", clock));
164 snd_assert(clock == ECHO_CLOCK_INTERNAL || clock == ECHO_CLOCK_SPDIF,
165 return -EINVAL);
166
167 chip->input_clock = clock;
168 return set_sample_rate(chip, chip->sample_rate);
169}
170
171
172
173/* This function routes the sound from a virtual channel to a real output */
174static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
175 int gain)
176{
177 int index;
178
179 snd_assert(pipe < num_pipes_out(chip) &&
180 output < num_busses_out(chip), return -EINVAL);
181
182 if (wait_handshake(chip))
183 return -EIO;
184
185 chip->vmixer_gain[output][pipe] = gain;
186 index = output * num_pipes_out(chip) + pipe;
187 chip->comm_page->vmixer[index] = gain;
188
189 DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
190 return 0;
191}
192
193
194
195/* Tell the DSP to read and update virtual mixer levels in comm page. */
196static int update_vmixer_level(struct echoaudio *chip)
197{
198 if (wait_handshake(chip))
199 return -EIO;
200 clear_handshake(chip);
201 return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
202}
203
204
205
206/* Tell the DSP to reread the flags from the comm page */
207static int update_flags(struct echoaudio *chip)
208{
209 if (wait_handshake(chip))
210 return -EIO;
211 clear_handshake(chip);
212 return send_vector(chip, DSP_VC_UPDATE_FLAGS);
213}
214
215
216
217static int set_professional_spdif(struct echoaudio *chip, char prof)
218{
219 DE_ACT(("set_professional_spdif %d\n", prof));
220 if (prof)
221 chip->comm_page->flags |=
222 __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
223 else
224 chip->comm_page->flags &=
225 ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
226 chip->professional_spdif = prof;
227 return update_flags(chip);
228}
229
diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c
new file mode 100644
index 000000000000..e31f0f11e3a8
--- /dev/null
+++ b/sound/pci/echoaudio/midi.c
@@ -0,0 +1,327 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32/******************************************************************************
33 MIDI lowlevel code
34******************************************************************************/
35
36/* Start and stop Midi input */
37static int enable_midi_input(struct echoaudio *chip, char enable)
38{
39 DE_MID(("enable_midi_input(%d)\n", enable));
40
41 if (wait_handshake(chip))
42 return -EIO;
43
44 if (enable) {
45 chip->mtc_state = MIDI_IN_STATE_NORMAL;
46 chip->comm_page->flags |=
47 __constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
48 } else
49 chip->comm_page->flags &=
50 ~__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
51
52 clear_handshake(chip);
53 return send_vector(chip, DSP_VC_UPDATE_FLAGS);
54}
55
56
57
58/* Send a buffer full of MIDI data to the DSP
59Returns how many actually written or < 0 on error */
60static int write_midi(struct echoaudio *chip, u8 *data, int bytes)
61{
62 snd_assert(bytes > 0 && bytes < MIDI_OUT_BUFFER_SIZE, return -EINVAL);
63
64 if (wait_handshake(chip))
65 return -EIO;
66
67 /* HF4 indicates that it is safe to write MIDI output data */
68 if (! (get_dsp_register(chip, CHI32_STATUS_REG) & CHI32_STATUS_REG_HF4))
69 return 0;
70
71 chip->comm_page->midi_output[0] = bytes;
72 memcpy(&chip->comm_page->midi_output[1], data, bytes);
73 chip->comm_page->midi_out_free_count = 0;
74 clear_handshake(chip);
75 send_vector(chip, DSP_VC_MIDI_WRITE);
76 DE_MID(("write_midi: %d\n", bytes));
77 return bytes;
78}
79
80
81
82/* Run the state machine for MIDI input data
83MIDI time code sync isn't supported by this code right now, but you still need
84this state machine to parse the incoming MIDI data stream. Every time the DSP
85sees a 0xF1 byte come in, it adds the DSP sample position to the MIDI data
86stream. The DSP sample position is represented as a 32 bit unsigned value,
87with the high 16 bits first, followed by the low 16 bits. Since these aren't
88real MIDI bytes, the following logic is needed to skip them. */
89static inline int mtc_process_data(struct echoaudio *chip, short midi_byte)
90{
91 switch (chip->mtc_state) {
92 case MIDI_IN_STATE_NORMAL:
93 if (midi_byte == 0xF1)
94 chip->mtc_state = MIDI_IN_STATE_TS_HIGH;
95 break;
96 case MIDI_IN_STATE_TS_HIGH:
97 chip->mtc_state = MIDI_IN_STATE_TS_LOW;
98 return MIDI_IN_SKIP_DATA;
99 break;
100 case MIDI_IN_STATE_TS_LOW:
101 chip->mtc_state = MIDI_IN_STATE_F1_DATA;
102 return MIDI_IN_SKIP_DATA;
103 break;
104 case MIDI_IN_STATE_F1_DATA:
105 chip->mtc_state = MIDI_IN_STATE_NORMAL;
106 break;
107 }
108 return 0;
109}
110
111
112
113/* This function is called from the IRQ handler and it reads the midi data
114from the DSP's buffer. It returns the number of bytes received. */
115static int midi_service_irq(struct echoaudio *chip)
116{
117 short int count, midi_byte, i, received;
118
119 /* The count is at index 0, followed by actual data */
120 count = le16_to_cpu(chip->comm_page->midi_input[0]);
121
122 snd_assert(count < MIDI_IN_BUFFER_SIZE, return 0);
123
124 /* Get the MIDI data from the comm page */
125 i = 1;
126 received = 0;
127 for (i = 1; i <= count; i++) {
128 /* Get the MIDI byte */
129 midi_byte = le16_to_cpu(chip->comm_page->midi_input[i]);
130
131 /* Parse the incoming MIDI stream. The incoming MIDI data
132 consists of MIDI bytes and timestamps for the MIDI time code
133 0xF1 bytes. mtc_process_data() is a little state machine that
134 parses the stream. If you get MIDI_IN_SKIP_DATA back, then
135 this is a timestamp byte, not a MIDI byte, so don't store it
136 in the MIDI input buffer. */
137 if (mtc_process_data(chip, midi_byte) == MIDI_IN_SKIP_DATA)
138 continue;
139
140 chip->midi_buffer[received++] = (u8)midi_byte;
141 }
142
143 return received;
144}
145
146
147
148
149/******************************************************************************
150 MIDI interface
151******************************************************************************/
152
153static int snd_echo_midi_input_open(struct snd_rawmidi_substream *substream)
154{
155 struct echoaudio *chip = substream->rmidi->private_data;
156
157 chip->midi_in = substream;
158 DE_MID(("rawmidi_iopen\n"));
159 return 0;
160}
161
162
163
164static void snd_echo_midi_input_trigger(struct snd_rawmidi_substream *substream,
165 int up)
166{
167 struct echoaudio *chip = substream->rmidi->private_data;
168
169 if (up != chip->midi_input_enabled) {
170 spin_lock_irq(&chip->lock);
171 enable_midi_input(chip, up);
172 spin_unlock_irq(&chip->lock);
173 chip->midi_input_enabled = up;
174 }
175}
176
177
178
179static int snd_echo_midi_input_close(struct snd_rawmidi_substream *substream)
180{
181 struct echoaudio *chip = substream->rmidi->private_data;
182
183 chip->midi_in = NULL;
184 DE_MID(("rawmidi_iclose\n"));
185 return 0;
186}
187
188
189
190static int snd_echo_midi_output_open(struct snd_rawmidi_substream *substream)
191{
192 struct echoaudio *chip = substream->rmidi->private_data;
193
194 chip->tinuse = 0;
195 chip->midi_full = 0;
196 chip->midi_out = substream;
197 DE_MID(("rawmidi_oopen\n"));
198 return 0;
199}
200
201
202
203static void snd_echo_midi_output_write(unsigned long data)
204{
205 struct echoaudio *chip = (struct echoaudio *)data;
206 unsigned long flags;
207 int bytes, sent, time;
208 unsigned char buf[MIDI_OUT_BUFFER_SIZE - 1];
209
210 DE_MID(("snd_echo_midi_output_write\n"));
211 /* No interrupts are involved: we have to check at regular intervals
212 if the card's output buffer has room for new data. */
213 sent = bytes = 0;
214 spin_lock_irqsave(&chip->lock, flags);
215 chip->midi_full = 0;
216 if (chip->midi_out && !snd_rawmidi_transmit_empty(chip->midi_out)) {
217 bytes = snd_rawmidi_transmit_peek(chip->midi_out, buf,
218 MIDI_OUT_BUFFER_SIZE - 1);
219 DE_MID(("Try to send %d bytes...\n", bytes));
220 sent = write_midi(chip, buf, bytes);
221 if (sent < 0) {
222 snd_printk(KERN_ERR "write_midi() error %d\n", sent);
223 /* retry later */
224 sent = 9000;
225 chip->midi_full = 1;
226 } else if (sent > 0) {
227 DE_MID(("%d bytes sent\n", sent));
228 snd_rawmidi_transmit_ack(chip->midi_out, sent);
229 } else {
230 /* Buffer is full. DSP's internal buffer is 64 (128 ?)
231 bytes long. Let's wait until half of them are sent */
232 DE_MID(("Full\n"));
233 sent = 32;
234 chip->midi_full = 1;
235 }
236 }
237
238 /* We restart the timer only if there is some data left to send */
239 if (!snd_rawmidi_transmit_empty(chip->midi_out) && chip->tinuse) {
240 /* The timer will expire slightly after the data has been
241 sent */
242 time = (sent << 3) / 25 + 1; /* 8/25=0.32ms to send a byte */
243 mod_timer(&chip->timer, jiffies + (time * HZ + 999) / 1000);
244 DE_MID(("Timer armed(%d)\n", ((time * HZ + 999) / 1000)));
245 }
246 spin_unlock_irqrestore(&chip->lock, flags);
247}
248
249
250
251static void snd_echo_midi_output_trigger(struct snd_rawmidi_substream *substream,
252 int up)
253{
254 struct echoaudio *chip = substream->rmidi->private_data;
255
256 DE_MID(("snd_echo_midi_output_trigger(%d)\n", up));
257 spin_lock_irq(&chip->lock);
258 if (up) {
259 if (!chip->tinuse) {
260 init_timer(&chip->timer);
261 chip->timer.function = snd_echo_midi_output_write;
262 chip->timer.data = (unsigned long)chip;
263 chip->tinuse = 1;
264 }
265 } else {
266 if (chip->tinuse) {
267 del_timer(&chip->timer);
268 chip->tinuse = 0;
269 DE_MID(("Timer removed\n"));
270 }
271 }
272 spin_unlock_irq(&chip->lock);
273
274 if (up && !chip->midi_full)
275 snd_echo_midi_output_write((unsigned long)chip);
276}
277
278
279
280static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream)
281{
282 struct echoaudio *chip = substream->rmidi->private_data;
283
284 chip->midi_out = NULL;
285 DE_MID(("rawmidi_oclose\n"));
286 return 0;
287}
288
289
290
291static struct snd_rawmidi_ops snd_echo_midi_input = {
292 .open = snd_echo_midi_input_open,
293 .close = snd_echo_midi_input_close,
294 .trigger = snd_echo_midi_input_trigger,
295};
296
297static struct snd_rawmidi_ops snd_echo_midi_output = {
298 .open = snd_echo_midi_output_open,
299 .close = snd_echo_midi_output_close,
300 .trigger = snd_echo_midi_output_trigger,
301};
302
303
304
305/* <--snd_echo_probe() */
306static int __devinit snd_echo_midi_create(struct snd_card *card,
307 struct echoaudio *chip)
308{
309 int err;
310
311 if ((err = snd_rawmidi_new(card, card->shortname, 0, 1, 1,
312 &chip->rmidi)) < 0)
313 return err;
314
315 strcpy(chip->rmidi->name, card->shortname);
316 chip->rmidi->private_data = chip;
317
318 snd_rawmidi_set_ops(chip->rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
319 &snd_echo_midi_input);
320 snd_rawmidi_set_ops(chip->rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
321 &snd_echo_midi_output);
322
323 chip->rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT |
324 SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX;
325 DE_INIT(("MIDI ok\n"));
326 return 0;
327}
diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c
new file mode 100644
index 000000000000..5dc512add372
--- /dev/null
+++ b/sound/pci/echoaudio/mona.c
@@ -0,0 +1,129 @@
1/*
2 * ALSA driver for Echoaudio soundcards.
3 * Copyright (C) 2003-2004 Giuliano Pochini <pochini@shiny.it>
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; version 2 of the License.
8 *
9 * This program is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 * GNU General Public License for more details.
13 *
14 * You should have received a copy of the GNU General Public License
15 * along with this program; if not, write to the Free Software
16 * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
17 */
18
19#define ECHO24_FAMILY
20#define ECHOCARD_MONA
21#define ECHOCARD_NAME "Mona"
22#define ECHOCARD_HAS_MONITOR
23#define ECHOCARD_HAS_ASIC
24#define ECHOCARD_HAS_SUPER_INTERLEAVE
25#define ECHOCARD_HAS_DIGITAL_IO
26#define ECHOCARD_HAS_DIGITAL_IN_AUTOMUTE
27#define ECHOCARD_HAS_DIGITAL_MODE_SWITCH
28#define ECHOCARD_HAS_EXTERNAL_CLOCK
29#define ECHOCARD_HAS_ADAT 6
30#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
31
32/* Pipe indexes */
33#define PX_ANALOG_OUT 0 /* 6 */
34#define PX_DIGITAL_OUT 6 /* 8 */
35#define PX_ANALOG_IN 14 /* 4 */
36#define PX_DIGITAL_IN 18 /* 8 */
37#define PX_NUM 26
38
39/* Bus indexes */
40#define BX_ANALOG_OUT 0 /* 6 */
41#define BX_DIGITAL_OUT 6 /* 8 */
42#define BX_ANALOG_IN 14 /* 4 */
43#define BX_DIGITAL_IN 18 /* 8 */
44#define BX_NUM 26
45
46
47#include <sound/driver.h>
48#include <linux/delay.h>
49#include <linux/init.h>
50#include <linux/interrupt.h>
51#include <linux/pci.h>
52#include <linux/slab.h>
53#include <linux/moduleparam.h>
54#include <linux/firmware.h>
55#include <sound/core.h>
56#include <sound/info.h>
57#include <sound/control.h>
58#include <sound/pcm.h>
59#include <sound/pcm_params.h>
60#include <sound/asoundef.h>
61#include <sound/initval.h>
62#include <asm/io.h>
63#include <asm/atomic.h>
64#include "echoaudio.h"
65
66#define FW_361_LOADER 0
67#define FW_MONA_301_DSP 1
68#define FW_MONA_361_DSP 2
69#define FW_MONA_301_1_ASIC48 3
70#define FW_MONA_301_1_ASIC96 4
71#define FW_MONA_361_1_ASIC48 5
72#define FW_MONA_361_1_ASIC96 6
73#define FW_MONA_2_ASIC 7
74
75static const struct firmware card_fw[] = {
76 {0, "loader_dsp.fw"},
77 {0, "mona_301_dsp.fw"},
78 {0, "mona_361_dsp.fw"},
79 {0, "mona_301_1_asic_48.fw"},
80 {0, "mona_301_1_asic_96.fw"},
81 {0, "mona_361_1_asic_48.fw"},
82 {0, "mona_361_1_asic_96.fw"},
83 {0, "mona_2_asic.fw"}
84};
85
86static struct pci_device_id snd_echo_ids[] = {
87 {0x1057, 0x1801, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56301 Mona rev.0 */
88 {0x1057, 0x1801, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56301 Mona rev.1 */
89 {0x1057, 0x1801, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56301 Mona rev.2 */
90 {0x1057, 0x3410, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56361 Mona rev.0 */
91 {0x1057, 0x3410, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56361 Mona rev.1 */
92 {0x1057, 0x3410, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56361 Mona rev.2 */
93 {0,}
94};
95
96static struct snd_pcm_hardware pcm_hardware_skel = {
97 .info = SNDRV_PCM_INFO_MMAP |
98 SNDRV_PCM_INFO_INTERLEAVED |
99 SNDRV_PCM_INFO_BLOCK_TRANSFER |
100 SNDRV_PCM_INFO_MMAP_VALID |
101 SNDRV_PCM_INFO_PAUSE |
102 SNDRV_PCM_INFO_SYNC_START,
103 .formats = SNDRV_PCM_FMTBIT_U8 |
104 SNDRV_PCM_FMTBIT_S16_LE |
105 SNDRV_PCM_FMTBIT_S24_3LE |
106 SNDRV_PCM_FMTBIT_S32_LE |
107 SNDRV_PCM_FMTBIT_S32_BE,
108 .rates = SNDRV_PCM_RATE_8000_48000 |
109 SNDRV_PCM_RATE_88200 |
110 SNDRV_PCM_RATE_96000,
111 .rate_min = 8000,
112 .rate_max = 96000,
113 .channels_min = 1,
114 .channels_max = 8,
115 .buffer_bytes_max = 262144,
116 .period_bytes_min = 32,
117 .period_bytes_max = 131072,
118 .periods_min = 2,
119 .periods_max = 220,
120 /* One page (4k) contains 512 instructions. I don't know if the hw
121 supports lists longer than this. In this case periods_max=220 is a
122 safe limit to make sure the list never exceeds 512 instructions. */
123};
124
125
126#include "mona_dsp.c"
127#include "echoaudio_dsp.c"
128#include "echoaudio_gml.c"
129#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c
new file mode 100644
index 000000000000..c0b4bf0be7d1
--- /dev/null
+++ b/sound/pci/echoaudio/mona_dsp.c
@@ -0,0 +1,428 @@
1/****************************************************************************
2
3 Copyright Echo Digital Audio Corporation (c) 1998 - 2004
4 All rights reserved
5 www.echoaudio.com
6
7 This file is part of Echo Digital Audio's generic driver library.
8
9 Echo Digital Audio's generic driver library is free software;
10 you can redistribute it and/or modify it under the terms of
11 the GNU General Public License as published by the Free Software
12 Foundation.
13
14 This program is distributed in the hope that it will be useful,
15 but WITHOUT ANY WARRANTY; without even the implied warranty of
16 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 GNU General Public License for more details.
18
19 You should have received a copy of the GNU General Public License
20 along with this program; if not, write to the Free Software
21 Foundation, Inc., 59 Temple Place - Suite 330, Boston,
22 MA 02111-1307, USA.
23
24 *************************************************************************
25
26 Translation from C++ and adaptation for use in ALSA-Driver
27 were made by Giuliano Pochini <pochini@shiny.it>
28
29****************************************************************************/
30
31
32static int write_control_reg(struct echoaudio *chip, u32 value, char force);
33static int set_input_clock(struct echoaudio *chip, u16 clock);
34static int set_professional_spdif(struct echoaudio *chip, char prof);
35static int set_digital_mode(struct echoaudio *chip, u8 mode);
36static int load_asic_generic(struct echoaudio *chip, u32 cmd,
37 const struct firmware *asic);
38static int check_asic_status(struct echoaudio *chip);
39
40
41static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
42{
43 int err;
44
45 DE_INIT(("init_hw() - Mona\n"));
46 snd_assert((subdevice_id & 0xfff0) == MONA, return -ENODEV);
47
48 if ((err = init_dsp_comm_page(chip))) {
49 DE_INIT(("init_hw - could not initialize DSP comm page\n"));
50 return err;
51 }
52
53 chip->device_id = device_id;
54 chip->subdevice_id = subdevice_id;
55 chip->bad_board = TRUE;
56 chip->input_clock_types =
57 ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF |
58 ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT;
59 chip->digital_modes =
60 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA |
61 ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL |
62 ECHOCAPS_HAS_DIGITAL_MODE_ADAT;
63
64 /* Mona comes in both '301 and '361 flavors */
65 if (chip->device_id == DEVICE_ID_56361)
66 chip->dsp_code_to_load = &card_fw[FW_MONA_361_DSP];
67 else
68 chip->dsp_code_to_load = &card_fw[FW_MONA_301_DSP];
69
70 chip->digital_mode = DIGITAL_MODE_SPDIF_RCA;
71 chip->professional_spdif = FALSE;
72 chip->digital_in_automute = TRUE;
73
74 if ((err = load_firmware(chip)) < 0)
75 return err;
76 chip->bad_board = FALSE;
77
78 if ((err = init_line_levels(chip)) < 0)
79 return err;
80
81 err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA);
82 snd_assert(err >= 0, return err);
83 err = set_professional_spdif(chip, TRUE);
84
85 DE_INIT(("init_hw done\n"));
86 return err;
87}
88
89
90
91static u32 detect_input_clocks(const struct echoaudio *chip)
92{
93 u32 clocks_from_dsp, clock_bits;
94
95 /* Map the DSP clock detect bits to the generic driver clock
96 detect bits */
97 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
98
99 clock_bits = ECHO_CLOCK_BIT_INTERNAL;
100
101 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF)
102 clock_bits |= ECHO_CLOCK_BIT_SPDIF;
103
104 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_ADAT)
105 clock_bits |= ECHO_CLOCK_BIT_ADAT;
106
107 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD)
108 clock_bits |= ECHO_CLOCK_BIT_WORD;
109
110 return clock_bits;
111}
112
113
114
115/* Mona has an ASIC on the PCI card and another ASIC in the external box;
116both need to be loaded. */
117static int load_asic(struct echoaudio *chip)
118{
119 u32 control_reg;
120 int err;
121 const struct firmware *asic;
122
123 if (chip->asic_loaded)
124 return 0;
125
126 mdelay(10);
127
128 if (chip->device_id == DEVICE_ID_56361)
129 asic = &card_fw[FW_MONA_361_1_ASIC48];
130 else
131 asic = &card_fw[FW_MONA_301_1_ASIC48];
132
133 err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC, asic);
134 if (err < 0)
135 return err;
136
137 chip->asic_code = asic;
138 mdelay(10);
139
140 /* Do the external one */
141 err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_EXTERNAL_ASIC,
142 &card_fw[FW_MONA_2_ASIC]);
143 if (err < 0)
144 return err;
145
146 mdelay(10);
147 err = check_asic_status(chip);
148
149 /* Set up the control register if the load succeeded -
150 48 kHz, internal clock, S/PDIF RCA mode */
151 if (!err) {
152 control_reg = GML_CONVERTER_ENABLE | GML_48KHZ;
153 err = write_control_reg(chip, control_reg, TRUE);
154 }
155
156 return err;
157}
158
159
160
161/* Depending on what digital mode you want, Mona needs different ASICs
162loaded. This function checks the ASIC needed for the new mode and sees
163if it matches the one already loaded. */
164static int switch_asic(struct echoaudio *chip, char double_speed)
165{
166 const struct firmware *asic;
167 int err;
168
169 /* Check the clock detect bits to see if this is
170 a single-speed clock or a double-speed clock; load
171 a new ASIC if necessary. */
172 if (chip->device_id == DEVICE_ID_56361) {
173 if (double_speed)
174 asic = &card_fw[FW_MONA_361_1_ASIC96];
175 else
176 asic = &card_fw[FW_MONA_361_1_ASIC48];
177 } else {
178 if (double_speed)
179 asic = &card_fw[FW_MONA_301_1_ASIC96];
180 else
181 asic = &card_fw[FW_MONA_301_1_ASIC48];
182 }
183
184 if (asic != chip->asic_code) {
185 /* Load the desired ASIC */
186 err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC,
187 asic);
188 if (err < 0)
189 return err;
190 chip->asic_code = asic;
191 }
192
193 return 0;
194}
195
196
197
198static int set_sample_rate(struct echoaudio *chip, u32 rate)
199{
200 u32 control_reg, clock;
201 const struct firmware *asic;
202 char force_write;
203
204 /* Only set the clock for internal mode. */
205 if (chip->input_clock != ECHO_CLOCK_INTERNAL) {
206 DE_ACT(("set_sample_rate: Cannot set sample rate - "
207 "clock not set to CLK_CLOCKININTERNAL\n"));
208 /* Save the rate anyhow */
209 chip->comm_page->sample_rate = cpu_to_le32(rate);
210 chip->sample_rate = rate;
211 return 0;
212 }
213
214 /* Now, check to see if the required ASIC is loaded */
215 if (rate >= 88200) {
216 if (chip->digital_mode == DIGITAL_MODE_ADAT)
217 return -EINVAL;
218 if (chip->device_id == DEVICE_ID_56361)
219 asic = &card_fw[FW_MONA_361_1_ASIC96];
220 else
221 asic = &card_fw[FW_MONA_301_1_ASIC96];
222 } else {
223 if (chip->device_id == DEVICE_ID_56361)
224 asic = &card_fw[FW_MONA_361_1_ASIC48];
225 else
226 asic = &card_fw[FW_MONA_301_1_ASIC48];
227 }
228
229 force_write = 0;
230 if (asic != chip->asic_code) {
231 int err;
232 /* Load the desired ASIC (load_asic_generic() can sleep) */
233 spin_unlock_irq(&chip->lock);
234 err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC,
235 asic);
236 spin_lock_irq(&chip->lock);
237
238 if (err < 0)
239 return err;
240 chip->asic_code = asic;
241 force_write = 1;
242 }
243
244 /* Compute the new control register value */
245 clock = 0;
246 control_reg = le32_to_cpu(chip->comm_page->control_register);
247 control_reg &= GML_CLOCK_CLEAR_MASK;
248 control_reg &= GML_SPDIF_RATE_CLEAR_MASK;
249
250 switch (rate) {
251 case 96000:
252 clock = GML_96KHZ;
253 break;
254 case 88200:
255 clock = GML_88KHZ;
256 break;
257 case 48000:
258 clock = GML_48KHZ | GML_SPDIF_SAMPLE_RATE1;
259 break;
260 case 44100:
261 clock = GML_44KHZ;
262 /* Professional mode */
263 if (control_reg & GML_SPDIF_PRO_MODE)
264 clock |= GML_SPDIF_SAMPLE_RATE0;
265 break;
266 case 32000:
267 clock = GML_32KHZ | GML_SPDIF_SAMPLE_RATE0 |
268 GML_SPDIF_SAMPLE_RATE1;
269 break;
270 case 22050:
271 clock = GML_22KHZ;
272 break;
273 case 16000:
274 clock = GML_16KHZ;
275 break;
276 case 11025:
277 clock = GML_11KHZ;
278 break;
279 case 8000:
280 clock = GML_8KHZ;
281 break;
282 default:
283 DE_ACT(("set_sample_rate: %d invalid!\n", rate));
284 return -EINVAL;
285 }
286
287 control_reg |= clock;
288
289 chip->comm_page->sample_rate = cpu_to_le32(rate); /* ignored by the DSP */
290 chip->sample_rate = rate;
291 DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock));
292
293 return write_control_reg(chip, control_reg, force_write);
294}
295
296
297
298static int set_input_clock(struct echoaudio *chip, u16 clock)
299{
300 u32 control_reg, clocks_from_dsp;
301 int err;
302
303 DE_ACT(("set_input_clock:\n"));
304
305 /* Prevent two simultaneous calls to switch_asic() */
306 if (atomic_read(&chip->opencount))
307 return -EAGAIN;
308
309 /* Mask off the clock select bits */
310 control_reg = le32_to_cpu(chip->comm_page->control_register) &
311 GML_CLOCK_CLEAR_MASK;
312 clocks_from_dsp = le32_to_cpu(chip->comm_page->status_clocks);
313
314 switch (clock) {
315 case ECHO_CLOCK_INTERNAL:
316 DE_ACT(("Set Mona clock to INTERNAL\n"));
317 chip->input_clock = ECHO_CLOCK_INTERNAL;
318 return set_sample_rate(chip, chip->sample_rate);
319 case ECHO_CLOCK_SPDIF:
320 if (chip->digital_mode == DIGITAL_MODE_ADAT)
321 return -EAGAIN;
322 spin_unlock_irq(&chip->lock);
323 err = switch_asic(chip, clocks_from_dsp &
324 GML_CLOCK_DETECT_BIT_SPDIF96);
325 spin_lock_irq(&chip->lock);
326 if (err < 0)
327 return err;
328 DE_ACT(("Set Mona clock to SPDIF\n"));
329 control_reg |= GML_SPDIF_CLOCK;
330 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_SPDIF96)
331 control_reg |= GML_DOUBLE_SPEED_MODE;
332 else
333 control_reg &= ~GML_DOUBLE_SPEED_MODE;
334 break;
335 case ECHO_CLOCK_WORD:
336 DE_ACT(("Set Mona clock to WORD\n"));
337 spin_unlock_irq(&chip->lock);
338 err = switch_asic(chip, clocks_from_dsp &
339 GML_CLOCK_DETECT_BIT_WORD96);
340 spin_lock_irq(&chip->lock);
341 if (err < 0)
342 return err;
343 control_reg |= GML_WORD_CLOCK;
344 if (clocks_from_dsp & GML_CLOCK_DETECT_BIT_WORD96)
345 control_reg |= GML_DOUBLE_SPEED_MODE;
346 else
347 control_reg &= ~GML_DOUBLE_SPEED_MODE;
348 break;
349 case ECHO_CLOCK_ADAT:
350 DE_ACT(("Set Mona clock to ADAT\n"));
351 if (chip->digital_mode != DIGITAL_MODE_ADAT)
352 return -EAGAIN;
353 control_reg |= GML_ADAT_CLOCK;
354 control_reg &= ~GML_DOUBLE_SPEED_MODE;
355 break;
356 default:
357 DE_ACT(("Input clock 0x%x not supported for Mona\n", clock));
358 return -EINVAL;
359 }
360
361 chip->input_clock = clock;
362 return write_control_reg(chip, control_reg, TRUE);
363}
364
365
366
367static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode)
368{
369 u32 control_reg;
370 int err, incompatible_clock;
371
372 /* Set clock to "internal" if it's not compatible with the new mode */
373 incompatible_clock = FALSE;
374 switch (mode) {
375 case DIGITAL_MODE_SPDIF_OPTICAL:
376 case DIGITAL_MODE_SPDIF_RCA:
377 if (chip->input_clock == ECHO_CLOCK_ADAT)
378 incompatible_clock = TRUE;
379 break;
380 case DIGITAL_MODE_ADAT:
381 if (chip->input_clock == ECHO_CLOCK_SPDIF)
382 incompatible_clock = TRUE;
383 break;
384 default:
385 DE_ACT(("Digital mode not supported: %d\n", mode));
386 return -EINVAL;
387 }
388
389 spin_lock_irq(&chip->lock);
390
391 if (incompatible_clock) { /* Switch to 48KHz, internal */
392 chip->sample_rate = 48000;
393 set_input_clock(chip, ECHO_CLOCK_INTERNAL);
394 }
395
396 /* Clear the current digital mode */
397 control_reg = le32_to_cpu(chip->comm_page->control_register);
398 control_reg &= GML_DIGITAL_MODE_CLEAR_MASK;
399
400 /* Tweak the control reg */
401 switch (mode) {
402 case DIGITAL_MODE_SPDIF_OPTICAL:
403 control_reg |= GML_SPDIF_OPTICAL_MODE;
404 break;
405 case DIGITAL_MODE_SPDIF_RCA:
406 /* GML_SPDIF_OPTICAL_MODE bit cleared */
407 break;
408 case DIGITAL_MODE_ADAT:
409 /* If the current ASIC is the 96KHz ASIC, switch the ASIC
410 and set to 48 KHz */
411 if (chip->asic_code == &card_fw[FW_MONA_361_1_ASIC96] ||
412 chip->asic_code == &card_fw[FW_MONA_301_1_ASIC96]) {
413 set_sample_rate(chip, 48000);
414 }
415 control_reg |= GML_ADAT_MODE;
416 control_reg &= ~GML_DOUBLE_SPEED_MODE;
417 break;
418 }
419
420 err = write_control_reg(chip, control_reg, FALSE);
421 spin_unlock_irq(&chip->lock);
422 if (err < 0)
423 return err;
424 chip->digital_mode = mode;
425
426 DE_ACT(("set_digital_mode to %d\n", mode));
427 return incompatible_clock;
428}
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8c2a8174ece1..23201f3eeb12 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -408,7 +408,9 @@ static const struct hda_codec_preset *find_codec_preset(struct hda_codec *codec)
408 u32 mask = preset->mask; 408 u32 mask = preset->mask;
409 if (! mask) 409 if (! mask)
410 mask = ~0; 410 mask = ~0;
411 if (preset->id == (codec->vendor_id & mask)) 411 if (preset->id == (codec->vendor_id & mask) &&
412 (! preset->rev ||
413 preset->rev == codec->revision_id))
412 return preset; 414 return preset;
413 } 415 }
414 } 416 }
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index dd4e00a82b55..33b7d5806469 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -799,6 +799,8 @@ static struct hda_board_config ad1986a_cfg_tbl[] = {
799 { .pci_subvendor = 0x1043, .pci_subdevice = 0x818f, 799 { .pci_subvendor = 0x1043, .pci_subdevice = 0x818f,
800 .config = AD1986A_LAPTOP }, /* ASUS P5GV-MX */ 800 .config = AD1986A_LAPTOP }, /* ASUS P5GV-MX */
801 { .modelname = "laptop-eapd", .config = AD1986A_LAPTOP_EAPD }, 801 { .modelname = "laptop-eapd", .config = AD1986A_LAPTOP_EAPD },
802 { .pci_subvendor = 0x144d, .pci_subdevice = 0xc023,
803 .config = AD1986A_LAPTOP_EAPD }, /* Samsung X60 Chane */
802 { .pci_subvendor = 0x144d, .pci_subdevice = 0xc024, 804 { .pci_subvendor = 0x144d, .pci_subdevice = 0xc024,
803 .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */ 805 .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */
804 { .pci_subvendor = 0x1043, .pci_subdevice = 0x1153, 806 { .pci_subvendor = 0x1043, .pci_subdevice = 0x1153,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 98b9f16c26ff..18d105263fea 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -78,6 +78,7 @@ enum {
78enum { 78enum {
79 ALC262_BASIC, 79 ALC262_BASIC,
80 ALC262_FUJITSU, 80 ALC262_FUJITSU,
81 ALC262_HP_BPC,
81 ALC262_AUTO, 82 ALC262_AUTO,
82 ALC262_MODEL_LAST /* last tag */ 83 ALC262_MODEL_LAST /* last tag */
83}; 84};
@@ -85,6 +86,7 @@ enum {
85/* ALC861 models */ 86/* ALC861 models */
86enum { 87enum {
87 ALC861_3ST, 88 ALC861_3ST,
89 ALC660_3ST,
88 ALC861_3ST_DIG, 90 ALC861_3ST_DIG,
89 ALC861_6ST_DIG, 91 ALC861_6ST_DIG,
90 ALC861_AUTO, 92 ALC861_AUTO,
@@ -99,6 +101,17 @@ enum {
99 ALC882_MODEL_LAST, 101 ALC882_MODEL_LAST,
100}; 102};
101 103
104/* ALC883 models */
105enum {
106 ALC883_3ST_2ch_DIG,
107 ALC883_3ST_6ch_DIG,
108 ALC883_3ST_6ch,
109 ALC883_6ST_DIG,
110 ALC888_DEMO_BOARD,
111 ALC883_AUTO,
112 ALC883_MODEL_LAST,
113};
114
102/* for GPIO Poll */ 115/* for GPIO Poll */
103#define GPIO_MASK 0x03 116#define GPIO_MASK 0x03
104 117
@@ -108,7 +121,8 @@ struct alc_spec {
108 unsigned int num_mixers; 121 unsigned int num_mixers;
109 122
110 const struct hda_verb *init_verbs[5]; /* initialization verbs 123 const struct hda_verb *init_verbs[5]; /* initialization verbs
111 * don't forget NULL termination! 124 * don't forget NULL
125 * termination!
112 */ 126 */
113 unsigned int num_init_verbs; 127 unsigned int num_init_verbs;
114 128
@@ -163,7 +177,9 @@ struct alc_spec {
163 * configuration template - to be copied to the spec instance 177 * configuration template - to be copied to the spec instance
164 */ 178 */
165struct alc_config_preset { 179struct alc_config_preset {
166 struct snd_kcontrol_new *mixers[5]; /* should be identical size with spec */ 180 struct snd_kcontrol_new *mixers[5]; /* should be identical size
181 * with spec
182 */
167 const struct hda_verb *init_verbs[5]; 183 const struct hda_verb *init_verbs[5];
168 unsigned int num_dacs; 184 unsigned int num_dacs;
169 hda_nid_t *dac_nids; 185 hda_nid_t *dac_nids;
@@ -184,7 +200,8 @@ struct alc_config_preset {
184/* 200/*
185 * input MUX handling 201 * input MUX handling
186 */ 202 */
187static int alc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) 203static int alc_mux_enum_info(struct snd_kcontrol *kcontrol,
204 struct snd_ctl_elem_info *uinfo)
188{ 205{
189 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 206 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
190 struct alc_spec *spec = codec->spec; 207 struct alc_spec *spec = codec->spec;
@@ -194,7 +211,8 @@ static int alc_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
194 return snd_hda_input_mux_info(&spec->input_mux[mux_idx], uinfo); 211 return snd_hda_input_mux_info(&spec->input_mux[mux_idx], uinfo);
195} 212}
196 213
197static int alc_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 214static int alc_mux_enum_get(struct snd_kcontrol *kcontrol,
215 struct snd_ctl_elem_value *ucontrol)
198{ 216{
199 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 217 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
200 struct alc_spec *spec = codec->spec; 218 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
204 return 0; 222 return 0;
205} 223}
206 224
207static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 225static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
226 struct snd_ctl_elem_value *ucontrol)
208{ 227{
209 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 228 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
210 struct alc_spec *spec = codec->spec; 229 struct alc_spec *spec = codec->spec;
211 unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); 230 unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
212 unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; 231 unsigned int mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
213 return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol, 232 return snd_hda_input_mux_put(codec, &spec->input_mux[mux_idx], ucontrol,
214 spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]); 233 spec->adc_nids[adc_idx],
234 &spec->cur_mux[adc_idx]);
215} 235}
216 236
217 237
218/* 238/*
219 * channel mode setting 239 * channel mode setting
220 */ 240 */
221static int alc_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) 241static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
242 struct snd_ctl_elem_info *uinfo)
222{ 243{
223 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 244 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
224 struct alc_spec *spec = codec->spec; 245 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
226 spec->num_channel_mode); 247 spec->num_channel_mode);
227} 248}
228 249
229static int alc_ch_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 250static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
251 struct snd_ctl_elem_value *ucontrol)
230{ 252{
231 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 253 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
232 struct alc_spec *spec = codec->spec; 254 struct alc_spec *spec = codec->spec;
233 return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode, 255 return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
234 spec->num_channel_mode, spec->multiout.max_channels); 256 spec->num_channel_mode,
257 spec->multiout.max_channels);
235} 258}
236 259
237static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 260static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
261 struct snd_ctl_elem_value *ucontrol)
238{ 262{
239 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 263 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
240 struct alc_spec *spec = codec->spec; 264 struct alc_spec *spec = codec->spec;
241 return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode, 265 return snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
242 spec->num_channel_mode, &spec->multiout.max_channels); 266 spec->num_channel_mode,
267 &spec->multiout.max_channels);
243} 268}
244 269
245/* 270/*
@@ -290,7 +315,8 @@ static signed char alc_pin_mode_dir_info[5][2] = {
290#define alc_pin_mode_n_items(_dir) \ 315#define alc_pin_mode_n_items(_dir) \
291 (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1) 316 (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
292 317
293static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) 318static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
319 struct snd_ctl_elem_info *uinfo)
294{ 320{
295 unsigned int item_num = uinfo->value.enumerated.item; 321 unsigned int item_num = uinfo->value.enumerated.item;
296 unsigned char dir = (kcontrol->private_value >> 16) & 0xff; 322 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_
305 return 0; 331 return 0;
306} 332}
307 333
308static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 334static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
335 struct snd_ctl_elem_value *ucontrol)
309{ 336{
310 unsigned int i; 337 unsigned int i;
311 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 338 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
312 hda_nid_t nid = kcontrol->private_value & 0xffff; 339 hda_nid_t nid = kcontrol->private_value & 0xffff;
313 unsigned char dir = (kcontrol->private_value >> 16) & 0xff; 340 unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
314 long *valp = ucontrol->value.integer.value; 341 long *valp = ucontrol->value.integer.value;
315 unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00); 342 unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
343 AC_VERB_GET_PIN_WIDGET_CONTROL,
344 0x00);
316 345
317 /* Find enumerated value for current pinctl setting */ 346 /* Find enumerated value for current pinctl setting */
318 i = alc_pin_mode_min(dir); 347 i = alc_pin_mode_min(dir);
319 while (alc_pin_mode_values[i]!=pinctl && i<=alc_pin_mode_max(dir)) 348 while (alc_pin_mode_values[i] != pinctl && i <= alc_pin_mode_max(dir))
320 i++; 349 i++;
321 *valp = i<=alc_pin_mode_max(dir)?i:alc_pin_mode_min(dir); 350 *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
322 return 0; 351 return 0;
323} 352}
324 353
325static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 354static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
355 struct snd_ctl_elem_value *ucontrol)
326{ 356{
327 signed int change; 357 signed int change;
328 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 358 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
329 hda_nid_t nid = kcontrol->private_value & 0xffff; 359 hda_nid_t nid = kcontrol->private_value & 0xffff;
330 unsigned char dir = (kcontrol->private_value >> 16) & 0xff; 360 unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
331 long val = *ucontrol->value.integer.value; 361 long val = *ucontrol->value.integer.value;
332 unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00); 362 unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
363 AC_VERB_GET_PIN_WIDGET_CONTROL,
364 0x00);
333 365
334 if (val<alc_pin_mode_min(dir) || val>alc_pin_mode_max(dir)) 366 if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
335 val = alc_pin_mode_min(dir); 367 val = alc_pin_mode_min(dir);
336 368
337 change = pinctl != alc_pin_mode_values[val]; 369 change = pinctl != alc_pin_mode_values[val];
338 if (change) { 370 if (change) {
339 /* Set pin mode to that requested */ 371 /* Set pin mode to that requested */
340 snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL, 372 snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL,
341 alc_pin_mode_values[val]); 373 alc_pin_mode_values[val]);
342 374
343 /* Also enable the retasking pin's input/output as required 375 /* Also enable the retasking pin's input/output as required
344 * for the requested pin mode. Enum values of 2 or less are 376 * 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
351 * this turns out to be necessary in the future. 383 * this turns out to be necessary in the future.
352 */ 384 */
353 if (val <= 2) { 385 if (val <= 2) {
354 snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, 386 snd_hda_codec_write(codec, nid, 0,
355 AMP_OUT_MUTE); 387 AC_VERB_SET_AMP_GAIN_MUTE,
356 snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, 388 AMP_OUT_MUTE);
357 AMP_IN_UNMUTE(0)); 389 snd_hda_codec_write(codec, nid, 0,
390 AC_VERB_SET_AMP_GAIN_MUTE,
391 AMP_IN_UNMUTE(0));
358 } else { 392 } else {
359 snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, 393 snd_hda_codec_write(codec, nid, 0,
360 AMP_IN_MUTE(0)); 394 AC_VERB_SET_AMP_GAIN_MUTE,
361 snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, 395 AMP_IN_MUTE(0));
362 AMP_OUT_UNMUTE); 396 snd_hda_codec_write(codec, nid, 0,
397 AC_VERB_SET_AMP_GAIN_MUTE,
398 AMP_OUT_UNMUTE);
363 } 399 }
364 } 400 }
365 return change; 401 return change;
@@ -378,7 +414,8 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
378 * needed for any "production" models. 414 * needed for any "production" models.
379 */ 415 */
380#ifdef CONFIG_SND_DEBUG 416#ifdef CONFIG_SND_DEBUG
381static int alc_gpio_data_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) 417static int alc_gpio_data_info(struct snd_kcontrol *kcontrol,
418 struct snd_ctl_elem_info *uinfo)
382{ 419{
383 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; 420 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
384 uinfo->count = 1; 421 uinfo->count = 1;
@@ -386,33 +423,38 @@ static int alc_gpio_data_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
386 uinfo->value.integer.max = 1; 423 uinfo->value.integer.max = 1;
387 return 0; 424 return 0;
388} 425}
389static int alc_gpio_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 426static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
427 struct snd_ctl_elem_value *ucontrol)
390{ 428{
391 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 429 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
392 hda_nid_t nid = kcontrol->private_value & 0xffff; 430 hda_nid_t nid = kcontrol->private_value & 0xffff;
393 unsigned char mask = (kcontrol->private_value >> 16) & 0xff; 431 unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
394 long *valp = ucontrol->value.integer.value; 432 long *valp = ucontrol->value.integer.value;
395 unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00); 433 unsigned int val = snd_hda_codec_read(codec, nid, 0,
434 AC_VERB_GET_GPIO_DATA, 0x00);
396 435
397 *valp = (val & mask) != 0; 436 *valp = (val & mask) != 0;
398 return 0; 437 return 0;
399} 438}
400static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 439static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
440 struct snd_ctl_elem_value *ucontrol)
401{ 441{
402 signed int change; 442 signed int change;
403 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 443 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
404 hda_nid_t nid = kcontrol->private_value & 0xffff; 444 hda_nid_t nid = kcontrol->private_value & 0xffff;
405 unsigned char mask = (kcontrol->private_value >> 16) & 0xff; 445 unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
406 long val = *ucontrol->value.integer.value; 446 long val = *ucontrol->value.integer.value;
407 unsigned int gpio_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00); 447 unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
448 AC_VERB_GET_GPIO_DATA,
449 0x00);
408 450
409 /* Set/unset the masked GPIO bit(s) as needed */ 451 /* Set/unset the masked GPIO bit(s) as needed */
410 change = (val==0?0:mask) != (gpio_data & mask); 452 change = (val == 0 ? 0 : mask) != (gpio_data & mask);
411 if (val==0) 453 if (val == 0)
412 gpio_data &= ~mask; 454 gpio_data &= ~mask;
413 else 455 else
414 gpio_data |= mask; 456 gpio_data |= mask;
415 snd_hda_codec_write(codec,nid,0,AC_VERB_SET_GPIO_DATA,gpio_data); 457 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_GPIO_DATA, gpio_data);
416 458
417 return change; 459 return change;
418} 460}
@@ -432,7 +474,8 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_
432 * necessary. 474 * necessary.
433 */ 475 */
434#ifdef CONFIG_SND_DEBUG 476#ifdef CONFIG_SND_DEBUG
435static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) 477static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol,
478 struct snd_ctl_elem_info *uinfo)
436{ 479{
437 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; 480 uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
438 uinfo->count = 1; 481 uinfo->count = 1;
@@ -440,33 +483,39 @@ static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ele
440 uinfo->value.integer.max = 1; 483 uinfo->value.integer.max = 1;
441 return 0; 484 return 0;
442} 485}
443static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 486static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
487 struct snd_ctl_elem_value *ucontrol)
444{ 488{
445 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 489 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
446 hda_nid_t nid = kcontrol->private_value & 0xffff; 490 hda_nid_t nid = kcontrol->private_value & 0xffff;
447 unsigned char mask = (kcontrol->private_value >> 16) & 0xff; 491 unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
448 long *valp = ucontrol->value.integer.value; 492 long *valp = ucontrol->value.integer.value;
449 unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00); 493 unsigned int val = snd_hda_codec_read(codec, nid, 0,
494 AC_VERB_GET_DIGI_CONVERT, 0x00);
450 495
451 *valp = (val & mask) != 0; 496 *valp = (val & mask) != 0;
452 return 0; 497 return 0;
453} 498}
454static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 499static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
500 struct snd_ctl_elem_value *ucontrol)
455{ 501{
456 signed int change; 502 signed int change;
457 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 503 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
458 hda_nid_t nid = kcontrol->private_value & 0xffff; 504 hda_nid_t nid = kcontrol->private_value & 0xffff;
459 unsigned char mask = (kcontrol->private_value >> 16) & 0xff; 505 unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
460 long val = *ucontrol->value.integer.value; 506 long val = *ucontrol->value.integer.value;
461 unsigned int ctrl_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00); 507 unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
508 AC_VERB_GET_DIGI_CONVERT,
509 0x00);
462 510
463 /* Set/unset the masked control bit(s) as needed */ 511 /* Set/unset the masked control bit(s) as needed */
464 change = (val==0?0:mask) != (ctrl_data & mask); 512 change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
465 if (val==0) 513 if (val==0)
466 ctrl_data &= ~mask; 514 ctrl_data &= ~mask;
467 else 515 else
468 ctrl_data |= mask; 516 ctrl_data |= mask;
469 snd_hda_codec_write(codec,nid,0,AC_VERB_SET_DIGI_CONVERT_1,ctrl_data); 517 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
518 ctrl_data);
470 519
471 return change; 520 return change;
472} 521}
@@ -481,14 +530,17 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
481/* 530/*
482 * set up from the preset table 531 * set up from the preset table
483 */ 532 */
484static void setup_preset(struct alc_spec *spec, const struct alc_config_preset *preset) 533static void setup_preset(struct alc_spec *spec,
534 const struct alc_config_preset *preset)
485{ 535{
486 int i; 536 int i;
487 537
488 for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++) 538 for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
489 spec->mixers[spec->num_mixers++] = preset->mixers[i]; 539 spec->mixers[spec->num_mixers++] = preset->mixers[i];
490 for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i]; i++) 540 for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
491 spec->init_verbs[spec->num_init_verbs++] = preset->init_verbs[i]; 541 i++)
542 spec->init_verbs[spec->num_init_verbs++] =
543 preset->init_verbs[i];
492 544
493 spec->channel_mode = preset->channel_mode; 545 spec->channel_mode = preset->channel_mode;
494 spec->num_channel_mode = preset->num_channel_mode; 546 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 *
517 * ALC880 3-stack model 569 * ALC880 3-stack model
518 * 570 *
519 * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e) 571 * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
520 * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18, F-Mic = 0x1b 572 * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18,
521 * HP = 0x19 573 * F-Mic = 0x1b, HP = 0x19
522 */ 574 */
523 575
524static hda_nid_t alc880_dac_nids[4] = { 576static hda_nid_t alc880_dac_nids[4] = {
@@ -662,7 +714,8 @@ static struct snd_kcontrol_new alc880_capture_alt_mixer[] = {
662/* 714/*
663 * ALC880 5-stack model 715 * ALC880 5-stack model
664 * 716 *
665 * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d), Side = 0x02 (0xd) 717 * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d),
718 * Side = 0x02 (0xd)
666 * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16 719 * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
667 * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19 720 * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
668 */ 721 */
@@ -700,7 +753,8 @@ static struct hda_channel_mode alc880_fivestack_modes[2] = {
700/* 753/*
701 * ALC880 6-stack model 754 * ALC880 6-stack model
702 * 755 *
703 * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e), Side = 0x05 (0x0f) 756 * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e),
757 * Side = 0x05 (0x0f)
704 * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17, 758 * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
705 * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b 759 * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
706 */ 760 */
@@ -811,7 +865,8 @@ static struct snd_kcontrol_new alc880_w810_base_mixer[] = {
811 * Z710V model 865 * Z710V model
812 * 866 *
813 * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d) 867 * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
814 * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?), Line = 0x1a 868 * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?),
869 * Line = 0x1a
815 */ 870 */
816 871
817static hda_nid_t alc880_z71v_dac_nids[1] = { 872static hda_nid_t alc880_z71v_dac_nids[1] = {
@@ -966,7 +1021,8 @@ static int alc_build_controls(struct hda_codec *codec)
966 } 1021 }
967 1022
968 if (spec->multiout.dig_out_nid) { 1023 if (spec->multiout.dig_out_nid) {
969 err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); 1024 err = snd_hda_create_spdif_out_ctls(codec,
1025 spec->multiout.dig_out_nid);
970 if (err < 0) 1026 if (err < 0)
971 return err; 1027 return err;
972 } 1028 }
@@ -999,8 +1055,8 @@ static struct hda_verb alc880_volume_init_verbs[] = {
999 1055
1000 /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback 1056 /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
1001 * mixer widget 1057 * mixer widget
1002 * Note: PASD motherboards uses the Line In 2 as the input for front panel 1058 * Note: PASD motherboards uses the Line In 2 as the input for front
1003 * mic (mic 2) 1059 * panel mic (mic 2)
1004 */ 1060 */
1005 /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ 1061 /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
1006 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, 1062 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -1154,8 +1210,8 @@ static struct hda_verb alc880_pin_z71v_init_verbs[] = {
1154 1210
1155/* 1211/*
1156 * 6-stack pin configuration: 1212 * 6-stack pin configuration:
1157 * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18, f-mic = 0x19, 1213 * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
1158 * line = 0x1a, HP = 0x1b 1214 * f-mic = 0x19, line = 0x1a, HP = 0x1b
1159 */ 1215 */
1160static struct hda_verb alc880_pin_6stack_init_verbs[] = { 1216static struct hda_verb alc880_pin_6stack_init_verbs[] = {
1161 {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ 1217 {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
@@ -1587,8 +1643,8 @@ static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
1587 struct snd_pcm_substream *substream) 1643 struct snd_pcm_substream *substream)
1588{ 1644{
1589 struct alc_spec *spec = codec->spec; 1645 struct alc_spec *spec = codec->spec;
1590 return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, 1646 return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
1591 format, substream); 1647 stream_tag, format, substream);
1592} 1648}
1593 1649
1594static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, 1650static 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,
1640{ 1696{
1641 struct alc_spec *spec = codec->spec; 1697 struct alc_spec *spec = codec->spec;
1642 1698
1643 snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0); 1699 snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
1700 0, 0, 0);
1644 return 0; 1701 return 0;
1645} 1702}
1646 1703
@@ -1822,7 +1879,8 @@ static struct hda_channel_mode alc880_test_modes[4] = {
1822 { 8, NULL }, 1879 { 8, NULL },
1823}; 1880};
1824 1881
1825static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) 1882static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
1883 struct snd_ctl_elem_info *uinfo)
1826{ 1884{
1827 static char *texts[] = { 1885 static char *texts[] = {
1828 "N/A", "Line Out", "HP Out", 1886 "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
1837 return 0; 1895 return 0;
1838} 1896}
1839 1897
1840static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 1898static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol,
1899 struct snd_ctl_elem_value *ucontrol)
1841{ 1900{
1842 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 1901 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
1843 hda_nid_t nid = (hda_nid_t)kcontrol->private_value; 1902 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
1863 return 0; 1922 return 0;
1864} 1923}
1865 1924
1866static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 1925static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
1926 struct snd_ctl_elem_value *ucontrol)
1867{ 1927{
1868 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 1928 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
1869 hda_nid_t nid = (hda_nid_t)kcontrol->private_value; 1929 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
1881 AC_VERB_GET_PIN_WIDGET_CONTROL, 0); 1941 AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
1882 new_ctl = ctls[ucontrol->value.enumerated.item[0]]; 1942 new_ctl = ctls[ucontrol->value.enumerated.item[0]];
1883 if (old_ctl != new_ctl) { 1943 if (old_ctl != new_ctl) {
1884 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl); 1944 snd_hda_codec_write(codec, nid, 0,
1945 AC_VERB_SET_PIN_WIDGET_CONTROL, new_ctl);
1885 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, 1946 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
1886 ucontrol->value.enumerated.item[0] >= 3 ? 0xb080 : 0xb000); 1947 (ucontrol->value.enumerated.item[0] >= 3 ?
1948 0xb080 : 0xb000));
1887 return 1; 1949 return 1;
1888 } 1950 }
1889 return 0; 1951 return 0;
1890} 1952}
1891 1953
1892static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) 1954static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
1955 struct snd_ctl_elem_info *uinfo)
1893{ 1956{
1894 static char *texts[] = { 1957 static char *texts[] = {
1895 "Front", "Surround", "CLFE", "Side" 1958 "Front", "Surround", "CLFE", "Side"
@@ -1903,7 +1966,8 @@ static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol, struct snd_ctl_e
1903 return 0; 1966 return 0;
1904} 1967}
1905 1968
1906static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 1969static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol,
1970 struct snd_ctl_elem_value *ucontrol)
1907{ 1971{
1908 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 1972 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
1909 hda_nid_t nid = (hda_nid_t)kcontrol->private_value; 1973 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
1914 return 0; 1978 return 0;
1915} 1979}
1916 1980
1917static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) 1981static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
1982 struct snd_ctl_elem_value *ucontrol)
1918{ 1983{
1919 struct hda_codec *codec = snd_kcontrol_chip(kcontrol); 1984 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
1920 hda_nid_t nid = (hda_nid_t)kcontrol->private_value; 1985 hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
@@ -2739,7 +2804,8 @@ static int patch_alc880(struct hda_codec *codec)
2739 2804
2740 board_config = snd_hda_check_board_config(codec, alc880_cfg_tbl); 2805 board_config = snd_hda_check_board_config(codec, alc880_cfg_tbl);
2741 if (board_config < 0 || board_config >= ALC880_MODEL_LAST) { 2806 if (board_config < 0 || board_config >= ALC880_MODEL_LAST) {
2742 printk(KERN_INFO "hda_codec: Unknown model for ALC880, trying auto-probe from BIOS...\n"); 2807 printk(KERN_INFO "hda_codec: Unknown model for ALC880, "
2808 "trying auto-probe from BIOS...\n");
2743 board_config = ALC880_AUTO; 2809 board_config = ALC880_AUTO;
2744 } 2810 }
2745 2811
@@ -2750,7 +2816,9 @@ static int patch_alc880(struct hda_codec *codec)
2750 alc_free(codec); 2816 alc_free(codec);
2751 return err; 2817 return err;
2752 } else if (! err) { 2818 } else if (! err) {
2753 printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using 3-stack mode...\n"); 2819 printk(KERN_INFO
2820 "hda_codec: Cannot set up configuration "
2821 "from BIOS. Using 3-stack mode...\n");
2754 board_config = ALC880_3ST; 2822 board_config = ALC880_3ST;
2755 } 2823 }
2756 } 2824 }
@@ -3947,7 +4015,8 @@ static int patch_alc260(struct hda_codec *codec)
3947 4015
3948 board_config = snd_hda_check_board_config(codec, alc260_cfg_tbl); 4016 board_config = snd_hda_check_board_config(codec, alc260_cfg_tbl);
3949 if (board_config < 0 || board_config >= ALC260_MODEL_LAST) { 4017 if (board_config < 0 || board_config >= ALC260_MODEL_LAST) {
3950 snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260\n"); 4018 snd_printd(KERN_INFO "hda_codec: Unknown model for ALC260, "
4019 "trying auto-probe from BIOS...\n");
3951 board_config = ALC260_AUTO; 4020 board_config = ALC260_AUTO;
3952 } 4021 }
3953 4022
@@ -3958,7 +4027,9 @@ static int patch_alc260(struct hda_codec *codec)
3958 alc_free(codec); 4027 alc_free(codec);
3959 return err; 4028 return err;
3960 } else if (! err) { 4029 } else if (! err) {
3961 printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n"); 4030 printk(KERN_INFO
4031 "hda_codec: Cannot set up configuration "
4032 "from BIOS. Using base mode...\n");
3962 board_config = ALC260_BASIC; 4033 board_config = ALC260_BASIC;
3963 } 4034 }
3964 } 4035 }
@@ -4320,9 +4391,12 @@ static struct snd_kcontrol_new alc882_capture_mixer[] = {
4320static struct hda_board_config alc882_cfg_tbl[] = { 4391static struct hda_board_config alc882_cfg_tbl[] = {
4321 { .modelname = "3stack-dig", .config = ALC882_3ST_DIG }, 4392 { .modelname = "3stack-dig", .config = ALC882_3ST_DIG },
4322 { .modelname = "6stack-dig", .config = ALC882_6ST_DIG }, 4393 { .modelname = "6stack-dig", .config = ALC882_6ST_DIG },
4323 { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* MSI */ 4394 { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668,
4324 { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* Foxconn */ 4395 .config = ALC882_6ST_DIG }, /* MSI */
4325 { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668, .config = ALC882_6ST_DIG }, /* ECS */ 4396 { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668,
4397 .config = ALC882_6ST_DIG }, /* Foxconn */
4398 { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668,
4399 .config = ALC882_6ST_DIG }, /* ECS to Intel*/
4326 { .modelname = "auto", .config = ALC882_AUTO }, 4400 { .modelname = "auto", .config = ALC882_AUTO },
4327 {} 4401 {}
4328}; 4402};
@@ -4439,10 +4513,6 @@ static void alc882_auto_init(struct hda_codec *codec)
4439 alc882_auto_init_analog_input(codec); 4513 alc882_auto_init_analog_input(codec);
4440} 4514}
4441 4515
4442/*
4443 * ALC882 Headphone poll in 3.5.1a or 3.5.2
4444 */
4445
4446static int patch_alc882(struct hda_codec *codec) 4516static int patch_alc882(struct hda_codec *codec)
4447{ 4517{
4448 struct alc_spec *spec; 4518 struct alc_spec *spec;
@@ -4457,7 +4527,8 @@ static int patch_alc882(struct hda_codec *codec)
4457 board_config = snd_hda_check_board_config(codec, alc882_cfg_tbl); 4527 board_config = snd_hda_check_board_config(codec, alc882_cfg_tbl);
4458 4528
4459 if (board_config < 0 || board_config >= ALC882_MODEL_LAST) { 4529 if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
4460 printk(KERN_INFO "hda_codec: Unknown model for ALC882, trying auto-probe from BIOS...\n"); 4530 printk(KERN_INFO "hda_codec: Unknown model for ALC882, "
4531 "trying auto-probe from BIOS...\n");
4461 board_config = ALC882_AUTO; 4532 board_config = ALC882_AUTO;
4462 } 4533 }
4463 4534
@@ -4468,7 +4539,9 @@ static int patch_alc882(struct hda_codec *codec)
4468 alc_free(codec); 4539 alc_free(codec);
4469 return err; 4540 return err;
4470 } else if (! err) { 4541 } else if (! err) {
4471 printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n"); 4542 printk(KERN_INFO
4543 "hda_codec: Cannot set up configuration "
4544 "from BIOS. Using base mode...\n");
4472 board_config = ALC882_3ST_DIG; 4545 board_config = ALC882_3ST_DIG;
4473 } 4546 }
4474 } 4547 }
@@ -4509,6 +4582,652 @@ static int patch_alc882(struct hda_codec *codec)
4509} 4582}
4510 4583
4511/* 4584/*
4585 * ALC883 support
4586 *
4587 * ALC883 is almost identical with ALC880 but has cleaner and more flexible
4588 * configuration. Each pin widget can choose any input DACs and a mixer.
4589 * Each ADC is connected from a mixer of all inputs. This makes possible
4590 * 6-channel independent captures.
4591 *
4592 * In addition, an independent DAC for the multi-playback (not used in this
4593 * driver yet).
4594 */
4595#define ALC883_DIGOUT_NID 0x06
4596#define ALC883_DIGIN_NID 0x0a
4597
4598static hda_nid_t alc883_dac_nids[4] = {
4599 /* front, rear, clfe, rear_surr */
4600 0x02, 0x04, 0x03, 0x05
4601};
4602
4603static hda_nid_t alc883_adc_nids[2] = {
4604 /* ADC1-2 */
4605 0x08, 0x09,
4606};
4607/* input MUX */
4608/* FIXME: should be a matrix-type input source selection */
4609
4610static struct hda_input_mux alc883_capture_source = {
4611 .num_items = 4,
4612 .items = {
4613 { "Mic", 0x0 },
4614 { "Front Mic", 0x1 },
4615 { "Line", 0x2 },
4616 { "CD", 0x4 },
4617 },
4618};
4619#define alc883_mux_enum_info alc_mux_enum_info
4620#define alc883_mux_enum_get alc_mux_enum_get
4621
4622static int alc883_mux_enum_put(struct snd_kcontrol *kcontrol,
4623 struct snd_ctl_elem_value *ucontrol)
4624{
4625 struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
4626 struct alc_spec *spec = codec->spec;
4627 const struct hda_input_mux *imux = spec->input_mux;
4628 unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
4629 static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
4630 hda_nid_t nid = capture_mixers[adc_idx];
4631 unsigned int *cur_val = &spec->cur_mux[adc_idx];
4632 unsigned int i, idx;
4633
4634 idx = ucontrol->value.enumerated.item[0];
4635 if (idx >= imux->num_items)
4636 idx = imux->num_items - 1;
4637 if (*cur_val == idx && ! codec->in_resume)
4638 return 0;
4639 for (i = 0; i < imux->num_items; i++) {
4640 unsigned int v = (i == idx) ? 0x7000 : 0x7080;
4641 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
4642 v | (imux->items[i].index << 8));
4643 }
4644 *cur_val = idx;
4645 return 1;
4646}
4647/*
4648 * 2ch mode
4649 */
4650static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
4651 { 2, NULL }
4652};
4653
4654/*
4655 * 2ch mode
4656 */
4657static struct hda_verb alc883_3ST_ch2_init[] = {
4658 { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
4659 { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
4660 { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
4661 { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
4662 { } /* end */
4663};
4664
4665/*
4666 * 6ch mode
4667 */
4668static struct hda_verb alc883_3ST_ch6_init[] = {
4669 { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
4670 { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
4671 { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
4672 { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
4673 { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
4674 { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
4675 { } /* end */
4676};
4677
4678static struct hda_channel_mode alc883_3ST_6ch_modes[2] = {
4679 { 2, alc883_3ST_ch2_init },
4680 { 6, alc883_3ST_ch6_init },
4681};
4682
4683/*
4684 * 6ch mode
4685 */
4686static struct hda_verb alc883_sixstack_ch6_init[] = {
4687 { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
4688 { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
4689 { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
4690 { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
4691 { } /* end */
4692};
4693
4694/*
4695 * 8ch mode
4696 */
4697static struct hda_verb alc883_sixstack_ch8_init[] = {
4698 { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
4699 { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
4700 { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
4701 { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
4702 { } /* end */
4703};
4704
4705static struct hda_channel_mode alc883_sixstack_modes[2] = {
4706 { 6, alc883_sixstack_ch6_init },
4707 { 8, alc883_sixstack_ch8_init },
4708};
4709
4710/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
4711 * Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
4712 */
4713
4714static struct snd_kcontrol_new alc883_base_mixer[] = {
4715 HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
4716 HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
4717 HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
4718 HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
4719 HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
4720 HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
4721 HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
4722 HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
4723 HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
4724 HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
4725 HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
4726 HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
4727 HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
4728 HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
4729 HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
4730 HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
4731 HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
4732 HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
4733 HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
4734 HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
4735 HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
4736 HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
4737 HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
4738 HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
4739 HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
4740 {
4741 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
4742 /* .name = "Capture Source", */
4743 .name = "Input Source",
4744 .count = 2,
4745 .info = alc883_mux_enum_info,
4746 .get = alc883_mux_enum_get,
4747 .put = alc883_mux_enum_put,
4748 },
4749 { } /* end */
4750};
4751
4752static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
4753 HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
4754 HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
4755 HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
4756 HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
4757 HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
4758 HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
4759 HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
4760 HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
4761 HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
4762 HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
4763 HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
4764 HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
4765 HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
4766 HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
4767 HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
4768 HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
4769 HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
4770 {
4771 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
4772 /* .name = "Capture Source", */
4773 .name = "Input Source",
4774 .count = 2,
4775 .info = alc883_mux_enum_info,
4776 .get = alc883_mux_enum_get,
4777 .put = alc883_mux_enum_put,
4778 },
4779 { } /* end */
4780};
4781
4782static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
4783 HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
4784 HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
4785 HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
4786 HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
4787 HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
4788 HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
4789 HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
4790 HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
4791 HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
4792 HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
4793 HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
4794 HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
4795 HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
4796 HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
4797 HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
4798 HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
4799 HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
4800 HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
4801 HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
4802 HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
4803 HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
4804 HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
4805 HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
4806 {
4807 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
4808 /* .name = "Capture Source", */
4809 .name = "Input Source",
4810 .count = 2,
4811 .info = alc883_mux_enum_info,
4812 .get = alc883_mux_enum_get,
4813 .put = alc883_mux_enum_put,
4814 },
4815 { } /* end */
4816};
4817
4818static struct snd_kcontrol_new alc883_chmode_mixer[] = {
4819 {
4820 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
4821 .name = "Channel Mode",
4822 .info = alc_ch_mode_info,
4823 .get = alc_ch_mode_get,
4824 .put = alc_ch_mode_put,
4825 },
4826 { } /* end */
4827};
4828
4829static struct hda_verb alc883_init_verbs[] = {
4830 /* ADC1: mute amp left and right */
4831 {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
4832 {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
4833 /* ADC2: mute amp left and right */
4834 {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
4835 {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
4836 /* Front mixer: unmute input/output amp left and right (volume = 0) */
4837 {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
4838 {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
4839 {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
4840 /* Rear mixer */
4841 {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
4842 {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
4843 {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
4844 /* CLFE mixer */
4845 {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
4846 {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
4847 {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
4848 /* Side mixer */
4849 {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
4850 {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
4851 {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
4852
4853 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4854 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4855 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
4856 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
4857 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
4858
4859 /* Front Pin: output 0 (0x0c) */
4860 {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
4861 {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
4862 {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
4863 /* Rear Pin: output 1 (0x0d) */
4864 {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
4865 {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
4866 {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
4867 /* CLFE Pin: output 2 (0x0e) */
4868 {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
4869 {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
4870 {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
4871 /* Side Pin: output 3 (0x0f) */
4872 {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
4873 {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
4874 {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
4875 /* Mic (rear) pin: input vref at 80% */
4876 {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
4877 {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
4878 /* Front Mic pin: input vref at 80% */
4879 {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
4880 {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
4881 /* Line In pin: input */
4882 {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
4883 {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
4884 /* Line-2 In: Headphone output (output 0 - 0x0c) */
4885 {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
4886 {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
4887 {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
4888 /* CD pin widget for input */
4889 {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
4890
4891 /* FIXME: use matrix-type input source selection */
4892 /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
4893 /* Input mixer2 */
4894 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4895 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4896 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
4897 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
4898 /* Input mixer3 */
4899 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4900 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4901 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
4902 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
4903 { }
4904};
4905
4906/*
4907 * generic initialization of ADC, input mixers and output mixers
4908 */
4909static struct hda_verb alc883_auto_init_verbs[] = {
4910 /*
4911 * Unmute ADC0-2 and set the default input to mic-in
4912 */
4913 {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
4914 {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4915 {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
4916 {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4917
4918 /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
4919 * mixer widget
4920 * Note: PASD motherboards uses the Line In 2 as the input for front panel
4921 * mic (mic 2)
4922 */
4923 /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
4924 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4925 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4926 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
4927 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
4928 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
4929
4930 /*
4931 * Set up output mixers (0x0c - 0x0f)
4932 */
4933 /* set vol=0 to output mixers */
4934 {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
4935 {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
4936 {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
4937 {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
4938 /* set up input amps for analog loopback */
4939 /* Amp Indices: DAC = 0, mixer = 1 */
4940 {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4941 {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4942 {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4943 {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4944 {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4945 {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4946 {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4947 {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4948 {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4949 {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4950
4951 /* FIXME: use matrix-type input source selection */
4952 /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
4953 /* Input mixer1 */
4954 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4955 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4956 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
4957 //{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
4958 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
4959 /* Input mixer2 */
4960 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
4961 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
4962 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
4963 //{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
4964 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
4965
4966 { }
4967};
4968
4969/* capture mixer elements */
4970static struct snd_kcontrol_new alc883_capture_mixer[] = {
4971 HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
4972 HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
4973 HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT),
4974 HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT),
4975 {
4976 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
4977 /* The multiple "Capture Source" controls confuse alsamixer
4978 * So call somewhat different..
4979 * FIXME: the controls appear in the "playback" view!
4980 */
4981 /* .name = "Capture Source", */
4982 .name = "Input Source",
4983 .count = 2,
4984 .info = alc882_mux_enum_info,
4985 .get = alc882_mux_enum_get,
4986 .put = alc882_mux_enum_put,
4987 },
4988 { } /* end */
4989};
4990
4991/* pcm configuration: identiacal with ALC880 */
4992#define alc883_pcm_analog_playback alc880_pcm_analog_playback
4993#define alc883_pcm_analog_capture alc880_pcm_analog_capture
4994#define alc883_pcm_digital_playback alc880_pcm_digital_playback
4995#define alc883_pcm_digital_capture alc880_pcm_digital_capture
4996
4997/*
4998 * configuration and preset
4999 */
5000static struct hda_board_config alc883_cfg_tbl[] = {
5001 { .modelname = "3stack-dig", .config = ALC883_3ST_2ch_DIG },
5002 { .modelname = "6stack-dig", .config = ALC883_6ST_DIG },
5003 { .modelname = "6stack-dig-demo", .config = ALC888_DEMO_BOARD },
5004 { .pci_subvendor = 0x1462, .pci_subdevice = 0x6668,
5005 .config = ALC883_6ST_DIG }, /* MSI */
5006 { .pci_subvendor = 0x105b, .pci_subdevice = 0x6668,
5007 .config = ALC883_6ST_DIG }, /* Foxconn */
5008 { .pci_subvendor = 0x1019, .pci_subdevice = 0x6668,
5009 .config = ALC883_3ST_6ch_DIG }, /* ECS to Intel*/
5010 { .pci_subvendor = 0x108e, .pci_subdevice = 0x534d,
5011 .config = ALC883_3ST_6ch },
5012 { .modelname = "auto", .config = ALC883_AUTO },
5013 {}
5014};
5015
5016static struct alc_config_preset alc883_presets[] = {
5017 [ALC883_3ST_2ch_DIG] = {
5018 .mixers = { alc883_3ST_2ch_mixer },
5019 .init_verbs = { alc883_init_verbs },
5020 .num_dacs = ARRAY_SIZE(alc883_dac_nids),
5021 .dac_nids = alc883_dac_nids,
5022 .dig_out_nid = ALC883_DIGOUT_NID,
5023 .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
5024 .adc_nids = alc883_adc_nids,
5025 .dig_in_nid = ALC883_DIGIN_NID,
5026 .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
5027 .channel_mode = alc883_3ST_2ch_modes,
5028 .input_mux = &alc883_capture_source,
5029 },
5030 [ALC883_3ST_6ch_DIG] = {
5031 .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
5032 .init_verbs = { alc883_init_verbs },
5033 .num_dacs = ARRAY_SIZE(alc883_dac_nids),
5034 .dac_nids = alc883_dac_nids,
5035 .dig_out_nid = ALC883_DIGOUT_NID,
5036 .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
5037 .adc_nids = alc883_adc_nids,
5038 .dig_in_nid = ALC883_DIGIN_NID,
5039 .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
5040 .channel_mode = alc883_3ST_6ch_modes,
5041 .input_mux = &alc883_capture_source,
5042 },
5043 [ALC883_3ST_6ch] = {
5044 .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
5045 .init_verbs = { alc883_init_verbs },
5046 .num_dacs = ARRAY_SIZE(alc883_dac_nids),
5047 .dac_nids = alc883_dac_nids,
5048 .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
5049 .adc_nids = alc883_adc_nids,
5050 .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
5051 .channel_mode = alc883_3ST_6ch_modes,
5052 .input_mux = &alc883_capture_source,
5053 },
5054 [ALC883_6ST_DIG] = {
5055 .mixers = { alc883_base_mixer, alc883_chmode_mixer },
5056 .init_verbs = { alc883_init_verbs },
5057 .num_dacs = ARRAY_SIZE(alc883_dac_nids),
5058 .dac_nids = alc883_dac_nids,
5059 .dig_out_nid = ALC883_DIGOUT_NID,
5060 .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
5061 .adc_nids = alc883_adc_nids,
5062 .dig_in_nid = ALC883_DIGIN_NID,
5063 .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
5064 .channel_mode = alc883_sixstack_modes,
5065 .input_mux = &alc883_capture_source,
5066 },
5067 [ALC888_DEMO_BOARD] = {
5068 .mixers = { alc883_base_mixer, alc883_chmode_mixer },
5069 .init_verbs = { alc883_init_verbs },
5070 .num_dacs = ARRAY_SIZE(alc883_dac_nids),
5071 .dac_nids = alc883_dac_nids,
5072 .dig_out_nid = ALC883_DIGOUT_NID,
5073 .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
5074 .adc_nids = alc883_adc_nids,
5075 .dig_in_nid = ALC883_DIGIN_NID,
5076 .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
5077 .channel_mode = alc883_sixstack_modes,
5078 .input_mux = &alc883_capture_source,
5079 },
5080};
5081
5082
5083/*
5084 * BIOS auto configuration
5085 */
5086static void alc883_auto_set_output_and_unmute(struct hda_codec *codec,
5087 hda_nid_t nid, int pin_type,
5088 int dac_idx)
5089{
5090 /* set as output */
5091 struct alc_spec *spec = codec->spec;
5092 int idx;
5093
5094 if (spec->multiout.dac_nids[dac_idx] == 0x25)
5095 idx = 4;
5096 else
5097 idx = spec->multiout.dac_nids[dac_idx] - 2;
5098
5099 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
5100 pin_type);
5101 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
5102 AMP_OUT_UNMUTE);
5103 snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
5104
5105}
5106
5107static void alc883_auto_init_multi_out(struct hda_codec *codec)
5108{
5109 struct alc_spec *spec = codec->spec;
5110 int i;
5111
5112 for (i = 0; i <= HDA_SIDE; i++) {
5113 hda_nid_t nid = spec->autocfg.line_out_pins[i];
5114 if (nid)
5115 alc883_auto_set_output_and_unmute(codec, nid, PIN_OUT, i);
5116 }
5117}
5118
5119static void alc883_auto_init_hp_out(struct hda_codec *codec)
5120{
5121 struct alc_spec *spec = codec->spec;
5122 hda_nid_t pin;
5123
5124 pin = spec->autocfg.hp_pin;
5125 if (pin) /* connect to front */
5126 /* use dac 0 */
5127 alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
5128}
5129
5130#define alc883_is_input_pin(nid) alc880_is_input_pin(nid)
5131#define ALC883_PIN_CD_NID ALC880_PIN_CD_NID
5132
5133static void alc883_auto_init_analog_input(struct hda_codec *codec)
5134{
5135 struct alc_spec *spec = codec->spec;
5136 int i;
5137
5138 for (i = 0; i < AUTO_PIN_LAST; i++) {
5139 hda_nid_t nid = spec->autocfg.input_pins[i];
5140 if (alc883_is_input_pin(nid)) {
5141 snd_hda_codec_write(codec, nid, 0,
5142 AC_VERB_SET_PIN_WIDGET_CONTROL,
5143 (i <= AUTO_PIN_FRONT_MIC ?
5144 PIN_VREF80 : PIN_IN));
5145 if (nid != ALC883_PIN_CD_NID)
5146 snd_hda_codec_write(codec, nid, 0,
5147 AC_VERB_SET_AMP_GAIN_MUTE,
5148 AMP_OUT_MUTE);
5149 }
5150 }
5151}
5152
5153/* almost identical with ALC880 parser... */
5154static int alc883_parse_auto_config(struct hda_codec *codec)
5155{
5156 struct alc_spec *spec = codec->spec;
5157 int err = alc880_parse_auto_config(codec);
5158
5159 if (err < 0)
5160 return err;
5161 else if (err > 0)
5162 /* hack - override the init verbs */
5163 spec->init_verbs[0] = alc883_auto_init_verbs;
5164 spec->mixers[spec->num_mixers] = alc883_capture_mixer;
5165 spec->num_mixers++;
5166 return err;
5167}
5168
5169/* additional initialization for auto-configuration model */
5170static void alc883_auto_init(struct hda_codec *codec)
5171{
5172 alc883_auto_init_multi_out(codec);
5173 alc883_auto_init_hp_out(codec);
5174 alc883_auto_init_analog_input(codec);
5175}
5176
5177static int patch_alc883(struct hda_codec *codec)
5178{
5179 struct alc_spec *spec;
5180 int err, board_config;
5181
5182 spec = kzalloc(sizeof(*spec), GFP_KERNEL);
5183 if (spec == NULL)
5184 return -ENOMEM;
5185
5186 codec->spec = spec;
5187
5188 board_config = snd_hda_check_board_config(codec, alc883_cfg_tbl);
5189 if (board_config < 0 || board_config >= ALC883_MODEL_LAST) {
5190 printk(KERN_INFO "hda_codec: Unknown model for ALC883, "
5191 "trying auto-probe from BIOS...\n");
5192 board_config = ALC883_AUTO;
5193 }
5194
5195 if (board_config == ALC883_AUTO) {
5196 /* automatic parse from the BIOS config */
5197 err = alc883_parse_auto_config(codec);
5198 if (err < 0) {
5199 alc_free(codec);
5200 return err;
5201 } else if (! err) {
5202 printk(KERN_INFO
5203 "hda_codec: Cannot set up configuration "
5204 "from BIOS. Using base mode...\n");
5205 board_config = ALC883_3ST_2ch_DIG;
5206 }
5207 }
5208
5209 if (board_config != ALC883_AUTO)
5210 setup_preset(spec, &alc883_presets[board_config]);
5211
5212 spec->stream_name_analog = "ALC883 Analog";
5213 spec->stream_analog_playback = &alc883_pcm_analog_playback;
5214 spec->stream_analog_capture = &alc883_pcm_analog_capture;
5215
5216 spec->stream_name_digital = "ALC883 Digital";
5217 spec->stream_digital_playback = &alc883_pcm_digital_playback;
5218 spec->stream_digital_capture = &alc883_pcm_digital_capture;
5219
5220 spec->adc_nids = alc883_adc_nids;
5221 spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
5222
5223 codec->patch_ops = alc_patch_ops;
5224 if (board_config == ALC883_AUTO)
5225 spec->init_hook = alc883_auto_init;
5226
5227 return 0;
5228}
5229
5230/*
4512 * ALC262 support 5231 * ALC262 support
4513 */ 5232 */
4514 5233
@@ -4542,6 +5261,28 @@ static struct snd_kcontrol_new alc262_base_mixer[] = {
4542 { } /* end */ 5261 { } /* end */
4543}; 5262};
4544 5263
5264static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
5265 HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
5266 HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
5267 HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
5268 HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
5269 HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
5270
5271 HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
5272 HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
5273 HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
5274 HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
5275 HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
5276 HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
5277 HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
5278 HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
5279 HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
5280 HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT),
5281 HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
5282 HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
5283 { } /* end */
5284};
5285
4545#define alc262_capture_mixer alc882_capture_mixer 5286#define alc262_capture_mixer alc882_capture_mixer
4546#define alc262_capture_alt_mixer alc882_capture_alt_mixer 5287#define alc262_capture_alt_mixer alc882_capture_alt_mixer
4547 5288
@@ -4645,6 +5386,17 @@ static struct hda_input_mux alc262_fujitsu_capture_source = {
4645 }, 5386 },
4646}; 5387};
4647 5388
5389static struct hda_input_mux alc262_HP_capture_source = {
5390 .num_items = 5,
5391 .items = {
5392 { "Mic", 0x0 },
5393 { "Front Mic", 0x3 },
5394 { "Line", 0x2 },
5395 { "CD", 0x4 },
5396 { "AUX IN", 0x6 },
5397 },
5398};
5399
4648/* mute/unmute internal speaker according to the hp jack and mute state */ 5400/* mute/unmute internal speaker according to the hp jack and mute state */
4649static void alc262_fujitsu_automute(struct hda_codec *codec, int force) 5401static void alc262_fujitsu_automute(struct hda_codec *codec, int force)
4650{ 5402{
@@ -4868,6 +5620,93 @@ static struct hda_verb alc262_volume_init_verbs[] = {
4868 { } 5620 { }
4869}; 5621};
4870 5622
5623static struct hda_verb alc262_HP_BPC_init_verbs[] = {
5624 /*
5625 * Unmute ADC0-2 and set the default input to mic-in
5626 */
5627 {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
5628 {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
5629 {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
5630 {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
5631 {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
5632 {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
5633
5634 /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
5635 * mixer widget
5636 * Note: PASD motherboards uses the Line In 2 as the input for front panel
5637 * mic (mic 2)
5638 */
5639 /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
5640 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
5641 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
5642 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
5643 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
5644 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
5645 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
5646 {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)},
5647
5648 /*
5649 * Set up output mixers (0x0c - 0x0e)
5650 */
5651 /* set vol=0 to output mixers */
5652 {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
5653 {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
5654 {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
5655
5656 /* set up input amps for analog loopback */
5657 /* Amp Indices: DAC = 0, mixer = 1 */
5658 {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
5659 {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
5660 {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
5661 {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
5662 {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
5663 {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
5664
5665 {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
5666 {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
5667 {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
5668
5669 {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
5670 {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
5671
5672 {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
5673 {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
5674
5675 {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
5676 {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
5677 {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
5678 {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
5679 {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
5680
5681 {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
5682 {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
5683 {0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
5684 {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x7023 },
5685 {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
5686 {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000 },
5687
5688
5689 /* FIXME: use matrix-type input source selection */
5690 /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
5691 /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
5692 {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
5693 {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
5694 {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
5695 {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
5696 /* Input mixer2 */
5697 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
5698 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
5699 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
5700 {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
5701 /* Input mixer3 */
5702 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
5703 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
5704 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
5705 {0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
5706
5707 { }
5708};
5709
4871/* pcm configuration: identiacal with ALC880 */ 5710/* pcm configuration: identiacal with ALC880 */
4872#define alc262_pcm_analog_playback alc880_pcm_analog_playback 5711#define alc262_pcm_analog_playback alc880_pcm_analog_playback
4873#define alc262_pcm_analog_capture alc880_pcm_analog_capture 5712#define alc262_pcm_analog_capture alc880_pcm_analog_capture
@@ -4928,7 +5767,16 @@ static void alc262_auto_init(struct hda_codec *codec)
4928static struct hda_board_config alc262_cfg_tbl[] = { 5767static struct hda_board_config alc262_cfg_tbl[] = {
4929 { .modelname = "basic", .config = ALC262_BASIC }, 5768 { .modelname = "basic", .config = ALC262_BASIC },
4930 { .modelname = "fujitsu", .config = ALC262_FUJITSU }, 5769 { .modelname = "fujitsu", .config = ALC262_FUJITSU },
4931 { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397, .config = ALC262_FUJITSU }, 5770 { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397,
5771 .config = ALC262_FUJITSU },
5772 { .pci_subvendor = 0x103c, .pci_subdevice = 0x208c,
5773 .config = ALC262_HP_BPC }, /* xw4400 */
5774 { .pci_subvendor = 0x103c, .pci_subdevice = 0x3014,
5775 .config = ALC262_HP_BPC }, /* xw6400 */
5776 { .pci_subvendor = 0x103c, .pci_subdevice = 0x3015,
5777 .config = ALC262_HP_BPC }, /* xw8400 */
5778 { .pci_subvendor = 0x103c, .pci_subdevice = 0x12fe,
5779 .config = ALC262_HP_BPC }, /* xw9400 */
4932 { .modelname = "auto", .config = ALC262_AUTO }, 5780 { .modelname = "auto", .config = ALC262_AUTO },
4933 {} 5781 {}
4934}; 5782};
@@ -4956,6 +5804,16 @@ static struct alc_config_preset alc262_presets[] = {
4956 .input_mux = &alc262_fujitsu_capture_source, 5804 .input_mux = &alc262_fujitsu_capture_source,
4957 .unsol_event = alc262_fujitsu_unsol_event, 5805 .unsol_event = alc262_fujitsu_unsol_event,
4958 }, 5806 },
5807 [ALC262_HP_BPC] = {
5808 .mixers = { alc262_HP_BPC_mixer },
5809 .init_verbs = { alc262_HP_BPC_init_verbs },
5810 .num_dacs = ARRAY_SIZE(alc262_dac_nids),
5811 .dac_nids = alc262_dac_nids,
5812 .hp_nid = 0x03,
5813 .num_channel_mode = ARRAY_SIZE(alc262_modes),
5814 .channel_mode = alc262_modes,
5815 .input_mux = &alc262_HP_capture_source,
5816 },
4959}; 5817};
4960 5818
4961static int patch_alc262(struct hda_codec *codec) 5819static int patch_alc262(struct hda_codec *codec)
@@ -4981,8 +5839,10 @@ static int patch_alc262(struct hda_codec *codec)
4981#endif 5839#endif
4982 5840
4983 board_config = snd_hda_check_board_config(codec, alc262_cfg_tbl); 5841 board_config = snd_hda_check_board_config(codec, alc262_cfg_tbl);
5842
4984 if (board_config < 0 || board_config >= ALC262_MODEL_LAST) { 5843 if (board_config < 0 || board_config >= ALC262_MODEL_LAST) {
4985 printk(KERN_INFO "hda_codec: Unknown model for ALC262, trying auto-probe from BIOS...\n"); 5844 printk(KERN_INFO "hda_codec: Unknown model for ALC262, "
5845 "trying auto-probe from BIOS...\n");
4986 board_config = ALC262_AUTO; 5846 board_config = ALC262_AUTO;
4987 } 5847 }
4988 5848
@@ -4993,7 +5853,9 @@ static int patch_alc262(struct hda_codec *codec)
4993 alc_free(codec); 5853 alc_free(codec);
4994 return err; 5854 return err;
4995 } else if (! err) { 5855 } else if (! err) {
4996 printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n"); 5856 printk(KERN_INFO
5857 "hda_codec: Cannot set up configuration "
5858 "from BIOS. Using base mode...\n");
4997 board_config = ALC262_BASIC; 5859 board_config = ALC262_BASIC;
4998 } 5860 }
4999 } 5861 }
@@ -5034,7 +5896,6 @@ static int patch_alc262(struct hda_codec *codec)
5034 return 0; 5896 return 0;
5035} 5897}
5036 5898
5037
5038/* 5899/*
5039 * ALC861 channel source setting (2/6 channel selection for 3-stack) 5900 * ALC861 channel source setting (2/6 channel selection for 3-stack)
5040 */ 5901 */
@@ -5049,9 +5910,11 @@ static struct hda_verb alc861_threestack_ch2_init[] = {
5049 /* set pin widget 18h (mic1/2) for input, for mic also enable the vref */ 5910 /* set pin widget 18h (mic1/2) for input, for mic also enable the vref */
5050 { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, 5911 { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
5051 5912
5052 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c }, 5913 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c },
5053 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, //mic 5914#if 0
5054 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, //line in 5915 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8)) }, /*mic*/
5916 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8)) }, /*line-in*/
5917#endif
5055 { } /* end */ 5918 { } /* end */
5056}; 5919};
5057/* 5920/*
@@ -5065,11 +5928,13 @@ static struct hda_verb alc861_threestack_ch6_init[] = {
5065 { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, 5928 { 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
5066 5929
5067 { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 }, 5930 { 0x0c, AC_VERB_SET_CONNECT_SEL, 0x00 },
5068 { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 }, 5931 { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00 },
5069 5932
5070 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 }, 5933 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
5071 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, //mic 5934#if 0
5072 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, //line in 5935 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8)) }, /*mic*/
5936 { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8)) }, /*line in*/
5937#endif
5073 { } /* end */ 5938 { } /* end */
5074}; 5939};
5075 5940
@@ -5353,6 +6218,11 @@ static hda_nid_t alc861_dac_nids[4] = {
5353 0x03, 0x06, 0x05, 0x04 6218 0x03, 0x06, 0x05, 0x04
5354}; 6219};
5355 6220
6221static hda_nid_t alc660_dac_nids[3] = {
6222 /* front, clfe, surround */
6223 0x03, 0x05, 0x06
6224};
6225
5356static hda_nid_t alc861_adc_nids[1] = { 6226static hda_nid_t alc861_adc_nids[1] = {
5357 /* ADC0-2 */ 6227 /* ADC0-2 */
5358 0x08, 6228 0x08,
@@ -5605,7 +6475,10 @@ static void alc861_auto_init(struct hda_codec *codec)
5605 */ 6475 */
5606static struct hda_board_config alc861_cfg_tbl[] = { 6476static struct hda_board_config alc861_cfg_tbl[] = {
5607 { .modelname = "3stack", .config = ALC861_3ST }, 6477 { .modelname = "3stack", .config = ALC861_3ST },
5608 { .pci_subvendor = 0x8086, .pci_subdevice = 0xd600, .config = ALC861_3ST }, 6478 { .pci_subvendor = 0x8086, .pci_subdevice = 0xd600,
6479 .config = ALC861_3ST },
6480 { .pci_subvendor = 0x1043, .pci_subdevice = 0x81e7,
6481 .config = ALC660_3ST },
5609 { .modelname = "3stack-dig", .config = ALC861_3ST_DIG }, 6482 { .modelname = "3stack-dig", .config = ALC861_3ST_DIG },
5610 { .modelname = "6stack-dig", .config = ALC861_6ST_DIG }, 6483 { .modelname = "6stack-dig", .config = ALC861_6ST_DIG },
5611 { .modelname = "auto", .config = ALC861_AUTO }, 6484 { .modelname = "auto", .config = ALC861_AUTO },
@@ -5648,6 +6521,17 @@ static struct alc_config_preset alc861_presets[] = {
5648 .adc_nids = alc861_adc_nids, 6521 .adc_nids = alc861_adc_nids,
5649 .input_mux = &alc861_capture_source, 6522 .input_mux = &alc861_capture_source,
5650 }, 6523 },
6524 [ALC660_3ST] = {
6525 .mixers = { alc861_3ST_mixer },
6526 .init_verbs = { alc861_threestack_init_verbs },
6527 .num_dacs = ARRAY_SIZE(alc660_dac_nids),
6528 .dac_nids = alc660_dac_nids,
6529 .num_channel_mode = ARRAY_SIZE(alc861_threestack_modes),
6530 .channel_mode = alc861_threestack_modes,
6531 .num_adc_nids = ARRAY_SIZE(alc861_adc_nids),
6532 .adc_nids = alc861_adc_nids,
6533 .input_mux = &alc861_capture_source,
6534 },
5651}; 6535};
5652 6536
5653 6537
@@ -5664,8 +6548,10 @@ static int patch_alc861(struct hda_codec *codec)
5664 codec->spec = spec; 6548 codec->spec = spec;
5665 6549
5666 board_config = snd_hda_check_board_config(codec, alc861_cfg_tbl); 6550 board_config = snd_hda_check_board_config(codec, alc861_cfg_tbl);
6551
5667 if (board_config < 0 || board_config >= ALC861_MODEL_LAST) { 6552 if (board_config < 0 || board_config >= ALC861_MODEL_LAST) {
5668 printk(KERN_INFO "hda_codec: Unknown model for ALC861, trying auto-probe from BIOS...\n"); 6553 printk(KERN_INFO "hda_codec: Unknown model for ALC861, "
6554 "trying auto-probe from BIOS...\n");
5669 board_config = ALC861_AUTO; 6555 board_config = ALC861_AUTO;
5670 } 6556 }
5671 6557
@@ -5676,7 +6562,9 @@ static int patch_alc861(struct hda_codec *codec)
5676 alc_free(codec); 6562 alc_free(codec);
5677 return err; 6563 return err;
5678 } else if (! err) { 6564 } else if (! err) {
5679 printk(KERN_INFO "hda_codec: Cannot set up configuration from BIOS. Using base mode...\n"); 6565 printk(KERN_INFO
6566 "hda_codec: Cannot set up configuration "
6567 "from BIOS. Using base mode...\n");
5680 board_config = ALC861_3ST_DIG; 6568 board_config = ALC861_3ST_DIG;
5681 } 6569 }
5682 } 6570 }
@@ -5707,8 +6595,12 @@ struct hda_codec_preset snd_hda_preset_realtek[] = {
5707 { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, 6595 { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
5708 { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, 6596 { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
5709 { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, 6597 { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
5710 { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 }, 6598 { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 },
5711 { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, 6599 { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
5712 { .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 }, 6600 { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
6601 { .id = 0x10ec0861, .rev = 0x100300, .name = "ALC861",
6602 .patch = patch_alc861 },
6603 { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
6604 .patch = patch_alc861 },
5713 {} /* terminator */ 6605 {} /* terminator */
5714}; 6606};
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 36f199442fdc..fb4bed0759d1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -42,6 +42,9 @@
42#define STAC_D945GTP3 1 42#define STAC_D945GTP3 1
43#define STAC_D945GTP5 2 43#define STAC_D945GTP5 2
44#define STAC_MACMINI 3 44#define STAC_MACMINI 3
45#define STAC_D965_2112 4
46#define STAC_D965_284B 5
47#define STAC_922X_MODELS 6 /* number of 922x models */
45 48
46struct sigmatel_spec { 49struct sigmatel_spec {
47 struct snd_kcontrol_new *mixers[4]; 50 struct snd_kcontrol_new *mixers[4];
@@ -107,10 +110,24 @@ static hda_nid_t stac922x_adc_nids[2] = {
107 0x06, 0x07, 110 0x06, 0x07,
108}; 111};
109 112
113static hda_nid_t stac9227_adc_nids[2] = {
114 0x07, 0x08,
115};
116
117#if 0
118static hda_nid_t d965_2112_dac_nids[3] = {
119 0x02, 0x03, 0x05,
120};
121#endif
122
110static hda_nid_t stac922x_mux_nids[2] = { 123static hda_nid_t stac922x_mux_nids[2] = {
111 0x12, 0x13, 124 0x12, 0x13,
112}; 125};
113 126
127static hda_nid_t stac9227_mux_nids[2] = {
128 0x15, 0x16,
129};
130
114static hda_nid_t stac927x_adc_nids[3] = { 131static hda_nid_t stac927x_adc_nids[3] = {
115 0x07, 0x08, 0x09 132 0x07, 0x08, 0x09
116}; 133};
@@ -173,6 +190,24 @@ static struct hda_verb stac922x_core_init[] = {
173 {} 190 {}
174}; 191};
175 192
193static struct hda_verb stac9227_core_init[] = {
194 /* set master volume and direct control */
195 { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
196 /* unmute node 0x1b */
197 { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
198 {}
199};
200
201static struct hda_verb d965_2112_core_init[] = {
202 /* set master volume and direct control */
203 { 0x16, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
204 /* unmute node 0x1b */
205 { 0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
206 /* select node 0x03 as DAC */
207 { 0x0b, AC_VERB_SET_CONNECT_SEL, 0x01},
208 {}
209};
210
176static struct hda_verb stac927x_core_init[] = { 211static struct hda_verb stac927x_core_init[] = {
177 /* set master volume and direct control */ 212 /* set master volume and direct control */
178 { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, 213 { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -212,6 +247,21 @@ static struct snd_kcontrol_new stac922x_mixer[] = {
212 { } /* end */ 247 { } /* end */
213}; 248};
214 249
250/* This needs to be generated dynamically based on sequence */
251static struct snd_kcontrol_new stac9227_mixer[] = {
252 {
253 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
254 .name = "Input Source",
255 .count = 1,
256 .info = stac92xx_mux_enum_info,
257 .get = stac92xx_mux_enum_get,
258 .put = stac92xx_mux_enum_put,
259 },
260 HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
261 HDA_CODEC_MUTE("Capture Switch", 0x1b, 0x0, HDA_OUTPUT),
262 { } /* end */
263};
264
215static snd_kcontrol_new_t stac927x_mixer[] = { 265static snd_kcontrol_new_t stac927x_mixer[] = {
216 { 266 {
217 .iface = SNDRV_CTL_ELEM_IFACE_MIXER, 267 .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -291,11 +341,17 @@ static unsigned int d945gtp5_pin_configs[10] = {
291 0x02a19320, 0x40000100, 341 0x02a19320, 0x40000100,
292}; 342};
293 343
294static unsigned int *stac922x_brd_tbl[] = { 344static unsigned int d965_2112_pin_configs[10] = {
295 ref922x_pin_configs, 345 0x0221401f, 0x40000100, 0x40000100, 0x01014011,
296 d945gtp3_pin_configs, 346 0x01a19021, 0x01813024, 0x01452130, 0x40000100,
297 d945gtp5_pin_configs, 347 0x02a19320, 0x40000100,
298 NULL, /* STAC_MACMINI */ 348};
349
350static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
351 [STAC_REF] = ref922x_pin_configs,
352 [STAC_D945GTP3] = d945gtp3_pin_configs,
353 [STAC_D945GTP5] = d945gtp5_pin_configs,
354 [STAC_D965_2112] = d965_2112_pin_configs,
299}; 355};
300 356
301static struct hda_board_config stac922x_cfg_tbl[] = { 357static struct hda_board_config stac922x_cfg_tbl[] = {
@@ -330,6 +386,12 @@ static struct hda_board_config stac922x_cfg_tbl[] = {
330 { .pci_subvendor = 0x8384, 386 { .pci_subvendor = 0x8384,
331 .pci_subdevice = 0x7680, 387 .pci_subdevice = 0x7680,
332 .config = STAC_MACMINI }, /* Apple Mac Mini (early 2006) */ 388 .config = STAC_MACMINI }, /* Apple Mac Mini (early 2006) */
389 { .pci_subvendor = PCI_VENDOR_ID_INTEL,
390 .pci_subdevice = 0x2112,
391 .config = STAC_D965_2112 },
392 { .pci_subvendor = PCI_VENDOR_ID_INTEL,
393 .pci_subdevice = 0x284b,
394 .config = STAC_D965_284B },
333 {} /* terminator */ 395 {} /* terminator */
334}; 396};
335 397
@@ -713,7 +775,8 @@ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cf
713 * A and B is not supported. 775 * A and B is not supported.
714 */ 776 */
715/* fill in the dac_nids table from the parsed pin configuration */ 777/* fill in the dac_nids table from the parsed pin configuration */
716static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, const struct auto_pin_cfg *cfg) 778static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec,
779 const struct auto_pin_cfg *cfg)
717{ 780{
718 struct sigmatel_spec *spec = codec->spec; 781 struct sigmatel_spec *spec = codec->spec;
719 hda_nid_t nid; 782 hda_nid_t nid;
@@ -732,10 +795,13 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, const struct aut
732} 795}
733 796
734/* add playback controls from the parsed DAC table */ 797/* add playback controls from the parsed DAC table */
735static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec, const struct auto_pin_cfg *cfg) 798static int stac92xx_auto_create_multi_out_ctls(struct sigmatel_spec *spec,
799 const struct auto_pin_cfg *cfg)
736{ 800{
737 char name[32]; 801 char name[32];
738 static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; 802 static const char *chname[4] = {
803 "Front", "Surround", NULL /*CLFE*/, "Side"
804 };
739 hda_nid_t nid; 805 hda_nid_t nid;
740 int i, err; 806 int i, err;
741 807
@@ -893,10 +959,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
893 return err; 959 return err;
894 if (! spec->autocfg.line_outs) 960 if (! spec->autocfg.line_outs)
895 return 0; /* can't find valid pin config */ 961 return 0; /* can't find valid pin config */
962
896 if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) 963 if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0)
897 return err; 964 return err;
898 if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) 965 if (spec->multiout.num_dacs == 0)
899 return err; 966 if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0)
967 return err;
900 968
901 if ((err = stac92xx_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || 969 if ((err = stac92xx_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 ||
902 (err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg)) < 0 || 970 (err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg)) < 0 ||
@@ -1194,7 +1262,8 @@ static int patch_stac922x(struct hda_codec *codec)
1194 codec->spec = spec; 1262 codec->spec = spec;
1195 spec->board_config = snd_hda_check_board_config(codec, stac922x_cfg_tbl); 1263 spec->board_config = snd_hda_check_board_config(codec, stac922x_cfg_tbl);
1196 if (spec->board_config < 0) 1264 if (spec->board_config < 0)
1197 snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, using BIOS defaults\n"); 1265 snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
1266 "using BIOS defaults\n");
1198 else if (stac922x_brd_tbl[spec->board_config] != NULL) { 1267 else if (stac922x_brd_tbl[spec->board_config] != NULL) {
1199 spec->num_pins = 10; 1268 spec->num_pins = 10;
1200 spec->pin_nids = stac922x_pin_nids; 1269 spec->pin_nids = stac922x_pin_nids;
@@ -1210,6 +1279,25 @@ static int patch_stac922x(struct hda_codec *codec)
1210 spec->mixer = stac922x_mixer; 1279 spec->mixer = stac922x_mixer;
1211 1280
1212 spec->multiout.dac_nids = spec->dac_nids; 1281 spec->multiout.dac_nids = spec->dac_nids;
1282
1283 switch (spec->board_config) {
1284 case STAC_D965_2112:
1285 spec->adc_nids = stac9227_adc_nids;
1286 spec->mux_nids = stac9227_mux_nids;
1287#if 0
1288 spec->multiout.dac_nids = d965_2112_dac_nids;
1289 spec->multiout.num_dacs = ARRAY_SIZE(d965_2112_dac_nids);
1290#endif
1291 spec->init = d965_2112_core_init;
1292 spec->mixer = stac9227_mixer;
1293 break;
1294 case STAC_D965_284B:
1295 spec->adc_nids = stac9227_adc_nids;
1296 spec->mux_nids = stac9227_mux_nids;
1297 spec->init = stac9227_core_init;
1298 spec->mixer = stac9227_mixer;
1299 break;
1300 }
1213 1301
1214 err = stac92xx_parse_auto_config(codec, 0x08, 0x09); 1302 err = stac92xx_parse_auto_config(codec, 0x08, 0x09);
1215 if (err < 0) { 1303 if (err < 0) {
diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c
index b5754b32b802..fec9440cb310 100644
--- a/sound/pci/ice1712/revo.c
+++ b/sound/pci/ice1712/revo.c
@@ -87,12 +87,25 @@ static void revo_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate)
87 * initialize the chips on M-Audio Revolution cards 87 * initialize the chips on M-Audio Revolution cards
88 */ 88 */
89 89
90static unsigned int revo71_num_stereo_front[] = {2};
91static char *revo71_channel_names_front[] = {"PCM Playback Volume"};
92
93static unsigned int revo71_num_stereo_surround[] = {1, 1, 2, 2};
94static char *revo71_channel_names_surround[] = {"PCM Center Playback Volume", "PCM LFE Playback Volume",
95 "PCM Side Playback Volume", "PCM Rear Playback Volume"};
96
97static unsigned int revo51_num_stereo[] = {2, 1, 1, 2};
98static char *revo51_channel_names[] = {"PCM Playback Volume", "PCM Center Playback Volume",
99 "PCM LFE Playback Volume", "PCM Rear Playback Volume"};
100
90static struct snd_akm4xxx akm_revo_front __devinitdata = { 101static struct snd_akm4xxx akm_revo_front __devinitdata = {
91 .type = SND_AK4381, 102 .type = SND_AK4381,
92 .num_dacs = 2, 103 .num_dacs = 2,
93 .ops = { 104 .ops = {
94 .set_rate_val = revo_set_rate_val 105 .set_rate_val = revo_set_rate_val
95 } 106 },
107 .num_stereo = revo71_num_stereo_front,
108 .channel_names = revo71_channel_names_front
96}; 109};
97 110
98static struct snd_ak4xxx_private akm_revo_front_priv __devinitdata = { 111static struct snd_ak4xxx_private akm_revo_front_priv __devinitdata = {
@@ -113,7 +126,9 @@ static struct snd_akm4xxx akm_revo_surround __devinitdata = {
113 .num_dacs = 6, 126 .num_dacs = 6,
114 .ops = { 127 .ops = {
115 .set_rate_val = revo_set_rate_val 128 .set_rate_val = revo_set_rate_val
116 } 129 },
130 .num_stereo = revo71_num_stereo_surround,
131 .channel_names = revo71_channel_names_surround
117}; 132};
118 133
119static struct snd_ak4xxx_private akm_revo_surround_priv __devinitdata = { 134static struct snd_ak4xxx_private akm_revo_surround_priv __devinitdata = {
@@ -133,7 +148,9 @@ static struct snd_akm4xxx akm_revo51 __devinitdata = {
133 .num_dacs = 6, 148 .num_dacs = 6,
134 .ops = { 149 .ops = {
135 .set_rate_val = revo_set_rate_val 150 .set_rate_val = revo_set_rate_val
136 } 151 },
152 .num_stereo = revo51_num_stereo,
153 .channel_names = revo51_channel_names
137}; 154};
138 155
139static struct snd_ak4xxx_private akm_revo51_priv __devinitdata = { 156static struct snd_ak4xxx_private akm_revo51_priv __devinitdata = {
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 627de9525a32..d32d83d970cc 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -3096,6 +3096,32 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev)
3096} 3096}
3097 3097
3098/* 3098/*
3099 * C-Media CM106/CM106+ have four 16-bit internal registers that are nicely
3100 * documented in the device's data sheet.
3101 */
3102static int snd_usb_cm106_write_int_reg(struct usb_device *dev, int reg, u16 value)
3103{
3104 u8 buf[4];
3105 buf[0] = 0x20;
3106 buf[1] = value & 0xff;
3107 buf[2] = (value >> 8) & 0xff;
3108 buf[3] = reg;
3109 return snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), USB_REQ_SET_CONFIGURATION,
3110 USB_DIR_OUT | USB_TYPE_CLASS | USB_RECIP_ENDPOINT,
3111 0, 0, &buf, 4, 1000);
3112}
3113
3114static int snd_usb_cm106_boot_quirk(struct usb_device *dev)
3115{
3116 /*
3117 * Enable line-out driver mode, set headphone source to front
3118 * channels, enable stereo mic.
3119 */
3120 return snd_usb_cm106_write_int_reg(dev, 2, 0x8004);
3121}
3122
3123
3124/*
3099 * Setup quirks 3125 * Setup quirks
3100 */ 3126 */
3101#define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */ 3127#define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */
@@ -3365,6 +3391,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
3365 goto __err_val; 3391 goto __err_val;
3366 } 3392 }
3367 3393
3394 /* C-Media CM106 / Turtle Beach Audio Advantage Roadie */
3395 if (id == USB_ID(0x10f5, 0x0200)) {
3396 if (snd_usb_cm106_boot_quirk(dev) < 0)
3397 goto __err_val;
3398 }
3399
3368 /* 3400 /*
3369 * found a config. now register to ALSA 3401 * found a config. now register to ALSA
3370 */ 3402 */
generated by cgit v1.2.2 (git 2.25.0) at 2024-11-10 04:48:48 -0500