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Message-ID: <s5hws767xxl.wl%tiwai@suse.de>
Date: Sun, 21 Jun 2009 11:12:22 +0200
From: Takashi Iwai <tiwai@...e.de>
To: Linus Torvalds <torvalds@...ux-foundation.org>
Cc: Andrew Morton <akpm@...ux-foundation.org>, perex@...ex.cz,
linux-kernel@...r.kernel.org
Subject: [GIT PULL] sound fixes for 2.6.31
Linus,
please pull ALSA fixes for v2.6.31 from:
git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git for-linus
containing the following changes.
Thanks!
Takashi
===
Daniel Mack (1):
ALSA: snd_usb_caiaq: fix legacy input streaming
Sasha Alexandr (3):
ALSA: HDA - Add pci-quirk for MSI MS-7350 motherboard.
ALSA: HDA - Name-fixes in code (tagra/targa)
ALSA: HDA - Correct trivial typos in comments.
Takashi Iwai (3):
ALSA: hda - Fix memory leak at codec creation
ASoC: Kill BUS_ID_SIZE
ALSA: hda - Add model=6530g option
Tony Vroon (1):
ALSA: hda - Acer Inspire 6530G model for Realtek ALC888
---
Documentation/sound/alsa/HD-Audio-Models.txt | 1 +
sound/pci/hda/hda_codec.c | 2 -
sound/pci/hda/patch_realtek.c | 134 ++++++++++++++++++++------
sound/soc/txx9/txx9aclc.c | 4 +-
sound/usb/caiaq/audio.c | 5 +-
sound/usb/caiaq/device.c | 2 +-
6 files changed, 110 insertions(+), 38 deletions(-)
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index de8e10a..0d8d235 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -139,6 +139,7 @@ ALC883/888
acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc)
acer-aspire Acer Aspire 9810
acer-aspire-4930g Acer Aspire 4930G
+ acer-aspire-6530g Acer Aspire 6530G
acer-aspire-8930g Acer Aspire 8930G
medion Medion Laptops
medion-md2 Medion MD2
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 562403a..462e2ce 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -972,8 +972,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
snd_hda_codec_read(codec, nid, 0,
AC_VERB_GET_SUBSYSTEM_ID, 0);
}
- if (bus->modelname)
- codec->modelname = kstrdup(bus->modelname, GFP_KERNEL);
/* power-up all before initialization */
hda_set_power_state(codec,
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d22b260..bf4b78a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -224,6 +224,7 @@ enum {
ALC883_ACER,
ALC883_ACER_ASPIRE,
ALC888_ACER_ASPIRE_4930G,
+ ALC888_ACER_ASPIRE_6530G,
ALC888_ACER_ASPIRE_8930G,
ALC883_MEDION,
ALC883_MEDION_MD2,
@@ -970,7 +971,7 @@ static void alc_automute_pin(struct hda_codec *codec)
}
}
-#if 0 /* it's broken in some acses -- temporarily disabled */
+#if 0 /* it's broken in some cases -- temporarily disabled */
static void alc_mic_automute(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
@@ -1170,7 +1171,7 @@ static int alc_subsystem_id(struct hda_codec *codec,
/* invalid SSID, check the special NID pin defcfg instead */
/*
- * 31~30 : port conetcivity
+ * 31~30 : port connectivity
* 29~21 : reserve
* 20 : PCBEEP input
* 19~16 : Check sum (15:1)
@@ -1471,6 +1472,25 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = {
};
/*
+ * ALC888 Acer Aspire 6530G model
+ */
+
+static struct hda_verb alc888_acer_aspire_6530g_verbs[] = {
+/* Bias voltage on for external mic port */
+ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80},
+/* Enable unsolicited event for HP jack */
+ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+/* Enable speaker output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+/* Enable headphone output */
+ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP},
+ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+ { }
+};
+
+/*
* ALC889 Acer Aspire 8930G model
*/
@@ -1544,6 +1564,25 @@ static struct hda_input_mux alc888_2_capture_sources[2] = {
}
};
+static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = {
+ /* Interal mic only available on one ADC */
+ {
+ .num_items = 3,
+ .items = {
+ { "Ext Mic", 0x0 },
+ { "CD", 0x4 },
+ { "Int Mic", 0xb },
+ },
+ },
+ {
+ .num_items = 2,
+ .items = {
+ { "Ext Mic", 0x0 },
+ { "CD", 0x4 },
+ },
+ }
+};
+
static struct hda_input_mux alc889_capture_sources[3] = {
/* Digital mic only available on first "ADC" */
{
@@ -6347,7 +6386,7 @@ static struct hda_channel_mode alc882_sixstack_modes[2] = {
};
/*
- * macbook pro ALC885 can switch LineIn to LineOut without loosing Mic
+ * macbook pro ALC885 can switch LineIn to LineOut without losing Mic
*/
/*
@@ -7047,7 +7086,7 @@ static struct hda_verb alc882_auto_init_verbs[] = {
#define alc882_loopbacks alc880_loopbacks
#endif
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
#define alc882_pcm_analog_playback alc880_pcm_analog_playback
#define alc882_pcm_analog_capture alc880_pcm_analog_capture
#define alc882_pcm_digital_playback alc880_pcm_digital_playback
@@ -8068,7 +8107,7 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
{ } /* end */
};
-static struct snd_kcontrol_new alc883_tagra_mixer[] = {
+static struct snd_kcontrol_new alc883_targa_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -8088,7 +8127,7 @@ static struct snd_kcontrol_new alc883_tagra_mixer[] = {
{ } /* end */
};
-static struct snd_kcontrol_new alc883_tagra_2ch_mixer[] = {
+static struct snd_kcontrol_new alc883_targa_2ch_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
@@ -8153,6 +8192,19 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = {
{ } /* end */
};
+static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = {
+ HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+ { } /* end */
+};
+
static struct snd_kcontrol_new alc888_lenovo_sky_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -8417,7 +8469,7 @@ static struct hda_verb alc883_2ch_fujitsu_pi2515_verbs[] = {
{ } /* end */
};
-static struct hda_verb alc883_tagra_verbs[] = {
+static struct hda_verb alc883_targa_verbs[] = {
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
@@ -8626,8 +8678,8 @@ static void alc883_medion_md2_init_hook(struct hda_codec *codec)
}
/* toggle speaker-output according to the hp-jack state */
-#define alc883_tagra_init_hook alc882_targa_init_hook
-#define alc883_tagra_unsol_event alc882_targa_unsol_event
+#define alc883_targa_init_hook alc882_targa_init_hook
+#define alc883_targa_unsol_event alc882_targa_unsol_event
static void alc883_clevo_m720_mic_automute(struct hda_codec *codec)
{
@@ -8957,7 +9009,7 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
#define alc883_loopbacks alc880_loopbacks
#endif
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
#define alc883_pcm_analog_playback alc880_pcm_analog_playback
#define alc883_pcm_analog_capture alc880_pcm_analog_capture
#define alc883_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
@@ -8978,6 +9030,7 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
[ALC883_ACER] = "acer",
[ALC883_ACER_ASPIRE] = "acer-aspire",
[ALC888_ACER_ASPIRE_4930G] = "acer-aspire-4930g",
+ [ALC888_ACER_ASPIRE_6530G] = "acer-aspire-6530g",
[ALC888_ACER_ASPIRE_8930G] = "acer-aspire-8930g",
[ALC883_MEDION] = "medion",
[ALC883_MEDION_MD2] = "medion-md2",
@@ -9021,7 +9074,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
ALC888_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
- ALC888_ACER_ASPIRE_4930G),
+ ALC888_ACER_ASPIRE_6530G),
/* default Acer -- disabled as it causes more problems.
* model=auto should work fine now
*/
@@ -9069,6 +9122,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
SND_PCI_QUIRK(0x1462, 0x7267, "MSI", ALC883_3ST_6ch_DIG),
SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
@@ -9165,8 +9219,8 @@ static struct alc_config_preset alc883_presets[] = {
.input_mux = &alc883_capture_source,
},
[ALC883_TARGA_DIG] = {
- .mixers = { alc883_tagra_mixer, alc883_chmode_mixer },
- .init_verbs = { alc883_init_verbs, alc883_tagra_verbs},
+ .mixers = { alc883_targa_mixer, alc883_chmode_mixer },
+ .init_verbs = { alc883_init_verbs, alc883_targa_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.dig_out_nid = ALC883_DIGOUT_NID,
@@ -9174,12 +9228,12 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_3ST_6ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
- .unsol_event = alc883_tagra_unsol_event,
- .init_hook = alc883_tagra_init_hook,
+ .unsol_event = alc883_targa_unsol_event,
+ .init_hook = alc883_targa_init_hook,
},
[ALC883_TARGA_2ch_DIG] = {
- .mixers = { alc883_tagra_2ch_mixer},
- .init_verbs = { alc883_init_verbs, alc883_tagra_verbs},
+ .mixers = { alc883_targa_2ch_mixer},
+ .init_verbs = { alc883_init_verbs, alc883_targa_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.adc_nids = alc883_adc_nids_alt,
@@ -9188,13 +9242,13 @@ static struct alc_config_preset alc883_presets[] = {
.num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
.channel_mode = alc883_3ST_2ch_modes,
.input_mux = &alc883_capture_source,
- .unsol_event = alc883_tagra_unsol_event,
- .init_hook = alc883_tagra_init_hook,
+ .unsol_event = alc883_targa_unsol_event,
+ .init_hook = alc883_targa_init_hook,
},
[ALC883_TARGA_8ch_DIG] = {
.mixers = { alc883_base_mixer, alc883_chmode_mixer },
.init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs,
- alc883_tagra_verbs },
+ alc883_targa_verbs },
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
@@ -9206,8 +9260,8 @@ static struct alc_config_preset alc883_presets[] = {
.channel_mode = alc883_4ST_8ch_modes,
.need_dac_fix = 1,
.input_mux = &alc883_capture_source,
- .unsol_event = alc883_tagra_unsol_event,
- .init_hook = alc883_tagra_init_hook,
+ .unsol_event = alc883_targa_unsol_event,
+ .init_hook = alc883_targa_init_hook,
},
[ALC883_ACER] = {
.mixers = { alc883_base_mixer },
@@ -9255,6 +9309,24 @@ static struct alc_config_preset alc883_presets[] = {
.unsol_event = alc_automute_amp_unsol_event,
.init_hook = alc888_acer_aspire_4930g_init_hook,
},
+ [ALC888_ACER_ASPIRE_6530G] = {
+ .mixers = { alc888_acer_aspire_6530_mixer },
+ .init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+ alc888_acer_aspire_6530g_verbs },
+ .num_dacs = ARRAY_SIZE(alc883_dac_nids),
+ .dac_nids = alc883_dac_nids,
+ .num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+ .adc_nids = alc883_adc_nids_rev,
+ .capsrc_nids = alc883_capsrc_nids_rev,
+ .dig_out_nid = ALC883_DIGOUT_NID,
+ .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes),
+ .channel_mode = alc883_3ST_2ch_modes,
+ .num_mux_defs =
+ ARRAY_SIZE(alc888_2_capture_sources),
+ .input_mux = alc888_acer_aspire_6530_sources,
+ .unsol_event = alc_automute_amp_unsol_event,
+ .init_hook = alc888_acer_aspire_4930g_init_hook,
+ },
[ALC888_ACER_ASPIRE_8930G] = {
.mixers = { alc888_base_mixer,
alc883_chmode_mixer },
@@ -9361,7 +9433,7 @@ static struct alc_config_preset alc883_presets[] = {
.init_hook = alc888_lenovo_ms7195_front_automute,
},
[ALC883_HAIER_W66] = {
- .mixers = { alc883_tagra_2ch_mixer},
+ .mixers = { alc883_targa_2ch_mixer},
.init_verbs = { alc883_init_verbs, alc883_haier_w66_verbs},
.num_dacs = ARRAY_SIZE(alc883_dac_nids),
.dac_nids = alc883_dac_nids,
@@ -11131,7 +11203,7 @@ static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = {
#define alc262_loopbacks alc880_loopbacks
#endif
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
#define alc262_pcm_analog_playback alc880_pcm_analog_playback
#define alc262_pcm_analog_capture alc880_pcm_analog_capture
#define alc262_pcm_digital_playback alc880_pcm_digital_playback
@@ -12286,7 +12358,7 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2);
}
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
#define alc268_pcm_analog_playback alc880_pcm_analog_playback
#define alc268_pcm_analog_capture alc880_pcm_analog_capture
#define alc268_pcm_analog_alt_capture alc880_pcm_analog_alt_capture
@@ -13197,7 +13269,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
#define alc269_loopbacks alc880_loopbacks
#endif
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
#define alc269_pcm_analog_playback alc880_pcm_analog_playback
#define alc269_pcm_analog_capture alc880_pcm_analog_capture
#define alc269_pcm_digital_playback alc880_pcm_digital_playback
@@ -14059,7 +14131,7 @@ static void alc861_toshiba_unsol_event(struct hda_codec *codec,
alc861_toshiba_automute(codec);
}
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
#define alc861_pcm_analog_playback alc880_pcm_analog_playback
#define alc861_pcm_analog_capture alc880_pcm_analog_capture
#define alc861_pcm_digital_playback alc880_pcm_digital_playback
@@ -14582,7 +14654,7 @@ static hda_nid_t alc861vd_dac_nids[4] = {
/* dac_nids for ALC660vd are in a different order - according to
* Realtek's driver.
- * This should probably tesult in a different mixer for 6stack models
+ * This should probably result in a different mixer for 6stack models
* of ALC660vd codecs, but for now there is only 3stack mixer
* - and it is the same as in 861vd.
* adc_nids in ALC660vd are (is) the same as in 861vd
@@ -15027,7 +15099,7 @@ static void alc861vd_dallas_init_hook(struct hda_codec *codec)
#define alc861vd_loopbacks alc880_loopbacks
#endif
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
#define alc861vd_pcm_analog_playback alc880_pcm_analog_playback
#define alc861vd_pcm_analog_capture alc880_pcm_analog_capture
#define alc861vd_pcm_digital_playback alc880_pcm_digital_playback
@@ -15206,7 +15278,7 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec)
hda_nid_t pin;
pin = spec->autocfg.hp_pins[0];
- if (pin) /* connect to front and use dac 0 */
+ if (pin) /* connect to front and use dac 0 */
alc861vd_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
pin = spec->autocfg.speaker_pins[0];
if (pin)
@@ -16669,7 +16741,7 @@ static struct snd_kcontrol_new alc272_nc10_mixer[] = {
#endif
-/* pcm configuration: identiacal with ALC880 */
+/* pcm configuration: identical with ALC880 */
#define alc662_pcm_analog_playback alc880_pcm_analog_playback
#define alc662_pcm_analog_capture alc880_pcm_analog_capture
#define alc662_pcm_digital_playback alc880_pcm_digital_playback
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index fa33661..938a58a 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -297,9 +297,9 @@ static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
static bool filter(struct dma_chan *chan, void *param)
{
struct txx9aclc_dmadata *dmadata = param;
- char devname[BUS_ID_SIZE + 2];
+ char devname[20 + 2]; /* FIXME: old BUS_ID_SIZE + 2 */
- sprintf(devname, "%s.%d", dmadata->dma_res->name,
+ snprintf(devname, sizeof(devname), "%s.%d", dmadata->dma_res->name,
(int)dmadata->dma_res->start);
if (strcmp(dev_name(chan->device->dev), devname) == 0) {
chan->private = &dmadata->dma_slave;
diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c
index b144513..8f9b60c 100644
--- a/sound/usb/caiaq/audio.c
+++ b/sound/usb/caiaq/audio.c
@@ -199,8 +199,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream)
dev->period_out_count[index] = BYTES_PER_SAMPLE + 1;
dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1;
} else {
- dev->period_in_count[index] = BYTES_PER_SAMPLE;
- dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE;
+ int in_pos = (dev->spec.data_alignment == 2) ? 0 : 2;
+ dev->period_in_count[index] = BYTES_PER_SAMPLE + in_pos;
+ dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE + in_pos;
}
if (dev->streaming)
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 2240624..0e5db71 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -35,7 +35,7 @@
#include "input.h"
MODULE_AUTHOR("Daniel Mack <daniel@...aq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
"{Native Instruments, RigKontrol3},"
--
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