October 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH0/2] jack: Add support for SND_JACK_LINEOUT
by Matthew Ranostay 21 Oct '08

21 Oct '08

Add support to the jack abstraction layer to report 'Line Out' presence insertions. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/include/linux/input.h b/include/linux/input.h index a5802c9..7323d2f 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -644,6 +644,7 @@ struct input_absinfo { #define SW_RADIO SW_RFKILL_ALL /* deprecated */ #define SW_MICROPHONE_INSERT 0x04 /* set = inserted */ #define SW_DOCK 0x05 /* set = plugged into dock */ +#define SW_LINEOUT_INSERT 0x06 /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index b1b2b8b..67270ee 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -35,6 +35,8 @@ enum snd_jack_types { SND_JACK_HEADPHONE = 0x0001, SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, + SND_JACK_LINEOUT = 0x0004, + SND_JACK_SWITCH = SND_JACK_HEADPHONE | SND_JACK_LINEOUT, }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index bd2d9e6..284432f 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,6 +34,7 @@ static int snd_jack_dev_free(struct snd_device *device) else input_free_device(jack->input_dev); + kfree(jack->id); kfree(jack); return 0; @@ -87,7 +88,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (jack == NULL) return -ENOMEM; - jack->id = id; + jack->id = kstrdup(id, GFP_KERNEL); jack->input_dev = input_allocate_device(); if (jack->input_dev == NULL) { @@ -102,6 +103,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_HEADPHONE) input_set_capability(jack->input_dev, EV_SW, SW_HEADPHONE_INSERT); + if (type & SND_JACK_LINEOUT) + input_set_capability(jack->input_dev, EV_SW, + SW_LINEOUT_INSERT); if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); @@ -153,6 +157,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_HEADPHONE) input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, status & SND_JACK_HEADPHONE); + if (jack->type & SND_JACK_LINEOUT) + input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, + status & SND_JACK_LINEOUT); if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE);

3 7

[alsa-devel] [PATCH2/4] hda: dynamic jack id
by Matthew Ranostay 21 Oct '08

21 Oct '08

Allow jack->id to be kstrdup()'ed to prevent scope issues from calling functions. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/core/jack.c b/sound/core/jack.c index 4978db1..284432f 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -34,6 +34,7 @@ static int snd_jack_dev_free(struct snd_device *device) else input_free_device(jack->input_dev); + kfree(jack->id); kfree(jack); return 0; @@ -87,7 +88,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (jack == NULL) return -ENOMEM; - jack->id = id; + jack->id = kstrdup(id, GFP_KERNEL); jack->input_dev = input_allocate_device(); if (jack->input_dev == NULL) {

2 1

[alsa-devel] [PATCH1/4] hda: lineout support to jack abstraction layer
by Matthew Ranostay 21 Oct '08

21 Oct '08

Add SND_JACK_LINEOUT reporting support to jack abstraction layer. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/include/linux/input.h b/include/linux/input.h index b86fb55..fa9127d 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -657,6 +657,7 @@ struct input_absinfo { #define SW_RADIO SW_RFKILL_ALL /* deprecated */ #define SW_MICROPHONE_INSERT 0x04 /* set = inserted */ #define SW_DOCK 0x05 /* set = plugged into dock */ +#define SW_LINEOUT_INSERT 0x06 /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index b1b2b8b..7cb25f4 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -35,6 +35,7 @@ enum snd_jack_types { SND_JACK_HEADPHONE = 0x0001, SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, + SND_JACK_LINEOUT = 0x0004, }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index bd2d9e6..4978db1 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -102,6 +102,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_HEADPHONE) input_set_capability(jack->input_dev, EV_SW, SW_HEADPHONE_INSERT); + if (type & SND_JACK_LINEOUT) + input_set_capability(jack->input_dev, EV_SW, + SW_LINEOUT_INSERT); if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); @@ -153,6 +156,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_HEADPHONE) input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, status & SND_JACK_HEADPHONE); + if (jack->type & SND_JACK_LINEOUT) + input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, + status & SND_JACK_LINEOUT); if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE);

2 1

[alsa-devel] [PATCH4/4] hda: add support for jack detection on IDT codecs.
by Matthew Ranostay 21 Oct '08

21 Oct '08

Add support to the IDT codec families to report jack status to the jack abstraction layer. This required some reorganization in the stac92xx_unsol_event function involving the STAC_VREF_EVENT event. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d106ea5..c24d22f 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -36,6 +36,7 @@ #include "hda_patch.h" #include "hda_beep.h" +#define STAC_INSERT_EVENT 0x10 #define STAC_PWR_EVENT 0x20 #define STAC_HP_EVENT 0x30 #define STAC_VREF_EVENT 0x40 @@ -129,6 +130,17 @@ enum { STAC_927X_MODELS }; +struct sigmatel_event { + hda_nid_t nid; + int data; +}; + +struct sigmatel_jack { + hda_nid_t nid; + int type; + struct snd_jack *jack; +}; + struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; @@ -161,6 +173,12 @@ struct sigmatel_spec { hda_nid_t *pwr_nids; hda_nid_t *dac_list; + /* jack detection */ + struct snd_array jacks; + + /* events */ + struct snd_array events; + /* playback */ struct hda_input_mux *mono_mux; struct hda_input_mux *amp_mux; @@ -216,9 +234,6 @@ struct sigmatel_spec { struct hda_pcm pcm_rec[2]; /* PCM information */ - /* jack detection */ - struct snd_jack *jack; - /* dynamic controls and input_mux */ struct auto_pin_cfg autocfg; struct snd_array kctls; @@ -2458,13 +2473,15 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int nid = cfg->hp_pins[cfg->hp_outs - 1]; spec->hp_switch = ucontrol->value.integer.value[0]; /* check to be sure that the ports are upto date with * switch changes */ - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + codec->patch_ops.unsol_event(codec, (STAC_HP_EVENT | nid) << 26); return 1; } @@ -2504,7 +2521,8 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ * appropriately according to the pin direction */ if (spec->hp_detect) - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + codec->patch_ops.unsol_event(codec, + (STAC_HP_EVENT | nid) << 26); return 1; } @@ -3574,13 +3592,70 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } +static int stac92xx_add_jack(struct hda_codec *codec, + hda_nid_t nid, int type) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_jack *jack; + int def_conf = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + int connectivity = get_defcfg_connect(def_conf); + char name[32]; + + if (connectivity && connectivity != AC_JACK_PORT_FIXED) + return 0; + + snd_array_init(&spec->jacks, sizeof(*jack), 32); + jack = snd_array_new(&spec->jacks); + if (!jack) + return -ENOMEM; + jack->nid = nid; + jack->type = type; + + sprintf(name, "%s at %s %s Jack", + snd_hda_get_jack_type(def_conf), + snd_hda_get_jack_connectivity(def_conf), + snd_hda_get_jack_location(def_conf)); + + return snd_jack_new(codec->bus->card, name, type, &jack->jack); +} + +static int stac92xx_add_event(struct sigmatel_spec *spec, hda_nid_t nid, + int data) +{ + struct sigmatel_event *event; + + snd_array_init(&spec->events, sizeof(*event), 32); + event = snd_array_new(&spec->events); + if (!event) + return -ENOMEM; + event->nid = nid; + event->data = data; + + return 0; +} + +static int stac92xx_event_data(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_event *events = spec->events.list; + if (events) { + int i; + for (i = 0; i < spec->events.used; i++) + if (events[i].nid == nid) + return events[i].data; + } + return 0; +} + static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, unsigned int event) { - if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | event)); + (AC_USRSP_EN | event | nid)); + } } static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) @@ -3623,27 +3698,36 @@ static int stac92xx_init(struct hda_codec *codec) /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ - for (i = 0; i < cfg->hp_outs; i++) - enable_pin_detect(codec, cfg->hp_pins[i], - STAC_HP_EVENT); + for (i = 0; i < cfg->hp_outs; i++) { + int type = SND_JACK_HEADPHONE; + hda_nid_t nid = cfg->hp_pins[i]; + enable_pin_detect(codec, nid, STAC_HP_EVENT | nid); + /* jack detection */ + if (cfg->hp_outs == i) + type |= SND_JACK_LINEOUT; + err = stac92xx_add_jack(codec, nid, type); + if (err < 0) + return err; + + } /* force to enable the first line-out; the others are set up * in unsol_event */ stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], - AC_PINCTL_OUT_EN); - stac92xx_auto_init_hp_out(codec); - /* jack detection */ - err = snd_jack_new(codec->bus->card, - "Headphone Jack", - SND_JACK_HEADPHONE, &spec->jack); - if (err < 0) - return err; + AC_PINCTL_OUT_EN); /* fake event to set up pins */ - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + codec->patch_ops.unsol_event(codec, + (STAC_HP_EVENT | spec->autocfg.hp_pins[0]) << 26); } else { stac92xx_auto_init_multi_out(codec); stac92xx_auto_init_hp_out(codec); } + for (i = 0; i < cfg->line_outs; i++) { + err = stac92xx_add_jack(codec, + cfg->line_out_pins[i], SND_JACK_LINEOUT); + if (err < 0) + return err; + } for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = cfg->input_pins[i]; if (nid) { @@ -3656,6 +3740,11 @@ static int stac92xx_init(struct hda_codec *codec) if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) pinctl |= stac92xx_get_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); + err = stac92xx_add_jack(codec, nid, + SND_JACK_MICROPHONE); + if (err < 0) + return err; + enable_pin_detect(codec, nid, STAC_INSERT_EVENT | nid); } } for (i = 0; i < spec->num_dmics; i++) @@ -3697,6 +3786,18 @@ static int stac92xx_init(struct hda_codec *codec) return 0; } +static void stac92xx_free_jacks(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + if (spec->jacks.list) { + struct sigmatel_jack *jacks = spec->jacks.list; + int i; + for (i = 0; i < spec->jacks.used; i++) + snd_device_free(codec->bus->card, &jacks[i].jack); + } + snd_array_free(&spec->jacks); +} + static void stac92xx_free_kctls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -3717,11 +3818,10 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; - if (spec->jack) - snd_device_free(codec->bus->card, spec->jack); - if (spec->bios_pin_configs) kfree(spec->bios_pin_configs); + stac92xx_free_jacks(codec); + snd_array_free(&spec->events); kfree(spec); snd_hda_detach_beep_device(codec); @@ -3804,8 +3904,6 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } - snd_jack_report(spec->jack, - presence ? SND_JACK_HEADPHONE : 0); if (presence) { /* disable lineouts, enable hp */ @@ -3862,24 +3960,57 @@ static void stac92xx_pin_sense(struct hda_codec *codec, int idx) /* power down unused output ports */ snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val); -}; +} + +static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_jack *jacks = spec->jacks.list; + + if (jacks) { + int i; + for (i = 0; i < spec->jacks.used; i++) { + if (jacks->nid == nid) { + unsigned int pin_ctl = + snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_PIN_WIDGET_CONTROL, + 0x00); + int type = jacks->type; + if (type == (SND_JACK_LINEOUT + | SND_JACK_HEADPHONE)) + type = (pin_ctl & AC_PINCTL_HP_EN) + ? SND_JACK_HEADPHONE : SND_JACK_LINEOUT; + snd_jack_report(jacks->jack, + get_hp_pin_presence(codec, nid) + ? type : 0); + } + jacks++; + } + } +} static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) { struct sigmatel_spec *spec = codec->spec; - int idx = res >> 26 & 0x0f; + int event = (res >> 26) & 0x70; + int nid = res >> 26 & 0x0f; - switch ((res >> 26) & 0x70) { + switch (event) { case STAC_HP_EVENT: stac92xx_hp_detect(codec, res); /* fallthru */ + case STAC_INSERT_EVENT: case STAC_PWR_EVENT: - if (spec->num_pwrs > 0) - stac92xx_pin_sense(codec, idx); + if (nid) { + if (spec->num_pwrs > 0) + stac92xx_pin_sense(codec, nid); + stac92xx_report_jack(codec, nid); + } break; case STAC_VREF_EVENT: { int data = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DATA, 0); + int idx = stac92xx_event_data(codec, nid); /* toggle VREF state based on GPIOx status */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, !!(data & (1 << idx))); @@ -4402,8 +4533,11 @@ again: snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | STAC_VREF_EVENT | 0x01)); + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_VREF_EVENT | codec->afg)); + err = stac92xx_add_event(spec, codec->afg, 0x02); + if (err < 0) + return err; spec->gpio_mask |= 0x02; break; } @@ -4802,8 +4936,11 @@ static int patch_stac9205(struct hda_codec *codec) snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | STAC_HP_EVENT)); + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_VREF_EVENT | codec->afg)); + err = stac92xx_add_event(spec, codec->afg, 0x01); + if (err < 0) + return err; spec->gpio_dir = 0x0b; spec->eapd_mask = 0x01;

1 0

[alsa-devel] [PATCH3/4] hda: add snd_hda_get_jack* functions
by Matthew Ranostay 21 Oct '08

21 Oct '08

Add snd_hda_get_jack* functions for reporting jack location, type, and connectivity type. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9a8adc6..eaa8b56 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -107,6 +107,52 @@ static void hda_keep_power_on(struct hda_codec *codec); static inline void hda_keep_power_on(struct hda_codec *codec) {} #endif +const char *snd_hda_get_jack_location(u32 cfg) +{ + static char *bases[7] = { + "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", + }; + static unsigned char specials_idx[] = { + 0x07, 0x08, + 0x17, 0x18, 0x19, + 0x37, 0x38 + }; + static char *specials[] = { + "Rear Panel", "Drive Bar", + "Riser", "HDMI", "ATAPI", + "Mobile-In", "Mobile-Out" + }; + int i; + cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT; + if ((cfg & 0x0f) < 7) + return bases[cfg & 0x0f]; + for (i = 0; i < ARRAY_SIZE(specials_idx); i++) { + if (cfg == specials_idx[i]) + return specials[i]; + } + return "UNKNOWN"; +} + +const char *snd_hda_get_jack_connectivity(u32 cfg) +{ + static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; + + return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3]; +} + +const char *snd_hda_get_jack_type(u32 cfg) +{ + static char *jack_types[16] = { + "Line Out", "Speaker", "HP Out", "CD", + "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", + "Line In", "Aux", "Mic", "Telephony", + "SPDIF In", "Digitial In", "Reserved", "Other" + }; + + return jack_types[(cfg & AC_DEFCFG_DEVICE) + >> AC_DEFCFG_DEVICE_SHIFT]; +} + /** * snd_hda_codec_read - send a command and get the response * @codec: the HDA codec diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index a77ba22..0a42034 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -859,6 +859,13 @@ int snd_hda_suspend(struct hda_bus *bus, pm_message_t state); int snd_hda_resume(struct hda_bus *bus); #endif +/* + * get widget information + */ +const char *get_jack_location(u32 cfg); +const char *get_jack_connectivity(u32 cfg); +const char *get_jack_type(u32 cfg); + /* * power saving */ diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 64ab19f..b36d4d0 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -145,32 +145,6 @@ static void print_pcm_caps(struct snd_info_buffer *buffer, print_pcm_formats(buffer, stream); } -static const char *get_jack_location(u32 cfg) -{ - static char *bases[7] = { - "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", - }; - static unsigned char specials_idx[] = { - 0x07, 0x08, - 0x17, 0x18, 0x19, - 0x37, 0x38 - }; - static char *specials[] = { - "Rear Panel", "Drive Bar", - "Riser", "HDMI", "ATAPI", - "Mobile-In", "Mobile-Out" - }; - int i; - cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT; - if ((cfg & 0x0f) < 7) - return bases[cfg & 0x0f]; - for (i = 0; i < ARRAY_SIZE(specials_idx); i++) { - if (cfg == specials_idx[i]) - return specials[i]; - } - return "UNKNOWN"; -} - static const char *get_jack_connection(u32 cfg) { static char *names[16] = { @@ -206,13 +180,6 @@ static void print_pin_caps(struct snd_info_buffer *buffer, int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; - static char *jack_types[16] = { - "Line Out", "Speaker", "HP Out", "CD", - "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", - "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" - }; - static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); @@ -274,9 +241,9 @@ static void print_pin_caps(struct snd_info_buffer *buffer, caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], - jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT], - jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3], - get_jack_location(caps)); + snd_hda_get_jack_type(caps), + snd_hda_get_jack_connectivity(caps), + snd_hda_get_jack_location(caps)); snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps), get_jack_color(caps));

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[alsa-devel] [PATCH2/2] hda: Add support for IDT codecs to report jack presence
by Matthew Ranostay 21 Oct '08

21 Oct '08

Add support for reporting line out, headphone, and microphone jack insertions via the jack abstraction layer for the Sigmatel/IDT codec family. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d106ea5..e3a5e7a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -36,6 +36,7 @@ #include "hda_patch.h" #include "hda_beep.h" +#define STAC_INSERT_EVENT 0x10 #define STAC_PWR_EVENT 0x20 #define STAC_HP_EVENT 0x30 #define STAC_VREF_EVENT 0x40 @@ -129,6 +130,17 @@ enum { STAC_927X_MODELS }; +struct sigmatel_event { + hda_nid_t nid; + int data; +}; + +struct sigmatel_jack { + hda_nid_t nid; + int type; + struct snd_jack *jack; +}; + struct sigmatel_spec { struct snd_kcontrol_new *mixers[4]; unsigned int num_mixers; @@ -161,6 +173,12 @@ struct sigmatel_spec { hda_nid_t *pwr_nids; hda_nid_t *dac_list; + /* jack detection */ + struct snd_array jacks; + + /* events */ + struct snd_array events; + /* playback */ struct hda_input_mux *mono_mux; struct hda_input_mux *amp_mux; @@ -216,9 +234,6 @@ struct sigmatel_spec { struct hda_pcm pcm_rec[2]; /* PCM information */ - /* jack detection */ - struct snd_jack *jack; - /* dynamic controls and input_mux */ struct auto_pin_cfg autocfg; struct snd_array kctls; @@ -2458,13 +2473,15 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int nid = cfg->hp_pins[cfg->hp_outs - 1]; spec->hp_switch = ucontrol->value.integer.value[0]; /* check to be sure that the ports are upto date with * switch changes */ - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + codec->patch_ops.unsol_event(codec, (STAC_HP_EVENT | nid) << 26); return 1; } @@ -2504,7 +2521,8 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ * appropriately according to the pin direction */ if (spec->hp_detect) - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + codec->patch_ops.unsol_event(codec, + (STAC_HP_EVENT | nid) << 26); return 1; } @@ -3574,13 +3592,70 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, AC_VERB_SET_GPIO_DATA, gpiostate); /* sync */ } +static int stac92xx_add_jack(struct hda_codec *codec, + hda_nid_t nid, int type) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_jack *jack; + int def_conf = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + int connectivity = get_defcfg_connect(def_conf); + char name[32]; + + if (connectivity && connectivity != AC_JACK_PORT_FIXED) + return 0; + + snd_array_init(&spec->jacks, sizeof(*jack), 32); + jack = snd_array_new(&spec->jacks); + if (!jack) + return -ENOMEM; + jack->nid = nid; + jack->type = type; + + sprintf(name, "%s at %s %s Jack", + get_jack_type(def_conf), + get_jack_connectivity(def_conf), + get_jack_location(def_conf)); + + return snd_jack_new(codec->bus->card, name, type, &jack->jack); +} + +static int stac92xx_add_event(struct sigmatel_spec *spec, hda_nid_t nid, + int data) +{ + struct sigmatel_event *event; + + snd_array_init(&spec->events, sizeof(*event), 32); + event = snd_array_new(&spec->events); + if (!event) + return -ENOMEM; + event->nid = nid; + event->data = data; + + return 0; +} + +static int stac92xx_event_data(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_event *events = spec->events.list; + if (events) { + int i; + for (i = 0; i < spec->events.used; i++) + if (events[i].nid == nid) + return events[i].data; + } + return 0; +} + static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, unsigned int event) { - if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | event)); + (AC_USRSP_EN | event | nid)); + } } static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid) @@ -3623,27 +3698,34 @@ static int stac92xx_init(struct hda_codec *codec) /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ - for (i = 0; i < cfg->hp_outs; i++) - enable_pin_detect(codec, cfg->hp_pins[i], - STAC_HP_EVENT); + for (i = 0; i < cfg->hp_outs; i++) { + hda_nid_t nid = cfg->hp_pins[i]; + enable_pin_detect(codec, nid, STAC_HP_EVENT | nid); + /* jack detection */ + err = stac92xx_add_jack(codec, nid, cfg->hp_outs == i + ? SND_JACK_SWITCH : SND_JACK_HEADPHONE); + if (err < 0) + return err; + + } /* force to enable the first line-out; the others are set up * in unsol_event */ stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0], - AC_PINCTL_OUT_EN); - stac92xx_auto_init_hp_out(codec); - /* jack detection */ - err = snd_jack_new(codec->bus->card, - "Headphone Jack", - SND_JACK_HEADPHONE, &spec->jack); - if (err < 0) - return err; + AC_PINCTL_OUT_EN); /* fake event to set up pins */ - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + codec->patch_ops.unsol_event(codec, + (STAC_HP_EVENT | spec->autocfg.hp_pins[0]) << 26); } else { stac92xx_auto_init_multi_out(codec); stac92xx_auto_init_hp_out(codec); } + for (i = 0; i < cfg->line_outs; i++) { + err = stac92xx_add_jack(codec, + cfg->line_out_pins[i], SND_JACK_LINEOUT); + if (err < 0) + return err; + } for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = cfg->input_pins[i]; if (nid) { @@ -3656,6 +3738,11 @@ static int stac92xx_init(struct hda_codec *codec) if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) pinctl |= stac92xx_get_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); + err = stac92xx_add_jack(codec, nid, + SND_JACK_MICROPHONE); + if (err < 0) + return err; + enable_pin_detect(codec, nid, STAC_INSERT_EVENT | nid); } } for (i = 0; i < spec->num_dmics; i++) @@ -3697,6 +3784,18 @@ static int stac92xx_init(struct hda_codec *codec) return 0; } +static void stac92xx_free_jacks(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + if (spec->jacks.list) { + struct sigmatel_jack *jacks = spec->jacks.list; + int i; + for (i = 0; i < spec->jacks.used; i++) + snd_device_free(codec->bus->card, &jacks[i].jack); + } + snd_array_free(&spec->jacks); +} + static void stac92xx_free_kctls(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -3717,11 +3816,10 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; - if (spec->jack) - snd_device_free(codec->bus->card, spec->jack); - if (spec->bios_pin_configs) kfree(spec->bios_pin_configs); + stac92xx_free_jacks(codec); + snd_array_free(&spec->events); kfree(spec); snd_hda_detach_beep_device(codec); @@ -3804,8 +3902,6 @@ static void stac92xx_hp_detect(struct hda_codec *codec, unsigned int res) break; presence = get_hp_pin_presence(codec, cfg->hp_pins[i]); } - snd_jack_report(spec->jack, - presence ? SND_JACK_HEADPHONE : 0); if (presence) { /* disable lineouts, enable hp */ @@ -3862,24 +3958,56 @@ static void stac92xx_pin_sense(struct hda_codec *codec, int idx) /* power down unused output ports */ snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, val); -}; +} + +static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + struct sigmatel_jack *jacks = spec->jacks.list; + + if (jacks) { + int i; + for (i = 0; i < spec->jacks.used; i++) { + if (jacks->nid == nid) { + unsigned int pin_ctl = + snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_PIN_WIDGET_CONTROL, + 0x00); + int type = jacks->type; + if (type == SND_JACK_SWITCH) + type = (pin_ctl & AC_PINCTL_HP_EN) + ? SND_JACK_HEADPHONE : SND_JACK_LINEOUT; + snd_jack_report(jacks->jack, + get_hp_pin_presence(codec, nid) + ? type : 0); + } + jacks++; + } + } +} static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) { struct sigmatel_spec *spec = codec->spec; - int idx = res >> 26 & 0x0f; + int event = (res >> 26) & 0x70; + int nid = res >> 26 & 0x0f; - switch ((res >> 26) & 0x70) { + switch (event) { case STAC_HP_EVENT: stac92xx_hp_detect(codec, res); /* fallthru */ + case STAC_INSERT_EVENT: case STAC_PWR_EVENT: - if (spec->num_pwrs > 0) - stac92xx_pin_sense(codec, idx); + if (nid) { + if (spec->num_pwrs > 0) + stac92xx_pin_sense(codec, nid); + stac92xx_report_jack(codec, nid); + } break; case STAC_VREF_EVENT: { int data = snd_hda_codec_read(codec, codec->afg, 0, AC_VERB_GET_GPIO_DATA, 0); + int idx = stac92xx_event_data(codec, nid); /* toggle VREF state based on GPIOx status */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, !!(data & (1 << idx))); @@ -4402,8 +4530,11 @@ again: snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | STAC_VREF_EVENT | 0x01)); + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_VREF_EVENT | codec->afg)); + err = stac92xx_add_event(spec, codec->afg, 0x02); + if (err < 0) + return err; spec->gpio_mask |= 0x02; break; } @@ -4802,8 +4933,11 @@ static int patch_stac9205(struct hda_codec *codec) snd_hda_codec_write(codec, codec->afg, 0, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x10); snd_hda_codec_write_cache(codec, codec->afg, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | STAC_HP_EVENT)); + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_VREF_EVENT | codec->afg)); + err = stac92xx_add_event(spec, codec->afg, 0x01); + if (err < 0) + return err; spec->gpio_dir = 0x0b; spec->eapd_mask = 0x01;

2 2

Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 21 Oct '08

21 Oct '08

At Tue, 21 Oct 2008 13:28:25 +0200, Nir Tzachar wrote: > > Hello. > > Updated to the latest linus-git tree, and pulled in the sound-2.6 tree. > > The good news are that by following Wei's instructions (adding the > device id in patch_nvhdmi.c), I managed to get audio through the hdmi. > Goodie. Could you give details? > The bad news are that mplayer misbehaves and will get stuck every time > I try to change the volume. However, this does not happen in vlc. Any > ideas? Well, we need to sort out where the problem comes from. Try the vanilla Linus tree without sound-2.6 whether the problem persists. thanks, Takashi > On Tue, Oct 21, 2008 at 10:10 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 21 Oct 2008 09:52:46 +0200, > > Nir Tzachar wrote: > >> > >> On Tue, Oct 21, 2008 at 9:50 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Tue, 21 Oct 2008 09:39:49 +0200, > >> > Nir Tzachar wrote: > >> >> > >> >> On Tue, Oct 21, 2008 at 9:34 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> > At Tue, 21 Oct 2008 09:30:21 +0200, > >> >> > Nir Tzachar wrote: > >> >> >> > >> >> >> Hello. > >> >> >> > >> >> >> On Tue, Oct 21, 2008 at 9:18 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> >> > At Mon, 20 Oct 2008 16:35:01 +0200, > >> >> >> > Nir Tzachar wrote: > >> >> >> >> > >> >> >> >> Hello, > >> >> >> >> > >> >> >> >> I have been trying to get audio through hdmi to work for some time > >> >> >> >> now. I have cloned the sound-2.6 repository to get the latest drivers, > >> >> >> >> which should support hdmi, AFAIK. > >> >> >> > > >> >> >> > Try more later version, from my sound git tree, or even Linus tree. > >> >> >> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > >> >> >> > >> >> >> I tried yesterday's version. Apart from not having hdmi output, it had > >> >> >> problems with volume control. > >> >> > > >> >> > Please elaborate? > >> >> > >> >> I performed a git-clone on the repository. The head was > >> >> b47d892509f2d908974781443d29b11dcd29d809. > >> > > >> > Better to pull onto the latest Linus tree. > >> > That is, clone linux-2.6.git, then pull from sound-2.6.git. > >> > >> ok, will do this. > >> > >> >> When I played a movie in mplayer and touch the volume controls, > >> >> mplayer got stuck. I can perform other tests. I'll also try to update > >> >> the tree. > >> > > >> > Which device are you using? HDMI or other? > >> > Also, please check whether this happens with Linus tree and 2.6.27.x, > >> > too. > >> > >> Using the analog device, and it works under 2.6.27.1 flawlessly. > > > > OK, then it must be a regression. > > If it works with Linus tree, the problem is the post 2.6.28 stuff. > > If it doesn't work with Linus tree, too, it's likely changes to > > patch_realtek.c... > > > > > > thanks, > > > > Takashi > > >

1 0

[alsa-devel] hda: broken PCM over spdif after AC3; suggestion
by Tomas Cernaj 21 Oct '08

21 Oct '08

Hello, trying to track down a long standing bug concerning PCM output over spdif and analog ports simultaneously after AC3 passthrough (see https://bugs.launchpad.net/bugs/25632 ), I've been digging into the inner workings of HD Audio... It seems to me that after playing DVDs with AC3 passthrough the stream associated with the default alsa device is only connected to the front Audio Output widget but not at the same time to the Digital output widget, as I would have expected. I've attached the contents of /proc/asound/Intel/codec#0 during music playback to this mail; you can see the different streams connected to nodes 0x02 and 0x03. I'm not familiar with the hda driver, but I suspect that after AC3 passthrough the multi-out configuration isn't reset to what it's supposed to be. Is there a way to work around this issue, or am I completely missing the point? Thanks for your help, Tomas

2 2

Re: [alsa-devel] Microphone array using ALSA
by Mike Mazarick 21 Oct '08

21 Oct '08

Ola Benderious had written: > Would it be possible to use several USB soundcards in GNU/Linux using > ALSA to build a microphone array? I have searched the Internet but > haven't found any straight answers. What I have read is that it might be > hard to sync the captured streams even if it would be possible. > > So, three questions. > 1. Is it theoretically possible? > 2. Can it be done using several USB soundcards? > 3. Do you know of any code or project in this area as it is? A friend of mine (Pete Mander) from England recently sent me the following information on ambisonic mics. I wrote: > On another front, I'm planning on using my GadgetLabs cards to work with Helmut Oellers creating a WaveFieldSynthesis system using 32 PC style speakers in my house (they have the amps built in). Since the GL cards would only need to be outputs, I believe it may be possible to get 4 of them to work in a linux system (however, that's to be tested). http://www.syntheticwave.de Pete replied: I did ponder whether I could do ambisonic recordings with the GL. The rigorously synchronised channels are perfect for this. I'll need to find time and a child safe workshop to build the soundfield mic from condenser capsules and the circuits: http://www.ambisonia.com/wiki/index.php/Microphones#Home_made_tetrahedrons Later on, Pete provided these additional fruits from his research: http://parts.digikey.co.uk/1/1/27254-microphone-cartridge-9-7mm-x-5mm-wm-55a 103.html http://homepage.ntlworld.com/henry01/cheap_soundfield/cheap_soundfield.htm http://www.kokkinizita.net/linuxaudio/tetra-pict.html http://en.wikipedia.org/wiki/Head-related_transfer_function If you want to be cc'd on the correspondence between Pete, Helmut, and me, send me a note off-line. To answer your questions: Q. Is it possible? A. Yes, but it doesn't quite make much sense to me unless there is some way to play back what the mics record. Q. Can it be done with several USB soundcards? A. The simple answer is no. My opinion is that you'd want to sync the samples so they would not drift apart. I don't know of an easy way to do this unless the soundcards are sync'd Q. Do you know of any code or project in this area as it is? No, I don't know of one. In addition, it will be a month or two before we even have a way to play back sources from multiple locations. We are trying to play a few 'tricks' by positioning the speakers in a room so that 32 channels sound like there are much more than 32 channels. If you are going to play games with an array of mics in order to find the position of a sound source, it seems that using multiple ambisonic mics is the way to go. Regards, Mike Mazarick (PS - if you had a slightly bigger budget, it would be quite possible to do the same type of thing and set up a WFS system with the RME 96/52 cards)

1 0

[alsa-devel] [PATCH 0/3] ASoC Blackfin part updates
by Bryan Wu 21 Oct '08

21 Oct '08

Hi Mark, These are some updates from Cliff and Mike for ASoC Blackfin part. I generated these patches against Takashi Iwai's sound-2.6.git tree. Thanks -Bryan

4 23

Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 21 Oct '08

21 Oct '08

At Tue, 21 Oct 2008 09:52:46 +0200, Nir Tzachar wrote: > > On Tue, Oct 21, 2008 at 9:50 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 21 Oct 2008 09:39:49 +0200, > > Nir Tzachar wrote: > >> > >> On Tue, Oct 21, 2008 at 9:34 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Tue, 21 Oct 2008 09:30:21 +0200, > >> > Nir Tzachar wrote: > >> >> > >> >> Hello. > >> >> > >> >> On Tue, Oct 21, 2008 at 9:18 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> > At Mon, 20 Oct 2008 16:35:01 +0200, > >> >> > Nir Tzachar wrote: > >> >> >> > >> >> >> Hello, > >> >> >> > >> >> >> I have been trying to get audio through hdmi to work for some time > >> >> >> now. I have cloned the sound-2.6 repository to get the latest drivers, > >> >> >> which should support hdmi, AFAIK. > >> >> > > >> >> > Try more later version, from my sound git tree, or even Linus tree. > >> >> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > >> >> > >> >> I tried yesterday's version. Apart from not having hdmi output, it had > >> >> problems with volume control. > >> > > >> > Please elaborate? > >> > >> I performed a git-clone on the repository. The head was > >> b47d892509f2d908974781443d29b11dcd29d809. > > > > Better to pull onto the latest Linus tree. > > That is, clone linux-2.6.git, then pull from sound-2.6.git. > > ok, will do this. > > >> When I played a movie in mplayer and touch the volume controls, > >> mplayer got stuck. I can perform other tests. I'll also try to update > >> the tree. > > > > Which device are you using? HDMI or other? > > Also, please check whether this happens with Linus tree and 2.6.27.x, > > too. > > Using the analog device, and it works under 2.6.27.1 flawlessly. OK, then it must be a regression. If it works with Linus tree, the problem is the post 2.6.28 stuff. If it doesn't work with Linus tree, too, it's likely changes to patch_realtek.c... thanks, Takashi

1 0

[alsa-devel] [PATCH] pci: use pci_ioremap_bar() in sound/
by Arjan van de Ven 21 Oct '08

21 Oct '08

From 6666b6edd4a0c496932ef1640b742cc72f9c57d8 Mon Sep 17 00:00:00 2001 From: Arjan van de Ven <arjan(a)linux.intel.com> Date: Sun, 28 Sep 2008 16:20:09 -0700 Subject: [PATCH] pci: use pci_ioremap_bar() in sound/ Use the newly introduced pci_ioremap_bar() function in sound/. pci_ioremap_bar() just takes a pci device and a bar number, with the goal of making it really hard to get wrong, while also having a central place to stick sanity checks. CC: tiwai(a)suse.de Signed-off-by: Arjan van de Ven <arjan(a)linux.intel.com> --- sound/pci/ad1889.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0.c | 3 +-- sound/pci/bt87x.c | 3 +-- sound/pci/cs4281.c | 4 ++-- sound/pci/cs5530.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/mixart/mixart.c | 3 +-- 9 files changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 92f3a97..a7f38e6 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -932,7 +932,7 @@ snd_ad1889_create(struct snd_card *card, goto free_and_ret; chip->bar = pci_resource_start(pci, 0); - chip->iobase = ioremap_nocache(chip->bar, pci_resource_len(pci, 0)); + chip->iobase = pci_ioremap_bar(pci, 0); if (chip->iobase == NULL) { printk(KERN_ERR PFX "unable to reserve region.\n"); err = -EBUSY; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 085a52b..226fe82 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1609,7 +1609,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 2f10630..0e6e5cc 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1252,7 +1252,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, return err; } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci, 0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR "AC'97 space ioremap problem\n"); snd_atiixp_free(chip); diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 68368e4..a36d4d1 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -180,8 +180,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) if ((err = pci_request_regions(pci, CARD_NAME_SHORT)) != 0) goto regions_out; - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { printk(KERN_ERR "MMIO area remap failed.\n"); err = -ENOMEM; diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 3aa8d97..1aa1c04 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -749,8 +749,7 @@ static int __devinit snd_bt87x_create(struct snd_card *card, pci_disable_device(pci); return err; } - chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), - pci_resource_len(pci, 0)); + chip->mmio = pci_ioremap_bar(pci, 0); if (!chip->mmio) { snd_printk(KERN_ERR "cannot remap io memory\n"); err = -ENOMEM; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index ef9308f..192e784 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -1382,8 +1382,8 @@ static int __devinit snd_cs4281_create(struct snd_card *card, chip->ba0_addr = pci_resource_start(pci, 0); chip->ba1_addr = pci_resource_start(pci, 1); - chip->ba0 = ioremap_nocache(chip->ba0_addr, pci_resource_len(pci, 0)); - chip->ba1 = ioremap_nocache(chip->ba1_addr, pci_resource_len(pci, 1)); + chip->ba0 = pci_ioremap_bar(pci, 0); + chip->ba1 = pci_ioremap_bar(pci, 1); if (!chip->ba0 || !chip->ba1) { snd_cs4281_free(chip); return -ENOMEM; diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 7ff8b68..4d9378d 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -132,7 +132,7 @@ static int __devinit snd_cs5530_create(struct snd_card *card, } chip->pci_base = pci_resource_start(pci, 0); - mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + mem = pci_ioremap_bar(pci, 0); if (mem == NULL) { kfree(chip); pci_disable_device(pci); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9f316c1..f080f8c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2158,7 +2158,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } chip->addr = pci_resource_start(pci, 0); - chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0)); + chip->remap_addr = pci_ioremap_bar(pci, 0); if (chip->remap_addr == NULL) { snd_printk(KERN_ERR SFX "ioremap error\n"); err = -ENXIO; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 2d0dce6..ae7601f 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1314,8 +1314,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, } for (i = 0; i < 2; i++) { mgr->mem[i].phys = pci_resource_start(pci, i); - mgr->mem[i].virt = ioremap_nocache(mgr->mem[i].phys, - pci_resource_len(pci, i)); + mgr->mem[i].virt = pci_ioremap_bar(pci, i); if (!mgr->mem[i].virt) { printk(KERN_ERR "unable to remap resource 0x%lx\n", mgr->mem[i].phys); -- 1.5.5.1 -- Arjan van de Ven Intel Open Source Technology Centre For development, discussion and tips for power savings, visit http://www.lesswatts.org

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Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 21 Oct '08

21 Oct '08

At Tue, 21 Oct 2008 09:39:49 +0200, Nir Tzachar wrote: > > On Tue, Oct 21, 2008 at 9:34 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 21 Oct 2008 09:30:21 +0200, > > Nir Tzachar wrote: > >> > >> Hello. > >> > >> On Tue, Oct 21, 2008 at 9:18 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Mon, 20 Oct 2008 16:35:01 +0200, > >> > Nir Tzachar wrote: > >> >> > >> >> Hello, > >> >> > >> >> I have been trying to get audio through hdmi to work for some time > >> >> now. I have cloned the sound-2.6 repository to get the latest drivers, > >> >> which should support hdmi, AFAIK. > >> > > >> > Try more later version, from my sound git tree, or even Linus tree. > >> > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > >> > >> I tried yesterday's version. Apart from not having hdmi output, it had > >> problems with volume control. > > > > Please elaborate? > > I performed a git-clone on the repository. The head was > b47d892509f2d908974781443d29b11dcd29d809. Better to pull onto the latest Linus tree. That is, clone linux-2.6.git, then pull from sound-2.6.git. > When I played a movie in mplayer and touch the volume controls, > mplayer got stuck. I can perform other tests. I'll also try to update > the tree. Which device are you using? HDMI or other? Also, please check whether this happens with Linus tree and 2.6.27.x, too. thanks, Takashi

1 0

Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 21 Oct '08

21 Oct '08

At Tue, 21 Oct 2008 09:30:21 +0200, Nir Tzachar wrote: > > Hello. > > On Tue, Oct 21, 2008 at 9:18 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 20 Oct 2008 16:35:01 +0200, > > Nir Tzachar wrote: > >> > >> Hello, > >> > >> I have been trying to get audio through hdmi to work for some time > >> now. I have cloned the sound-2.6 repository to get the latest drivers, > >> which should support hdmi, AFAIK. > > > > Try more later version, from my sound git tree, or even Linus tree. > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > I tried yesterday's version. Apart from not having hdmi output, it had > problems with volume control. Please elaborate? Takashi

1 0

[alsa-devel] Nvidia hdmi problems
by Nir Tzachar 21 Oct '08

21 Oct '08

Hello, I have been trying to get audio through hdmi to work for some time now. I have cloned the sound-2.6 repository to get the latest drivers, which should support hdmi, AFAIK. Motherboard is Gigabyte GA-73PVM-S2H, with nforce 630i and realtek alc889a. aplay lists the hdmi device: > aplay -L default:CARD=NVidia HDA NVidia, ALC885 Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, ALC885 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) but, > aplay -D hdmi test.wav aplay: main:583: audio open error: No such file or directory fails when trying to open /dev/snd/pcmC0D3p only the following devices exists: /dev/snd/seq /dev/snd/controlC0 /dev/snd/pcmC0D0c /dev/snd/pcmC0D0p /dev/snd/pcmC0D1c /dev/snd/pcmC0D1p /dev/snd/pcmC0D2c /dev/snd/timer I am using alsa-lib and alsa-utils 1.0.17 . Does anyone have an idea? Thanks.

2 1

[alsa-devel] snd_pcsp locking mess
by Stas Sergeev 21 Oct '08

21 Oct '08

bugme-daemon(a)bugzilla.kernel.org wrote: > ------- Comment #17 from tiwai(a)suse.de 2008-05-18 10:22 ------- > Anyway, what I meant is that the part touching I/O ports could be a faster path > than softirq. And the hrtimer code can use a bit simpler locks. The hrtimer > callback needs just one thing about DMA buffer - the buffer is allocated and > the address doesn't change during the callback. So, we'd need the lock for the > hr callback and for the PCM prepare, hwparams and free callbacks, i.e. to > assure that the hrtimer callback is finished when these PCM callbacks may > change the DMA buffer. Thus this lock doesn't have to be PCM substream lock. I wonder if it would be better to introduce such a lock in an alsa core rather than in a driver. The same way as snd_pcm_stream_lock(). > The current lock mess may come from the PCM trigger (STOP) called from > snd_pcm_period_elapsed(). So, putting snd_pcm_period_elapsed() out of hrtimer > lock is anyway necessary. A tasklet for snd_pcm_period_elapsed() is needed for > this reason, too. Wouldn't it be better to just make snd_pcm_period_elapsed() to not take the stream lock? There is a requirement right now to drop the stream lock before calling this. I still beleive this is the root of all the headache. If you need to drop it, then you either accept the race or need a second lock, and the troubles ensue.

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[alsa-devel] [PATCH1/2] hda: add access to jack functions
by Matthew Ranostay 21 Oct '08

21 Oct '08

Removed 'static' declaration from get_jack_location() to be accessed via the codec patches. Also added jack location and type reporting functions for use in the jack abstraction layer implementations. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 4a08c31..10f26e0 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -383,6 +383,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, /* * get widget capabilities */ +const char *get_jack_connectivity(u32 cfg); +const char *get_jack_type(u32 cfg); +const char *get_jack_location(u32 cfg); static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) { if (nid < codec->start_nid || diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 64ab19f..b77c1b1 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -145,7 +145,7 @@ static void print_pcm_caps(struct snd_info_buffer *buffer, print_pcm_formats(buffer, stream); } -static const char *get_jack_location(u32 cfg) +const char *get_jack_location(u32 cfg) { static char *bases[7] = { "N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom", @@ -171,6 +171,26 @@ static const char *get_jack_location(u32 cfg) return "UNKNOWN"; } +const char *get_jack_connectivity(u32 cfg) +{ + static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; + + return jack_locations[(cfg >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3]; +} + +const char *get_jack_type(u32 cfg) +{ + static char *jack_types[16] = { + "Line Out", "Speaker", "HP Out", "CD", + "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", + "Line In", "Aux", "Mic", "Telephony", + "SPDIF In", "Digitial In", "Reserved", "Other" + }; + + return jack_types[(cfg & AC_DEFCFG_DEVICE) + >> AC_DEFCFG_DEVICE_SHIFT]; +} + static const char *get_jack_connection(u32 cfg) { static char *names[16] = { @@ -206,13 +226,6 @@ static void print_pin_caps(struct snd_info_buffer *buffer, int *supports_vref) { static char *jack_conns[4] = { "Jack", "N/A", "Fixed", "Both" }; - static char *jack_types[16] = { - "Line Out", "Speaker", "HP Out", "CD", - "SPDIF Out", "Digital Out", "Modem Line", "Modem Hand", - "Line In", "Aux", "Mic", "Telephony", - "SPDIF In", "Digitial In", "Reserved", "Other" - }; - static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); @@ -274,8 +287,8 @@ static void print_pin_caps(struct snd_info_buffer *buffer, caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); snd_iprintf(buffer, " Pin Default 0x%08x: [%s] %s at %s %s\n", caps, jack_conns[(caps & AC_DEFCFG_PORT_CONN) >> AC_DEFCFG_PORT_CONN_SHIFT], - jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT], - jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3], + get_jack_type(caps), + get_jack_connectivity(caps), get_jack_location(caps)); snd_iprintf(buffer, " Conn = %s, Color = %s\n", get_jack_connection(caps),

2 1

Re: [alsa-devel] Audio on Linux is a mess
by Marek Vasut 20 Oct '08

20 Oct '08

You got what you paid for. Maybe if you change your attitude and supply some information, someone might consider looking in your issue, othervise it would be a problem since I forgot my crystal ball back at home. On Monday 20 of October 2008 16:41:45 Fred . wrote: > Yeah, fix the Linux audio so that if an application behaves wrong, or > does something stupid, then after I close the application, the system > should be silent. > It shouldn't continue make sound even after I quit the application. > And PulseAudio shouldn't crash because one application did something > even if it was stupid. > > On Mon, Oct 20, 2008 at 8:35 AM, Marek Vasut <marek.vasut(a)gmail.com> wrote: > > On Friday 17 of October 2008 15:24:20 Fred . wrote: > >> ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: > >> Timeout ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to > >> connect: Timeout AL lib: alsa.c:344: Could not open playback device > >> 'default': Connection refused AL lib: oss.c:179: Could not open > >> /dev/dsp: Device or resource busy File > >> "/var/lib/python-support/python2.5/pyglet/media/drivers/openal/__init__. > >>py" , line 340, in driver_init_device = alc.alcOpenDevice(device_name) > >> > >> Audio on Linux is a mess. > >> I launch an Python script that uses audio then it breaks, and when the > >> Python script is closed, I still hear the audio skipping and repeating > >> like, > >> "kakakakakakaka" in infinite loop. > >> And PulseAudio seems to be frozen. > >> Then I kill PulseAudio and finally the audio noise stops. > >> > >> Oh, and I heard OpenAL downsamples stereo to mono on Linux due to a bug. > >> > >> Whole audio on Linux is just a big mess. > > > > Please behave yourself. > > Is there any way we can help you (which I doubt, but ...)? > > > >> _______________________________________________ > >> Alsa-devel mailing list > >> Alsa-devel(a)alsa-project.org > >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

2 1

[alsa-devel] Audio on Linux is a mess
by Fred . 20 Oct '08

20 Oct '08

ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: Timeout ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: Timeout AL lib: alsa.c:344: Could not open playback device 'default': Connection refused AL lib: oss.c:179: Could not open /dev/dsp: Device or resource busy File "/var/lib/python-support/python2.5/pyglet/media/drivers/openal/__init__.py", line 340, in driver_init_device = alc.alcOpenDevice(device_name) Audio on Linux is a mess. I launch an Python script that uses audio then it breaks, and when the Python script is closed, I still hear the audio skipping and repeating like, "kakakakakakaka" in infinite loop. And PulseAudio seems to be frozen. Then I kill PulseAudio and finally the audio noise stops. Oh, and I heard OpenAL downsamples stereo to mono on Linux due to a bug. Whole audio on Linux is just a big mess.

3 2

[alsa-devel] No pcm device?
by John Kacur 20 Oct '08

20 Oct '08

I'm not sure if the hardware I have makes this problem simple for me, or difficult, because it seems to differ from other ASoC code. Basically, I have a i2s as transport, connected to max5556 driving the output. (and a few other chips related to amplification and muting) There is however no sound card, thus no PCM device. Now the question is how to proceed? In other words what kind of minimal PCM layer do I need if any? Thanks

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[alsa-devel] [PATCH] snd-sbxfi: implement Master volume control
by William Pitcock 20 Oct '08

20 Oct '08

Unfortunately, the X-Fi does not have a hardware master volume control, so we have to create a virtual master and bind all playback controls to it. This patch depends on the patch that renames Master to PCM. Signed-off-by: William Pitcock <nenolod(a)sacredspiral.co.uk> --- sound/pci/sbxfi/sbxfi.c | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/sound/pci/sbxfi/sbxfi.c b/sound/pci/sbxfi/sbxfi.c index 3e99a97..d8b32aa 100644 --- a/sound/pci/sbxfi/sbxfi.c +++ b/sound/pci/sbxfi/sbxfi.c @@ -1685,15 +1685,45 @@ static struct snd_kcontrol_new mixers[] __devinitdata = { }, }; +/* list of controls to attach to the master volume control. */ +static const char *slave_vols[] __devinitdata = { + "PCM Playback Volume", +}; + static int __devinit sbxfi_create_mixer(struct sbxfi *chip) { int i, err; + struct snd_kcontrol *vmaster; for (i = 0; i < ARRAY_SIZE(mixers); i++) { err = snd_ctl_add(chip->card, snd_ctl_new1(&mixers[i], chip)); if (err < 0) return err; } + + /* + * Unfortunately, it seems that there is no master control, unlike + * on the emu10k1 series, so we create a virtual master, and bind + * our playback controls to it. + */ + vmaster = snd_ctl_make_virtual_master("Master Playback Volume", NULL); + if (vmaster == NULL) + return -ENOMEM; + + /* attach the master to the card, and attach all slaves to it. */ + snd_ctl_add(chip->card, vmaster); + for (i = 0; i < ARRAY_SIZE(slave_vols); i++) { + struct snd_kcontrol *slave; + struct snd_ctl_elem_id sid; + + memset(&sid, 0, sizeof(sid)); + strlcpy(sid.name, slave_vols[i], 44); /* XXX this should be a constant defined somewhere */ + sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + + if ((slave = snd_ctl_find_id(chip->card, &sid)) != NULL) + snd_ctl_add_slave(vmaster, slave); + } + return 0; } -- 1.5.5.4

3 6

[alsa-devel] Microphone array using ALSA
by Ola Benderius 20 Oct '08

20 Oct '08

Hi! http://en.wikipedia.org/wiki/Microphone_array Would it be possible to use several USB soundcards in GNU/Linux using ALSA to build a microphone array? I have searched the Internet but haven't found any straight answers. What I have read is that it might be hard to sync the captured streams even if it would be possible. So, three questions. 1. Is it theoretically possible? 2. Can it be done using several USB soundcards? 3. Do you know of any code or project in this area as it is? Think this might be a necessary advancement for the future, the applications are many. Even Windows Vista has support for it. Other thoughts, comments? Regards, Ola

1 0

[alsa-devel] ALSA SOC in the ARM linux kernel - pxa2xx-pcm oops on invalid dma_ch.
by Oliver Ford 20 Oct '08

20 Oct '08

Hi, I had been working on a kernel for the iPAQ 21x series (PXA310 based with a WM9713) which I had originally (June 08) based on arm.linux.org.uk's git tree with wolfsonmicro.com/linux-2.6-asoc/asoc-v2-dev merged in. Recently (last week) I tried to pull the latest changes from arm.linux.org.uk and got too many conflicts in the ALSA SOC code, the mainline kernel code seems to have been cleaned up a lot compared to asoc-v2-dev. Due to this I'm now working from the ALSA SOC code as it comes in arm.linux.org.uk. This mostly works but I'm having trouble with oopses due to invalid DMA channels in pxa2xx-pcm.c. Is this the right place to ask about this, if so I'll post the details here (which I already posted on the arm-linux mailing list). Is there a branch on wolfsonmicro.com that would be closer to what is now in the main kernel? Thanks, Oliver Ford

2 1

[alsa-devel] davinci_request_dma()
by John Kacur 20 Oct '08

20 Oct '08

I'm looking for a little bit more help reading the SOC code. In particular, I'm reading davinci-pcm.c, and function davinci_pcm_dma_request, which makes two calls to davinci_request_dma. However, when I grep for davinci_request_dma, the only places I can find it are in davinci_pcm_dma_request itself! [jkacur@mydcore linux-2.6.26]$ grep -rn davinci_request_dma * sound/soc/davinci/davinci-pcm.c:143: ret = davinci_request_dma(prtd->params->channel, prtd->params->name, sound/soc/davinci/davinci-pcm.c:150: ret = davinci_request_dma(PARAM_ANY, "Link", Either I'm missing something, or this code wouldn't even compile. I would appreciate any help in understanding this. Thank you.

2 10

[alsa-devel] snd-xfi info
by James Courtier-Dutton 20 Oct '08

20 Oct '08

Hi, Trying to answer some question raised by comments in the current code. 1) The card should be able to DMA to anywhere in a 64-bit address space. It is set to 32bit DMA by default. A bit has to be set to turn on 64-bit. 2) GPIOs > GPIOs > SB046x / SB067x / SB0760 > Bit In/Out Def Use > 0: Not used > 1: Out 0 Reset DAC > 2: Out 0 Int_Drv_SpdifBypas (1 = Bypass) > 3: Not Used > 4: Not Used > 5: Out 0 In_Drv_DAC_SR (0=48kHz, 1=96kHz) > 6: Out 0 Back_Bracket_Lineouts_Mute (0=Mute) > 7: Out 0 Mic/Line select( 0=Line, 1=Mic) > 8: Out 1 Digital IO Select (0 = Digital, 1 = Analog) > 9: In - Int_Drv_Headphone_Det (1 = headphone detected) > 10: Not used > 11: In - FP_Headphone_Det (0 = headphone detected) > 12: Not used > 13: In - LineOut3 detect (1 = jack inserted) > 14: In - LineOut2 detect (1 = jack inserted) > 15: In - LineOut1 detect (1 = jack inserted) > > SB073x > Bit In/Out Def Use > 0: Not used > 1: Out 0 Reset DAC > 2: Out 0 Int_Drv_SpdifBypas (1 = Bypass) > 3: Not Used > 4: Not Used > 5: Out 0 In_Drv_DAC_SR (0=48kHz, 1=96kHz) > 6: Out 0 Back_Bracket_Lineouts_Mute (0=Mute) > 7: Out 0 Mic/Line select( 0=Line, 1=Mic) > 8: In - Line/Mic In detect (0 = jack inserted???) > 9: In - Int_Drv_Headphone_Det (1 = headphone detected) > 10: Not used > 11: In - FP_Headphone_Det (0 = headphone detected) > 12: In - Digital Out detect (1 = detected) > 13: In - LineOut3 detect (1 = jack inserted) > 14: In - LineOut2 detect (1 = jack inserted) > 15: In - LineOut1 detect (1 = jack inserted) > > Vista compatible > Bit In/Out Def Use > 0: In - FP_Linein_detect (1 = jack inserted) > 1: Out 0 Reset DAC > 2: Not Used > 3: Not Used > 4: Not Used > 5: Not Used > 6: Out 0 Back_Bracket_Lineouts_Mute (0=Mute) > 7: Out 0 Mic/Line select( 0=Line, 1=Mic) > 8: In - Line/Mic In detect (1 = jack inserted???) > 9: In - FP_MicIn_Detect (1 = jack inserted) > 10: Not used > 11: In - FP_Headphone_Det (0 = headphone detected) > 12: In - LineOut4 detect (1 = detected) > 13: In - LineOut3 detect (1 = jack inserted) > 14: In - LineOut2 detect (1 = jack inserted) > 15: In - LineOut1 detect (1 = jack inserted) > > SB055x > Bit In/Out Def Use > 0: In - FP_Mic_detect (1 = jack inserted) > 1: Out 0 Mute all analog inputs(0=Mute) > 2: Out 0 ADC_Gain (0=boost off, 1=boost on) > 3: Out 1 SPDIFOut0 Unmute (0=Mute, 1=Unmute) > 4: Out 0 FP_Mic_select( 1=FPMic) > 5: Out 0 Ext_Drv_DAC_SR (0=48kHz, 1=96kHz) > 6: Out 0 Lineout1_Mute (1=Mute) > 7: Out 0 Mic power on(1=Power on) > 8: Out 0 Digital IO Select (0 = Analog, 1=Digital) > 9: Out 0 Mic_sw(0=Line/Aux, 1=FP/BP Mic) > 10: Not used > 11: In - FP_Headphone_Det (0 = headphone detected) > 12: Out 0 AuxMicLine sw(0=MicLine) > 13: In - LineOut3 detect (1 = jack inserted) > 14: In - LineOut2 detect (1 = jack inserted) > 15: In - LineOut1 detect (1 = jack inserted) > > Card Family > SB046x / > SB067x / > SB076x > = > Subsystem ID > 0x0021 > 0x0023 > 0x0024 > 0x0025 > 0x0026 > 0x0027 > 0x0028 > 0x002A > 0x002B > 0x002C > 0x002D > 0x002E > 0x0032 > 0x0033 > > Card Family > SB073x > = > Subsystem ID > 0x0029 > 0x0031 > > Card Family > SB055x > = > Subsystem ID > 0x0022 > 0x002F > > Card Family > Vista Compatible Hardware > = > Subsystem ID > Range from 0x6000 to 0x6FFF > More later James

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