aboutsummaryrefslogtreecommitdiff
path: root/sound/pci/hda
diff options
context:
space:
mode:
Diffstat (limited to 'sound/pci/hda')
-rw-r--r--sound/pci/hda/hda_codec.c52
-rw-r--r--sound/pci/hda/hda_codec.h2
-rw-r--r--sound/pci/hda/hda_intel.c72
-rw-r--r--sound/pci/hda/hda_local.h22
-rw-r--r--sound/pci/hda/hda_proc.c5
-rw-r--r--sound/pci/hda/patch_analog.c133
-rw-r--r--sound/pci/hda/patch_realtek.c274
-rw-r--r--sound/pci/hda/patch_si3054.c2
8 files changed, 401 insertions, 161 deletions
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3815403ed09..0dbeeaf6113 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -518,6 +518,13 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
return -ENODEV;
}
+ if (! codec->subsystem_id) {
+ hda_nid_t nid = codec->afg ? codec->afg : codec->mfg;
+ codec->subsystem_id = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_SUBSYSTEM_ID,
+ 0);
+ }
+
codec->preset = find_codec_preset(codec);
if (! *bus->card->mixername)
snd_hda_get_codec_name(codec, bus->card->mixername,
@@ -814,6 +821,51 @@ int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t
}
/*
+ * bound volume controls
+ *
+ * bind multiple volumes (# indices, from 0)
+ */
+
+#define AMP_VAL_IDX_SHIFT 19
+#define AMP_VAL_IDX_MASK (0x0f<<19)
+
+int snd_hda_mixer_bind_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned long pval;
+ int err;
+
+ down(&codec->spdif_mutex); /* reuse spdif_mutex */
+ pval = kcontrol->private_value;
+ kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */
+ err = snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
+ kcontrol->private_value = pval;
+ up(&codec->spdif_mutex);
+ return err;
+}
+
+int snd_hda_mixer_bind_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned long pval;
+ int i, indices, err = 0, change = 0;
+
+ down(&codec->spdif_mutex); /* reuse spdif_mutex */
+ pval = kcontrol->private_value;
+ indices = (pval & AMP_VAL_IDX_MASK) >> AMP_VAL_IDX_SHIFT;
+ for (i = 0; i < indices; i++) {
+ kcontrol->private_value = (pval & ~AMP_VAL_IDX_MASK) | (i << AMP_VAL_IDX_SHIFT);
+ err = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+ if (err < 0)
+ break;
+ change |= err;
+ }
+ kcontrol->private_value = pval;
+ up(&codec->spdif_mutex);
+ return err < 0 ? err : change;
+}
+
+/*
* SPDIF out controls
*/
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index bb53bcf7674..1179d6cfa82 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -79,6 +79,8 @@ enum {
#define AC_VERB_GET_GPIO_MASK 0x0f16
#define AC_VERB_GET_GPIO_DIRECTION 0x0f17
#define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c
+/* f20: AFG/MFG */
+#define AC_VERB_GET_SUBSYSTEM_ID 0x0f20
/*
* SET verbs
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 6fe696e53ea..9d1412a9f2f 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -47,23 +47,24 @@
#include "hda_codec.h"
-static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
-static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
-static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
-static char *model[SNDRV_CARDS];
-static int position_fix[SNDRV_CARDS];
+static int index = SNDRV_DEFAULT_IDX1;
+static char *id = SNDRV_DEFAULT_STR1;
+static char *model;
+static int position_fix;
-module_param_array(index, int, NULL, 0444);
+module_param(index, int, 0444);
MODULE_PARM_DESC(index, "Index value for Intel HD audio interface.");
-module_param_array(id, charp, NULL, 0444);
+module_param(id, charp, 0444);
MODULE_PARM_DESC(id, "ID string for Intel HD audio interface.");
-module_param_array(enable, bool, NULL, 0444);
-MODULE_PARM_DESC(enable, "Enable Intel HD audio interface.");
-module_param_array(model, charp, NULL, 0444);
+module_param(model, charp, 0444);
MODULE_PARM_DESC(model, "Use the given board model.");
-module_param_array(position_fix, int, NULL, 0444);
+module_param(position_fix, int, 0444);
MODULE_PARM_DESC(position_fix, "Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size).");
+/* just for backward compatibility */
+static int enable;
+module_param(enable, bool, 0444);
+
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, ICH6M},"
@@ -223,6 +224,9 @@ enum {
#define ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR 0x42
#define ATI_SB450_HDAUDIO_ENABLE_SNOOP 0x02
+/* Defines for Nvidia HDA support */
+#define NVIDIA_HDA_TRANSREG_ADDR 0x4e
+#define NVIDIA_HDA_ENABLE_COHBITS 0x0f
/*
* Use CORB/RIRB for communication from/to codecs.
@@ -328,6 +332,7 @@ enum {
AZX_DRIVER_VIA,
AZX_DRIVER_SIS,
AZX_DRIVER_ULI,
+ AZX_DRIVER_NVIDIA,
};
static char *driver_short_names[] __devinitdata = {
@@ -335,7 +340,8 @@ static char *driver_short_names[] __devinitdata = {
[AZX_DRIVER_ATI] = "HDA ATI SB",
[AZX_DRIVER_VIA] = "HDA VIA VT82xx",
[AZX_DRIVER_SIS] = "HDA SIS966",
- [AZX_DRIVER_ULI] = "HDA ULI M5461"
+ [AZX_DRIVER_ULI] = "HDA ULI M5461",
+ [AZX_DRIVER_NVIDIA] = "HDA NVidia",
};
/*
@@ -710,14 +716,14 @@ static void azx_stream_stop(azx_t *chip, azx_dev_t *azx_dev)
*/
static void azx_init_chip(azx_t *chip)
{
- unsigned char tcsel_reg, ati_misc_cntl2;
+ unsigned char reg;
/* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
* TCSEL == Traffic Class Select Register, which sets PCI express QOS
* Ensuring these bits are 0 clears playback static on some HD Audio codecs
*/
- pci_read_config_byte (chip->pci, ICH6_PCIREG_TCSEL, &tcsel_reg);
- pci_write_config_byte(chip->pci, ICH6_PCIREG_TCSEL, tcsel_reg & 0xf8);
+ pci_read_config_byte (chip->pci, ICH6_PCIREG_TCSEL, &reg);
+ pci_write_config_byte(chip->pci, ICH6_PCIREG_TCSEL, reg & 0xf8);
/* reset controller */
azx_reset(chip);
@@ -733,13 +739,21 @@ static void azx_init_chip(azx_t *chip)
azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr));
- /* For ATI SB450 azalia HD audio, we need to enable snoop */
- if (chip->driver_type == AZX_DRIVER_ATI) {
+ switch (chip->driver_type) {
+ case AZX_DRIVER_ATI:
+ /* For ATI SB450 azalia HD audio, we need to enable snoop */
pci_read_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
- &ati_misc_cntl2);
+ &reg);
pci_write_config_byte(chip->pci, ATI_SB450_HDAUDIO_MISC_CNTR2_ADDR,
- (ati_misc_cntl2 & 0xf8) | ATI_SB450_HDAUDIO_ENABLE_SNOOP);
- }
+ (reg & 0xf8) | ATI_SB450_HDAUDIO_ENABLE_SNOOP);
+ break;
+ case AZX_DRIVER_NVIDIA:
+ /* For NVIDIA HDA, enable snoop */
+ pci_read_config_byte(chip->pci,NVIDIA_HDA_TRANSREG_ADDR, &reg);
+ pci_write_config_byte(chip->pci,NVIDIA_HDA_TRANSREG_ADDR,
+ (reg & 0xf0) | NVIDIA_HDA_ENABLE_COHBITS);
+ break;
+ }
}
@@ -1264,6 +1278,7 @@ static int __devinit azx_pcm_create(azx_t *chip)
err = create_codec_pcm(chip, codec, &codec->pcm_info[c], pcm_dev);
if (err < 0)
return err;
+ chip->pcm[pcm_dev]->dev_class = SNDRV_PCM_CLASS_MODEM;
pcm_dev++;
}
}
@@ -1530,32 +1545,24 @@ static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci,
static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
{
- static int dev;
snd_card_t *card;
azx_t *chip;
int err = 0;
- if (dev >= SNDRV_CARDS)
- return -ENODEV;
- if (! enable[dev]) {
- dev++;
- return -ENOENT;
- }
-
- card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ card = snd_card_new(index, id, THIS_MODULE, 0);
if (NULL == card) {
snd_printk(KERN_ERR SFX "Error creating card!\n");
return -ENOMEM;
}
- if ((err = azx_create(card, pci, position_fix[dev], pci_id->driver_data,
+ if ((err = azx_create(card, pci, position_fix, pci_id->driver_data,
&chip)) < 0) {
snd_card_free(card);
return err;
}
/* create codec instances */
- if ((err = azx_codec_create(chip, model[dev])) < 0) {
+ if ((err = azx_codec_create(chip, model)) < 0) {
snd_card_free(card);
return err;
}
@@ -1581,7 +1588,6 @@ static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *
}
pci_set_drvdata(pci, card);
- dev++;
return err;
}
@@ -1601,6 +1607,8 @@ static struct pci_device_id azx_ids[] = {
{ 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */
{ 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */
{ 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */
+ { 0x10de, 0x026c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA 026c */
+ { 0x10de, 0x0371, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA 0371 */
{ 0, }
};
MODULE_DEVICE_TABLE(pci, azx_ids);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 810cfd2d9bb..f51a56f813c 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -27,28 +27,36 @@
* for mixer controls
*/
#define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19))
+/* mono volume with index (index=0,1,...) (channel=1,2) */
#define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
.info = snd_hda_mixer_amp_volume_info, \
.get = snd_hda_mixer_amp_volume_get, \
.put = snd_hda_mixer_amp_volume_put, \
.private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
+/* stereo volume with index */
#define HDA_CODEC_VOLUME_IDX(xname, xcidx, nid, xindex, direction) \
HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, 3, xindex, direction)
+/* mono volume */
#define HDA_CODEC_VOLUME_MONO(xname, nid, channel, xindex, direction) \
HDA_CODEC_VOLUME_MONO_IDX(xname, 0, nid, channel, xindex, direction)
+/* stereo volume */
#define HDA_CODEC_VOLUME(xname, nid, xindex, direction) \
HDA_CODEC_VOLUME_MONO(xname, nid, 3, xindex, direction)
+/* mono mute switch with index (index=0,1,...) (channel=1,2) */
#define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
.info = snd_hda_mixer_amp_switch_info, \
.get = snd_hda_mixer_amp_switch_get, \
.put = snd_hda_mixer_amp_switch_put, \
.private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
+/* stereo mute switch with index */
#define HDA_CODEC_MUTE_IDX(xname, xcidx, nid, xindex, direction) \
HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, 3, xindex, direction)
+/* mono mute switch */
#define HDA_CODEC_MUTE_MONO(xname, nid, channel, xindex, direction) \
HDA_CODEC_MUTE_MONO_IDX(xname, 0, nid, channel, xindex, direction)
+/* stereo mute switch */
#define HDA_CODEC_MUTE(xname, nid, xindex, direction) \
HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction)
@@ -59,6 +67,20 @@ int snd_hda_mixer_amp_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t
int snd_hda_mixer_amp_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol);
int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol);
+/* mono switch binding multiple inputs */
+#define HDA_BIND_MUTE_MONO(xname, nid, channel, indices, direction) \
+ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
+ .info = snd_hda_mixer_amp_switch_info, \
+ .get = snd_hda_mixer_bind_switch_get, \
+ .put = snd_hda_mixer_bind_switch_put, \
+ .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, indices, direction) }
+
+/* stereo switch binding multiple inputs */
+#define HDA_BIND_MUTE(xname,nid,indices,dir) HDA_BIND_MUTE_MONO(xname,nid,3,indices,dir)
+
+int snd_hda_mixer_bind_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol);
+int snd_hda_mixer_bind_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol);
+
int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 08f6a6efc5e..39ddf1cd901 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -281,6 +281,11 @@ static void print_codec_info(snd_info_entry_t *entry, snd_info_buffer_t *buffer)
print_pcm_caps(buffer, codec, nid);
}
+ if (wid_caps & AC_WCAP_POWER)
+ snd_iprintf(buffer, " Power: 0x%x\n",
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_POWER_STATE, 0));
+
if (wid_caps & AC_WCAP_CONN_LIST) {
int c, curr = -1;
if (conn_len > 1 && wid_type != AC_WID_AUD_MIX)
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index da6874d3988..d7d636decef 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -28,15 +28,38 @@
#include "hda_local.h"
struct ad198x_spec {
- struct semaphore amp_mutex; /* PCM volume/mute control mutex */
- struct hda_multi_out multiout; /* playback */
- hda_nid_t adc_nid;
+ snd_kcontrol_new_t *mixers[5];
+ int num_mixers;
+
+ const struct hda_verb *init_verbs[3]; /* initialization verbs
+ * don't forget NULL termination!
+ */
+ unsigned int num_init_verbs;
+
+ /* playback */
+ struct hda_multi_out multiout; /* playback set-up
+ * max_channels, dacs must be set
+ * dig_out_nid and hp_nid are optional
+ */
+
+ /* capture */
+ unsigned int num_adc_nids;
+ hda_nid_t *adc_nids;
+ hda_nid_t dig_in_nid; /* digital-in NID; optional */
+
+ /* capture source */
const struct hda_input_mux *input_mux;
- unsigned int cur_mux; /* capture source */
+ unsigned int cur_mux[3];
+
+ /* channel model */
+ const struct alc_channel_mode *channel_mode;
+ int num_channel_mode;
+
+ /* PCM information */
+ struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */
+
+ struct semaphore amp_mutex; /* PCM volume/mute control mutex */
unsigned int spdif_route;
- snd_kcontrol_new_t *mixers;
- const struct hda_verb *init_verbs;
- struct hda_pcm pcm_rec[2]; /* PCM information */
};
/*
@@ -54,8 +77,9 @@ static int ad198x_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *u
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct ad198x_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
- ucontrol->value.enumerated.item[0] = spec->cur_mux;
+ ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
return 0;
}
@@ -63,9 +87,10 @@ static int ad198x_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *u
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
struct ad198x_spec *spec = codec->spec;
+ unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
- spec->adc_nid, &spec->cur_mux);
+ spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]);
}
/*
@@ -74,22 +99,34 @@ static int ad198x_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *u
static int ad198x_init(struct hda_codec *codec)
{
struct ad198x_spec *spec = codec->spec;
- snd_hda_sequence_write(codec, spec->init_verbs);
+ int i;
+
+ for (i = 0; i < spec->num_init_verbs; i++)
+ snd_hda_sequence_write(codec, spec->init_verbs[i]);
return 0;
}
static int ad198x_build_controls(struct hda_codec *codec)
{
struct ad198x_spec *spec = codec->spec;
+ unsigned int i;
int err;
- err = snd_hda_add_new_ctls(codec, spec->mixers);
- if (err < 0)
- return err;
- if (spec->multiout.dig_out_nid)
+ for (i = 0; i < spec->num_mixers; i++) {
+ err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
+ if (err < 0)
+ return err;
+ }
+ if (spec->multiout.dig_out_nid) {
err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
- if (err < 0)
- return err;
+ if (err < 0)
+ return err;
+ }
+ if (spec->dig_in_nid) {
+ err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
+ if (err < 0)
+ return err;
+ }
return 0;
}
@@ -152,7 +189,8 @@ static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
snd_pcm_substream_t *substream)
{
struct ad198x_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nid, stream_tag, 0, format);
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+ stream_tag, 0, format);
return 0;
}
@@ -161,7 +199,8 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
snd_pcm_substream_t *substream)
{
struct ad198x_spec *spec = codec->spec;
- snd_hda_codec_setup_stream(codec, spec->adc_nid, 0, 0, 0);
+ snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+ 0, 0, 0);
return 0;
}
@@ -171,7 +210,7 @@ static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
static struct hda_pcm_stream ad198x_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
- .channels_max = 6,
+ .channels_max = 6, /* changed later */
.nid = 0, /* fill later */
.ops = {
.open = ad198x_playback_pcm_open,
@@ -181,7 +220,7 @@ static struct hda_pcm_stream ad198x_pcm_analog_playback = {
};
static struct hda_pcm_stream ad198x_pcm_analog_capture = {
- .substreams = 2,
+ .substreams = 1,
.channels_min = 2,
.channels_max = 2,
.nid = 0, /* fill later */
@@ -202,6 +241,13 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = {
},
};
+static struct hda_pcm_stream ad198x_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ /* NID is set in alc_build_pcms */
+};
+
static int ad198x_build_pcms(struct hda_codec *codec)
{
struct ad198x_spec *spec = codec->spec;
@@ -215,7 +261,8 @@ static int ad198x_build_pcms(struct hda_codec *codec)
info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture;
- info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nid;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
if (spec->multiout.dig_out_nid) {
info++;
@@ -223,6 +270,10 @@ static int ad198x_build_pcms(struct hda_codec *codec)
info->name = "AD198x Digital";
info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback;
info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
+ if (spec->dig_in_nid) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
+ }
}
return 0;
@@ -237,10 +288,15 @@ static void ad198x_free(struct hda_codec *codec)
static int ad198x_resume(struct hda_codec *codec)
{
struct ad198x_spec *spec = codec->spec;
+ int i;
ad198x_init(codec);
- snd_hda_resume_ctls(codec, spec->mixers);
- snd_hda_resume_spdif_out(codec);
+ for (i = 0; i < spec->num_mixers; i++)
+ snd_hda_resume_ctls(codec, spec->mixers[i]);
+ if (spec->multiout.dig_out_nid)
+ snd_hda_resume_spdif_out(codec);
+ if (spec->dig_in_nid)
+ snd_hda_resume_spdif_in(codec);
return 0;
}
#endif
@@ -269,6 +325,7 @@ static struct hda_codec_ops ad198x_patch_ops = {
static hda_nid_t ad1986a_dac_nids[3] = {
AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC
};
+static hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC };
static struct hda_input_mux ad1986a_capture_source = {
.num_items = 7,
@@ -476,10 +533,13 @@ static int patch_ad1986a(struct hda_codec *codec)
spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids);
spec->multiout.dac_nids = ad1986a_dac_nids;
spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT;
- spec->adc_nid = AD1986A_ADC;
+ spec->num_adc_nids = 1;
+ spec->adc_nids = ad1986a_adc_nids;
spec->input_mux = &ad1986a_capture_source;
- spec->mixers = ad1986a_mixers;
- spec->init_verbs = ad1986a_init_verbs;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1986a_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1986a_init_verbs;
codec->patch_ops = ad198x_patch_ops;
@@ -495,6 +555,7 @@ static int patch_ad1986a(struct hda_codec *codec)
#define AD1983_ADC 0x04
static hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
+static hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC };
static struct hda_input_mux ad1983_capture_source = {
.num_items = 4,
@@ -619,6 +680,7 @@ static struct hda_verb ad1983_init_verbs[] = {
{ } /* end */
};
+
static int patch_ad1983(struct hda_codec *codec)
{
struct ad198x_spec *spec;
@@ -634,10 +696,13 @@ static int patch_ad1983(struct hda_codec *codec)
spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids);
spec->multiout.dac_nids = ad1983_dac_nids;
spec->multiout.dig_out_nid = AD1983_SPDIF_OUT;
- spec->adc_nid = AD1983_ADC;
+ spec->num_adc_nids = 1;
+ spec->adc_nids = ad1983_adc_nids;
spec->input_mux = &ad1983_capture_source;
- spec->mixers = ad1983_mixers;
- spec->init_verbs = ad1983_init_verbs;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1983_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1983_init_verbs;
spec->spdif_route = 0;
codec->patch_ops = ad198x_patch_ops;
@@ -655,6 +720,7 @@ static int patch_ad1983(struct hda_codec *codec)
#define AD1981_ADC 0x04
static hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
+static hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC };
/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */
static struct hda_input_mux ad1981_capture_source = {
@@ -775,10 +841,13 @@ static int patch_ad1981(struct hda_codec *codec)
spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids);
spec->multiout.dac_nids = ad1981_dac_nids;
spec->multiout.dig_out_nid = AD1981_SPDIF_OUT;
- spec->adc_nid = AD1981_ADC;
+ spec->num_adc_nids = 1;
+ spec->adc_nids = ad1981_adc_nids;
spec->input_mux = &ad1981_capture_source;
- spec->mixers = ad1981_mixers;
- spec->init_verbs = ad1981_init_verbs;
+ spec->num_mixers = 1;
+ spec->mixers[0] = ad1981_mixers;
+ spec->num_init_verbs = 1;
+ spec->init_verbs[0] = ad1981_init_verbs;
spec->spdif_route = 0;
codec->patch_ops = ad198x_patch_ops;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7327deb6df9..cffb83fdcff 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -57,6 +57,7 @@ enum {
enum {
ALC260_BASIC,
ALC260_HP,
+ ALC260_FUJITSU_S702x,
ALC260_MODEL_LAST /* last tag */
};
@@ -72,6 +73,7 @@ enum {
#define PIN_VREF50 0x21
#define PIN_OUT 0x40
#define PIN_HP 0xc0
+#define PIN_HP_AMP 0x80
struct alc_spec {
/* codec parameterization */
@@ -113,8 +115,6 @@ struct alc_spec {
/* PCM information */
struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */
- struct semaphore bind_mutex; /* for bound controls */
-
/* dynamic controls, init_verbs and input_mux */
struct auto_pin_cfg autocfg;
unsigned int num_kctl_alloc, num_kctl_used;
@@ -218,72 +218,53 @@ static int alc880_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *uc
/*
- * bound volume controls
- *
- * bind multiple volumes (# indices, from 0)
+ * Control of pin widget settings via the mixer. Only boolean settings are
+ * supported, so VrefEn can't be controlled using these functions as they
+ * stand.
*/
-
-#define AMP_VAL_IDX_SHIFT 19
-#define AMP_VAL_IDX_MASK (0x0f<<19)
-
-static int alc_bind_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+static int alc_pinctl_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- unsigned long pval;
-
- down(&spec->bind_mutex);
- pval = kcontrol->private_value;
- kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */
- snd_hda_mixer_amp_switch_info(kcontrol, uinfo);
- kcontrol->private_value = pval;
- up(&spec->bind_mutex);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
return 0;
}
-static int alc_bind_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int alc_pinctl_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- unsigned long pval;
-
- down(&spec->bind_mutex);
- pval = kcontrol->private_value;
- kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */
- snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
- kcontrol->private_value = pval;
- up(&spec->bind_mutex);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ long mask = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+
+ *valp = 0;
+ if (snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00) & mask)
+ *valp = 1;
return 0;
}
-static int alc_bind_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+static int alc_pinctl_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
{
struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- unsigned long pval;
- int i, indices, change = 0;
-
- down(&spec->bind_mutex);
- pval = kcontrol->private_value;
- indices = (pval & AMP_VAL_IDX_MASK) >> AMP_VAL_IDX_SHIFT;
- for (i = 0; i < indices; i++) {
- kcontrol->private_value = (pval & ~AMP_VAL_IDX_MASK) | (i << AMP_VAL_IDX_SHIFT);
- change |= snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
- }
- kcontrol->private_value = pval;
- up(&spec->bind_mutex);
+ hda_nid_t nid = kcontrol->private_value & 0xffff;
+ long mask = (kcontrol->private_value >> 16) & 0xff;
+ long *valp = ucontrol->value.integer.value;
+ unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00);
+ int change = ((pinctl & mask)!=0) != *valp;
+
+ if (change)
+ snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL,
+ *valp?(pinctl|mask):(pinctl&~mask));
return change;
}
-#define ALC_BIND_MUTE_MONO(xname, nid, channel, indices, direction) \
+#define ALC_PINCTL_SWITCH(xname, nid, mask) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .info = alc_bind_switch_info, \
- .get = alc_bind_switch_get, \
- .put = alc_bind_switch_put, \
- .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, indices, direction) }
-
-#define ALC_BIND_MUTE(xname,nid,indices,dir) ALC_BIND_MUTE_MONO(xname,nid,3,indices,dir)
-
+ .info = alc_pinctl_switch_info, \
+ .get = alc_pinctl_switch_get, \
+ .put = alc_pinctl_switch_put, \
+ .private_value = (nid) | (mask<<16) }
/*
* ALC880 3-stack model
@@ -354,13 +335,13 @@ static struct alc_channel_mode alc880_threestack_modes[2] = {
static snd_kcontrol_new_t alc880_three_stack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
@@ -441,7 +422,7 @@ static snd_kcontrol_new_t alc880_capture_alt_mixer[] = {
/* additional mixers to alc880_three_stack_mixer */
static snd_kcontrol_new_t alc880_five_stack_mixer[] = {
HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
{ } /* end */
};
@@ -498,15 +479,15 @@ static struct alc_channel_mode alc880_sixstack_modes[1] = {
static snd_kcontrol_new_t alc880_six_stack_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Side 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("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
@@ -566,13 +547,13 @@ static struct alc_channel_mode alc880_w810_modes[1] = {
/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
static snd_kcontrol_new_t alc880_w810_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
{ } /* end */
};
@@ -597,9 +578,9 @@ static struct alc_channel_mode alc880_2_jack_modes[1] = {
static snd_kcontrol_new_t alc880_z71v_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Headphone Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0d, 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),
@@ -623,9 +604,9 @@ static hda_nid_t alc880_f1734_dac_nids[1] = {
static snd_kcontrol_new_t alc880_f1734_mixer[] = {
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Internal Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x0d, 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),
@@ -648,13 +629,13 @@ static snd_kcontrol_new_t alc880_f1734_mixer[] = {
static snd_kcontrol_new_t alc880_asus_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
@@ -1383,10 +1364,10 @@ static snd_kcontrol_new_t alc880_test_mixer[] = {
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CLFE Playback Volume", 0x0e, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- ALC_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT),
- ALC_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("CLFE Playback Switch", 0x0e, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
PIN_CTL_TEST("Front Pin Mode", 0x14),
PIN_CTL_TEST("Surround Pin Mode", 0x15),
PIN_CTL_TEST("CLFE Pin Mode", 0x16),
@@ -1769,7 +1750,7 @@ enum {
static snd_kcontrol_new_t alc880_control_templates[] = {
HDA_CODEC_VOLUME(NULL, 0, 0, 0),
HDA_CODEC_MUTE(NULL, 0, 0, 0),
- ALC_BIND_MUTE(NULL, 0, 0, 0),
+ HDA_BIND_MUTE(NULL, 0, 0, 0),
};
/* add dynamic controls */
@@ -2087,7 +2068,6 @@ static int patch_alc880(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
- init_MUTEX(&spec->bind_mutex);
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, alc880_cfg_tbl);
@@ -2205,6 +2185,17 @@ static struct hda_input_mux alc260_capture_source = {
},
};
+/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack
+ * and the internal CD lines.
+ */
+static struct hda_input_mux alc260_fujitsu_capture_source = {
+ .num_items = 2,
+ .items = {
+ { "Mic/Line", 0x0 },
+ { "CD", 0x4 },
+ },
+};
+
/*
* This is just place-holder, so there's something for alc_build_pcms to look
* at when it calculates the maximum number of channels. ALC260 has no mixer
@@ -2217,7 +2208,7 @@ static struct alc_channel_mode alc260_modes[1] = {
static snd_kcontrol_new_t alc260_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
@@ -2229,9 +2220,9 @@ static snd_kcontrol_new_t alc260_base_mixer[] = {
HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
{
@@ -2246,7 +2237,7 @@ static snd_kcontrol_new_t alc260_base_mixer[] = {
static snd_kcontrol_new_t alc260_hp_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
@@ -2256,9 +2247,9 @@ static snd_kcontrol_new_t alc260_hp_mixer[] = {
HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x05, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Capture Switch", 0x05, 0x0, HDA_INPUT),
{
@@ -2271,6 +2262,30 @@ static snd_kcontrol_new_t alc260_hp_mixer[] = {
{ } /* end */
};
+static snd_kcontrol_new_t alc260_fujitsu_mixer[] = {
+ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
+ ALC_PINCTL_SWITCH("Headphone Amp Switch", 0x14, PIN_HP_AMP),
+ HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+ HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
+ HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
+ HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
+ HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+ HDA_BIND_MUTE("Internal Speaker Playback Switch", 0x09, 2, HDA_INPUT),
+ HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
+ HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .info = alc_mux_enum_info,
+ .get = alc_mux_enum_get,
+ .put = alc_mux_enum_put,
+ },
+ { } /* end */
+};
+
static struct hda_verb alc260_init_verbs[] = {
/* Line In pin widget for input */
{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
@@ -2332,6 +2347,60 @@ static struct hda_verb alc260_init_verbs[] = {
{ }
};
+/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
+ * laptops.
+ */
+static struct hda_verb alc260_fujitsu_init_verbs[] = {
+ /* Disable all GPIOs */
+ {0x01, AC_VERB_SET_GPIO_MASK, 0},
+ /* Internal speaker is connected to headphone pin */
+ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+ /* Headphone/Line-out jack connects to Line1 pin; make it an output */
+ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+ /* Mic/Line-in jack is connected to mic1 pin, so make it an input */
+ {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+ /* Ensure all other unused pins are disabled and muted.
+ * Note: trying to set widget 0x15 to anything blocks all audio
+ * output for some reason, so just leave that at the default.
+ */
+ {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+ {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Disable digital (SPDIF) pins */
+ {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+ {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+ /* Start with mixer outputs muted */
+ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+ {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+ /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */
+ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute Line1 pin widget amp left and right (no equiv mixer ctrl) */
+ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+ /* Unmute pin widget used for Line-in (no equiv mixer ctrl) */
+ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+ /* Mute capture amp left and right */
+ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+ /* Set ADC connection select to line in (on mic1 pin) */
+ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+ /* Mute all inputs to mixer widget (even unconnected ones) */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+};
+
static struct hda_pcm_stream alc260_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
@@ -2347,6 +2416,8 @@ static struct hda_pcm_stream alc260_pcm_analog_capture = {
static struct hda_board_config alc260_cfg_tbl[] = {
{ .modelname = "hp", .config = ALC260_HP },
{ .pci_subvendor = 0x103c, .config = ALC260_HP },
+ { .modelname = "fujitsu", .config = ALC260_FUJITSU_S702x },
+ { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1326, .config = ALC260_FUJITSU_S702x },
{}
};
@@ -2359,7 +2430,6 @@ static int patch_alc260(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
- init_MUTEX(&spec->bind_mutex);
codec->spec = spec;
board_config = snd_hda_check_board_config(codec, alc260_cfg_tbl);
@@ -2373,14 +2443,23 @@ static int patch_alc260(struct hda_codec *codec)
spec->mixers[spec->num_mixers] = alc260_hp_mixer;
spec->num_mixers++;
break;
+ case ALC260_FUJITSU_S702x:
+ spec->mixers[spec->num_mixers] = alc260_fujitsu_mixer;
+ spec->num_mixers++;
+ break;
default:
spec->mixers[spec->num_mixers] = alc260_base_mixer;
spec->num_mixers++;
break;
}
- spec->init_verbs[0] = alc260_init_verbs;
- spec->num_init_verbs = 1;
+ if (board_config != ALC260_FUJITSU_S702x) {
+ spec->init_verbs[0] = alc260_init_verbs;
+ spec->num_init_verbs = 1;
+ } else {
+ spec->init_verbs[0] = alc260_fujitsu_init_verbs;
+ spec->num_init_verbs = 1;
+ }
spec->channel_mode = alc260_modes;
spec->num_channel_mode = ARRAY_SIZE(alc260_modes);
@@ -2393,7 +2472,11 @@ static int patch_alc260(struct hda_codec *codec)
spec->multiout.num_dacs = ARRAY_SIZE(alc260_dac_nids);
spec->multiout.dac_nids = alc260_dac_nids;
- spec->input_mux = &alc260_capture_source;
+ if (board_config != ALC260_FUJITSU_S702x) {
+ spec->input_mux = &alc260_capture_source;
+ } else {
+ spec->input_mux = &alc260_fujitsu_capture_source;
+ }
switch (board_config) {
case ALC260_HP:
spec->num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids);
@@ -2483,15 +2566,15 @@ static int alc882_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *u
*/
static snd_kcontrol_new_t alc882_base_mixer[] = {
HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- ALC_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+ HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- ALC_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
+ HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
@@ -2609,7 +2692,6 @@ static int patch_alc882(struct hda_codec *codec)
if (spec == NULL)
return -ENOMEM;
- init_MUTEX(&spec->bind_mutex);
codec->spec = spec;
spec->mixers[spec->num_mixers] = alc882_base_mixer;
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index d014b7bb70d..9c7fe0b3200 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -3,7 +3,7 @@
*
* HD audio interface patch for Silicon Labs 3054/5 modem codec
*
- * Copyright (c) 2005 Sasha Khapyorsky <sashak@smlink.com>
+ * Copyright (c) 2005 Sasha Khapyorsky <sashak@alsa-project.org>
* Takashi Iwai <tiwai@suse.de>
*
*
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-03 11:02:13 +0000