summaryrefslogtreecommitdiff
path: root/sound
diff options
context:
space:
mode:
Diffstat (limited to 'sound')
-rw-r--r--sound/aoa/codecs/onyx.c13
-rw-r--r--sound/aoa/codecs/tas.c13
-rw-r--r--sound/arm/aaci.c13
-rw-r--r--sound/core/compress_offload.c13
-rw-r--r--sound/core/control.c2
-rw-r--r--sound/core/init.c170
-rw-r--r--sound/core/jack.c4
-rw-r--r--sound/core/misc.c2
-rw-r--r--sound/core/pcm.c100
-rw-r--r--sound/core/pcm_lib.c3
-rw-r--r--sound/core/pcm_native.c15
-rw-r--r--sound/core/seq/seq.c1
-rw-r--r--sound/core/timer.c1
-rw-r--r--sound/core/vmaster.c46
-rw-r--r--sound/firewire/isight.c4
-rw-r--r--sound/firewire/speakers.c4
-rw-r--r--sound/i2c/other/tea575x-tuner.c169
-rw-r--r--sound/isa/sb/emu8000_patch.c1
-rw-r--r--sound/oss/os.h1
-rw-r--r--sound/oss/vidc.c1
-rw-r--r--sound/oss/waveartist.c1
-rw-r--r--sound/pci/asihpi/hpios.h1
-rw-r--r--sound/pci/au88x0/au88x0.h13
-rw-r--r--sound/pci/au88x0/au88x0_core.c20
-rw-r--r--sound/pci/au88x0/au88x0_pcm.c127
-rw-r--r--sound/pci/aw2/aw2-saa7146.c1
-rw-r--r--sound/pci/azt3328.c3
-rw-r--r--sound/pci/ctxfi/ctvmem.c2
-rw-r--r--sound/pci/es1968.c15
-rw-r--r--sound/pci/fm801.c20
-rw-r--r--sound/pci/hda/alc260_quirks.c968
-rw-r--r--sound/pci/hda/alc880_quirks.c1700
-rw-r--r--sound/pci/hda/alc882_quirks.c861
-rw-r--r--sound/pci/hda/alc_quirks.c480
-rw-r--r--sound/pci/hda/hda_codec.c206
-rw-r--r--sound/pci/hda/hda_codec.h4
-rw-r--r--sound/pci/hda/hda_eld.c4
-rw-r--r--sound/pci/hda/hda_intel.c54
-rw-r--r--sound/pci/hda/hda_jack.c40
-rw-r--r--sound/pci/hda/hda_jack.h13
-rw-r--r--sound/pci/hda/hda_local.h30
-rw-r--r--sound/pci/hda/patch_analog.c72
-rw-r--r--sound/pci/hda/patch_ca0132.c33
-rw-r--r--sound/pci/hda/patch_cirrus.c10
-rw-r--r--sound/pci/hda/patch_conexant.c150
-rw-r--r--sound/pci/hda/patch_hdmi.c2
-rw-r--r--sound/pci/hda/patch_realtek.c1955
-rw-r--r--sound/pci/hda/patch_sigmatel.c224
-rw-r--r--sound/pci/hda/patch_via.c335
-rw-r--r--sound/pci/ice1712/ice1724.c23
-rw-r--r--sound/pci/intel8x0.c6
-rw-r--r--sound/pci/oxygen/oxygen_mixer.c25
-rw-r--r--sound/pci/rme9652/hdspm.c1
-rw-r--r--sound/pci/ymfpci/ymfpci.c21
-rw-r--r--sound/pci/ymfpci/ymfpci_main.c30
-rw-r--r--sound/soc/Kconfig3
-rw-r--r--sound/soc/Makefile3
-rw-r--r--sound/soc/atmel/atmel-pcm.c4
-rw-r--r--sound/soc/atmel/snd-soc-afeb9260.c37
-rw-r--r--sound/soc/blackfin/bf5xx-ad1836.c17
-rw-r--r--sound/soc/blackfin/bf5xx-ad193x.c17
-rw-r--r--sound/soc/blackfin/bf5xx-ad73311.c29
-rw-r--r--sound/soc/blackfin/bf5xx-ssm2602.c20
-rw-r--r--sound/soc/blackfin/bfin-eval-adau1373.c13
-rw-r--r--sound/soc/blackfin/bfin-eval-adau1701.c16
-rw-r--r--sound/soc/blackfin/bfin-eval-adav80x.c13
-rw-r--r--sound/soc/codecs/Kconfig8
-rw-r--r--sound/soc/codecs/Makefile4
-rw-r--r--sound/soc/codecs/ad1836.c6
-rw-r--r--sound/soc/codecs/ad1980.c2
-rw-r--r--sound/soc/codecs/adau1373.c7
-rw-r--r--sound/soc/codecs/adau1701.c2
-rw-r--r--sound/soc/codecs/ak4104.c174
-rw-r--r--sound/soc/codecs/ak4535.c98
-rw-r--r--sound/soc/codecs/ak4535.h2
-rw-r--r--sound/soc/codecs/ak4642.c33
-rw-r--r--sound/soc/codecs/ak4671.c2
-rw-r--r--sound/soc/codecs/alc5623.c12
-rw-r--r--sound/soc/codecs/alc5632.c197
-rw-r--r--sound/soc/codecs/alc5632.h1
-rw-r--r--sound/soc/codecs/cq93vc.c4
-rw-r--r--sound/soc/codecs/cs4270.c4
-rw-r--r--sound/soc/codecs/cs4271.c2
-rw-r--r--sound/soc/codecs/cs42l73.c2
-rw-r--r--sound/soc/codecs/da7210.c146
-rw-r--r--sound/soc/codecs/lm4857.c2
-rw-r--r--sound/soc/codecs/max9768.c247
-rw-r--r--sound/soc/codecs/max98088.c4
-rw-r--r--sound/soc/codecs/max98095.c6
-rw-r--r--sound/soc/codecs/max9877.c2
-rw-r--r--sound/soc/codecs/sgtl5000.c36
-rw-r--r--sound/soc/codecs/sn95031.c5
-rw-r--r--sound/soc/codecs/ssm2602.c2
-rw-r--r--sound/soc/codecs/stac9766.c2
-rw-r--r--sound/soc/codecs/tlv320aic23.c2
-rw-r--r--sound/soc/codecs/tlv320aic26.c2
-rw-r--r--sound/soc/codecs/tlv320aic32x4.c112
-rw-r--r--sound/soc/codecs/tlv320aic3x.c66
-rw-r--r--sound/soc/codecs/tlv320aic3x.h9
-rw-r--r--sound/soc/codecs/tlv320dac33.c10
-rw-r--r--sound/soc/codecs/tpa6130a2.c4
-rw-r--r--sound/soc/codecs/twl4030.c42
-rw-r--r--sound/soc/codecs/twl6040.c31
-rw-r--r--sound/soc/codecs/twl6040.h1
-rw-r--r--sound/soc/codecs/uda134x.c6
-rw-r--r--sound/soc/codecs/wl1273.c2
-rw-r--r--sound/soc/codecs/wm2000.c31
-rw-r--r--sound/soc/codecs/wm2200.c2286
-rw-r--r--sound/soc/codecs/wm2200.h3674
-rw-r--r--sound/soc/codecs/wm5100.c648
-rw-r--r--sound/soc/codecs/wm8731.c109
-rw-r--r--sound/soc/codecs/wm8737.c2
-rw-r--r--sound/soc/codecs/wm8753.c195
-rw-r--r--sound/soc/codecs/wm8770.c5
-rw-r--r--sound/soc/codecs/wm8776.c8
-rw-r--r--sound/soc/codecs/wm8804.c154
-rw-r--r--sound/soc/codecs/wm8904.c856
-rw-r--r--sound/soc/codecs/wm8904.h11
-rw-r--r--sound/soc/codecs/wm8940.c16
-rw-r--r--sound/soc/codecs/wm8955.c247
-rw-r--r--sound/soc/codecs/wm8958-dsp2.c16
-rw-r--r--sound/soc/codecs/wm8960.c2
-rw-r--r--sound/soc/codecs/wm8961.c2
-rw-r--r--sound/soc/codecs/wm8962.c2151
-rw-r--r--sound/soc/codecs/wm8971.c37
-rw-r--r--sound/soc/codecs/wm8974.c45
-rw-r--r--sound/soc/codecs/wm8978.c185
-rw-r--r--sound/soc/codecs/wm8978.h2
-rw-r--r--sound/soc/codecs/wm8983.c5
-rw-r--r--sound/soc/codecs/wm8985.c315
-rw-r--r--sound/soc/codecs/wm8988.c171
-rw-r--r--sound/soc/codecs/wm8990.c2
-rw-r--r--sound/soc/codecs/wm8991.c2
-rw-r--r--sound/soc/codecs/wm8993.c649
-rw-r--r--sound/soc/codecs/wm8993.h9
-rw-r--r--sound/soc/codecs/wm8994.c558
-rw-r--r--sound/soc/codecs/wm8994.h14
-rw-r--r--sound/soc/codecs/wm8995.c4
-rw-r--r--sound/soc/codecs/wm8996.c258
-rw-r--r--sound/soc/codecs/wm8996.h4
-rw-r--r--sound/soc/codecs/wm9081.c80
-rw-r--r--sound/soc/codecs/wm9090.c272
-rw-r--r--sound/soc/codecs/wm9705.c2
-rw-r--r--sound/soc/codecs/wm9712.c16
-rw-r--r--sound/soc/codecs/wm9713.c2
-rw-r--r--sound/soc/codecs/wm_hubs.c158
-rw-r--r--sound/soc/codecs/wm_hubs.h12
-rw-r--r--sound/soc/davinci/davinci-pcm.c4
-rw-r--r--sound/soc/ep93xx/Kconfig1
-rw-r--r--sound/soc/ep93xx/edb93xx.c4
-rw-r--r--sound/soc/ep93xx/ep93xx-pcm.c151
-rw-r--r--sound/soc/ep93xx/snappercl15.c4
-rw-r--r--sound/soc/fsl/fsl_dma.c10
-rw-r--r--sound/soc/fsl/fsl_ssi.c6
-rw-r--r--sound/soc/fsl/mpc5200_dma.c17
-rw-r--r--sound/soc/fsl/mpc8610_hpcd.c18
-rw-r--r--sound/soc/fsl/p1022_ds.c60
-rw-r--r--sound/soc/imx/Kconfig25
-rw-r--r--sound/soc/imx/Makefile15
-rw-r--r--sound/soc/imx/eukrea-tlv320.c40
-rw-r--r--sound/soc/imx/imx-audmux.c314
-rw-r--r--sound/soc/imx/imx-audmux.h60
-rw-r--r--sound/soc/imx/imx-pcm-dma-mx2.c224
-rw-r--r--sound/soc/imx/imx-pcm.c105
-rw-r--r--sound/soc/imx/imx-pcm.h32
-rw-r--r--sound/soc/imx/imx-ssi.c120
-rw-r--r--sound/soc/imx/imx-ssi.h16
-rw-r--r--sound/soc/imx/mx27vis-aic32x4.c159
-rw-r--r--sound/soc/imx/phycore-ac97.c27
-rw-r--r--sound/soc/imx/wm1133-ev1.c25
-rw-r--r--sound/soc/jz4740/qi_lb60.c56
-rw-r--r--sound/soc/kirkwood/kirkwood-dma.c4
-rw-r--r--sound/soc/kirkwood/kirkwood-openrd.c46
-rw-r--r--sound/soc/kirkwood/kirkwood-t5325.c47
-rw-r--r--sound/soc/mid-x86/mfld_machine.c2
-rw-r--r--sound/soc/mxs/Kconfig2
-rw-r--r--sound/soc/mxs/mxs-pcm.c157
-rw-r--r--sound/soc/mxs/mxs-pcm.h16
-rw-r--r--sound/soc/mxs/mxs-saif.c56
-rw-r--r--sound/soc/omap/Kconfig15
-rw-r--r--sound/soc/omap/Makefile6
-rw-r--r--sound/soc/omap/am3517evm.c2
-rw-r--r--sound/soc/omap/ams-delta.c38
-rw-r--r--sound/soc/omap/igep0020.c2
-rw-r--r--sound/soc/omap/mcbsp.c1040
-rw-r--r--sound/soc/omap/mcbsp.h346
-rw-r--r--sound/soc/omap/n810.c19
-rw-r--r--sound/soc/omap/omap-abe-twl6040.c349
-rw-r--r--sound/soc/omap/omap-dmic.c7
-rw-r--r--sound/soc/omap/omap-mcbsp.c321
-rw-r--r--sound/soc/omap/omap-mcbsp.h2
-rw-r--r--sound/soc/omap/omap-mcpdm.c2
-rw-r--r--sound/soc/omap/omap-pcm.h2
-rw-r--r--sound/soc/omap/omap3beagle.c2
-rw-r--r--sound/soc/omap/omap3evm.c2
-rw-r--r--sound/soc/omap/omap3pandora.c4
-rw-r--r--sound/soc/omap/osk5912.c2
-rw-r--r--sound/soc/omap/overo.c2
-rw-r--r--sound/soc/omap/rx51.c27
-rw-r--r--sound/soc/omap/sdp3430.c4
-rw-r--r--sound/soc/omap/sdp4430.c279
-rw-r--r--sound/soc/omap/zoom2.c4
-rw-r--r--sound/soc/pxa/corgi.c14
-rw-r--r--sound/soc/pxa/magician.c2
-rw-r--r--sound/soc/pxa/poodle.c14
-rw-r--r--sound/soc/pxa/pxa-ssp.c64
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c10
-rw-r--r--sound/soc/pxa/raumfeld.c2
-rw-r--r--sound/soc/pxa/spitz.c14
-rw-r--r--sound/soc/pxa/tosa.c2
-rw-r--r--sound/soc/s6000/s6000-pcm.c5
-rw-r--r--sound/soc/samsung/Kconfig12
-rw-r--r--sound/soc/samsung/ac97.c4
-rw-r--r--sound/soc/samsung/dma.c2
-rw-r--r--sound/soc/samsung/i2s.c27
-rw-r--r--sound/soc/samsung/i2s.h2
-rw-r--r--sound/soc/samsung/littlemill.c3
-rw-r--r--sound/soc/samsung/neo1973_wm8753.c73
-rw-r--r--sound/soc/samsung/pcm.c4
-rw-r--r--sound/soc/samsung/s3c24xx_simtec.c6
-rw-r--r--sound/soc/samsung/smdk_wm8580.c4
-rw-r--r--sound/soc/samsung/smdk_wm9713.c4
-rw-r--r--sound/soc/sh/fsi.c918
-rw-r--r--sound/soc/soc-core.c410
-rw-r--r--sound/soc/soc-dapm.c412
-rw-r--r--sound/soc/soc-dmaengine-pcm.c288
-rw-r--r--sound/soc/soc-io.c7
-rw-r--r--sound/soc/soc-pcm.c104
-rw-r--r--sound/soc/soc-utils.c20
-rw-r--r--sound/soc/tegra/tegra_alc5632.c129
-rw-r--r--sound/soc/tegra/tegra_pcm.c2
-rw-r--r--sound/spi/at73c213.c12
-rw-r--r--sound/usb/6fire/chip.c3
-rw-r--r--sound/usb/6fire/chip.h1
-rw-r--r--sound/usb/6fire/comm.c1
-rw-r--r--sound/usb/6fire/comm.h1
-rw-r--r--sound/usb/6fire/common.h1
-rw-r--r--sound/usb/6fire/control.c341
-rw-r--r--sound/usb/6fire/control.h7
-rw-r--r--sound/usb/6fire/firmware.c1
-rw-r--r--sound/usb/6fire/midi.c1
-rw-r--r--sound/usb/6fire/midi.h1
-rw-r--r--sound/usb/6fire/pcm.c1
-rw-r--r--sound/usb/6fire/pcm.h1
-rw-r--r--sound/usb/Kconfig1
-rw-r--r--sound/usb/caiaq/audio.c5
-rw-r--r--sound/usb/card.h1
-rw-r--r--sound/usb/format.c4
-rw-r--r--sound/usb/pcm.c6
-rw-r--r--sound/usb/quirks-table.h8
-rw-r--r--sound/usb/quirks.c6
-rw-r--r--sound/usb/usx2y/usbusx2yaudio.c4
-rw-r--r--sound/usb/usx2y/usx2yhwdeppcm.c2
253 files changed, 18224 insertions, 11755 deletions
diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c
index 762af68c8996..270790d384e2 100644
--- a/sound/aoa/codecs/onyx.c
+++ b/sound/aoa/codecs/onyx.c
@@ -1132,15 +1132,4 @@ static struct i2c_driver onyx_driver = {
.id_table = onyx_i2c_id,
};
-static int __init onyx_init(void)
-{
- return i2c_add_driver(&onyx_driver);
-}
-
-static void __exit onyx_exit(void)
-{
- i2c_del_driver(&onyx_driver);
-}
-
-module_init(onyx_init);
-module_exit(onyx_exit);
+module_i2c_driver(onyx_driver);
diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c
index fd2188c3df2b..8e63d1f35ce1 100644
--- a/sound/aoa/codecs/tas.c
+++ b/sound/aoa/codecs/tas.c
@@ -1026,15 +1026,4 @@ static struct i2c_driver tas_driver = {
.id_table = tas_i2c_id,
};
-static int __init tas_init(void)
-{
- return i2c_add_driver(&tas_driver);
-}
-
-static void __exit tas_exit(void)
-{
- i2c_del_driver(&tas_driver);
-}
-
-module_init(tas_init);
-module_exit(tas_exit);
+module_i2c_driver(tas_driver);
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index b37b702a3a6a..5119fdabcb98 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -1110,18 +1110,7 @@ static struct amba_driver aaci_driver = {
.id_table = aaci_ids,
};
-static int __init aaci_init(void)
-{
- return amba_driver_register(&aaci_driver);
-}
-
-static void __exit aaci_exit(void)
-{
- amba_driver_unregister(&aaci_driver);
-}
-
-module_init(aaci_init);
-module_exit(aaci_exit);
+module_amba_driver(aaci_driver);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("ARM PrimeCell PL041 Advanced Audio CODEC Interface driver");
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index dac3633507c9..a68aed7fce02 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -441,19 +441,22 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg)
params = kmalloc(sizeof(*params), GFP_KERNEL);
if (!params)
return -ENOMEM;
- if (copy_from_user(params, (void __user *)arg, sizeof(*params)))
- return -EFAULT;
+ if (copy_from_user(params, (void __user *)arg, sizeof(*params))) {
+ retval = -EFAULT;
+ goto out;
+ }
retval = snd_compr_allocate_buffer(stream, params);
if (retval) {
- kfree(params);
- return -ENOMEM;
+ retval = -ENOMEM;
+ goto out;
}
retval = stream->ops->set_params(stream, params);
if (retval)
goto out;
stream->runtime->state = SNDRV_PCM_STATE_SETUP;
- } else
+ } else {
return -EPERM;
+ }
out:
kfree(params);
return retval;
diff --git a/sound/core/control.c b/sound/core/control.c
index 819a5c579a39..2487a6bb1c54 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1313,7 +1313,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file,
err = -EPERM;
goto __kctl_end;
}
- err = kctl->tlv.c(kctl, op_flag, tlv.length, _tlv->tlv);
+ err = kctl->tlv.c(kctl, op_flag, tlv.length, _tlv->tlv);
if (err > 0) {
up_read(&card->controls_rwsem);
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_TLV, &kctl->id);
diff --git a/sound/core/init.c b/sound/core/init.c
index 3ac49b1b7cb8..d8ec849af128 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -22,6 +22,7 @@
#include <linux/init.h>
#include <linux/sched.h>
#include <linux/module.h>
+#include <linux/device.h>
#include <linux/file.h>
#include <linux/slab.h>
#include <linux/time.h>
@@ -480,74 +481,104 @@ int snd_card_free(struct snd_card *card)
EXPORT_SYMBOL(snd_card_free);
-static void snd_card_set_id_no_lock(struct snd_card *card, const char *nid)
+/* retrieve the last word of shortname or longname */
+static const char *retrieve_id_from_card_name(const char *name)
{
- int i, len, idx_flag = 0, loops = SNDRV_CARDS;
- const char *spos, *src;
- char *id;
-
- if (nid == NULL) {
- id = card->shortname;
- spos = src = id;
- while (*id != '\0') {
- if (*id == ' ')
- spos = id + 1;
- id++;
- }
- } else {
- spos = src = nid;
+ const char *spos = name;
+
+ while (*name) {
+ if (isspace(*name) && isalnum(name[1]))
+ spos = name + 1;
+ name++;
}
- id = card->id;
- while (*spos != '\0' && !isalnum(*spos))
- spos++;
- if (isdigit(*spos))
- *id++ = isalpha(src[0]) ? src[0] : 'D';
- while (*spos != '\0' && (size_t)(id - card->id) < sizeof(card->id) - 1) {
- if (isalnum(*spos))
- *id++ = *spos;
- spos++;
+ return spos;
+}
+
+/* return true if the given id string doesn't conflict any other card ids */
+static bool card_id_ok(struct snd_card *card, const char *id)
+{
+ int i;
+ if (!snd_info_check_reserved_words(id))
+ return false;
+ for (i = 0; i < snd_ecards_limit; i++) {
+ if (snd_cards[i] && snd_cards[i] != card &&
+ !strcmp(snd_cards[i]->id, id))
+ return false;
}
- *id = '\0';
+ return true;
+}
- id = card->id;
+/* copy to card->id only with valid letters from nid */
+static void copy_valid_id_string(struct snd_card *card, const char *src,
+ const char *nid)
+{
+ char *id = card->id;
+
+ while (*nid && !isalnum(*nid))
+ nid++;
+ if (isdigit(*nid))
+ *id++ = isalpha(*src) ? *src : 'D';
+ while (*nid && (size_t)(id - card->id) < sizeof(card->id) - 1) {
+ if (isalnum(*nid))
+ *id++ = *nid;
+ nid++;
+ }
+ *id = 0;
+}
+
+/* Set card->id from the given string
+ * If the string conflicts with other ids, add a suffix to make it unique.
+ */
+static void snd_card_set_id_no_lock(struct snd_card *card, const char *src,
+ const char *nid)
+{
+ int len, loops;
+ bool with_suffix;
+ bool is_default = false;
+ char *id;
- if (*id == '\0')
+ copy_valid_id_string(card, src, nid);
+ id = card->id;
+
+ again:
+ /* use "Default" for obviously invalid strings
+ * ("card" conflicts with proc directories)
+ */
+ if (!*id || !strncmp(id, "card", 4)) {
strcpy(id, "Default");
+ is_default = true;
+ }
- while (1) {
- if (loops-- == 0) {
- snd_printk(KERN_ERR "unable to set card id (%s)\n", id);
- strcpy(card->id, card->proc_root->name);
- return;
- }
- if (!snd_info_check_reserved_words(id))
- goto __change;
- for (i = 0; i < snd_ecards_limit; i++) {
- if (snd_cards[i] && !strcmp(snd_cards[i]->id, id))
- goto __change;
- }
- break;
+ with_suffix = false;
+ for (loops = 0; loops < SNDRV_CARDS; loops++) {
+ if (card_id_ok(card, id))
+ return; /* OK */
- __change:
len = strlen(id);
- if (idx_flag) {
- if (id[len-1] != '9')
- id[len-1]++;
- else
- id[len-1] = 'A';
- } else if ((size_t)len <= sizeof(card->id) - 3) {
- strcat(id, "_1");
- idx_flag++;
+ if (!with_suffix) {
+ /* add the "_X" suffix */
+ char *spos = id + len;
+ if (len > sizeof(card->id) - 3)
+ spos = id + sizeof(card->id) - 3;
+ strcpy(spos, "_1");
+ with_suffix = true;
} else {
- spos = id + len - 2;
- if ((size_t)len <= sizeof(card->id) - 2)
- spos++;
- *(char *)spos++ = '_';
- *(char *)spos++ = '1';
- *(char *)spos++ = '\0';
- idx_flag++;
+ /* modify the existing suffix */
+ if (id[len - 1] != '9')
+ id[len - 1]++;
+ else
+ id[len - 1] = 'A';
}
}
+ /* fallback to the default id */
+ if (!is_default) {
+ *id = 0;
+ goto again;
+ }
+ /* last resort... */
+ snd_printk(KERN_ERR "unable to set card id (%s)\n", id);
+ if (card->proc_root->name)
+ strcpy(card->id, card->proc_root->name);
}
/**
@@ -564,7 +595,7 @@ void snd_card_set_id(struct snd_card *card, const char *nid)
if (card->id[0] != '\0')
return;
mutex_lock(&snd_card_mutex);
- snd_card_set_id_no_lock(card, nid);
+ snd_card_set_id_no_lock(card, nid, nid);
mutex_unlock(&snd_card_mutex);
}
EXPORT_SYMBOL(snd_card_set_id);
@@ -596,22 +627,12 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr,
memcpy(buf1, buf, copy);
buf1[copy] = '\0';
mutex_lock(&snd_card_mutex);
- if (!snd_info_check_reserved_words(buf1)) {
- __exist:
+ if (!card_id_ok(NULL, buf1)) {
mutex_unlock(&snd_card_mutex);
return -EEXIST;
}
- for (idx = 0; idx < snd_ecards_limit; idx++) {
- if (snd_cards[idx] && !strcmp(snd_cards[idx]->id, buf1)) {
- if (card == snd_cards[idx])
- goto __ok;
- else
- goto __exist;
- }
- }
strcpy(card->id, buf1);
snd_info_card_id_change(card);
-__ok:
mutex_unlock(&snd_card_mutex);
return count;
@@ -665,7 +686,18 @@ int snd_card_register(struct snd_card *card)
mutex_unlock(&snd_card_mutex);
return 0;
}
- snd_card_set_id_no_lock(card, card->id[0] == '\0' ? NULL : card->id);
+ if (*card->id) {
+ /* make a unique id name from the given string */
+ char tmpid[sizeof(card->id)];
+ memcpy(tmpid, card->id, sizeof(card->id));
+ snd_card_set_id_no_lock(card, tmpid, tmpid);
+ } else {
+ /* create an id from either shortname or longname */
+ const char *src;
+ src = *card->shortname ? card->shortname : card->longname;
+ snd_card_set_id_no_lock(card, src,
+ retrieve_id_from_card_name(src));
+ }
snd_cards[card->number] = card;
mutex_unlock(&snd_card_mutex);
init_info_for_card(card);
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 26edf63b265f..471e1e3b0a99 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -25,7 +25,7 @@
#include <sound/jack.h>
#include <sound/core.h>
-static int jack_switch_types[] = {
+static int jack_switch_types[SND_JACK_SWITCH_TYPES] = {
SW_HEADPHONE_INSERT,
SW_MICROPHONE_INSERT,
SW_LINEOUT_INSERT,
@@ -128,7 +128,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
jack->type = type;
- for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++)
+ for (i = 0; i < SND_JACK_SWITCH_TYPES; i++)
if (type & (1 << i))
input_set_capability(jack->input_dev, EV_SW,
jack_switch_types[i]);
diff --git a/sound/core/misc.c b/sound/core/misc.c
index 465f0ce772cb..768167925409 100644
--- a/sound/core/misc.c
+++ b/sound/core/misc.c
@@ -72,7 +72,7 @@ void __snd_printk(unsigned int level, const char *path, int line,
char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV";
#endif
-#ifdef CONFIG_SND_DEBUG
+#ifdef CONFIG_SND_DEBUG
if (debug < level)
return;
#endif
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 8928ca871c22..1a3070b4e5b5 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -24,6 +24,7 @@
#include <linux/module.h>
#include <linux/time.h>
#include <linux/mutex.h>
+#include <linux/device.h>
#include <sound/core.h>
#include <sound/minors.h>
#include <sound/pcm.h>
@@ -650,7 +651,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
pstr->stream = stream;
pstr->pcm = pcm;
pstr->substream_count = substream_count;
- if (substream_count > 0) {
+ if (substream_count > 0 && !pcm->internal) {
err = snd_pcm_stream_proc_init(pstr);
if (err < 0) {
snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
@@ -674,15 +675,18 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
pstr->substream = substream;
else
prev->next = substream;
- err = snd_pcm_substream_proc_init(substream);
- if (err < 0) {
- snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
- if (prev == NULL)
- pstr->substream = NULL;
- else
- prev->next = NULL;
- kfree(substream);
- return err;
+
+ if (!pcm->internal) {
+ err = snd_pcm_substream_proc_init(substream);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Error in snd_pcm_stream_proc_init\n");
+ if (prev == NULL)
+ pstr->substream = NULL;
+ else
+ prev->next = NULL;
+ kfree(substream);
+ return err;
+ }
}
substream->group = &substream->self_group;
spin_lock_init(&substream->self_group.lock);
@@ -696,25 +700,9 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
EXPORT_SYMBOL(snd_pcm_new_stream);
-/**
- * snd_pcm_new - create a new PCM instance
- * @card: the card instance
- * @id: the id string
- * @device: the device index (zero based)
- * @playback_count: the number of substreams for playback
- * @capture_count: the number of substreams for capture
- * @rpcm: the pointer to store the new pcm instance
- *
- * Creates a new PCM instance.
- *
- * The pcm operators have to be set afterwards to the new instance
- * via snd_pcm_set_ops().
- *
- * Returns zero if successful, or a negative error code on failure.
- */
-int snd_pcm_new(struct snd_card *card, const char *id, int device,
- int playback_count, int capture_count,
- struct snd_pcm ** rpcm)
+static int _snd_pcm_new(struct snd_card *card, const char *id, int device,
+ int playback_count, int capture_count, bool internal,
+ struct snd_pcm **rpcm)
{
struct snd_pcm *pcm;
int err;
@@ -735,6 +723,7 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device,
}
pcm->card = card;
pcm->device = device;
+ pcm->internal = internal;
if (id)
strlcpy(pcm->id, id, sizeof(pcm->id));
if ((err = snd_pcm_new_stream(pcm, SNDRV_PCM_STREAM_PLAYBACK, playback_count)) < 0) {
@@ -756,8 +745,59 @@ int snd_pcm_new(struct snd_card *card, const char *id, int device,
return 0;
}
+/**
+ * snd_pcm_new - create a new PCM instance
+ * @card: the card instance
+ * @id: the id string
+ * @device: the device index (zero based)
+ * @playback_count: the number of substreams for playback
+ * @capture_count: the number of substreams for capture
+ * @rpcm: the pointer to store the new pcm instance
+ *
+ * Creates a new PCM instance.
+ *
+ * The pcm operators have to be set afterwards to the new instance
+ * via snd_pcm_set_ops().
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_new(struct snd_card *card, const char *id, int device,
+ int playback_count, int capture_count, struct snd_pcm **rpcm)
+{
+ return _snd_pcm_new(card, id, device, playback_count, capture_count,
+ false, rpcm);
+}
EXPORT_SYMBOL(snd_pcm_new);
+/**
+ * snd_pcm_new_internal - create a new internal PCM instance
+ * @card: the card instance
+ * @id: the id string
+ * @device: the device index (zero based - shared with normal PCMs)
+ * @playback_count: the number of substreams for playback
+ * @capture_count: the number of substreams for capture
+ * @rpcm: the pointer to store the new pcm instance
+ *
+ * Creates a new internal PCM instance with no userspace device or procfs
+ * entries. This is used by ASoC Back End PCMs in order to create a PCM that
+ * will only be used internally by kernel drivers. i.e. it cannot be opened
+ * by userspace. It provides existing ASoC components drivers with a substream
+ * and access to any private data.
+ *
+ * The pcm operators have to be set afterwards to the new instance
+ * via snd_pcm_set_ops().
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ */
+int snd_pcm_new_internal(struct snd_card *card, const char *id, int device,
+ int playback_count, int capture_count,
+ struct snd_pcm **rpcm)
+{
+ return _snd_pcm_new(card, id, device, playback_count, capture_count,
+ true, rpcm);
+}
+EXPORT_SYMBOL(snd_pcm_new_internal);
+
static void snd_pcm_free_stream(struct snd_pcm_str * pstr)
{
struct snd_pcm_substream *substream, *substream_next;
@@ -994,7 +1034,7 @@ static int snd_pcm_dev_register(struct snd_device *device)
}
for (cidx = 0; cidx < 2; cidx++) {
int devtype = -1;
- if (pcm->streams[cidx].substream == NULL)
+ if (pcm->streams[cidx].substream == NULL || pcm->internal)
continue;
switch (cidx) {
case SNDRV_PCM_STREAM_PLAYBACK:
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 3420bd3da5d7..4d18941178e6 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1029,7 +1029,8 @@ static int snd_interval_ratden(struct snd_interval *i,
*
* Returns non-zero if the value is changed, zero if not changed.
*/
-int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask)
+int snd_interval_list(struct snd_interval *i, unsigned int count,
+ const unsigned int *list, unsigned int mask)
{
unsigned int k;
struct snd_interval list_range;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 25ed9fe41b89..3fe99e644eb8 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -1586,12 +1586,18 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
struct file *file;
struct snd_pcm_file *pcm_file;
struct snd_pcm_substream *substream1;
+ struct snd_pcm_group *group;
file = snd_pcm_file_fd(fd);
if (!file)
return -EBADFD;
pcm_file = file->private_data;
substream1 = pcm_file->substream;
+ group = kmalloc(sizeof(*group), GFP_KERNEL);
+ if (!group) {
+ res = -ENOMEM;
+ goto _nolock;
+ }
down_write(&snd_pcm_link_rwsem);
write_lock_irq(&snd_pcm_link_rwlock);
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN ||
@@ -1604,11 +1610,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
goto _end;
}
if (!snd_pcm_stream_linked(substream)) {
- substream->group = kmalloc(sizeof(struct snd_pcm_group), GFP_ATOMIC);
- if (substream->group == NULL) {
- res = -ENOMEM;
- goto _end;
- }
+ substream->group = group;
spin_lock_init(&substream->group->lock);
INIT_LIST_HEAD(&substream->group->substreams);
list_add_tail(&substream->link_list, &substream->group->substreams);
@@ -1620,7 +1622,10 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd)
_end:
write_unlock_irq(&snd_pcm_link_rwlock);
up_write(&snd_pcm_link_rwsem);
+ _nolock:
fput(file);
+ if (res < 0)
+ kfree(group);
return res;
}
diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c
index 9d8379aedf40..712110561082 100644
--- a/sound/core/seq/seq.c
+++ b/sound/core/seq/seq.c
@@ -21,6 +21,7 @@
#include <linux/init.h>
#include <linux/module.h>
+#include <linux/device.h>
#include <sound/core.h>
#include <sound/initval.h>
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 8e7561dfc5fc..6ddcf06f52f9 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -24,6 +24,7 @@
#include <linux/slab.h>
#include <linux/time.h>
#include <linux/mutex.h>
+#include <linux/device.h>
#include <linux/module.h>
#include <linux/string.h>
#include <sound/core.h>
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 130cfe677d60..14a286a7bf2b 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -37,6 +37,8 @@ struct link_master {
struct link_ctl_info info;
int val; /* the master value */
unsigned int tlv[4];
+ void (*hook)(void *private_data, int);
+ void *hook_private_data;
};
/*
@@ -126,7 +128,9 @@ static int master_init(struct link_master *master)
master->info.count = 1; /* always mono */
/* set full volume as default (= no attenuation) */
master->val = master->info.max_val;
- return 0;
+ if (master->hook)
+ master->hook(master->hook_private_data, master->val);
+ return 1;
}
return -ENOENT;
}
@@ -329,6 +333,8 @@ static int master_put(struct snd_kcontrol *kcontrol,
slave_put_val(slave, uval);
}
kfree(uval);
+ if (master->hook && !err)
+ master->hook(master->hook_private_data, master->val);
return 1;
}
@@ -408,3 +414,41 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
return kctl;
}
EXPORT_SYMBOL(snd_ctl_make_virtual_master);
+
+/**
+ * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control
+ * @kcontrol: vmaster kctl element
+ * @hook: the hook function
+ *
+ * Adds the given hook to the vmaster control element so that it's called
+ * at each time when the value is changed.
+ */
+int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kcontrol,
+ void (*hook)(void *private_data, int),
+ void *private_data)
+{
+ struct link_master *master = snd_kcontrol_chip(kcontrol);
+ master->hook = hook;
+ master->hook_private_data = private_data;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_ctl_add_vmaster_hook);
+
+/**
+ * snd_ctl_sync_vmaster_hook - Sync the vmaster hook
+ * @kcontrol: vmaster kctl element
+ *
+ * Call the hook function to synchronize with the current value of the given
+ * vmaster element. NOP when NULL is passed to @kcontrol or the hook doesn't
+ * exist.
+ */
+void snd_ctl_sync_vmaster_hook(struct snd_kcontrol *kcontrol)
+{
+ struct link_master *master;
+ if (!kcontrol)
+ return;
+ master = snd_kcontrol_chip(kcontrol);
+ if (master->hook)
+ master->hook(master->hook_private_data, master->val);
+}
+EXPORT_SYMBOL_GPL(snd_ctl_sync_vmaster_hook);
diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c
index cd094ecaca3b..d428ffede4f3 100644
--- a/sound/firewire/isight.c
+++ b/sound/firewire/isight.c
@@ -611,7 +611,6 @@ static void isight_card_free(struct snd_card *card)
fw_iso_resources_destroy(&isight->resources);
fw_unit_put(isight->unit);
- fw_device_put(isight->device);
mutex_destroy(&isight->mutex);
}
@@ -644,7 +643,7 @@ static int isight_probe(struct device *unit_dev)
isight->card = card;
mutex_init(&isight->mutex);
isight->unit = fw_unit_get(unit);
- isight->device = fw_device_get(fw_dev);
+ isight->device = fw_dev;
isight->audio_base = get_unit_base(unit);
if (!isight->audio_base) {
dev_err(&unit->device, "audio unit base not found\n");
@@ -681,7 +680,6 @@ static int isight_probe(struct device *unit_dev)
err_unit:
fw_unit_put(isight->unit);
- fw_device_put(isight->device);
mutex_destroy(&isight->mutex);
error:
snd_card_free(card);
diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c
index cbe6bb9e53b6..297244e658d9 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -656,12 +656,10 @@ static u32 fwspk_read_firmware_version(struct fw_unit *unit)
static void fwspk_card_free(struct snd_card *card)
{
struct fwspk *fwspk = card->private_data;
- struct fw_device *dev = fw_parent_device(fwspk->unit);
amdtp_out_stream_destroy(&fwspk->stream);
cmp_connection_destroy(&fwspk->connection);
fw_unit_put(fwspk->unit);
- fw_device_put(dev);
mutex_destroy(&fwspk->mutex);
}
@@ -718,7 +716,6 @@ static int __devinit fwspk_probe(struct device *unit_dev)
fwspk = card->private_data;
fwspk->card = card;
mutex_init(&fwspk->mutex);
- fw_device_get(fw_dev);
fwspk->unit = fw_unit_get(unit);
fwspk->device_info = fwspk_detect(fw_dev);
if (!fwspk->device_info) {
@@ -767,7 +764,6 @@ err_connection:
cmp_connection_destroy(&fwspk->connection);
err_unit:
fw_unit_put(fwspk->unit);
- fw_device_put(fw_dev);
mutex_destroy(&fwspk->mutex);
error:
snd_card_free(card);
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index 6b68c8206805..a63faec5e7fd 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -25,21 +25,20 @@
#include <linux/module.h>
#include <linux/init.h>
#include <linux/slab.h>
-#include <linux/version.h>
+#include <linux/sched.h>
+#include <media/v4l2-device.h>
#include <media/v4l2-dev.h>
+#include <media/v4l2-fh.h>
#include <media/v4l2-ioctl.h>
+#include <media/v4l2-event.h>
#include <sound/tea575x-tuner.h>
MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips");
MODULE_LICENSE("GPL");
-static int radio_nr = -1;
-module_param(radio_nr, int, 0);
-
-#define RADIO_VERSION KERNEL_VERSION(0, 0, 2)
-#define FREQ_LO (50UL * 16000)
-#define FREQ_HI (150UL * 16000)
+#define FREQ_LO (76U * 16000)
+#define FREQ_HI (108U * 16000)
/*
* definitions
@@ -90,7 +89,7 @@ static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val)
tea->ops->set_pins(tea, 0);
}
-static unsigned int snd_tea575x_read(struct snd_tea575x *tea)
+static u32 snd_tea575x_read(struct snd_tea575x *tea)
{
u16 l, rdata;
u32 data = 0;
@@ -121,11 +120,13 @@ static unsigned int snd_tea575x_read(struct snd_tea575x *tea)
return data;
}
-static void snd_tea575x_get_freq(struct snd_tea575x *tea)
+static u32 snd_tea575x_get_freq(struct snd_tea575x *tea)
{
- unsigned long freq;
+ u32 freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK;
+
+ if (freq == 0)
+ return freq;
- freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK;
/* freq *= 12.5 */
freq *= 125;
freq /= 10;
@@ -135,14 +136,13 @@ static void snd_tea575x_get_freq(struct snd_tea575x *tea)
else
freq -= TEA575X_FMIF;
- tea->freq = freq * 16; /* from kHz */
+ return clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */
}
static void snd_tea575x_set_freq(struct snd_tea575x *tea)
{
- unsigned long freq;
+ u32 freq = tea->freq;
- freq = clamp(tea->freq, FREQ_LO, FREQ_HI);
freq /= 16; /* to kHz */
/* crystal fixup */
if (tea->tea5759)
@@ -167,12 +167,14 @@ static int vidioc_querycap(struct file *file, void *priv,
{
struct snd_tea575x *tea = video_drvdata(file);
- strlcpy(v->driver, "tea575x-tuner", sizeof(v->driver));
+ strlcpy(v->driver, tea->v4l2_dev->name, sizeof(v->driver));
strlcpy(v->card, tea->card, sizeof(v->card));
strlcat(v->card, tea->tea5759 ? " TEA5759" : " TEA5757", sizeof(v->card));
strlcpy(v->bus_info, tea->bus_info, sizeof(v->bus_info));
- v->version = RADIO_VERSION;
- v->capabilities = V4L2_CAP_TUNER | V4L2_CAP_RADIO;
+ v->device_caps = V4L2_CAP_TUNER | V4L2_CAP_RADIO;
+ if (!tea->cannot_read_data)
+ v->device_caps |= V4L2_CAP_HW_FREQ_SEEK;
+ v->capabilities = v->device_caps | V4L2_CAP_DEVICE_CAPS;
return 0;
}
@@ -191,18 +193,24 @@ static int vidioc_g_tuner(struct file *file, void *priv,
v->capability = V4L2_TUNER_CAP_LOW | V4L2_TUNER_CAP_STEREO;
v->rangelow = FREQ_LO;
v->rangehigh = FREQ_HI;
- v->rxsubchans = V4L2_TUNER_SUB_MONO | V4L2_TUNER_SUB_STEREO;
- v->audmode = tea->stereo ? V4L2_TUNER_MODE_STEREO : V4L2_TUNER_MODE_MONO;
+ v->rxsubchans = tea->stereo ? V4L2_TUNER_SUB_STEREO : V4L2_TUNER_SUB_MONO;
+ v->audmode = (tea->val & TEA575X_BIT_MONO) ?
+ V4L2_TUNER_MODE_MONO : V4L2_TUNER_MODE_STEREO;
v->signal = tea->tuned ? 0xffff : 0;
-
return 0;
}
static int vidioc_s_tuner(struct file *file, void *priv,
struct v4l2_tuner *v)
{
- if (v->index > 0)
+ struct snd_tea575x *tea = video_drvdata(file);
+
+ if (v->index)
return -EINVAL;
+ tea->val &= ~TEA575X_BIT_MONO;
+ if (v->audmode == V4L2_TUNER_MODE_MONO)
+ tea->val |= TEA575X_BIT_MONO;
+ snd_tea575x_write(tea, tea->val);
return 0;
}
@@ -214,7 +222,6 @@ static int vidioc_g_frequency(struct file *file, void *priv,
if (f->tuner != 0)
return -EINVAL;
f->type = V4L2_TUNER_RADIO;
- snd_tea575x_get_freq(tea);
f->frequency = tea->freq;
return 0;
}
@@ -227,33 +234,72 @@ static int vidioc_s_frequency(struct file *file, void *priv,
if (f->tuner != 0 || f->type != V4L2_TUNER_RADIO)
return -EINVAL;
- if (f->frequency < FREQ_LO || f->frequency > FREQ_HI)
- return -EINVAL;
-
- tea->freq = f->frequency;
-
+ tea->val &= ~TEA575X_BIT_SEARCH;
+ tea->freq = clamp(f->frequency, FREQ_LO, FREQ_HI);
snd_tea575x_set_freq(tea);
-
return 0;
}
-static int vidioc_g_audio(struct file *file, void *priv,
- struct v4l2_audio *a)
+static int vidioc_s_hw_freq_seek(struct file *file, void *fh,
+ struct v4l2_hw_freq_seek *a)
{
- if (a->index > 1)
- return -EINVAL;
-
- strcpy(a->name, "Radio");
- a->capability = V4L2_AUDCAP_STEREO;
- return 0;
-}
+ struct snd_tea575x *tea = video_drvdata(file);
+ unsigned long timeout;
+ int i;
-static int vidioc_s_audio(struct file *file, void *priv,
- struct v4l2_audio *a)
-{
- if (a->index != 0)
+ if (tea->cannot_read_data)
+ return -ENOTTY;
+ if (a->tuner || a->wrap_around)
return -EINVAL;
- return 0;
+
+ /* clear the frequency, HW will fill it in */
+ tea->val &= ~TEA575X_BIT_FREQ_MASK;
+ tea->val |= TEA575X_BIT_SEARCH;
+ if (a->seek_upward)
+ tea->val |= TEA575X_BIT_UPDOWN;
+ else
+ tea->val &= ~TEA575X_BIT_UPDOWN;
+ snd_tea575x_write(tea, tea->val);
+ timeout = jiffies + msecs_to_jiffies(10000);
+ for (;;) {
+ if (time_after(jiffies, timeout))
+ break;
+ if (schedule_timeout_interruptible(msecs_to_jiffies(10))) {
+ /* some signal arrived, stop search */
+ tea->val &= ~TEA575X_BIT_SEARCH;
+ snd_tea575x_set_freq(tea);
+ return -ERESTARTSYS;
+ }
+ if (!(snd_tea575x_read(tea) & TEA575X_BIT_SEARCH)) {
+ u32 freq;
+
+ /* Found a frequency, wait until it can be read */
+ for (i = 0; i < 100; i++) {
+ msleep(10);
+ freq = snd_tea575x_get_freq(tea);
+ if (freq) /* available */
+ break;
+ }
+ if (freq == 0) /* shouldn't happen */
+ break;
+ /*
+ * if we moved by less than 50 kHz, or in the wrong
+ * direction, continue seeking
+ */
+ if (abs(tea->freq - freq) < 16 * 50 ||
+ (a->seek_upward && freq < tea->freq) ||
+ (!a->seek_upward && freq > tea->freq)) {
+ snd_tea575x_write(tea, tea->val);
+ continue;
+ }
+ tea->freq = freq;
+ tea->val &= ~TEA575X_BIT_SEARCH;
+ return 0;
+ }
+ }
+ tea->val &= ~TEA575X_BIT_SEARCH;
+ snd_tea575x_set_freq(tea);
+ return -EAGAIN;
}
static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl)
@@ -273,23 +319,27 @@ static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl)
static const struct v4l2_file_operations tea575x_fops = {
.owner = THIS_MODULE,
.unlocked_ioctl = video_ioctl2,
+ .open = v4l2_fh_open,
+ .release = v4l2_fh_release,
+ .poll = v4l2_ctrl_poll,
};
static const struct v4l2_ioctl_ops tea575x_ioctl_ops = {
.vidioc_querycap = vidioc_querycap,
.vidioc_g_tuner = vidioc_g_tuner,
.vidioc_s_tuner = vidioc_s_tuner,
- .vidioc_g_audio = vidioc_g_audio,
- .vidioc_s_audio = vidioc_s_audio,
.vidioc_g_frequency = vidioc_g_frequency,
.vidioc_s_frequency = vidioc_s_frequency,
+ .vidioc_s_hw_freq_seek = vidioc_s_hw_freq_seek,
+ .vidioc_log_status = v4l2_ctrl_log_status,
+ .vidioc_subscribe_event = v4l2_ctrl_subscribe_event,
+ .vidioc_unsubscribe_event = v4l2_event_unsubscribe,
};
-static struct video_device tea575x_radio = {
- .name = "tea575x-tuner",
+static const struct video_device tea575x_radio = {
.fops = &tea575x_fops,
.ioctl_ops = &tea575x_ioctl_ops,
- .release = video_device_release_empty,
+ .release = video_device_release_empty,
};
static const struct v4l2_ctrl_ops tea575x_ctrl_ops = {
@@ -303,27 +353,34 @@ int snd_tea575x_init(struct snd_tea575x *tea)
{
int retval;
- tea->mute = 1;
+ tea->mute = true;
- snd_tea575x_write(tea, 0x55AA);
- if (snd_tea575x_read(tea) != 0x55AA)
- return -ENODEV;
+ /* Not all devices can or know how to read the data back.
+ Such devices can set cannot_read_data to true. */
+ if (!tea->cannot_read_data) {
+ snd_tea575x_write(tea, 0x55AA);
+ if (snd_tea575x_read(tea) != 0x55AA)
+ return -ENODEV;
+ }
- tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40;
+ tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_5_28;
tea->freq = 90500 * 16; /* 90.5Mhz default */
snd_tea575x_set_freq(tea);
tea->vd = tea575x_radio;
video_set_drvdata(&tea->vd, tea);
mutex_init(&tea->mutex);
+ strlcpy(tea->vd.name, tea->v4l2_dev->name, sizeof(tea->vd.name));
tea->vd.lock = &tea->mutex;
+ tea->vd.v4l2_dev = tea->v4l2_dev;
+ tea->vd.ctrl_handler = &tea->ctrl_handler;
+ set_bit(V4L2_FL_USE_FH_PRIO, &tea->vd.flags);
v4l2_ctrl_handler_init(&tea->ctrl_handler, 1);
- tea->vd.ctrl_handler = &tea->ctrl_handler;
v4l2_ctrl_new_std(&tea->ctrl_handler, &tea575x_ctrl_ops, V4L2_CID_AUDIO_MUTE, 0, 1, 1, 1);
retval = tea->ctrl_handler.error;
if (retval) {
- printk(KERN_ERR "tea575x-tuner: can't initialize controls\n");
+ v4l2_err(tea->v4l2_dev, "can't initialize controls\n");
v4l2_ctrl_handler_free(&tea->ctrl_handler);
return retval;
}
@@ -338,9 +395,9 @@ int snd_tea575x_init(struct snd_tea575x *tea)
v4l2_ctrl_handler_setup(&tea->ctrl_handler);
- retval = video_register_device(&tea->vd, VFL_TYPE_RADIO, radio_nr);
+ retval = video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->radio_nr);
if (retval) {
- printk(KERN_ERR "tea575x-tuner: can't register video device!\n");
+ v4l2_err(tea->v4l2_dev, "can't register video device!\n");
v4l2_ctrl_handler_free(&tea->ctrl_handler);
return retval;
}
diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c
index e09f144177f5..c99c6078be33 100644
--- a/sound/isa/sb/emu8000_patch.c
+++ b/sound/isa/sb/emu8000_patch.c
@@ -22,7 +22,6 @@
#include "emu8000_local.h"
#include <asm/uaccess.h>
#include <linux/moduleparam.h>
-#include <linux/moduleparam.h>
static int emu8000_reset_addr;
module_param(emu8000_reset_addr, int, 0444);
diff --git a/sound/oss/os.h b/sound/oss/os.h
index a1a962d7f67d..75ad0cd0c0ab 100644
--- a/sound/oss/os.h
+++ b/sound/oss/os.h
@@ -16,7 +16,6 @@
#include <linux/slab.h>
#include <linux/ioport.h>
#include <asm/page.h>
-#include <asm/system.h>
#include <linux/vmalloc.h>
#include <asm/uaccess.h>
#include <linux/poll.h>
diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c
index 12ba28e7b933..92ca5bee1860 100644
--- a/sound/oss/vidc.c
+++ b/sound/oss/vidc.c
@@ -28,7 +28,6 @@
#include <asm/io.h>
#include <asm/hardware/iomd.h>
#include <asm/irq.h>
-#include <asm/system.h>
#include "sound_config.h"
#include "vidc.h"
diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c
index 52468742d9f2..24c430f721d4 100644
--- a/sound/oss/waveartist.c
+++ b/sound/oss/waveartist.c
@@ -42,7 +42,6 @@
#include <linux/spinlock.h>
#include <linux/bitops.h>
-#include <asm/system.h>
#include "sound_config.h"
#include "waveartist.h"
diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h
index c5cef113c209..d3fbd0d76c37 100644
--- a/sound/pci/asihpi/hpios.h
+++ b/sound/pci/asihpi/hpios.h
@@ -30,7 +30,6 @@ HPI Operating System Specific macros for Linux Kernel driver
#define HPI_BUILD_KERNEL_MODE
#include <linux/io.h>
-#include <asm/system.h>
#include <linux/ioctl.h>
#include <linux/kernel.h>
#include <linux/string.h>
diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h
index bb938153a964..466a5c8e8354 100644
--- a/sound/pci/au88x0/au88x0.h
+++ b/sound/pci/au88x0/au88x0.h
@@ -26,7 +26,7 @@
#include <sound/mpu401.h>
#include <sound/hwdep.h>
#include <sound/ac97_codec.h>
-
+#include <sound/tlv.h>
#endif
#ifndef CHIP_AU8820
@@ -107,6 +107,14 @@
#define NR_WTPB 0x20 /* WT channels per each bank. */
#define NR_PCM 0x10
+struct pcm_vol {
+ struct snd_kcontrol *kctl;
+ int active;
+ int dma;
+ int mixin[4];
+ int vol[4];
+};
+
/* Structs */
typedef struct {
//int this_08; /* Still unknown */
@@ -168,6 +176,7 @@ struct snd_vortex {
/* Xtalk canceler */
int xt_mode; /* 1: speakers, 0:headphones. */
#endif
+ struct pcm_vol pcm_vol[NR_PCM];
int isquad; /* cache of extended ID codec flag. */
@@ -239,7 +248,7 @@ static int vortex_alsafmt_aspfmt(int alsafmt);
/* Connection stuff. */
static void vortex_connect_default(vortex_t * vortex, int en);
static int vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch,
- int dir, int type);
+ int dir, int type, int subdev);
static char vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out,
int restype);
#ifndef CHIP_AU8810
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index 6933a27a5d76..525f881f0409 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -2050,8 +2050,6 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype)
}
/* Default Connections */
-static int
-vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type);
static void vortex_connect_default(vortex_t * vortex, int en)
{
@@ -2111,15 +2109,13 @@ static void vortex_connect_default(vortex_t * vortex, int en)
Return: Return allocated DMA or same DMA passed as "dma" when dma >= 0.
*/
static int
-vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type)
+vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir,
+ int type, int subdev)
{
stream_t *stream;
int i, en;
+ struct pcm_vol *p;
- if ((nr_ch == 3)
- || ((dir == SNDRV_PCM_STREAM_CAPTURE) && (nr_ch > 2)))
- return -EBUSY;
-
if (dma >= 0) {
en = 0;
vortex_adb_checkinout(vortex,
@@ -2250,6 +2246,14 @@ vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type)
MIX_DEFIGAIN);
#endif
}
+ if (stream->type == VORTEX_PCM_ADB && en) {
+ p = &vortex->pcm_vol[subdev];
+ p->dma = dma;
+ for (i = 0; i < nr_ch; i++)
+ p->mixin[i] = mix[i];
+ for (i = 0; i < ch_top; i++)
+ p->vol[i] = 0;
+ }
}
#ifndef CHIP_AU8820
else {
@@ -2473,7 +2477,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id)
hwread(vortex->mmio, VORTEX_IRQ_STAT);
handled = 1;
}
- if (source & IRQ_MIDI) {
+ if ((source & IRQ_MIDI) && vortex->rmidi) {
snd_mpu401_uart_interrupt(vortex->irq,
vortex->rmidi->private_data);
handled = 1;
diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c
index 0ef2f9712208..e59f120742a4 100644
--- a/sound/pci/au88x0/au88x0_pcm.c
+++ b/sound/pci/au88x0/au88x0_pcm.c
@@ -122,6 +122,18 @@ static struct snd_pcm_hw_constraint_list hw_constraints_au8830_channels = {
.mask = 0,
};
#endif
+
+static void vortex_notify_pcm_vol_change(struct snd_card *card,
+ struct snd_kcontrol *kctl, int activate)
+{
+ if (activate)
+ kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ else
+ kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE |
+ SNDRV_CTL_EVENT_MASK_INFO, &(kctl->id));
+}
+
/* open callback */
static int snd_vortex_pcm_open(struct snd_pcm_substream *substream)
{
@@ -230,12 +242,14 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream,
if (stream != NULL)
vortex_adb_allocroute(chip, stream->dma,
stream->nr_ch, stream->dir,
- stream->type);
+ stream->type,
+ substream->number);
/* Alloc routes. */
dma =
vortex_adb_allocroute(chip, -1,
params_channels(hw_params),
- substream->stream, type);
+ substream->stream, type,
+ substream->number);
if (dma < 0) {
spin_unlock_irq(&chip->lock);
return dma;
@@ -246,6 +260,11 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream,
vortex_adbdma_setbuffers(chip, dma,
params_period_bytes(hw_params),
params_periods(hw_params));
+ if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) {
+ chip->pcm_vol[substream->number].active = 1;
+ vortex_notify_pcm_vol_change(chip->card,
+ chip->pcm_vol[substream->number].kctl, 1);
+ }
}
#ifndef CHIP_AU8810
else {
@@ -275,10 +294,18 @@ static int snd_vortex_pcm_hw_free(struct snd_pcm_substream *substream)
spin_lock_irq(&chip->lock);
// Delete audio routes.
if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) {
- if (stream != NULL)
+ if (stream != NULL) {
+ if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) {
+ chip->pcm_vol[substream->number].active = 0;
+ vortex_notify_pcm_vol_change(chip->card,
+ chip->pcm_vol[substream->number].kctl,
+ 0);
+ }
vortex_adb_allocroute(chip, stream->dma,
stream->nr_ch, stream->dir,
- stream->type);
+ stream->type,
+ substream->number);
+ }
}
#ifndef CHIP_AU8810
else {
@@ -506,6 +533,83 @@ static struct snd_kcontrol_new snd_vortex_mixer_spdif[] __devinitdata = {
},
};
+/* subdevice PCM Volume control */
+
+static int snd_vortex_pcm_vol_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = (VORTEX_IS_QUAD(vortex) ? 4 : 2);
+ uinfo->value.integer.min = -128;
+ uinfo->value.integer.max = 32;
+ return 0;
+}
+
+static int snd_vortex_pcm_vol_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int i;
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ int subdev = kcontrol->id.subdevice;
+ struct pcm_vol *p = &vortex->pcm_vol[subdev];
+ int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2);
+ for (i = 0; i < max_chn; i++)
+ ucontrol->value.integer.value[i] = p->vol[i];
+ return 0;
+}
+
+static int snd_vortex_pcm_vol_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int i;
+ int changed = 0;
+ int mixin;
+ unsigned char vol;
+ vortex_t *vortex = snd_kcontrol_chip(kcontrol);
+ int subdev = kcontrol->id.subdevice;
+ struct pcm_vol *p = &vortex->pcm_vol[subdev];
+ int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2);
+ for (i = 0; i < max_chn; i++) {
+ if (p->vol[i] != ucontrol->value.integer.value[i]) {
+ p->vol[i] = ucontrol->value.integer.value[i];
+ if (p->active) {
+ switch (vortex->dma_adb[p->dma].nr_ch) {
+ case 1:
+ mixin = p->mixin[0];
+ break;
+ case 2:
+ default:
+ mixin = p->mixin[(i < 2) ? i : (i - 2)];
+ break;
+ case 4:
+ mixin = p->mixin[i];
+ break;
+ };
+ vol = p->vol[i];
+ vortex_mix_setinputvolumebyte(vortex,
+ vortex->mixplayb[i], mixin, vol);
+ }
+ changed = 1;
+ }
+ }
+ return changed;
+}
+
+static const DECLARE_TLV_DB_MINMAX(vortex_pcm_vol_db_scale, -9600, 2400);
+
+static struct snd_kcontrol_new snd_vortex_pcm_vol __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+ .name = "PCM Playback Volume",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+ SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_INACTIVE,
+ .info = snd_vortex_pcm_vol_info,
+ .get = snd_vortex_pcm_vol_get,
+ .put = snd_vortex_pcm_vol_put,
+ .tlv = { .p = vortex_pcm_vol_db_scale },
+};
+
/* create a pcm device */
static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
{
@@ -555,5 +659,20 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr)
return err;
}
}
+ if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_ADB) {
+ for (i = 0; i < NR_PCM; i++) {
+ chip->pcm_vol[i].active = 0;
+ chip->pcm_vol[i].dma = -1;
+ kctl = snd_ctl_new1(&snd_vortex_pcm_vol, chip);
+ if (!kctl)
+ return -ENOMEM;
+ chip->pcm_vol[i].kctl = kctl;
+ kctl->id.device = 0;
+ kctl->id.subdevice = i;
+ err = snd_ctl_add(chip->card, kctl);
+ if (err < 0)
+ return err;
+ }
+ }
return 0;
}
diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c
index 8afd8b5d1ac7..4439636971eb 100644
--- a/sound/pci/aw2/aw2-saa7146.c
+++ b/sound/pci/aw2/aw2-saa7146.c
@@ -27,7 +27,6 @@
#include <linux/pci.h>
#include <linux/interrupt.h>
#include <linux/delay.h>
-#include <asm/system.h>
#include <asm/io.h>
#include <sound/core.h>
#include <sound/initval.h>
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 95ffa6a9db6e..496f14c1a731 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2684,10 +2684,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
if (err < 0)
goto out_err;
+ opl3->private_data = chip;
}
- opl3->private_data = chip;
-
sprintf(card->longname, "%s at 0x%lx, irq %i",
card->shortname, chip->ctrl_io, chip->irq);
diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c
index b78f3fc3c33c..6109490b83e8 100644
--- a/sound/pci/ctxfi/ctvmem.c
+++ b/sound/pci/ctxfi/ctvmem.c
@@ -36,7 +36,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size)
size = CT_PAGE_ALIGN(size);
if (size > vm->size) {
- printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural "
+ printk(KERN_ERR "ctxfi: Fail! No sufficient device virtual "
"memory space available!\n");
return NULL;
}
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index cb557c603a80..a8faae1c85e4 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -142,6 +142,7 @@ static int enable_mpu[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2};
#ifdef SUPPORT_JOYSTICK
static bool joystick[SNDRV_CARDS];
#endif
+static int radio_nr[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1};
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
@@ -165,6 +166,9 @@ MODULE_PARM_DESC(enable_mpu, "Enable MPU401. (0 = off, 1 = on, 2 = auto)");
module_param_array(joystick, bool, NULL, 0444);
MODULE_PARM_DESC(joystick, "Enable joystick.");
#endif
+module_param_array(radio_nr, int, NULL, 0444);
+MODULE_PARM_DESC(radio_nr, "Radio device numbers");
+
#define NR_APUS 64
@@ -558,6 +562,7 @@ struct es1968 {
struct work_struct hwvol_work;
#ifdef CONFIG_SND_ES1968_RADIO
+ struct v4l2_device v4l2_dev;
struct snd_tea575x tea;
#endif
};
@@ -2613,6 +2618,7 @@ static int snd_es1968_free(struct es1968 *chip)
#ifdef CONFIG_SND_ES1968_RADIO
snd_tea575x_exit(&chip->tea);
+ v4l2_device_unregister(&chip->v4l2_dev);
#endif
if (chip->irq >= 0)
@@ -2655,6 +2661,7 @@ static int __devinit snd_es1968_create(struct snd_card *card,
int capt_streams,
int chip_type,
int do_pm,
+ int radio_nr,
struct es1968 **chip_ret)
{
static struct snd_device_ops ops = {
@@ -2751,7 +2758,14 @@ static int __devinit snd_es1968_create(struct snd_card *card,
snd_card_set_dev(card, &pci->dev);
#ifdef CONFIG_SND_ES1968_RADIO
+ err = v4l2_device_register(&pci->dev, &chip->v4l2_dev);
+ if (err < 0) {
+ snd_es1968_free(chip);
+ return err;
+ }
+ chip->tea.v4l2_dev = &chip->v4l2_dev;
chip->tea.private_data = chip;
+ chip->tea.radio_nr = radio_nr;
chip->tea.ops = &snd_es1968_tea_ops;
strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card));
sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci));
@@ -2797,6 +2811,7 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci,
pcm_substreams_c[dev],
pci_id->driver_data,
use_pm[dev],
+ radio_nr[dev],
&chip)) < 0) {
snd_card_free(card);
return err;
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index 9597ef1eccca..a416ea8af3e9 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -58,6 +58,7 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card
* High 16-bits are video (radio) device number + 1
*/
static int tea575x_tuner[SNDRV_CARDS];
+static int radio_nr[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1};
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for the FM801 soundcard.");
@@ -67,6 +68,9 @@ module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable FM801 soundcard.");
module_param_array(tea575x_tuner, int, NULL, 0444);
MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only).");
+module_param_array(radio_nr, int, NULL, 0444);
+MODULE_PARM_DESC(radio_nr, "Radio device numbers");
+
#define TUNER_DISABLED (1<<3)
#define TUNER_ONLY (1<<4)
@@ -197,6 +201,7 @@ struct fm801 {
struct snd_info_entry *proc_entry;
#ifdef CONFIG_SND_FM801_TEA575X_BOOL
+ struct v4l2_device v4l2_dev;
struct snd_tea575x tea;
#endif
@@ -1154,8 +1159,10 @@ static int snd_fm801_free(struct fm801 *chip)
__end_hw:
#ifdef CONFIG_SND_FM801_TEA575X_BOOL
- if (!(chip->tea575x_tuner & TUNER_DISABLED))
+ if (!(chip->tea575x_tuner & TUNER_DISABLED)) {
snd_tea575x_exit(&chip->tea);
+ v4l2_device_unregister(&chip->v4l2_dev);
+ }
#endif
if (chip->irq >= 0)
free_irq(chip->irq, chip);
@@ -1175,6 +1182,7 @@ static int snd_fm801_dev_free(struct snd_device *device)
static int __devinit snd_fm801_create(struct snd_card *card,
struct pci_dev * pci,
int tea575x_tuner,
+ int radio_nr,
struct fm801 ** rchip)
{
struct fm801 *chip;
@@ -1234,6 +1242,13 @@ static int __devinit snd_fm801_create(struct snd_card *card,
snd_card_set_dev(card, &pci->dev);
#ifdef CONFIG_SND_FM801_TEA575X_BOOL
+ err = v4l2_device_register(&pci->dev, &chip->v4l2_dev);
+ if (err < 0) {
+ snd_fm801_free(chip);
+ return err;
+ }
+ chip->tea.v4l2_dev = &chip->v4l2_dev;
+ chip->tea.radio_nr = radio_nr;
chip->tea.private_data = chip;
chip->tea.ops = &snd_fm801_tea_ops;
sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci));
@@ -1241,6 +1256,7 @@ static int __devinit snd_fm801_create(struct snd_card *card,
(tea575x_tuner & TUNER_TYPE_MASK) < 4) {
if (snd_tea575x_init(&chip->tea)) {
snd_printk(KERN_ERR "TEA575x radio not found\n");
+ snd_fm801_free(chip);
return -ENODEV;
}
} else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) {
@@ -1287,7 +1303,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
if (err < 0)
return err;
- if ((err = snd_fm801_create(card, pci, tea575x_tuner[dev], &chip)) < 0) {
+ if ((err = snd_fm801_create(card, pci, tea575x_tuner[dev], radio_nr[dev], &chip)) < 0) {
snd_card_free(card);
return err;
}
diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c
deleted file mode 100644
index 3b5170b9700f..000000000000
--- a/sound/pci/hda/alc260_quirks.c
+++ /dev/null
@@ -1,968 +0,0 @@
-/*
- * ALC260 quirk models
- * included by patch_realtek.c
- */
-
-/* ALC260 models */
-enum {
- ALC260_AUTO,
- ALC260_BASIC,
- ALC260_FUJITSU_S702X,
- ALC260_ACER,
- ALC260_WILL,
- ALC260_REPLACER_672V,
- ALC260_FAVORIT100,
-#ifdef CONFIG_SND_DEBUG
- ALC260_TEST,
-#endif
- ALC260_MODEL_LAST /* last tag */
-};
-
-static const hda_nid_t alc260_dac_nids[1] = {
- /* front */
- 0x02,
-};
-
-static const hda_nid_t alc260_adc_nids[1] = {
- /* ADC0 */
- 0x04,
-};
-
-static const hda_nid_t alc260_adc_nids_alt[1] = {
- /* ADC1 */
- 0x05,
-};
-
-/* NIDs used when simultaneous access to both ADCs makes sense. Note that
- * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC.
- */
-static const hda_nid_t alc260_dual_adc_nids[2] = {
- /* ADC0, ADC1 */
- 0x04, 0x05
-};
-
-#define ALC260_DIGOUT_NID 0x03
-#define ALC260_DIGIN_NID 0x06
-
-static const struct hda_input_mux alc260_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack,
- * headphone jack and the internal CD lines since these are the only pins at
- * which audio can appear. For flexibility, also allow the option of
- * recording the mixer output on the second ADC (ADC0 doesn't have a
- * connection to the mixer output).
- */
-static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = {
- {
- .num_items = 3,
- .items = {
- { "Mic/Line", 0x0 },
- { "CD", 0x4 },
- { "Headphone", 0x2 },
- },
- },
- {
- .num_items = 4,
- .items = {
- { "Mic/Line", 0x0 },
- { "CD", 0x4 },
- { "Headphone", 0x2 },
- { "Mixer", 0x5 },
- },
- },
-
-};
-
-/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to
- * the Fujitsu S702x, but jacks are marked differently.
- */
-static const struct hda_input_mux alc260_acer_capture_sources[2] = {
- {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Headphone", 0x5 },
- },
- },
- {
- .num_items = 5,
- .items = {
- { "Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- { "Headphone", 0x6 },
- { "Mixer", 0x5 },
- },
- },
-};
-
-/* Maxdata Favorit 100XS */
-static const struct hda_input_mux alc260_favorit100_capture_sources[2] = {
- {
- .num_items = 2,
- .items = {
- { "Line/Mic", 0x0 },
- { "CD", 0x4 },
- },
- },
- {
- .num_items = 3,
- .items = {
- { "Line/Mic", 0x0 },
- { "CD", 0x4 },
- { "Mixer", 0x5 },
- },
- },
-};
-
-/*
- * 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
- * element which allows changing the channel mode, so the verb list is
- * never used.
- */
-static const struct hda_channel_mode alc260_modes[1] = {
- { 2, NULL },
-};
-
-
-/* Mixer combinations
- *
- * basic: base_output + input + pc_beep + capture
- * fujitsu: fujitsu + capture
- * acer: acer + capture
- */
-
-static const struct snd_kcontrol_new alc260_base_output_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc260_input_mixer[] = {
- 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),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
- { } /* end */
-};
-
-/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12,
- * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10.
- */
-static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT),
- ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- 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),
- ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
- { } /* end */
-};
-
-/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current
- * versions of the ALC260 don't act on requests to enable mic bias from NID
- * 0x0f (used to drive the headphone jack in these laptops). The ALC260
- * datasheet doesn't mention this restriction. At this stage it's not clear
- * whether this behaviour is intentional or is a hardware bug in chip
- * revisions available in early 2006. Therefore for now allow the
- * "Headphone Jack Mode" control to span all choices, but if it turns out
- * that the lack of mic bias for this NID is intentional we could change the
- * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
- *
- * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006
- * don't appear to make the mic bias available from the "line" jack, even
- * though the NID used for this jack (0x14) can supply it. The theory is
- * that perhaps Acer have included blocking capacitors between the ALC260
- * and the output jack. If this turns out to be the case for all such
- * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT
- * to ALC_PIN_DIR_INOUT_NOMICBIAS.
- *
- * The C20x Tablet series have a mono internal speaker which is controlled
- * via the chip's Mono sum widget and pin complex, so include the necessary
- * controls for such models. On models without a "mono speaker" the control
- * won't do anything.
- */
-static const struct snd_kcontrol_new alc260_acer_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
- ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0,
- HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 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("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- { } /* end */
-};
-
-/* Maxdata Favorit 100XS: one output and one input (0x12) jack
- */
-static const struct snd_kcontrol_new alc260_favorit100_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
- ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- { } /* end */
-};
-
-/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12,
- * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17.
- */
-static const struct snd_kcontrol_new alc260_will_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
- { } /* end */
-};
-
-/* Replacer 672V ALC260 pin usage: Mic jack = 0x12,
- * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f.
- */
-static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = {
- HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
- ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
- HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
- ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
- { } /* end */
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb alc260_init_verbs[] = {
- /* Line In pin widget for input */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* CD pin widget for input */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /* Mic2 (front panel) pin widget for input and vref at 80% */
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- /* LINE-2 is used for line-out in rear */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* select line-out */
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* LINE-OUT pin */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* enable HP */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* enable Mono */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* set connection select to line in (default select for this ADC) */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* mute capture amp left and right */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* set connection select to line in (default select for this ADC) */
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* set vol=0 Line-Out mixer amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* unmute pin widget amp left and right (no gain on this amp) */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* set vol=0 HP mixer amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* unmute pin widget amp left and right (no gain on this amp) */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* set vol=0 Mono mixer amp left and right */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* unmute pin widget amp left and right (no gain on this amp) */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* unmute LINE-2 out pin */
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 &
- * Line In 2 = 0x03
- */
- /* mute analog inputs */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
- /* mute Front out path */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* mute Headphone out path */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* mute Mono out path */
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- { }
-};
-
-/* Initialisation sequence for ALC260 as configured in Fujitsu S702x
- * laptops. ALC260 pin usage: Mic/Line jack = 0x12, HP jack = 0x14, CD
- * audio = 0x16, internal speaker = 0x10.
- */
-static const 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. */
- {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)},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x15, 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},
-
- /* Ensure Line1 pin widget takes its input from the OUT1 sum bus
- * when acting as an output.
- */
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* 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 output buffer since it starts as an output.
- * If the pin mode is changed by the user the pin mode control will
- * take care of enabling the pin's input/output buffers as needed.
- * Therefore there's no need to enable the input buffer at this
- * stage.
- */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute input buffer of 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 match default mixer setting - line
- * in (on mic1 pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do the same for the second ADC: mute capture input amp and
- * set ADC connection to line in (on mic1 pin)
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, 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 */
-
- { }
-};
-
-/* Initialisation sequence for ALC260 as configured in Acer TravelMate and
- * similar laptops (adapted from Fujitsu init verbs).
- */
-static const struct hda_verb alc260_acer_init_verbs[] = {
- /* On TravelMate laptops, GPIO 0 enables the internal speaker and
- * the headphone jack. Turn this on and rely on the standard mute
- * methods whenever the user wants to turn these outputs off.
- */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
- /* Internal speaker/Headphone jack is connected to Line-out pin */
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Internal microphone/Mic jack is connected to Mic1 pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- /* Line In jack is connected to Line1 pin */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Some Acers (eg: C20x Tablets) use Mono pin for internal speaker */
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Ensure all other unused pins are disabled and muted. */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x10, 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)},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x15, 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},
-
- /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
- * bus when acting as outputs.
- */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute Line-out pin widget amp left and right
- * (no equiv mixer ctrl)
- */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute mono pin widget amp output (no equiv mixer ctrl) */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute Mic1 and Line1 pin widget input buffers since they start as
- * inputs. If the pin mode is changed by the user the pin mode control
- * will take care of enabling the pin's input/output buffers as needed.
- * Therefore there's no need to enable the input buffer at this
- * stage.
- */
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x14, 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 match default mixer setting - mic
- * (on mic1 pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do similar with the second ADC: mute capture input amp and
- * set ADC connection to mic to match ALSA's default state.
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, 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 */
-
- { }
-};
-
-/* Initialisation sequence for Maxdata Favorit 100XS
- * (adapted from Acer init verbs).
- */
-static const struct hda_verb alc260_favorit100_init_verbs[] = {
- /* GPIO 0 enables the output jack.
- * Turn this on and rely on the standard mute
- * methods whenever the user wants to turn these outputs off.
- */
- {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x01},
- /* Line/Mic input jack is connected to Mic1 pin */
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- /* Ensure all other unused pins are disabled and muted. */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x10, 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)},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
- {0x15, 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},
-
- /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
- * bus when acting as outputs.
- */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute Line-out pin widget amp left and right
- * (no equiv mixer ctrl)
- */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Unmute Mic1 and Line1 pin widget input buffers since they start as
- * inputs. If the pin mode is changed by the user the pin mode control
- * will take care of enabling the pin's input/output buffers as needed.
- * Therefore there's no need to enable the input buffer at this
- * stage.
- */
- {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 match default mixer setting - mic
- * (on mic1 pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do similar with the second ADC: mute capture input amp and
- * set ADC connection to mic to match ALSA's default state.
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, 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 const struct hda_verb alc260_will_verbs[] = {
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x1a, AC_VERB_SET_PROC_COEF, 0x3040},
- {}
-};
-
-static const struct hda_verb alc260_replacer_672v_verbs[] = {
- {0x0f, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
- {0x1a, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x1a, AC_VERB_SET_PROC_COEF, 0x3050},
-
- {0x01, AC_VERB_SET_GPIO_MASK, 0x01},
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
-
- {0x0f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {}
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc260_replacer_672v_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */
- present = snd_hda_jack_detect(codec, 0x0f);
- if (present) {
- snd_hda_codec_write_cache(codec, 0x01, 0,
- AC_VERB_SET_GPIO_DATA, 1);
- snd_hda_codec_write_cache(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_HP);
- } else {
- snd_hda_codec_write_cache(codec, 0x01, 0,
- AC_VERB_SET_GPIO_DATA, 0);
- snd_hda_codec_write_cache(codec, 0x0f, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- PIN_OUT);
- }
-}
-
-static void alc260_replacer_672v_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- if ((res >> 26) == ALC_HP_EVENT)
- alc260_replacer_672v_automute(codec);
-}
-
-static const struct hda_verb alc260_hp_dc7600_verbs[] = {
- {0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {}
-};
-
-/* Test configuration for debugging, modelled after the ALC880 test
- * configuration.
- */
-#ifdef CONFIG_SND_DEBUG
-static const hda_nid_t alc260_test_dac_nids[1] = {
- 0x02,
-};
-static const hda_nid_t alc260_test_adc_nids[2] = {
- 0x04, 0x05,
-};
-/* For testing the ALC260, each input MUX needs its own definition since
- * the signal assignments are different. This assumes that the first ADC
- * is NID 0x04.
- */
-static const struct hda_input_mux alc260_test_capture_sources[2] = {
- {
- .num_items = 7,
- .items = {
- { "MIC1 pin", 0x0 },
- { "MIC2 pin", 0x1 },
- { "LINE1 pin", 0x2 },
- { "LINE2 pin", 0x3 },
- { "CD pin", 0x4 },
- { "LINE-OUT pin", 0x5 },
- { "HP-OUT pin", 0x6 },
- },
- },
- {
- .num_items = 8,
- .items = {
- { "MIC1 pin", 0x0 },
- { "MIC2 pin", 0x1 },
- { "LINE1 pin", 0x2 },
- { "LINE2 pin", 0x3 },
- { "CD pin", 0x4 },
- { "Mixer", 0x5 },
- { "LINE-OUT pin", 0x6 },
- { "HP-OUT pin", 0x7 },
- },
- },
-};
-static const struct snd_kcontrol_new alc260_test_mixer[] = {
- /* Output driver widgets */
- HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT),
-
- /* Modes for retasking pin widgets
- * Note: the ALC260 doesn't seem to act on requests to enable mic
- * bias from NIDs 0x0f and 0x10. The ALC260 datasheet doesn't
- * mention this restriction. At this stage it's not clear whether
- * this behaviour is intentional or is a hardware bug in chip
- * revisions available at least up until early 2006. Therefore for
- * now allow the "HP-OUT" and "LINE-OUT" Mode controls to span all
- * choices, but if it turns out that the lack of mic bias for these
- * NIDs is intentional we could change their modes from
- * ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS.
- */
- ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT),
- ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT),
-
- /* Loopback mixer controls */
- HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, 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-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
- HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
- HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT),
-
- /* Controls for GPIO pins, assuming they are configured as outputs */
- ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01),
- ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02),
- ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04),
- ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08),
-
- /* Switches to allow the digital IO pins to be enabled. The datasheet
- * is ambigious as to which NID is which; testing on laptops which
- * make this output available should provide clarification.
- */
- ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01),
- ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01),
-
- /* A switch allowing EAPD to be enabled. Some laptops seem to use
- * this output to turn on an external amplifier.
- */
- ALC_EAPD_CTRL_SWITCH("LINE-OUT EAPD Enable Switch", 0x0f, 0x02),
- ALC_EAPD_CTRL_SWITCH("HP-OUT EAPD Enable Switch", 0x10, 0x02),
-
- { } /* end */
-};
-static const struct hda_verb alc260_test_init_verbs[] = {
- /* Enable all GPIOs as outputs with an initial value of 0 */
- {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f},
- {0x01, AC_VERB_SET_GPIO_DATA, 0x00},
- {0x01, AC_VERB_SET_GPIO_MASK, 0x0f},
-
- /* Enable retasking pins as output, initially without power amp */
- {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-
- /* Disable digital (SPDIF) pins initially, but users can enable
- * them via a mixer switch. In the case of SPDIF-out, this initverb
- * payload also sets the generation to 0, output to be in "consumer"
- * PCM format, copyright asserted, no pre-emphasis and no validity
- * control.
- */
- {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
- {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
-
- /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the
- * OUT1 sum bus when acting as an output.
- */
- {0x0b, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0c, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0d, AC_VERB_SET_CONNECT_SEL, 0},
- {0x0e, AC_VERB_SET_CONNECT_SEL, 0},
-
- /* Start with output sum widgets muted and their output gains at min */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-
- /* Unmute retasking pin widget output buffers since the default
- * state appears to be output. As the pin mode is changed by the
- * user the pin mode control will take care of enabling the pin's
- * input/output buffers as needed.
- */
- {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Also unmute the mono-out pin widget */
- {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Mute capture amp left and right */
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- /* Set ADC connection select to match default mixer setting (mic1
- * pin)
- */
- {0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Do the same for the second ADC: mute capture input amp and
- * set ADC connection to mic1 pin
- */
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x05, 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 */
-
- { }
-};
-#endif
-
-/*
- * ALC260 configurations
- */
-static const char * const alc260_models[ALC260_MODEL_LAST] = {
- [ALC260_BASIC] = "basic",
- [ALC260_FUJITSU_S702X] = "fujitsu",
- [ALC260_ACER] = "acer",
- [ALC260_WILL] = "will",
- [ALC260_REPLACER_672V] = "replacer",
- [ALC260_FAVORIT100] = "favorit100",
-#ifdef CONFIG_SND_DEBUG
- [ALC260_TEST] = "test",
-#endif
- [ALC260_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc260_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
- SND_PCI_QUIRK(0x1025, 0x007f, "Acer", ALC260_WILL),
- SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
- SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
- SND_PCI_QUIRK(0x104d, 0x81bb, "Sony VAIO", ALC260_BASIC),
- SND_PCI_QUIRK(0x104d, 0x81cc, "Sony VAIO", ALC260_BASIC),
- SND_PCI_QUIRK(0x104d, 0x81cd, "Sony VAIO", ALC260_BASIC),
- SND_PCI_QUIRK(0x10cf, 0x1326, "Fujitsu S702X", ALC260_FUJITSU_S702X),
- SND_PCI_QUIRK(0x152d, 0x0729, "CTL U553W", ALC260_BASIC),
- SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_REPLACER_672V),
- SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_WILL),
- {}
-};
-
-static const struct alc_config_preset alc260_presets[] = {
- [ALC260_BASIC] = {
- .mixers = { alc260_base_output_mixer,
- alc260_input_mixer },
- .init_verbs = { alc260_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- },
- [ALC260_FUJITSU_S702X] = {
- .mixers = { alc260_fujitsu_mixer },
- .init_verbs = { alc260_fujitsu_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_fujitsu_capture_sources),
- .input_mux = alc260_fujitsu_capture_sources,
- },
- [ALC260_ACER] = {
- .mixers = { alc260_acer_mixer },
- .init_verbs = { alc260_acer_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
- .input_mux = alc260_acer_capture_sources,
- },
- [ALC260_FAVORIT100] = {
- .mixers = { alc260_favorit100_mixer },
- .init_verbs = { alc260_favorit100_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
- .adc_nids = alc260_dual_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
- .input_mux = alc260_favorit100_capture_sources,
- },
- [ALC260_WILL] = {
- .mixers = { alc260_will_mixer },
- .init_verbs = { alc260_init_verbs, alc260_will_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
- .adc_nids = alc260_adc_nids,
- .dig_out_nid = ALC260_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- },
- [ALC260_REPLACER_672V] = {
- .mixers = { alc260_replacer_672v_mixer },
- .init_verbs = { alc260_init_verbs, alc260_replacer_672v_verbs },
- .num_dacs = ARRAY_SIZE(alc260_dac_nids),
- .dac_nids = alc260_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_adc_nids),
- .adc_nids = alc260_adc_nids,
- .dig_out_nid = ALC260_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .input_mux = &alc260_capture_source,
- .unsol_event = alc260_replacer_672v_unsol_event,
- .init_hook = alc260_replacer_672v_automute,
- },
-#ifdef CONFIG_SND_DEBUG
- [ALC260_TEST] = {
- .mixers = { alc260_test_mixer },
- .init_verbs = { alc260_test_init_verbs },
- .num_dacs = ARRAY_SIZE(alc260_test_dac_nids),
- .dac_nids = alc260_test_dac_nids,
- .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids),
- .adc_nids = alc260_test_adc_nids,
- .num_channel_mode = ARRAY_SIZE(alc260_modes),
- .channel_mode = alc260_modes,
- .num_mux_defs = ARRAY_SIZE(alc260_test_capture_sources),
- .input_mux = alc260_test_capture_sources,
- },
-#endif
-};
-
diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c
deleted file mode 100644
index 5b68435d195b..000000000000
--- a/sound/pci/hda/alc880_quirks.c
+++ /dev/null
@@ -1,1700 +0,0 @@
-/*
- * ALC880 quirk models
- * included by patch_realtek.c
- */
-
-/* ALC880 board config type */
-enum {
- ALC880_AUTO,
- ALC880_3ST,
- ALC880_3ST_DIG,
- ALC880_5ST,
- ALC880_5ST_DIG,
- ALC880_W810,
- ALC880_Z71V,
- ALC880_6ST,
- ALC880_6ST_DIG,
- ALC880_F1734,
- ALC880_ASUS,
- ALC880_ASUS_DIG,
- ALC880_ASUS_W1V,
- ALC880_ASUS_DIG2,
- ALC880_FUJITSU,
- ALC880_UNIWILL_DIG,
- ALC880_UNIWILL,
- ALC880_UNIWILL_P53,
- ALC880_CLEVO,
- ALC880_TCL_S700,
- ALC880_LG,
-#ifdef CONFIG_SND_DEBUG
- ALC880_TEST,
-#endif
- ALC880_MODEL_LAST /* last tag */
-};
-
-/*
- * ALC880 3-stack model
- *
- * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e)
- * Pin assignment: Front = 0x14, Line-In/Surr = 0x1a, Mic/CLFE = 0x18,
- * F-Mic = 0x1b, HP = 0x19
- */
-
-static const hda_nid_t alc880_dac_nids[4] = {
- /* front, rear, clfe, rear_surr */
- 0x02, 0x05, 0x04, 0x03
-};
-
-static const hda_nid_t alc880_adc_nids[3] = {
- /* ADC0-2 */
- 0x07, 0x08, 0x09,
-};
-
-/* The datasheet says the node 0x07 is connected from inputs,
- * but it shows zero connection in the real implementation on some devices.
- * Note: this is a 915GAV bug, fixed on 915GLV
- */
-static const hda_nid_t alc880_adc_nids_alt[2] = {
- /* ADC1-2 */
- 0x08, 0x09,
-};
-
-#define ALC880_DIGOUT_NID 0x06
-#define ALC880_DIGIN_NID 0x0a
-#define ALC880_PIN_CD_NID 0x1c
-
-static const struct hda_input_mux alc880_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x3 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-/* channel source setting (2/6 channel selection for 3-stack) */
-/* 2ch mode */
-static const struct hda_verb alc880_threestack_ch2_init[] = {
- /* set line-in to input, mute it */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- /* set mic-in to input vref 80%, mute it */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/* 6ch mode */
-static const struct hda_verb alc880_threestack_ch6_init[] = {
- /* set line-in to output, unmute it */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- /* set mic-in to output, unmute it */
- { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc880_threestack_modes[2] = {
- { 2, alc880_threestack_ch2_init },
- { 6, alc880_threestack_ch6_init },
-};
-
-static const struct snd_kcontrol_new alc880_three_stack_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- 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),
- 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),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-/*
- * ALC880 5-stack model
- *
- * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0d),
- * Side = 0x02 (0xd)
- * Pin assignment: Front = 0x14, Surr = 0x17, CLFE = 0x16
- * Line-In/Side = 0x1a, Mic = 0x18, F-Mic = 0x1b, HP = 0x19
- */
-
-/* additional mixers to alc880_three_stack_mixer */
-static const struct snd_kcontrol_new alc880_five_stack_mixer[] = {
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0d, 2, HDA_INPUT),
- { } /* end */
-};
-
-/* channel source setting (6/8 channel selection for 5-stack) */
-/* 6ch mode */
-static const struct hda_verb alc880_fivestack_ch6_init[] = {
- /* set line-in to input, mute it */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
- { } /* end */
-};
-
-/* 8ch mode */
-static const struct hda_verb alc880_fivestack_ch8_init[] = {
- /* set line-in to output, unmute it */
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
- { } /* end */
-};
-
-static const struct hda_channel_mode alc880_fivestack_modes[2] = {
- { 6, alc880_fivestack_ch6_init },
- { 8, alc880_fivestack_ch8_init },
-};
-
-
-/*
- * ALC880 6-stack model
- *
- * DAC: Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e),
- * Side = 0x05 (0x0f)
- * Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, Side = 0x17,
- * Mic = 0x18, F-Mic = 0x19, Line = 0x1a, HP = 0x1b
- */
-
-static const hda_nid_t alc880_6st_dac_nids[4] = {
- /* front, rear, clfe, rear_surr */
- 0x02, 0x03, 0x04, 0x05
-};
-
-static const struct hda_input_mux alc880_6stack_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-/* fixed 8-channels */
-static const struct hda_channel_mode alc880_sixstack_modes[1] = {
- { 8, NULL },
-};
-
-static const struct snd_kcontrol_new alc880_six_stack_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-
-/*
- * ALC880 W810 model
- *
- * W810 has rear IO for:
- * Front (DAC 02)
- * Surround (DAC 03)
- * Center/LFE (DAC 04)
- * Digital out (06)
- *
- * The system also has a pair of internal speakers, and a headphone jack.
- * These are both connected to Line2 on the codec, hence to DAC 02.
- *
- * There is a variable resistor to control the speaker or headphone
- * volume. This is a hardware-only device without a software API.
- *
- * Plugging headphones in will disable the internal speakers. This is
- * implemented in hardware, not via the driver using jack sense. In
- * a similar fashion, plugging into the rear socket marked "front" will
- * disable both the speakers and headphones.
- *
- * For input, there's a microphone jack, and an "audio in" jack.
- * These may not do anything useful with this driver yet, because I
- * haven't setup any initialization verbs for these yet...
- */
-
-static const hda_nid_t alc880_w810_dac_nids[3] = {
- /* front, rear/surround, clfe */
- 0x02, 0x03, 0x04
-};
-
-/* fixed 6 channels */
-static const struct hda_channel_mode alc880_w810_modes[1] = {
- { 6, NULL }
-};
-
-/* Pin assignment: Front = 0x14, Surr = 0x15, CLFE = 0x16, HP = 0x1b */
-static const struct snd_kcontrol_new alc880_w810_base_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- { } /* end */
-};
-
-
-/*
- * Z710V model
- *
- * DAC: Front = 0x02 (0x0c), HP = 0x03 (0x0d)
- * Pin assignment: Front = 0x14, HP = 0x15, Mic = 0x18, Mic2 = 0x19(?),
- * Line = 0x1a
- */
-
-static const hda_nid_t alc880_z71v_dac_nids[1] = {
- 0x02
-};
-#define ALC880_Z71V_HP_DAC 0x03
-
-/* fixed 2 channels */
-static const struct hda_channel_mode alc880_2_jack_modes[1] = {
- { 2, NULL }
-};
-
-static const struct snd_kcontrol_new alc880_z71v_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- 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),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-
-/*
- * ALC880 F1734 model
- *
- * DAC: HP = 0x02 (0x0c), Front = 0x03 (0x0d)
- * Pin assignment: HP = 0x14, Front = 0x15, Mic = 0x18
- */
-
-static const hda_nid_t alc880_f1734_dac_nids[1] = {
- 0x03
-};
-#define ALC880_F1734_HP_DAC 0x02
-
-static const struct snd_kcontrol_new alc880_f1734_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("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, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct hda_input_mux alc880_f1734_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x1 },
- { "CD", 0x4 },
- },
-};
-
-
-/*
- * ALC880 ASUS model
- *
- * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
- * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
- * Mic = 0x18, Line = 0x1a
- */
-
-#define alc880_asus_dac_nids alc880_w810_dac_nids /* identical with w810 */
-#define alc880_asus_modes alc880_threestack_modes /* 2/6 channel mode */
-
-static const struct snd_kcontrol_new alc880_asus_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-/*
- * ALC880 ASUS W1V model
- *
- * DAC: HP/Front = 0x02 (0x0c), Surr = 0x03 (0x0d), CLFE = 0x04 (0x0e)
- * Pin assignment: HP/Front = 0x14, Surr = 0x15, CLFE = 0x16,
- * Mic = 0x18, Line = 0x1a, Line2 = 0x1b
- */
-
-/* additional mixers to alc880_asus_mixer */
-static const struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
- HDA_CODEC_VOLUME("Line2 Playback Volume", 0x0b, 0x03, HDA_INPUT),
- HDA_CODEC_MUTE("Line2 Playback Switch", 0x0b, 0x03, HDA_INPUT),
- { } /* end */
-};
-
-/* TCL S700 */
-static const struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0B, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0B, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0B, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0B, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-/* Uniwill */
-static const struct snd_kcontrol_new alc880_uniwill_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc880_fujitsu_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("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),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc880_uniwill_p53_mixer[] = {
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Headphone Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- { } /* end */
-};
-
-/*
- * initialize the codec volumes, etc
- */
-
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static const struct hda_verb alc880_volume_init_verbs[] = {
- /*
- * Unmute ADC0-2 and set the default input to mic-in
- */
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
- /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback
- * mixer widget
- * Note: PASD motherboards uses the Line In 2 as the input for front
- * panel mic (mic 2)
- */
- /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-
- /*
- * Set up output mixers (0x0c - 0x0f)
- */
- /* set vol=0 to output mixers */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* set up input amps for analog loopback */
- /* Amp Indices: DAC = 0, mixer = 1 */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- { }
-};
-
-/*
- * 3-stack pin configuration:
- * front = 0x14, mic/clfe = 0x18, HP = 0x19, line/surr = 0x1a, f-mic = 0x1b
- */
-static const struct hda_verb alc880_pin_3stack_init_verbs[] = {
- /*
- * preset connection lists of input pins
- * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
- */
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */
-
- /*
- * Set pin mode and muting
- */
- /* set front pin widgets 0x14 for output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mic2 (as headphone out) for HP output */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Line In pin widget for input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line2 (as front mic) pin widget for input and vref at 80% */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/*
- * 5-stack pin configuration:
- * front = 0x14, surround = 0x17, clfe = 0x16, mic = 0x18, HP = 0x19,
- * line-in/side = 0x1a, f-mic = 0x1b
- */
-static const struct hda_verb alc880_pin_5stack_init_verbs[] = {
- /*
- * preset connection lists of input pins
- * 0 = front, 1 = rear_surr, 2 = CLFE, 3 = surround
- */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/side */
-
- /*
- * Set pin mode and muting
- */
- /* set pin widgets 0x14-0x17 for output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* unmute pins for output (no gain on this amp) */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Mic2 (as headphone out) for HP output */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Line In pin widget for input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line2 (as front mic) pin widget for input and vref at 80% */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/*
- * W810 pin configuration:
- * front = 0x14, surround = 0x15, clfe = 0x16, HP = 0x1b
- */
-static const struct hda_verb alc880_pin_w810_init_verbs[] = {
- /* hphone/speaker input selector: front DAC */
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- { }
-};
-
-/*
- * Z71V pin configuration:
- * Speaker-out = 0x14, HP = 0x15, Mic = 0x18, Line-in = 0x1a, Mic2 = 0x1b (?)
- */
-static const struct hda_verb alc880_pin_z71v_init_verbs[] = {
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/*
- * 6-stack pin configuration:
- * front = 0x14, surr = 0x15, clfe = 0x16, side = 0x17, mic = 0x18,
- * f-mic = 0x19, line = 0x1a, HP = 0x1b
- */
-static const struct hda_verb alc880_pin_6stack_init_verbs[] = {
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/*
- * Uniwill pin configuration:
- * HP = 0x14, InternalSpeaker = 0x15, mic = 0x18, internal mic = 0x19,
- * line = 0x1a
- */
-static const struct hda_verb alc880_uniwill_init_verbs[] = {
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, */
- /* {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_MIC_EVENT},
-
- { }
-};
-
-/*
-* Uniwill P53
-* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19,
- */
-static const struct hda_verb alc880_uniwill_p53_init_verbs[] = {
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_DCVOL_EVENT},
-
- { }
-};
-
-static const struct hda_verb alc880_beep_init_verbs[] = {
- { 0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) },
- { }
-};
-
-static void alc880_uniwill_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x16;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
-}
-
-static void alc880_uniwill_init_hook(struct hda_codec *codec)
-{
- alc_hp_automute(codec);
- alc88x_simple_mic_automute(codec);
-}
-
-static void alc880_uniwill_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- /* Looks like the unsol event is incompatible with the standard
- * definition. 4bit tag is placed at 28 bit!
- */
- switch (res >> 28) {
- case ALC_MIC_EVENT:
- alc88x_simple_mic_automute(codec);
- break;
- default:
- alc_sku_unsol_event(codec, res);
- break;
- }
-}
-
-static void alc880_uniwill_p53_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x15;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
-}
-
-static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- present = snd_hda_codec_read(codec, 0x21, 0,
- AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
- present &= HDA_AMP_VOLMASK;
- snd_hda_codec_amp_stereo(codec, 0x0c, HDA_OUTPUT, 0,
- HDA_AMP_VOLMASK, present);
- snd_hda_codec_amp_stereo(codec, 0x0d, HDA_OUTPUT, 0,
- HDA_AMP_VOLMASK, present);
-}
-
-static void alc880_uniwill_p53_unsol_event(struct hda_codec *codec,
- unsigned int res)
-{
- /* Looks like the unsol event is incompatible with the standard
- * definition. 4bit tag is placed at 28 bit!
- */
- if ((res >> 28) == ALC_DCVOL_EVENT)
- alc880_uniwill_p53_dcvol_automute(codec);
- else
- alc_sku_unsol_event(codec, res);
-}
-
-/*
- * F1734 pin configuration:
- * HP = 0x14, speaker-out = 0x15, mic = 0x18
- */
-static const struct hda_verb alc880_pin_f1734_init_verbs[] = {
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_HP_EVENT},
- {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|ALC_DCVOL_EVENT},
-
- { }
-};
-
-/*
- * ASUS pin configuration:
- * HP/front = 0x14, surr = 0x15, clfe = 0x16, mic = 0x18, line = 0x1a
- */
-static const struct hda_verb alc880_pin_asus_init_verbs[] = {
- {0x10, AC_VERB_SET_CONNECT_SEL, 0x02},
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x12, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- { }
-};
-
-/* Enable GPIO mask and set output */
-#define alc880_gpio1_init_verbs alc_gpio1_init_verbs
-#define alc880_gpio2_init_verbs alc_gpio2_init_verbs
-#define alc880_gpio3_init_verbs alc_gpio3_init_verbs
-
-/* Clevo m520g init */
-static const struct hda_verb alc880_pin_clevo_init_verbs[] = {
- /* headphone output */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* line-out */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Line-in */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* CD */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Mic1 (rear panel) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Mic2 (front panel) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* headphone */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* change to EAPD mode */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
-
- { }
-};
-
-static const struct hda_verb alc880_pin_tcl_S700_init_verbs[] = {
- /* change to EAPD mode */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3060},
-
- /* Headphone output */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- /* Front output*/
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- /* Line In pin widget for input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Mic1 (rear panel) pin widget for input and vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-
- /* change to EAPD mode */
- {0x20, AC_VERB_SET_COEF_INDEX, 0x07},
- {0x20, AC_VERB_SET_PROC_COEF, 0x3070},
-
- { }
-};
-
-/*
- * LG m1 express dual
- *
- * Pin assignment:
- * Rear Line-In/Out (blue): 0x14
- * Build-in Mic-In: 0x15
- * Speaker-out: 0x17
- * HP-Out (green): 0x1b
- * Mic-In/Out (red): 0x19
- * SPDIF-Out: 0x1e
- */
-
-/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */
-static const hda_nid_t alc880_lg_dac_nids[3] = {
- 0x05, 0x02, 0x03
-};
-
-/* seems analog CD is not working */
-static const struct hda_input_mux alc880_lg_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x1 },
- { "Line", 0x5 },
- { "Internal Mic", 0x6 },
- },
-};
-
-/* 2,4,6 channel modes */
-static const struct hda_verb alc880_lg_ch2_init[] = {
- /* set line-in and mic-in to input */
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { }
-};
-
-static const struct hda_verb alc880_lg_ch4_init[] = {
- /* set line-in to out and mic-in to input */
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
- { }
-};
-
-static const struct hda_verb alc880_lg_ch6_init[] = {
- /* set line-in and mic-in to output */
- { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
- { }
-};
-
-static const struct hda_channel_mode alc880_lg_ch_modes[3] = {
- { 2, alc880_lg_ch2_init },
- { 4, alc880_lg_ch4_init },
- { 6, alc880_lg_ch6_init },
-};
-
-static const struct snd_kcontrol_new alc880_lg_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0f, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT),
- HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb alc880_lg_init_verbs[] = {
- /* set capture source to mic-in */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* mute all amp mixer inputs */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
- /* line-in to input */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* built-in mic */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* speaker-out */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* mic-in to input */
- {0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* HP-out */
- {0x13, AC_VERB_SET_CONNECT_SEL, 0x03},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* jack sense */
- {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT},
- { }
-};
-
-/* toggle speaker-output according to the hp-jack state */
-static void alc880_lg_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x1b;
- spec->autocfg.speaker_pins[0] = 0x17;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
-}
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list alc880_lg_loopbacks[] = {
- { 0x0b, HDA_INPUT, 1 },
- { 0x0b, HDA_INPUT, 6 },
- { 0x0b, HDA_INPUT, 7 },
- { } /* end */
-};
-#endif
-
-/*
- * Test configuration for debugging
- *
- * Almost all inputs/outputs are enabled. I/O pins can be configured via
- * enum controls.
- */
-#ifdef CONFIG_SND_DEBUG
-static const hda_nid_t alc880_test_dac_nids[4] = {
- 0x02, 0x03, 0x04, 0x05
-};
-
-static const struct hda_input_mux alc880_test_capture_source = {
- .num_items = 7,
- .items = {
- { "In-1", 0x0 },
- { "In-2", 0x1 },
- { "In-3", 0x2 },
- { "In-4", 0x3 },
- { "CD", 0x4 },
- { "Front", 0x5 },
- { "Surround", 0x6 },
- },
-};
-
-static const struct hda_channel_mode alc880_test_modes[4] = {
- { 2, NULL },
- { 4, NULL },
- { 6, NULL },
- { 8, NULL },
-};
-
-static int alc_test_pin_ctl_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- static const char * const texts[] = {
- "N/A", "Line Out", "HP Out",
- "In Hi-Z", "In 50%", "In Grd", "In 80%", "In 100%"
- };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 8;
- if (uinfo->value.enumerated.item >= 8)
- uinfo->value.enumerated.item = 7;
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
- return 0;
-}
-
-static int alc_test_pin_ctl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
- unsigned int pin_ctl, item = 0;
-
- pin_ctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
- if (pin_ctl & AC_PINCTL_OUT_EN) {
- if (pin_ctl & AC_PINCTL_HP_EN)
- item = 2;
- else
- item = 1;
- } else if (pin_ctl & AC_PINCTL_IN_EN) {
- switch (pin_ctl & AC_PINCTL_VREFEN) {
- case AC_PINCTL_VREF_HIZ: item = 3; break;
- case AC_PINCTL_VREF_50: item = 4; break;
- case AC_PINCTL_VREF_GRD: item = 5; break;
- case AC_PINCTL_VREF_80: item = 6; break;
- case AC_PINCTL_VREF_100: item = 7; break;
- }
- }
- ucontrol->value.enumerated.item[0] = item;
- return 0;
-}
-
-static int alc_test_pin_ctl_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
- static const unsigned int ctls[] = {
- 0, AC_PINCTL_OUT_EN, AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_50,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_80,
- AC_PINCTL_IN_EN | AC_PINCTL_VREF_100,
- };
- unsigned int old_ctl, new_ctl;
-
- old_ctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
- new_ctl = ctls[ucontrol->value.enumerated.item[0]];
- if (old_ctl != new_ctl) {
- int val;
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- new_ctl);
- val = ucontrol->value.enumerated.item[0] >= 3 ?
- HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, val);
- return 1;
- }
- return 0;
-}
-
-static int alc_test_pin_src_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- static const char * const texts[] = {
- "Front", "Surround", "CLFE", "Side"
- };
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = 4;
- if (uinfo->value.enumerated.item >= 4)
- uinfo->value.enumerated.item = 3;
- strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
- return 0;
-}
-
-static int alc_test_pin_src_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
- unsigned int sel;
-
- sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0);
- ucontrol->value.enumerated.item[0] = sel & 3;
- return 0;
-}
-
-static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = (hda_nid_t)kcontrol->private_value;
- unsigned int sel;
-
- sel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0) & 3;
- if (ucontrol->value.enumerated.item[0] != sel) {
- sel = ucontrol->value.enumerated.item[0] & 3;
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_CONNECT_SEL, sel);
- return 1;
- }
- return 0;
-}
-
-#define PIN_CTL_TEST(xname,nid) { \
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = xname, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_test_pin_ctl_info, \
- .get = alc_test_pin_ctl_get, \
- .put = alc_test_pin_ctl_put, \
- .private_value = nid \
- }
-
-#define PIN_SRC_TEST(xname,nid) { \
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
- .name = xname, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_test_pin_src_info, \
- .get = alc_test_pin_src_get, \
- .put = alc_test_pin_src_put, \
- .private_value = nid \
- }
-
-static const struct snd_kcontrol_new alc880_test_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- 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),
- 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),
- PIN_CTL_TEST("Side Pin Mode", 0x17),
- PIN_CTL_TEST("In-1 Pin Mode", 0x18),
- PIN_CTL_TEST("In-2 Pin Mode", 0x19),
- PIN_CTL_TEST("In-3 Pin Mode", 0x1a),
- PIN_CTL_TEST("In-4 Pin Mode", 0x1b),
- PIN_SRC_TEST("In-1 Pin Source", 0x18),
- PIN_SRC_TEST("In-2 Pin Source", 0x19),
- PIN_SRC_TEST("In-3 Pin Source", 0x1a),
- PIN_SRC_TEST("In-4 Pin Source", 0x1b),
- HDA_CODEC_VOLUME("In-1 Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_MUTE("In-1 Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("In-2 Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_MUTE("In-2 Playback Switch", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("In-3 Playback Volume", 0x0b, 0x2, HDA_INPUT),
- HDA_CODEC_MUTE("In-3 Playback Switch", 0x0b, 0x2, HDA_INPUT),
- HDA_CODEC_VOLUME("In-4 Playback Volume", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_MUTE("In-4 Playback Switch", 0x0b, 0x3, HDA_INPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x4, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x4, HDA_INPUT),
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb alc880_test_init_verbs[] = {
- /* Unmute inputs of 0x0c - 0x0f */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /* Vol output for 0x0c-0x0f */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- /* Set output pins 0x14-0x17 */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- /* Unmute output pins 0x14-0x17 */
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Set input pins 0x18-0x1c */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- /* Mute input pins 0x18-0x1b */
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* ADC set up */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Analog input/passthru */
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- { }
-};
-#endif
-
-/*
- */
-
-static const char * const alc880_models[ALC880_MODEL_LAST] = {
- [ALC880_3ST] = "3stack",
- [ALC880_TCL_S700] = "tcl",
- [ALC880_3ST_DIG] = "3stack-digout",
- [ALC880_CLEVO] = "clevo",
- [ALC880_5ST] = "5stack",
- [ALC880_5ST_DIG] = "5stack-digout",
- [ALC880_W810] = "w810",
- [ALC880_Z71V] = "z71v",
- [ALC880_6ST] = "6stack",
- [ALC880_6ST_DIG] = "6stack-digout",
- [ALC880_ASUS] = "asus",
- [ALC880_ASUS_W1V] = "asus-w1v",
- [ALC880_ASUS_DIG] = "asus-dig",
- [ALC880_ASUS_DIG2] = "asus-dig2",
- [ALC880_UNIWILL_DIG] = "uniwill",
- [ALC880_UNIWILL_P53] = "uniwill-p53",
- [ALC880_FUJITSU] = "fujitsu",
- [ALC880_F1734] = "F1734",
- [ALC880_LG] = "lg",
-#ifdef CONFIG_SND_DEBUG
- [ALC880_TEST] = "test",
-#endif
- [ALC880_AUTO] = "auto",
-};
-
-static const struct snd_pci_quirk alc880_cfg_tbl[] = {
- SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_W810),
- SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_6ST),
- SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST),
- SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V),
- SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x1113, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x1123, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x1173, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_Z71V),
- /* SND_PCI_QUIRK(0x1043, 0x1964, "ASUS", ALC880_ASUS_DIG), */
- SND_PCI_QUIRK(0x1043, 0x1973, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x814e, "ASUS P5GD1 w/SPDIF", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
- SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
- SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
- SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */
- SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST),
- SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST),
- SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST),
- SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_5ST),
- SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_5ST),
- SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x1558, 0x0520, "Clevo m520G", ALC880_CLEVO),
- SND_PCI_QUIRK(0x1558, 0x0660, "Clevo m655n", ALC880_CLEVO),
- SND_PCI_QUIRK(0x1558, 0x5401, "ASUS", ALC880_ASUS_DIG2),
- SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_UNIWILL_DIG),
- SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_F1734),
- SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_UNIWILL),
- SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_UNIWILL_P53),
- SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_W810),
- SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_F1734),
- SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FUJITSU),
- SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_F1734),
- SND_PCI_QUIRK(0x1734, 0x10b0, "Fujitsu", ALC880_FUJITSU),
- SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_LG),
- SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_LG),
- SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_LG),
- SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_TCL_S700),
- SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_6ST_DIG), /* broken BIOS */
- SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_6ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_3ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG),
- /* default Intel */
- SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST),
- SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG),
- SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG),
- {}
-};
-
-/*
- * ALC880 codec presets
- */
-static const struct alc_config_preset alc880_presets[] = {
- [ALC880_3ST] = {
- .mixers = { alc880_three_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_3stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_3ST_DIG] = {
- .mixers = { alc880_three_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_3stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_TCL_S700] = {
- .mixers = { alc880_tcl_s700_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_tcl_S700_init_verbs,
- alc880_gpio2_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .adc_nids = alc880_adc_nids_alt, /* FIXME: correct? */
- .num_adc_nids = 1, /* single ADC */
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
- .channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_5ST] = {
- .mixers = { alc880_three_stack_mixer,
- alc880_five_stack_mixer},
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_5stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
- .channel_mode = alc880_fivestack_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_5ST_DIG] = {
- .mixers = { alc880_three_stack_mixer,
- alc880_five_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_5stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes),
- .channel_mode = alc880_fivestack_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_6ST] = {
- .mixers = { alc880_six_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_6stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
- .dac_nids = alc880_6st_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
- .channel_mode = alc880_sixstack_modes,
- .input_mux = &alc880_6stack_capture_source,
- },
- [ALC880_6ST_DIG] = {
- .mixers = { alc880_six_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_6stack_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_6st_dac_nids),
- .dac_nids = alc880_6st_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_sixstack_modes),
- .channel_mode = alc880_sixstack_modes,
- .input_mux = &alc880_6stack_capture_source,
- },
- [ALC880_W810] = {
- .mixers = { alc880_w810_base_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_w810_init_verbs,
- alc880_gpio2_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_w810_dac_nids),
- .dac_nids = alc880_w810_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
- .channel_mode = alc880_w810_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_Z71V] = {
- .mixers = { alc880_z71v_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_z71v_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_z71v_dac_nids),
- .dac_nids = alc880_z71v_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
- .channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_F1734] = {
- .mixers = { alc880_f1734_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_f1734_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_f1734_dac_nids),
- .dac_nids = alc880_f1734_dac_nids,
- .hp_nid = 0x02,
- .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
- .channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_f1734_capture_source,
- .unsol_event = alc880_uniwill_p53_unsol_event,
- .setup = alc880_uniwill_p53_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC880_ASUS] = {
- .mixers = { alc880_asus_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_ASUS_DIG] = {
- .mixers = { alc880_asus_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_ASUS_DIG2] = {
- .mixers = { alc880_asus_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs,
- alc880_gpio2_init_verbs }, /* use GPIO2 */
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_ASUS_W1V] = {
- .mixers = { alc880_asus_mixer, alc880_asus_w1v_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_UNIWILL_DIG] = {
- .mixers = { alc880_asus_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_asus_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_asus_modes),
- .channel_mode = alc880_asus_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_UNIWILL] = {
- .mixers = { alc880_uniwill_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_uniwill_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- .unsol_event = alc880_uniwill_unsol_event,
- .setup = alc880_uniwill_setup,
- .init_hook = alc880_uniwill_init_hook,
- },
- [ALC880_UNIWILL_P53] = {
- .mixers = { alc880_uniwill_p53_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_uniwill_p53_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
- .dac_nids = alc880_asus_dac_nids,
- .num_channel_mode = ARRAY_SIZE(alc880_w810_modes),
- .channel_mode = alc880_threestack_modes,
- .input_mux = &alc880_capture_source,
- .unsol_event = alc880_uniwill_p53_unsol_event,
- .setup = alc880_uniwill_p53_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC880_FUJITSU] = {
- .mixers = { alc880_fujitsu_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_uniwill_p53_init_verbs,
- alc880_beep_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes),
- .channel_mode = alc880_2_jack_modes,
- .input_mux = &alc880_capture_source,
- .unsol_event = alc880_uniwill_p53_unsol_event,
- .setup = alc880_uniwill_p53_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC880_CLEVO] = {
- .mixers = { alc880_three_stack_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_pin_clevo_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_dac_nids),
- .dac_nids = alc880_dac_nids,
- .hp_nid = 0x03,
- .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
- .channel_mode = alc880_threestack_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_capture_source,
- },
- [ALC880_LG] = {
- .mixers = { alc880_lg_mixer },
- .init_verbs = { alc880_volume_init_verbs,
- alc880_lg_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids),
- .dac_nids = alc880_lg_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes),
- .channel_mode = alc880_lg_ch_modes,
- .need_dac_fix = 1,
- .input_mux = &alc880_lg_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc880_lg_setup,
- .init_hook = alc_hp_automute,
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- .loopbacks = alc880_lg_loopbacks,
-#endif
- },
-#ifdef CONFIG_SND_DEBUG
- [ALC880_TEST] = {
- .mixers = { alc880_test_mixer },
- .init_verbs = { alc880_test_init_verbs },
- .num_dacs = ARRAY_SIZE(alc880_test_dac_nids),
- .dac_nids = alc880_test_dac_nids,
- .dig_out_nid = ALC880_DIGOUT_NID,
- .num_channel_mode = ARRAY_SIZE(alc880_test_modes),
- .channel_mode = alc880_test_modes,
- .input_mux = &alc880_test_capture_source,
- },
-#endif
-};
-
diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c
deleted file mode 100644
index bdf0ed4ab3e2..000000000000
--- a/sound/pci/hda/alc882_quirks.c
+++ /dev/null
@@ -1,861 +0,0 @@
-/*
- * ALC882/ALC883/ALC888/ALC889 quirk models
- * included by patch_realtek.c
- */
-
-/* ALC882 models */
-enum {
- ALC882_AUTO,
- ALC885_MBA21,
- ALC885_MBP3,
- ALC885_MB5,
- ALC885_MACMINI3,
- ALC885_IMAC91,
- ALC889A_MB31,
- ALC882_MODEL_LAST,
-};
-
-#define ALC882_DIGOUT_NID 0x06
-#define ALC882_DIGIN_NID 0x0a
-#define ALC883_DIGOUT_NID ALC882_DIGOUT_NID
-#define ALC883_DIGIN_NID ALC882_DIGIN_NID
-#define ALC1200_DIGOUT_NID 0x10
-
-
-static const struct hda_channel_mode alc882_ch_modes[1] = {
- { 8, NULL }
-};
-
-/* DACs */
-static const hda_nid_t alc882_dac_nids[4] = {
- /* front, rear, clfe, rear_surr */
- 0x02, 0x03, 0x04, 0x05
-};
-#define alc883_dac_nids alc882_dac_nids
-
-/* ADCs */
-#define alc882_adc_nids alc880_adc_nids
-#define alc882_adc_nids_alt alc880_adc_nids_alt
-#define alc883_adc_nids alc882_adc_nids_alt
-
-static const hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
-#define alc883_capsrc_nids alc882_capsrc_nids_alt
-
-/* input MUX */
-/* FIXME: should be a matrix-type input source selection */
-
-static const struct hda_input_mux alc882_capture_source = {
- .num_items = 4,
- .items = {
- { "Mic", 0x0 },
- { "Front Mic", 0x1 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-#define alc883_capture_source alc882_capture_source
-
-static const struct hda_input_mux mb5_capture_source = {
- .num_items = 3,
- .items = {
- { "Mic", 0x1 },
- { "Line", 0x7 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux macmini3_capture_source = {
- .num_items = 2,
- .items = {
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux alc883_3stack_6ch_intel = {
- .num_items = 4,
- .items = {
- { "Mic", 0x1 },
- { "Front Mic", 0x0 },
- { "Line", 0x2 },
- { "CD", 0x4 },
- },
-};
-
-static const struct hda_input_mux alc889A_mb31_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x0 },
- /* Front Mic (0x01) unused */
- { "Line", 0x2 },
- /* Line 2 (0x03) unused */
- /* CD (0x04) unused? */
- },
-};
-
-static const struct hda_input_mux alc889A_imac91_capture_source = {
- .num_items = 2,
- .items = {
- { "Mic", 0x01 },
- { "Line", 0x2 }, /* Not sure! */
- },
-};
-
-/* Macbook Air 2,1 */
-
-static const struct hda_channel_mode alc885_mba21_ch_modes[1] = {
- { 2, NULL },
-};
-
-/*
- * macbook pro ALC885 can switch LineIn to LineOut without losing Mic
- */
-
-/*
- * 2ch mode
- */
-static const struct hda_verb alc885_mbp_ch2_init[] = {
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- { } /* end */
-};
-
-/*
- * 4ch mode
- */
-static const struct hda_verb alc885_mbp_ch4_init[] = {
- { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
- { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
- { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- { } /* end */
-};
-
-static const struct hda_channel_mode alc885_mbp_4ch_modes[2] = {
- { 2, alc885_mbp_ch2_init },
- { 4, alc885_mbp_ch4_init },
-};
-
-/*
- * 2ch
- * Speakers/Woofer/HP = Front
- * LineIn = Input
- */
-static const struct hda_verb alc885_mb5_ch2_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- { } /* end */
-};
-
-/*
- * 6ch mode
- * Speakers/HP = Front
- * Woofer = LFE
- * LineIn = Surround
- */
-static const struct hda_verb alc885_mb5_ch6_init[] = {
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- { } /* end */
-};
-
-static const struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
- { 2, alc885_mb5_ch2_init },
- { 6, alc885_mb5_ch6_init },
-};
-
-#define alc885_macmini3_6ch_modes alc885_mb5_6ch_modes
-
-/* Macbook Air 2,1 same control for HP and internal Speaker */
-
-static const struct snd_kcontrol_new alc885_mba21_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_OUTPUT),
- { }
-};
-
-
-static const struct snd_kcontrol_new alc885_mbp3_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Headphone Playback Switch", 0x0e, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc885_mb5_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_MUTE ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x19, 0x00, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc885_macmini3_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x00, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc885_imac91_mixer[] = {
- HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
- { } /* end */
-};
-
-
-static const struct snd_kcontrol_new alc882_chmode_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb alc882_base_init_verbs[] = {
- /* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Rear mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* CLFE mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Side mixer */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
- /* Front Pin: output 0 (0x0c) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* Rear Pin: output 1 (0x0d) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
- /* CLFE Pin: output 2 (0x0e) */
- {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* Side Pin: output 3 (0x0f) */
- {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
- /* Mic (rear) pin: input vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin: input */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line-2 In: Headphone output (output 0 - 0x0c) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* CD pin widget for input */
- {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- /* ADC2: mute amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* ADC3: mute amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- { }
-};
-
-#define alc883_init_verbs alc882_base_init_verbs
-
-/* Macbook 5,1 */
-static const struct hda_verb alc885_mb5_init_verbs[] = {
- /* DACs */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Front mixer */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Surround mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* LFE mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* HP mixer */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Front Pin (0x0c) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* LFE Pin (0x0e) */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* HP Pin (0x0f) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
- { }
-};
-
-/* Macmini 3,1 */
-static const struct hda_verb alc885_macmini3_init_verbs[] = {
- /* DACs */
- {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- /* Front mixer */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Surround mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* LFE mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* HP mixer */
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Front Pin (0x0c) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* LFE Pin (0x0e) */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x01},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02},
- /* HP Pin (0x0f) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x03},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
- /* Line In pin */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- { }
-};
-
-
-static const struct hda_verb alc885_mba21_init_verbs[] = {
- /*Internal and HP Speaker Mixer*/
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
- /*Internal Speaker Pin (0x0c)*/
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP Pin: output 0 (0x0e) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
- /* Line in (is hp when jack connected)*/
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-
- { }
- };
-
-
-/* Macbook Pro rev3 */
-static const struct hda_verb alc885_mbp3_init_verbs[] = {
- /* Front mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Rear mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* HP mixer */
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Front Pin: output 0 (0x0c) */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP Pin: output 0 (0x0e) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x02},
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
- /* Mic (rear) pin: input vref at 80% */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Line In pin: use output 1 when in LineOut mode */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-
- /* FIXME: use matrix-type input source selection */
- /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
- /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Input mixer2 */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* Input mixer3 */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* ADC1: mute amp left and right */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* ADC2: mute amp left and right */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* ADC3: mute amp left and right */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
- { }
-};
-
-/* iMac 9,1 */
-static const struct hda_verb alc885_imac91_init_verbs[] = {
- /* Internal Speaker Pin (0x0c) */
- {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
- {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* HP Pin: Rear */
- {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
- {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
- {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
- {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC_HP_EVENT | AC_USRSP_EN)},
- /* Line in Rear */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
- {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Front Mic pin: input vref at 80% */
- {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
- {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
- /* Rear mixer */
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
- /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
- {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
- /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
- {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
- {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
- /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
- {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
- {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
- { }
-};
-
-/* Toggle speaker-output according to the hp-jack state */
-static void alc885_imac24_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x18;
- spec->autocfg.speaker_pins[1] = 0x1a;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
-}
-
-#define alc885_mb5_setup alc885_imac24_setup
-#define alc885_macmini3_setup alc885_imac24_setup
-
-/* Macbook Air 2,1 */
-static void alc885_mba21_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x18;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
-}
-
-
-
-static void alc885_mbp3_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x15;
- spec->autocfg.speaker_pins[0] = 0x14;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
-}
-
-static void alc885_imac91_setup(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
-
- spec->autocfg.hp_pins[0] = 0x14;
- spec->autocfg.speaker_pins[0] = 0x18;
- spec->autocfg.speaker_pins[1] = 0x1a;
- alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP);
-}
-
-/* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
-static const struct hda_verb alc889A_mb31_ch2_init[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
- { } /* end */
-};
-
-/* 4ch mode (Speaker:front, Subwoofer:CLFE, Line:CLFE, Headphones:front) */
-static const struct hda_verb alc889A_mb31_ch4_init[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP as front */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
- { } /* end */
-};
-
-/* 5ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:rear) */
-static const struct hda_verb alc889A_mb31_ch5_init[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as rear */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Subwoofer on */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Line as input */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Line off */
- { } /* end */
-};
-
-/* 6ch mode (Speaker:front, Subwoofer:off, Line:CLFE, Headphones:Rear) */
-static const struct hda_verb alc889A_mb31_ch6_init[] = {
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* HP as front */
- {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Subwoofer off */
- {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Line as output */
- {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Line on */
- { } /* end */
-};
-
-static const struct hda_channel_mode alc889A_mb31_6ch_modes[4] = {
- { 2, alc889A_mb31_ch2_init },
- { 4, alc889A_mb31_ch4_init },
- { 5, alc889A_mb31_ch5_init },
- { 6, alc889A_mb31_ch6_init },
-};
-
-static const struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
- HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
- HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
- HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x19, 0, HDA_INPUT),
- HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc889A_mb31_mixer[] = {
- /* Output mixers */
- HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE("Front Playback Switch", 0x0c, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x00,
- HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 0x02, HDA_INPUT),
- HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x00, HDA_OUTPUT),
- HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 0x02, HDA_INPUT),
- /* Output switches */
- HDA_CODEC_MUTE("Enable Speaker", 0x14, 0x00, HDA_OUTPUT),
- HDA_CODEC_MUTE("Enable Headphones", 0x15, 0x00, HDA_OUTPUT),
- HDA_CODEC_MUTE_MONO("Enable LFE", 0x16, 2, 0x00, HDA_OUTPUT),
- /* Boost mixers */
- HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Boost Volume", 0x1a, 0x00, HDA_INPUT),
- /* Input mixers */
- HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
- HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
- HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
- { } /* end */
-};
-
-static const struct snd_kcontrol_new alc883_chmode_mixer[] = {
- {
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Channel Mode",
- .info = alc_ch_mode_info,
- .get = alc_ch_mode_get,
- .put = alc_ch_mode_put,
- },
- { } /* end */
-};
-
-static const struct hda_verb alc889A_mb31_verbs[] = {
- /* Init rear pin (used as headphone output) */
- {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc4}, /* Apple Headphones */
- {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Connect to front */
- {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC_HP_EVENT | AC_USRSP_EN},
- /* Init line pin (used as output in 4ch and 6ch mode) */
- {0x1a, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Connect to CLFE */
- /* Init line 2 pin (used as headphone out by default) */
- {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Use as input */
- {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Mute output */
- { } /* end */
-};
-
-/* Mute speakers according to the headphone jack state */
-static void alc889A_mb31_automute(struct hda_codec *codec)
-{
- unsigned int present;
-
- /* Mute only in 2ch or 4ch mode */
- if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0)
- == 0x00) {
- present = snd_hda_jack_detect(codec, 0x15);
- snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
- }
-}
-
-static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
-{
- if ((res >> 26) == ALC_HP_EVENT)
- alc889A_mb31_automute(codec);
-}
-
-/*
- * configuration and preset
- */
-static const char * const alc882_models[ALC882_MODEL_LAST] = {
- [ALC885_MB5] = "mb5",
- [ALC885_MACMINI3] = "macmini3",
- [ALC885_MBA21] = "mba21",
- [ALC885_MBP3] = "mbp3",
- [ALC885_IMAC91] = "imac91",
- [ALC889A_MB31] = "mb31",
- [ALC882_AUTO] = "auto",
-};
-
-/* codec SSID table for Intel Mac */
-static const struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
- SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31),
- SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21),
- SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
- SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
- SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91),
- SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
- SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5),
- /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
- * so apparently no perfect solution yet
- */
- SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
- SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC885_MB5),
- SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC885_MACMINI3),
- {} /* terminator */
-};
-
-static const struct alc_config_preset alc882_presets[] = {
- [ALC885_MBA21] = {
- .mixers = { alc885_mba21_mixer },
- .init_verbs = { alc885_mba21_init_verbs, alc880_gpio1_init_verbs },
- .num_dacs = 2,
- .dac_nids = alc882_dac_nids,
- .channel_mode = alc885_mba21_ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
- .input_mux = &alc882_capture_source,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_mba21_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC885_MBP3] = {
- .mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
- .init_verbs = { alc885_mbp3_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = 2,
- .dac_nids = alc882_dac_nids,
- .hp_nid = 0x04,
- .channel_mode = alc885_mbp_4ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
- .input_mux = &alc882_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_mbp3_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC885_MB5] = {
- .mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
- .init_verbs = { alc885_mb5_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .channel_mode = alc885_mb5_6ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
- .input_mux = &mb5_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_mb5_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC885_MACMINI3] = {
- .mixers = { alc885_macmini3_mixer, alc882_chmode_mixer },
- .init_verbs = { alc885_macmini3_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .channel_mode = alc885_macmini3_6ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_macmini3_6ch_modes),
- .input_mux = &macmini3_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_macmini3_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC885_IMAC91] = {
- .mixers = {alc885_imac91_mixer},
- .init_verbs = { alc885_imac91_init_verbs,
- alc880_gpio1_init_verbs },
- .num_dacs = ARRAY_SIZE(alc882_dac_nids),
- .dac_nids = alc882_dac_nids,
- .channel_mode = alc885_mba21_ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
- .input_mux = &alc889A_imac91_capture_source,
- .dig_out_nid = ALC882_DIGOUT_NID,
- .dig_in_nid = ALC882_DIGIN_NID,
- .unsol_event = alc_sku_unsol_event,
- .setup = alc885_imac91_setup,
- .init_hook = alc_hp_automute,
- },
- [ALC889A_MB31] = {
- .mixers = { alc889A_mb31_mixer, alc883_chmode_mixer},
- .init_verbs = { alc883_init_verbs, alc889A_mb31_verbs,
- alc880_gpio1_init_verbs },
- .adc_nids = alc883_adc_nids,
- .num_adc_nids = ARRAY_SIZE(alc883_adc_nids),
- .capsrc_nids = alc883_capsrc_nids,
- .dac_nids = alc883_dac_nids,
- .num_dacs = ARRAY_SIZE(alc883_dac_nids),
- .channel_mode = alc889A_mb31_6ch_modes,
- .num_channel_mode = ARRAY_SIZE(alc889A_mb31_6ch_modes),
- .input_mux = &alc889A_mb31_capture_source,
- .dig_out_nid = ALC883_DIGOUT_NID,
- .unsol_event = alc889A_mb31_unsol_event,
- .init_hook = alc889A_mb31_automute,
- },
-};
-
-
diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c
deleted file mode 100644
index a18952ed4311..000000000000
--- a/sound/pci/hda/alc_quirks.c
+++ /dev/null
@@ -1,480 +0,0 @@
-/*
- * Common codes for Realtek codec quirks
- * included by patch_realtek.c
- */
-
-/*
- * configuration template - to be copied to the spec instance
- */
-struct alc_config_preset {
- const struct snd_kcontrol_new *mixers[5]; /* should be identical size
- * with spec
- */
- const struct snd_kcontrol_new *cap_mixer; /* capture mixer */
- const struct hda_verb *init_verbs[5];
- unsigned int num_dacs;
- const hda_nid_t *dac_nids;
- hda_nid_t dig_out_nid; /* optional */
- hda_nid_t hp_nid; /* optional */
- const hda_nid_t *slave_dig_outs;
- unsigned int num_adc_nids;
- const hda_nid_t *adc_nids;
- const hda_nid_t *capsrc_nids;
- hda_nid_t dig_in_nid;
- unsigned int num_channel_mode;
- const struct hda_channel_mode *channel_mode;
- int need_dac_fix;
- int const_channel_count;
- unsigned int num_mux_defs;
- const struct hda_input_mux *input_mux;
- void (*unsol_event)(struct hda_codec *, unsigned int);
- void (*setup)(struct hda_codec *);
- void (*init_hook)(struct hda_codec *);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- const struct hda_amp_list *loopbacks;
- void (*power_hook)(struct hda_codec *codec);
-#endif
-};
-
-/*
- * channel mode setting
- */
-static int alc_ch_mode_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- return snd_hda_ch_mode_info(codec, uinfo, spec->channel_mode,
- spec->num_channel_mode);
-}
-
-static int alc_ch_mode_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- return snd_hda_ch_mode_get(codec, ucontrol, spec->channel_mode,
- spec->num_channel_mode,
- spec->ext_channel_count);
-}
-
-static int alc_ch_mode_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- struct alc_spec *spec = codec->spec;
- int err = snd_hda_ch_mode_put(codec, ucontrol, spec->channel_mode,
- spec->num_channel_mode,
- &spec->ext_channel_count);
- if (err >= 0 && !spec->const_channel_count) {
- spec->multiout.max_channels = spec->ext_channel_count;
- if (spec->need_dac_fix)
- spec->multiout.num_dacs = spec->multiout.max_channels / 2;
- }
- return err;
-}
-
-/*
- * Control the mode of pin widget settings via the mixer. "pc" is used
- * instead of "%" to avoid consequences of accidentally treating the % as
- * being part of a format specifier. Maximum allowed length of a value is
- * 63 characters plus NULL terminator.
- *
- * Note: some retasking pin complexes seem to ignore requests for input
- * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these
- * are requested. Therefore order this list so that this behaviour will not
- * cause problems when mixer clients move through the enum sequentially.
- * NIDs 0x0f and 0x10 have been observed to have this behaviour as of
- * March 2006.
- */
-static const char * const alc_pin_mode_names[] = {
- "Mic 50pc bias", "Mic 80pc bias",
- "Line in", "Line out", "Headphone out",
-};
-static const unsigned char alc_pin_mode_values[] = {
- PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP,
-};
-/* The control can present all 5 options, or it can limit the options based
- * in the pin being assumed to be exclusively an input or an output pin. In
- * addition, "input" pins may or may not process the mic bias option
- * depending on actual widget capability (NIDs 0x0f and 0x10 don't seem to
- * accept requests for bias as of chip versions up to March 2006) and/or
- * wiring in the computer.
- */
-#define ALC_PIN_DIR_IN 0x00
-#define ALC_PIN_DIR_OUT 0x01
-#define ALC_PIN_DIR_INOUT 0x02
-#define ALC_PIN_DIR_IN_NOMICBIAS 0x03
-#define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04
-
-/* Info about the pin modes supported by the different pin direction modes.
- * For each direction the minimum and maximum values are given.
- */
-static const signed char alc_pin_mode_dir_info[5][2] = {
- { 0, 2 }, /* ALC_PIN_DIR_IN */
- { 3, 4 }, /* ALC_PIN_DIR_OUT */
- { 0, 4 }, /* ALC_PIN_DIR_INOUT */
- { 2, 2 }, /* ALC_PIN_DIR_IN_NOMICBIAS */
- { 2, 4 }, /* ALC_PIN_DIR_INOUT_NOMICBIAS */
-};
-#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0])
-#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1])
-#define alc_pin_mode_n_items(_dir) \
- (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1)
-
-static int alc_pin_mode_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- unsigned int item_num = uinfo->value.enumerated.item;
- unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
-
- uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
- uinfo->count = 1;
- uinfo->value.enumerated.items = alc_pin_mode_n_items(dir);
-
- if (item_num<alc_pin_mode_min(dir) || item_num>alc_pin_mode_max(dir))
- item_num = alc_pin_mode_min(dir);
- strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]);
- return 0;
-}
-
-static int alc_pin_mode_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- unsigned int i;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL,
- 0x00);
-
- /* Find enumerated value for current pinctl setting */
- i = alc_pin_mode_min(dir);
- while (i <= alc_pin_mode_max(dir) && alc_pin_mode_values[i] != pinctl)
- i++;
- *valp = i <= alc_pin_mode_max(dir) ? i: alc_pin_mode_min(dir);
- return 0;
-}
-
-static int alc_pin_mode_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- signed int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char dir = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int pinctl = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_PIN_WIDGET_CONTROL,
- 0x00);
-
- if (val < alc_pin_mode_min(dir) || val > alc_pin_mode_max(dir))
- val = alc_pin_mode_min(dir);
-
- change = pinctl != alc_pin_mode_values[val];
- if (change) {
- /* Set pin mode to that requested */
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- alc_pin_mode_values[val]);
-
- /* Also enable the retasking pin's input/output as required
- * for the requested pin mode. Enum values of 2 or less are
- * input modes.
- *
- * Dynamically switching the input/output buffers probably
- * reduces noise slightly (particularly on input) so we'll
- * do it. However, having both input and output buffers
- * enabled simultaneously doesn't seem to be problematic if
- * this turns out to be necessary in the future.
- */
- if (val <= 2) {
- snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
- HDA_AMP_MUTE, 0);
- } else {
- snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, 0,
- HDA_AMP_MUTE, HDA_AMP_MUTE);
- snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
- HDA_AMP_MUTE, 0);
- }
- }
- return change;
-}
-
-#define ALC_PIN_MODE(xname, nid, dir) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_pin_mode_info, \
- .get = alc_pin_mode_get, \
- .put = alc_pin_mode_put, \
- .private_value = nid | (dir<<16) }
-
-/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged
- * together using a mask with more than one bit set. This control is
- * currently used only by the ALC260 test model. At this stage they are not
- * needed for any "production" models.
- */
-#ifdef CONFIG_SND_DEBUG
-#define alc_gpio_data_info snd_ctl_boolean_mono_info
-
-static int alc_gpio_data_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_GPIO_DATA, 0x00);
-
- *valp = (val & mask) != 0;
- return 0;
-}
-static int alc_gpio_data_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- signed int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int gpio_data = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_GPIO_DATA,
- 0x00);
-
- /* Set/unset the masked GPIO bit(s) as needed */
- change = (val == 0 ? 0 : mask) != (gpio_data & mask);
- if (val == 0)
- gpio_data &= ~mask;
- else
- gpio_data |= mask;
- snd_hda_codec_write_cache(codec, nid, 0,
- AC_VERB_SET_GPIO_DATA, gpio_data);
-
- return change;
-}
-#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_gpio_data_info, \
- .get = alc_gpio_data_get, \
- .put = alc_gpio_data_put, \
- .private_value = nid | (mask<<16) }
-#endif /* CONFIG_SND_DEBUG */
-
-/* A switch control to allow the enabling of the digital IO pins on the
- * ALC260. This is incredibly simplistic; the intention of this control is
- * to provide something in the test model allowing digital outputs to be
- * identified if present. If models are found which can utilise these
- * outputs a more complete mixer control can be devised for those models if
- * necessary.
- */
-#ifdef CONFIG_SND_DEBUG
-#define alc_spdif_ctrl_info snd_ctl_boolean_mono_info
-
-static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_DIGI_CONVERT_1, 0x00);
-
- *valp = (val & mask) != 0;
- return 0;
-}
-static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- signed int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_DIGI_CONVERT_1,
- 0x00);
-
- /* Set/unset the masked control bit(s) as needed */
- change = (val == 0 ? 0 : mask) != (ctrl_data & mask);
- if (val==0)
- ctrl_data &= ~mask;
- else
- ctrl_data |= mask;
- snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1,
- ctrl_data);
-
- return change;
-}
-#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_spdif_ctrl_info, \
- .get = alc_spdif_ctrl_get, \
- .put = alc_spdif_ctrl_put, \
- .private_value = nid | (mask<<16) }
-#endif /* CONFIG_SND_DEBUG */
-
-/* A switch control to allow the enabling EAPD digital outputs on the ALC26x.
- * Again, this is only used in the ALC26x test models to help identify when
- * the EAPD line must be asserted for features to work.
- */
-#ifdef CONFIG_SND_DEBUG
-#define alc_eapd_ctrl_info snd_ctl_boolean_mono_info
-
-static int alc_eapd_ctrl_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long *valp = ucontrol->value.integer.value;
- unsigned int val = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_EAPD_BTLENABLE, 0x00);
-
- *valp = (val & mask) != 0;
- return 0;
-}
-
-static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- int change;
- struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
- hda_nid_t nid = kcontrol->private_value & 0xffff;
- unsigned char mask = (kcontrol->private_value >> 16) & 0xff;
- long val = *ucontrol->value.integer.value;
- unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_EAPD_BTLENABLE,
- 0x00);
-
- /* Set/unset the masked control bit(s) as needed */
- change = (!val ? 0 : mask) != (ctrl_data & mask);
- if (!val)
- ctrl_data &= ~mask;
- else
- ctrl_data |= mask;
- snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE,
- ctrl_data);
-
- return change;
-}
-
-#define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \
- { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \
- .subdevice = HDA_SUBDEV_NID_FLAG | nid, \
- .info = alc_eapd_ctrl_info, \
- .get = alc_eapd_ctrl_get, \
- .put = alc_eapd_ctrl_put, \
- .private_value = nid | (mask<<16) }
-#endif /* CONFIG_SND_DEBUG */
-
-static void alc_fixup_autocfg_pin_nums(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct auto_pin_cfg *cfg = &spec->autocfg;
-
- if (!cfg->line_outs) {
- while (cfg->line_outs < AUTO_CFG_MAX_OUTS &&
- cfg->line_out_pins[cfg->line_outs])
- cfg->line_outs++;
- }
- if (!cfg->speaker_outs) {
- while (cfg->speaker_outs < AUTO_CFG_MAX_OUTS &&
- cfg->speaker_pins[cfg->speaker_outs])
- cfg->speaker_outs++;
- }
- if (!cfg->hp_outs) {
- while (cfg->hp_outs < AUTO_CFG_MAX_OUTS &&
- cfg->hp_pins[cfg->hp_outs])
- cfg->hp_outs++;
- }
-}
-
-/*
- * set up from the preset table
- */
-static void setup_preset(struct hda_codec *codec,
- const struct alc_config_preset *preset)
-{
- struct alc_spec *spec = codec->spec;
- int i;
-
- for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
- add_mixer(spec, preset->mixers[i]);
- spec->cap_mixer = preset->cap_mixer;
- for (i = 0; i < ARRAY_SIZE(preset->init_verbs) && preset->init_verbs[i];
- i++)
- add_verb(spec, preset->init_verbs[i]);
-
- spec->channel_mode = preset->channel_mode;
- spec->num_channel_mode = preset->num_channel_mode;
- spec->need_dac_fix = preset->need_dac_fix;
- spec->const_channel_count = preset->const_channel_count;
-
- if (preset->const_channel_count)
- spec->multiout.max_channels = preset->const_channel_count;
- else
- spec->multiout.max_channels = spec->channel_mode[0].channels;
- spec->ext_channel_count = spec->channel_mode[0].channels;
-
- spec->multiout.num_dacs = preset->num_dacs;
- spec->multiout.dac_nids = preset->dac_nids;
- spec->multiout.dig_out_nid = preset->dig_out_nid;
- spec->multiout.slave_dig_outs = preset->slave_dig_outs;
- spec->multiout.hp_nid = preset->hp_nid;
-
- spec->num_mux_defs = preset->num_mux_defs;
- if (!spec->num_mux_defs)
- spec->num_mux_defs = 1;
- spec->input_mux = preset->input_mux;
-
- spec->num_adc_nids = preset->num_adc_nids;
- spec->adc_nids = preset->adc_nids;
- spec->capsrc_nids = preset->capsrc_nids;
- spec->dig_in_nid = preset->dig_in_nid;
-
- spec->unsol_event = preset->unsol_event;
- spec->init_hook = preset->init_hook;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- spec->power_hook = preset->power_hook;
- spec->loopback.amplist = preset->loopbacks;
-#endif
-
- if (preset->setup)
- preset->setup(codec);
-
- alc_fixup_autocfg_pin_nums(codec);
-}
-
-static void alc_simple_setup_automute(struct alc_spec *spec, int mode)
-{
- int lo_pin = spec->autocfg.line_out_pins[0];
-
- if (lo_pin == spec->autocfg.speaker_pins[0] ||
- lo_pin == spec->autocfg.hp_pins[0])
- lo_pin = 0;
- spec->automute_mode = mode;
- spec->detect_hp = !!spec->autocfg.hp_pins[0];
- spec->detect_lo = !!lo_pin;
- spec->automute_lo = spec->automute_lo_possible = !!lo_pin;
- spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0];
-}
-
-/* auto-toggle front mic */
-static void alc88x_simple_mic_automute(struct hda_codec *codec)
-{
- unsigned int present;
- unsigned char bits;
-
- present = snd_hda_jack_detect(codec, 0x18);
- bits = present ? HDA_AMP_MUTE : 0;
- snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
-}
-
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 4df72c0e8c37..7a8fcc4c15f8 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -19,6 +19,7 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
+#include <linux/mm.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/slab.h>
@@ -1447,7 +1448,7 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid,
for (i = 0; i < c->cvt_setups.used; i++) {
p = snd_array_elem(&c->cvt_setups, i);
if (!p->active && p->stream_tag == stream_tag &&
- get_wcaps_type(get_wcaps(codec, p->nid)) == type)
+ get_wcaps_type(get_wcaps(c, p->nid)) == type)
p->dirty = 1;
}
}
@@ -1759,7 +1760,11 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT;
parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT;
parm |= index << AC_AMP_SET_INDEX_SHIFT;
- parm |= val;
+ if ((val & HDA_AMP_MUTE) && !(info->amp_caps & AC_AMPCAP_MUTE) &&
+ (info->amp_caps & AC_AMPCAP_MIN_MUTE))
+ ; /* set the zero value as a fake mute */
+ else
+ parm |= val;
snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
info->vol[ch] = val;
}
@@ -2026,7 +2031,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
val1 += ofs;
val1 = ((int)val1) * ((int)val2);
- if (min_mute)
+ if (min_mute || (caps & AC_AMPCAP_MIN_MUTE))
val2 |= TLV_DB_SCALE_MUTE;
if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
return -EFAULT;
@@ -2300,7 +2305,7 @@ typedef int (*map_slave_func_t)(void *, struct snd_kcontrol *);
/* apply the function to all matching slave ctls in the mixer list */
static int map_slaves(struct hda_codec *codec, const char * const *slaves,
- map_slave_func_t func, void *data)
+ const char *suffix, map_slave_func_t func, void *data)
{
struct hda_nid_item *items;
const char * const *s;
@@ -2313,7 +2318,14 @@ static int map_slaves(struct hda_codec *codec, const char * const *slaves,
sctl->id.iface != SNDRV_CTL_ELEM_IFACE_MIXER)
continue;
for (s = slaves; *s; s++) {
- if (!strcmp(sctl->id.name, *s)) {
+ char tmpname[sizeof(sctl->id.name)];
+ const char *name = *s;
+ if (suffix) {
+ snprintf(tmpname, sizeof(tmpname), "%s %s",
+ name, suffix);
+ name = tmpname;
+ }
+ if (!strcmp(sctl->id.name, name)) {
err = func(data, sctl);
if (err)
return err;
@@ -2329,12 +2341,65 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl)
return 1;
}
+/* guess the value corresponding to 0dB */
+static int get_kctl_0dB_offset(struct snd_kcontrol *kctl)
+{
+ int _tlv[4];
+ const int *tlv = NULL;
+ int val = -1;
+
+ if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) {
+ /* FIXME: set_fs() hack for obtaining user-space TLV data */
+ mm_segment_t fs = get_fs();
+ set_fs(get_ds());
+ if (!kctl->tlv.c(kctl, 0, sizeof(_tlv), _tlv))
+ tlv = _tlv;
+ set_fs(fs);
+ } else if (kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_TLV_READ)
+ tlv = kctl->tlv.p;
+ if (tlv && tlv[0] == SNDRV_CTL_TLVT_DB_SCALE)
+ val = -tlv[2] / tlv[3];
+ return val;
+}
+
+/* call kctl->put with the given value(s) */
+static int put_kctl_with_value(struct snd_kcontrol *kctl, int val)
+{
+ struct snd_ctl_elem_value *ucontrol;
+ ucontrol = kzalloc(sizeof(*ucontrol), GFP_KERNEL);
+ if (!ucontrol)
+ return -ENOMEM;
+ ucontrol->value.integer.value[0] = val;
+ ucontrol->value.integer.value[1] = val;
+ kctl->put(kctl, ucontrol);
+ kfree(ucontrol);
+ return 0;
+}
+
+/* initialize the slave volume with 0dB */
+static int init_slave_0dB(void *data, struct snd_kcontrol *slave)
+{
+ int offset = get_kctl_0dB_offset(slave);
+ if (offset > 0)
+ put_kctl_with_value(slave, offset);
+ return 0;
+}
+
+/* unmute the slave */
+static int init_slave_unmute(void *data, struct snd_kcontrol *slave)
+{
+ return put_kctl_with_value(slave, 1);
+}
+
/**
* snd_hda_add_vmaster - create a virtual master control and add slaves
* @codec: HD-audio codec
* @name: vmaster control name
* @tlv: TLV data (optional)
* @slaves: slave control names (optional)
+ * @suffix: suffix string to each slave name (optional)
+ * @init_slave_vol: initialize slaves to unmute/0dB
+ * @ctl_ret: store the vmaster kcontrol in return
*
* Create a virtual master control with the given name. The TLV data
* must be either NULL or a valid data.
@@ -2345,13 +2410,18 @@ static int check_slave_present(void *data, struct snd_kcontrol *sctl)
*
* This function returns zero if successful or a negative error code.
*/
-int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
- unsigned int *tlv, const char * const *slaves)
+int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
+ unsigned int *tlv, const char * const *slaves,
+ const char *suffix, bool init_slave_vol,
+ struct snd_kcontrol **ctl_ret)
{
struct snd_kcontrol *kctl;
int err;
- err = map_slaves(codec, slaves, check_slave_present, NULL);
+ if (ctl_ret)
+ *ctl_ret = NULL;
+
+ err = map_slaves(codec, slaves, suffix, check_slave_present, NULL);
if (err != 1) {
snd_printdd("No slave found for %s\n", name);
return 0;
@@ -2363,13 +2433,119 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
if (err < 0)
return err;
- err = map_slaves(codec, slaves, (map_slave_func_t)snd_ctl_add_slave,
- kctl);
+ err = map_slaves(codec, slaves, suffix,
+ (map_slave_func_t)snd_ctl_add_slave, kctl);
if (err < 0)
return err;
+
+ /* init with master mute & zero volume */
+ put_kctl_with_value(kctl, 0);
+ if (init_slave_vol)
+ map_slaves(codec, slaves, suffix,
+ tlv ? init_slave_0dB : init_slave_unmute, kctl);
+
+ if (ctl_ret)
+ *ctl_ret = kctl;
+ return 0;
+}
+EXPORT_SYMBOL_HDA(__snd_hda_add_vmaster);
+
+/*
+ * mute-LED control using vmaster
+ */
+static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ static const char * const texts[] = {
+ "Off", "On", "Follow Master"
+ };
+ unsigned int index;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 3;
+ index = uinfo->value.enumerated.item;
+ if (index >= 3)
+ index = 2;
+ strcpy(uinfo->value.enumerated.name, texts[index]);
+ return 0;
+}
+
+static int vmaster_mute_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.enumerated.item[0] = hook->mute_mode;
return 0;
}
-EXPORT_SYMBOL_HDA(snd_hda_add_vmaster);
+
+static int vmaster_mute_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_vmaster_mute_hook *hook = snd_kcontrol_chip(kcontrol);
+ unsigned int old_mode = hook->mute_mode;
+
+ hook->mute_mode = ucontrol->value.enumerated.item[0];
+ if (hook->mute_mode > HDA_VMUTE_FOLLOW_MASTER)
+ hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER;
+ if (old_mode == hook->mute_mode)
+ return 0;
+ snd_hda_sync_vmaster_hook(hook);
+ return 1;
+}
+
+static struct snd_kcontrol_new vmaster_mute_mode = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Mute-LED Mode",
+ .info = vmaster_mute_mode_info,
+ .get = vmaster_mute_mode_get,
+ .put = vmaster_mute_mode_put,
+};
+
+/*
+ * Add a mute-LED hook with the given vmaster switch kctl
+ * "Mute-LED Mode" control is automatically created and associated with
+ * the given hook.
+ */
+int snd_hda_add_vmaster_hook(struct hda_codec *codec,
+ struct hda_vmaster_mute_hook *hook,
+ bool expose_enum_ctl)
+{
+ struct snd_kcontrol *kctl;
+
+ if (!hook->hook || !hook->sw_kctl)
+ return 0;
+ snd_ctl_add_vmaster_hook(hook->sw_kctl, hook->hook, codec);
+ hook->codec = codec;
+ hook->mute_mode = HDA_VMUTE_FOLLOW_MASTER;
+ if (!expose_enum_ctl)
+ return 0;
+ kctl = snd_ctl_new1(&vmaster_mute_mode, hook);
+ if (!kctl)
+ return -ENOMEM;
+ return snd_hda_ctl_add(codec, 0, kctl);
+}
+EXPORT_SYMBOL_HDA(snd_hda_add_vmaster_hook);
+
+/*
+ * Call the hook with the current value for synchronization
+ * Should be called in init callback
+ */
+void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook)
+{
+ if (!hook->hook || !hook->codec)
+ return;
+ switch (hook->mute_mode) {
+ case HDA_VMUTE_FOLLOW_MASTER:
+ snd_ctl_sync_vmaster_hook(hook->sw_kctl);
+ break;
+ default:
+ hook->hook(hook->codec, hook->mute_mode);
+ break;
+ }
+}
+EXPORT_SYMBOL_HDA(snd_hda_sync_vmaster_hook);
+
/**
* snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch
@@ -5114,7 +5290,7 @@ static int fill_audio_out_name(struct hda_codec *codec, hda_nid_t nid,
const char *pfx = "", *sfx = "";
/* handle as a speaker if it's a fixed line-out */
- if (!strcmp(name, "Line-Out") && attr == INPUT_PIN_ATTR_INT)
+ if (!strcmp(name, "Line Out") && attr == INPUT_PIN_ATTR_INT)
name = "Speaker";
/* check the location */
switch (attr) {
@@ -5173,7 +5349,7 @@ int snd_hda_get_pin_label(struct hda_codec *codec, hda_nid_t nid,
switch (get_defcfg_device(def_conf)) {
case AC_JACK_LINE_OUT:
- return fill_audio_out_name(codec, nid, cfg, "Line-Out",
+ return fill_audio_out_name(codec, nid, cfg, "Line Out",
label, maxlen, indexp);
case AC_JACK_SPEAKER:
return fill_audio_out_name(codec, nid, cfg, "Speaker",
@@ -5268,6 +5444,10 @@ int snd_hda_suspend(struct hda_bus *bus)
list_for_each_entry(codec, &bus->codec_list, list) {
if (hda_codec_is_power_on(codec))
hda_call_codec_suspend(codec);
+ else /* forcibly change the power to D3 even if not used */
+ hda_set_power_state(codec,
+ codec->afg ? codec->afg : codec->mfg,
+ AC_PWRST_D3);
if (codec->patch_ops.post_suspend)
codec->patch_ops.post_suspend(codec);
}
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index e9f71dc0d464..9a9f372e1be4 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -298,6 +298,9 @@ enum {
#define AC_AMPCAP_MUTE (1<<31) /* mute capable */
#define AC_AMPCAP_MUTE_SHIFT 31
+/* driver-specific amp-caps: using bits 24-30 */
+#define AC_AMPCAP_MIN_MUTE (1 << 30) /* min-volume = mute */
+
/* Connection list */
#define AC_CLIST_LENGTH (0x7f<<0)
#define AC_CLIST_LONG (1<<7)
@@ -852,6 +855,7 @@ struct hda_codec {
unsigned int pins_shutup:1; /* pins are shut up */
unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
unsigned int ignore_misc_bit:1; /* ignore MISC_NO_PRESENCE bit */
+ unsigned int no_jack_detect:1; /* Machine has no jack-detection */
#ifdef CONFIG_SND_HDA_POWER_SAVE
unsigned int power_on :1; /* current (global) power-state */
unsigned int power_transition :1; /* power-state in transition */
diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c
index c1da422e085a..b58b4b1687fa 100644
--- a/sound/pci/hda/hda_eld.c
+++ b/sound/pci/hda/hda_eld.c
@@ -385,8 +385,8 @@ error:
static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen)
{
static unsigned int alsa_rates[] = {
- 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200,
- 96000, 176400, 192000, 384000
+ 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
+ 88200, 96000, 176400, 192000, 384000
};
int i, j;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index fb35474c1203..c19e71a94e1b 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -84,7 +84,7 @@ module_param_array(model, charp, NULL, 0444);
MODULE_PARM_DESC(model, "Use the given board model.");
module_param_array(position_fix, int, NULL, 0444);
MODULE_PARM_DESC(position_fix, "DMA pointer read method."
- "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO).");
+ "(0 = auto, 1 = LPIB, 2 = POSBUF, 3 = VIACOMBO, 4 = COMBO).");
module_param_array(bdl_pos_adj, int, NULL, 0644);
MODULE_PARM_DESC(bdl_pos_adj, "BDL position adjustment offset.");
module_param_array(probe_mask, int, NULL, 0444);
@@ -94,7 +94,7 @@ MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization.");
module_param(single_cmd, bool, 0444);
MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
"(for debugging only).");
-module_param(enable_msi, int, 0444);
+module_param(enable_msi, bint, 0444);
MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
#ifdef CONFIG_SND_HDA_PATCH_LOADER
module_param_array(patch, charp, NULL, 0444);
@@ -121,8 +121,8 @@ module_param(power_save_controller, bool, 0644);
MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode.");
#endif
-static bool align_buffer_size = 1;
-module_param(align_buffer_size, bool, 0644);
+static int align_buffer_size = -1;
+module_param(align_buffer_size, bint, 0644);
MODULE_PARM_DESC(align_buffer_size,
"Force buffer and period sizes to be multiple of 128 bytes.");
@@ -148,6 +148,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
"{Intel, PCH},"
"{Intel, CPT},"
"{Intel, PPT},"
+ "{Intel, LPT},"
"{Intel, PBG},"
"{Intel, SCH},"
"{ATI, SB450},"
@@ -329,6 +330,7 @@ enum {
POS_FIX_LPIB,
POS_FIX_POSBUF,
POS_FIX_VIACOMBO,
+ POS_FIX_COMBO,
};
/* Defines for ATI HD Audio support in SB450 south bridge */
@@ -469,6 +471,7 @@ struct azx {
unsigned int irq_pending_warned :1;
unsigned int probing :1; /* codec probing phase */
unsigned int snoop:1;
+ unsigned int align_buffer_size:1;
/* for debugging */
unsigned int last_cmd[AZX_MAX_CODECS];
@@ -514,6 +517,7 @@ enum {
#define AZX_DCAPS_SYNC_WRITE (1 << 19) /* sync each cmd write */
#define AZX_DCAPS_OLD_SSYNC (1 << 20) /* Old SSYNC reg for ICH */
#define AZX_DCAPS_BUFSIZE (1 << 21) /* no buffer size alignment */
+#define AZX_DCAPS_ALIGN_BUFSIZE (1 << 22) /* buffer size alignment */
/* quirks for ATI SB / AMD Hudson */
#define AZX_DCAPS_PRESET_ATI_SB \
@@ -526,7 +530,8 @@ enum {
/* quirks for Nvidia */
#define AZX_DCAPS_PRESET_NVIDIA \
- (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI)
+ (AZX_DCAPS_NVIDIA_SNOOP | AZX_DCAPS_RIRB_DELAY | AZX_DCAPS_NO_MSI |\
+ AZX_DCAPS_ALIGN_BUFSIZE)
static char *driver_short_names[] __devinitdata = {
[AZX_DRIVER_ICH] = "HDA Intel",
@@ -1690,7 +1695,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
runtime->hw.rates = hinfo->rates;
snd_pcm_limit_hw_rates(runtime);
snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
- if (align_buffer_size)
+ if (chip->align_buffer_size)
/* constrain buffer sizes to be multiple of 128
bytes. This is more efficient in terms of memory
access but isn't required by the HDA spec and
@@ -2346,17 +2351,6 @@ static void azx_power_notify(struct hda_bus *bus)
* power management
*/
-static int snd_hda_codecs_inuse(struct hda_bus *bus)
-{
- struct hda_codec *codec;
-
- list_for_each_entry(codec, &bus->codec_list, list) {
- if (snd_hda_codec_needs_resume(codec))
- return 1;
- }
- return 0;
-}
-
static int azx_suspend(struct pci_dev *pci, pm_message_t state)
{
struct snd_card *card = pci_get_drvdata(pci);
@@ -2403,8 +2397,7 @@ static int azx_resume(struct pci_dev *pci)
return -EIO;
azx_init_pci(chip);
- if (snd_hda_codecs_inuse(chip->bus))
- azx_init_chip(chip, 1);
+ azx_init_chip(chip, 1);
snd_hda_resume(chip->bus);
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
@@ -2516,6 +2509,7 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
case POS_FIX_LPIB:
case POS_FIX_POSBUF:
case POS_FIX_VIACOMBO:
+ case POS_FIX_COMBO:
return fix;
}
@@ -2695,6 +2689,12 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
chip->position_fix[0] = chip->position_fix[1] =
check_position_fix(chip, position_fix[dev]);
+ /* combo mode uses LPIB for playback */
+ if (chip->position_fix[0] == POS_FIX_COMBO) {
+ chip->position_fix[0] = POS_FIX_LPIB;
+ chip->position_fix[1] = POS_FIX_AUTO;
+ }
+
check_probe_mask(chip, dev);
chip->single_cmd = single_cmd;
@@ -2773,8 +2773,16 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
}
/* disable buffer size rounding to 128-byte multiples if supported */
- if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
- align_buffer_size = 0;
+ if (align_buffer_size >= 0)
+ chip->align_buffer_size = !!align_buffer_size;
+ else {
+ if (chip->driver_caps & AZX_DCAPS_BUFSIZE)
+ chip->align_buffer_size = 0;
+ else if (chip->driver_caps & AZX_DCAPS_ALIGN_BUFSIZE)
+ chip->align_buffer_size = 1;
+ else
+ chip->align_buffer_size = 1;
+ }
/* allow 64bit DMA address if supported by H/W */
if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64)))
@@ -2990,6 +2998,10 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = {
{ PCI_DEVICE(0x8086, 0x1e20),
.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
AZX_DCAPS_BUFSIZE},
+ /* Lynx Point */
+ { PCI_DEVICE(0x8086, 0x8c20),
+ .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
+ AZX_DCAPS_BUFSIZE},
/* SCH */
{ PCI_DEVICE(0x8086, 0x811b),
.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index d8a35da0803f..d68948499fbc 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -19,6 +19,22 @@
#include "hda_local.h"
#include "hda_jack.h"
+bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
+{
+ if (codec->no_jack_detect)
+ return false;
+ if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT))
+ return false;
+ if (!codec->ignore_misc_bit &&
+ (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
+ AC_DEFCFG_MISC_NO_PRESENCE))
+ return false;
+ if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
+ return false;
+ return true;
+}
+EXPORT_SYMBOL_HDA(is_jack_detectable);
+
/* execute pin sense measurement */
static u32 read_pin_sense(struct hda_codec *codec, hda_nid_t nid)
{
@@ -282,7 +298,8 @@ int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl);
static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
- const struct auto_pin_cfg *cfg)
+ const struct auto_pin_cfg *cfg,
+ char *lastname, int *lastidx)
{
unsigned int def_conf, conn;
char name[44];
@@ -298,6 +315,10 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
return 0;
snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx);
+ if (!strcmp(name, lastname) && idx == *lastidx)
+ idx++;
+ strncpy(lastname, name, 44);
+ *lastidx = idx;
err = snd_hda_jack_add_kctl(codec, nid, name, idx);
if (err < 0)
return err;
@@ -311,41 +332,42 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec,
const struct auto_pin_cfg *cfg)
{
const hda_nid_t *p;
- int i, err;
+ int i, err, lastidx = 0;
+ char lastname[44] = "";
for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) {
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) {
if (*p == *cfg->line_out_pins) /* might be duplicated */
break;
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) {
if (*p == *cfg->line_out_pins) /* might be duplicated */
break;
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0; i < cfg->num_inputs; i++) {
- err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg);
+ err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) {
- err = add_jack_kctl(codec, *p, cfg);
+ err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
if (err < 0)
return err;
}
- err = add_jack_kctl(codec, cfg->dig_in_pin, cfg);
+ err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx);
if (err < 0)
return err;
- err = add_jack_kctl(codec, cfg->mono_out_pin, cfg);
+ err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx);
if (err < 0)
return err;
return 0;
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index f8f97c71c9c1..c66655cf413a 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -62,18 +62,7 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid,
u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
-static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid)
-{
- if (!(snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT))
- return false;
- if (!codec->ignore_misc_bit &&
- (get_defcfg_misc(snd_hda_codec_get_pincfg(codec, nid)) &
- AC_DEFCFG_MISC_NO_PRESENCE))
- return false;
- if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
- return false;
- return true;
-}
+bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);
int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
const char *name, int idx);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index aca8d3193b95..0ec9248165bc 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -139,10 +139,36 @@ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir,
unsigned int *tlv);
struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
const char *name);
-int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
- unsigned int *tlv, const char * const *slaves);
+int __snd_hda_add_vmaster(struct hda_codec *codec, char *name,
+ unsigned int *tlv, const char * const *slaves,
+ const char *suffix, bool init_slave_vol,
+ struct snd_kcontrol **ctl_ret);
+#define snd_hda_add_vmaster(codec, name, tlv, slaves, suffix) \
+ __snd_hda_add_vmaster(codec, name, tlv, slaves, suffix, true, NULL)
int snd_hda_codec_reset(struct hda_codec *codec);
+enum {
+ HDA_VMUTE_OFF,
+ HDA_VMUTE_ON,
+ HDA_VMUTE_FOLLOW_MASTER,
+};
+
+struct hda_vmaster_mute_hook {
+ /* below two fields must be filled by the caller of
+ * snd_hda_add_vmaster_hook() beforehand
+ */
+ struct snd_kcontrol *sw_kctl;
+ void (*hook)(void *, int);
+ /* below are initialized automatically */
+ unsigned int mute_mode; /* HDA_VMUTE_XXX */
+ struct hda_codec *codec;
+};
+
+int snd_hda_add_vmaster_hook(struct hda_codec *codec,
+ struct hda_vmaster_mute_hook *hook,
+ bool expose_enum_ctl);
+void snd_hda_sync_vmaster_hook(struct hda_vmaster_mute_hook *hook);
+
/* amp value bits */
#define HDA_AMP_MUTE 0x80
#define HDA_AMP_UNMUTE 0x00
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 9cb14b42dfff..7143393927da 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -82,6 +82,7 @@ struct ad198x_spec {
unsigned int inv_jack_detect: 1;/* inverted jack-detection */
unsigned int inv_eapd: 1; /* inverted EAPD implementation */
unsigned int analog_beep: 1; /* analog beep input present */
+ unsigned int avoid_init_slave_vol:1;
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
@@ -137,51 +138,17 @@ static int ad198x_init(struct hda_codec *codec)
return 0;
}
-static const char * const ad_slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "Headphone Playback Volume",
- "Mono Playback Volume",
- "Speaker Playback Volume",
- "IEC958 Playback Volume",
+static const char * const ad_slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side",
+ "Headphone", "Mono", "Speaker", "IEC958",
NULL
};
-static const char * const ad_slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "Headphone Playback Switch",
- "Mono Playback Switch",
- "Speaker Playback Switch",
- "IEC958 Playback Switch",
+static const char * const ad1988_6stack_fp_slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side", "IEC958",
NULL
};
-static const char * const ad1988_6stack_fp_slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "IEC958 Playback Volume",
- NULL
-};
-
-static const char * const ad1988_6stack_fp_slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "IEC958 Playback Switch",
- NULL
-};
static void ad198x_free_kctls(struct hda_codec *codec);
#ifdef CONFIG_SND_HDA_INPUT_BEEP
@@ -257,10 +224,12 @@ static int ad198x_build_controls(struct hda_codec *codec)
unsigned int vmaster_tlv[4];
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
- err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ err = __snd_hda_add_vmaster(codec, "Master Playback Volume",
vmaster_tlv,
(spec->slave_vols ?
- spec->slave_vols : ad_slave_vols));
+ spec->slave_vols : ad_slave_pfxs),
+ "Playback Volume",
+ !spec->avoid_init_slave_vol, NULL);
if (err < 0)
return err;
}
@@ -268,7 +237,8 @@ static int ad198x_build_controls(struct hda_codec *codec)
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
NULL,
(spec->slave_sws ?
- spec->slave_sws : ad_slave_sws));
+ spec->slave_sws : ad_slave_pfxs),
+ "Playback Switch");
if (err < 0)
return err;
}
@@ -3385,8 +3355,8 @@ static int patch_ad1988(struct hda_codec *codec)
if (spec->autocfg.hp_pins[0]) {
spec->mixers[spec->num_mixers++] = ad1988_hp_mixers;
- spec->slave_vols = ad1988_6stack_fp_slave_vols;
- spec->slave_sws = ad1988_6stack_fp_slave_sws;
+ spec->slave_vols = ad1988_6stack_fp_slave_pfxs;
+ spec->slave_sws = ad1988_6stack_fp_slave_pfxs;
spec->alt_dac_nid = ad1988_alt_dac_nid;
spec->stream_analog_alt_playback =
&ad198x_pcm_analog_alt_playback;
@@ -3594,16 +3564,8 @@ static const struct hda_amp_list ad1884_loopbacks[] = {
#endif
static const char * const ad1884_slave_vols[] = {
- "PCM Playback Volume",
- "Mic Playback Volume",
- "Mono Playback Volume",
- "Front Mic Playback Volume",
- "Mic Playback Volume",
- "CD Playback Volume",
- "Internal Mic Playback Volume",
- "Docking Mic Playback Volume",
- /* "Beep Playback Volume", */
- "IEC958 Playback Volume",
+ "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
+ "Internal Mic", "Docking Mic", /* "Beep", */ "IEC958",
NULL
};
@@ -3644,6 +3606,8 @@ static int patch_ad1884(struct hda_codec *codec)
spec->vmaster_nid = 0x04;
/* we need to cover all playback volumes */
spec->slave_vols = ad1884_slave_vols;
+ /* slaves may contain input volumes, so we can't raise to 0dB blindly */
+ spec->avoid_init_slave_vol = 1;
codec->patch_ops = ad198x_patch_ops;
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 35abe3c62908..21d91d580da8 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -728,18 +728,19 @@ static int ca0132_hp_switch_put(struct snd_kcontrol *kcontrol,
err = chipio_read(codec, REG_CODEC_MUTE, &data);
if (err < 0)
- return err;
+ goto exit;
/* *valp 0 is mute, 1 is unmute */
data = (data & 0x7f) | (*valp ? 0 : 0x80);
- chipio_write(codec, REG_CODEC_MUTE, data);
+ err = chipio_write(codec, REG_CODEC_MUTE, data);
if (err < 0)
- return err;
+ goto exit;
spec->curr_hp_switch = *valp;
+ exit:
snd_hda_power_down(codec);
- return 1;
+ return err < 0 ? err : 1;
}
static int ca0132_speaker_switch_get(struct snd_kcontrol *kcontrol,
@@ -770,18 +771,19 @@ static int ca0132_speaker_switch_put(struct snd_kcontrol *kcontrol,
err = chipio_read(codec, REG_CODEC_MUTE, &data);
if (err < 0)
- return err;
+ goto exit;
/* *valp 0 is mute, 1 is unmute */
data = (data & 0xef) | (*valp ? 0 : 0x10);
- chipio_write(codec, REG_CODEC_MUTE, data);
+ err = chipio_write(codec, REG_CODEC_MUTE, data);
if (err < 0)
- return err;
+ goto exit;
spec->curr_speaker_switch = *valp;
+ exit:
snd_hda_power_down(codec);
- return 1;
+ return err < 0 ? err : 1;
}
static int ca0132_hp_volume_get(struct snd_kcontrol *kcontrol,
@@ -819,25 +821,26 @@ static int ca0132_hp_volume_put(struct snd_kcontrol *kcontrol,
err = chipio_read(codec, REG_CODEC_HP_VOL_L, &data);
if (err < 0)
- return err;
+ goto exit;
val = 31 - left_vol;
data = (data & 0xe0) | val;
- chipio_write(codec, REG_CODEC_HP_VOL_L, data);
+ err = chipio_write(codec, REG_CODEC_HP_VOL_L, data);
if (err < 0)
- return err;
+ goto exit;
val = 31 - right_vol;
data = (data & 0xe0) | val;
- chipio_write(codec, REG_CODEC_HP_VOL_R, data);
+ err = chipio_write(codec, REG_CODEC_HP_VOL_R, data);
if (err < 0)
- return err;
+ goto exit;
spec->curr_hp_volume[0] = left_vol;
spec->curr_hp_volume[1] = right_vol;
+ exit:
snd_hda_power_down(codec);
- return 1;
+ return err < 0 ? err : 1;
}
static int add_hp_switch(struct hda_codec *codec, hda_nid_t nid)
@@ -936,6 +939,8 @@ static int ca0132_build_controls(struct hda_codec *codec)
if (err < 0)
return err;
err = add_in_volume(codec, spec->dig_in, "IEC958");
+ if (err < 0)
+ return err;
}
return 0;
}
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 0e99357e822c..c83ccdba1e5a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -609,7 +609,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
"Front Speaker", "Surround Speaker", "Bass Speaker"
};
static const char * const line_outs[] = {
- "Front Line-Out", "Surround Line-Out", "Bass Line-Out"
+ "Front Line Out", "Surround Line Out", "Bass Line Out"
};
fix_volume_caps(codec, dac);
@@ -635,7 +635,7 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
if (num_ctls > 1)
name = line_outs[idx];
else
- name = "Line-Out";
+ name = "Line Out";
break;
}
@@ -988,8 +988,10 @@ static void cs_automic(struct hda_codec *codec)
change_cur_input(codec, !spec->automic_idx, 0);
} else {
if (present) {
- spec->last_input = spec->cur_input;
- spec->cur_input = spec->automic_idx;
+ if (spec->cur_input != spec->automic_idx) {
+ spec->last_input = spec->cur_input;
+ spec->cur_input = spec->automic_idx;
+ }
} else {
spec->cur_input = spec->last_input;
}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 8a32a69c83c3..8c6523bbc797 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -70,6 +70,8 @@ struct conexant_spec {
const struct snd_kcontrol_new *mixers[5];
int num_mixers;
hda_nid_t vmaster_nid;
+ struct hda_vmaster_mute_hook vmaster_mute;
+ bool vmaster_mute_led;
const struct hda_verb *init_verbs[5]; /* initialization verbs
* don't forget NULL
@@ -465,21 +467,8 @@ static const struct snd_kcontrol_new cxt_beep_mixer[] = {
};
#endif
-static const char * const slave_vols[] = {
- "Headphone Playback Volume",
- "Speaker Playback Volume",
- "Front Playback Volume",
- "Surround Playback Volume",
- "CLFE Playback Volume",
- NULL
-};
-
-static const char * const slave_sws[] = {
- "Headphone Playback Switch",
- "Speaker Playback Switch",
- "Front Playback Switch",
- "Surround Playback Switch",
- "CLFE Playback Switch",
+static const char * const slave_pfxs[] = {
+ "Headphone", "Speaker", "Front", "Surround", "CLFE",
NULL
};
@@ -519,14 +508,17 @@ static int conexant_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv, slave_vols);
+ vmaster_tlv, slave_pfxs,
+ "Playback Volume");
if (err < 0)
return err;
}
if (spec->vmaster_nid &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
- err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, slave_sws);
+ err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, slave_pfxs,
+ "Playback Switch", true,
+ &spec->vmaster_mute.sw_kctl);
if (err < 0)
return err;
}
@@ -1643,7 +1635,7 @@ static void cxt5051_update_speaker(struct hda_codec *codec)
pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
pinctl);
- /* on ideapad there is an aditional speaker (subwoofer) to mute */
+ /* on ideapad there is an additional speaker (subwoofer) to mute */
if (spec->ideapad)
snd_hda_codec_write(codec, 0x1b, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
@@ -3027,14 +3019,13 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS),
- SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
+ SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T510", CXT5066_AUTO),
SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520 & W520", CXT5066_AUTO),
SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD),
SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS),
SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO),
- SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */
SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO),
{}
};
@@ -3482,7 +3473,7 @@ static int cx_automute_mode_info(struct snd_kcontrol *kcontrol,
"Disabled", "Enabled"
};
static const char * const texts3[] = {
- "Disabled", "Speaker Only", "Line-Out+Speaker"
+ "Disabled", "Speaker Only", "Line Out+Speaker"
};
const char * const *texts;
@@ -3943,6 +3934,63 @@ static void enable_unsol_pins(struct hda_codec *codec, int num_pins,
snd_hda_jack_detect_enable(codec, pins[i], action);
}
+static bool found_in_nid_list(hda_nid_t nid, const hda_nid_t *list, int nums)
+{
+ int i;
+ for (i = 0; i < nums; i++)
+ if (list[i] == nid)
+ return true;
+ return false;
+}
+
+/* is the given NID found in any of autocfg items? */
+static bool found_in_autocfg(struct auto_pin_cfg *cfg, hda_nid_t nid)
+{
+ int i;
+
+ if (found_in_nid_list(nid, cfg->line_out_pins, cfg->line_outs) ||
+ found_in_nid_list(nid, cfg->hp_pins, cfg->hp_outs) ||
+ found_in_nid_list(nid, cfg->speaker_pins, cfg->speaker_outs) ||
+ found_in_nid_list(nid, cfg->dig_out_pins, cfg->dig_outs))
+ return true;
+ for (i = 0; i < cfg->num_inputs; i++)
+ if (cfg->inputs[i].pin == nid)
+ return true;
+ if (cfg->dig_in_pin == nid)
+ return true;
+ return false;
+}
+
+/* clear unsol-event tags on unused pins; Conexant codecs seem to leave
+ * invalid unsol tags by some reason
+ */
+static void clear_unsol_on_unused_pins(struct hda_codec *codec)
+{
+ struct conexant_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+
+ for (i = 0; i < codec->init_pins.used; i++) {
+ struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+ if (!found_in_autocfg(cfg, pin->nid))
+ snd_hda_codec_write(codec, pin->nid, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE, 0);
+ }
+}
+
+/* turn on/off EAPD according to Master switch */
+static void cx_auto_vmaster_hook(void *private_data, int enabled)
+{
+ struct hda_codec *codec = private_data;
+ struct conexant_spec *spec = codec->spec;
+
+ if (enabled && spec->pin_eapd_ctrls) {
+ cx_auto_update_speakers(codec);
+ return;
+ }
+ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, enabled);
+}
+
static void cx_auto_init_output(struct hda_codec *codec)
{
struct conexant_spec *spec = codec->spec;
@@ -3983,6 +4031,7 @@ static void cx_auto_init_output(struct hda_codec *codec)
/* turn on all EAPDs if no individual EAPD control is available */
if (!spec->pin_eapd_ctrls)
cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+ clear_unsol_on_unused_pins(codec);
}
static void cx_auto_init_input(struct hda_codec *codec)
@@ -4046,11 +4095,13 @@ static void cx_auto_init_digital(struct hda_codec *codec)
static int cx_auto_init(struct hda_codec *codec)
{
+ struct conexant_spec *spec = codec->spec;
/*snd_hda_sequence_write(codec, cx_auto_init_verbs);*/
cx_auto_init_output(codec);
cx_auto_init_input(codec);
cx_auto_init_digital(codec);
snd_hda_jack_report_sync(codec);
+ snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
return 0;
}
@@ -4079,7 +4130,8 @@ static int cx_auto_add_volume_idx(struct hda_codec *codec, const char *basename,
err = snd_hda_ctl_add(codec, nid, kctl);
if (err < 0)
return err;
- if (!(query_amp_caps(codec, nid, hda_dir) & AC_AMPCAP_MUTE))
+ if (!(query_amp_caps(codec, nid, hda_dir) &
+ (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)))
break;
}
return 0;
@@ -4295,6 +4347,13 @@ static int cx_auto_build_controls(struct hda_codec *codec)
err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
if (err < 0)
return err;
+ if (spec->vmaster_mute.sw_kctl) {
+ spec->vmaster_mute.hook = cx_auto_vmaster_hook;
+ err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute,
+ spec->vmaster_mute_led);
+ if (err < 0)
+ return err;
+ }
return 0;
}
@@ -4319,7 +4378,6 @@ static int cx_auto_search_adcs(struct hda_codec *codec)
return 0;
}
-
static const struct hda_codec_ops cx_auto_patch_ops = {
.build_controls = cx_auto_build_controls,
.build_pcms = conexant_build_pcms,
@@ -4367,6 +4425,7 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = {
{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
{ 0x17, 0x21a11000 }, /* dock-mic */
{ 0x19, 0x2121103f }, /* dock-HP */
+ { 0x1c, 0x21440100 }, /* dock SPDIF out */
{}
};
@@ -4379,6 +4438,22 @@ static const struct snd_pci_quirk cxt_fixups[] = {
{}
};
+/* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches
+ * can be created (bko#42825)
+ */
+static void add_cx5051_fake_mutes(struct hda_codec *codec)
+{
+ static hda_nid_t out_nids[] = {
+ 0x10, 0x11, 0
+ };
+ hda_nid_t *p;
+
+ for (p = out_nids; *p; p++)
+ snd_hda_override_amp_caps(codec, *p, HDA_OUTPUT,
+ AC_AMPCAP_MIN_MUTE |
+ query_amp_caps(codec, *p, HDA_OUTPUT));
+}
+
static int patch_conexant_auto(struct hda_codec *codec)
{
struct conexant_spec *spec;
@@ -4397,10 +4472,25 @@ static int patch_conexant_auto(struct hda_codec *codec)
case 0x14f15045:
spec->single_adc_amp = 1;
break;
+ case 0x14f15051:
+ add_cx5051_fake_mutes(codec);
+ break;
}
apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl);
+ /* Show mute-led control only on HP laptops
+ * This is a sort of white-list: on HP laptops, EAPD corresponds
+ * only to the mute-LED without actualy amp function. Meanwhile,
+ * others may use EAPD really as an amp switch, so it might be
+ * not good to expose it blindly.
+ */
+ switch (codec->subsystem_id >> 16) {
+ case 0x103c:
+ spec->vmaster_mute_led = 1;
+ break;
+ }
+
err = cx_auto_search_adcs(codec);
if (err < 0)
return err;
@@ -4414,6 +4504,18 @@ static int patch_conexant_auto(struct hda_codec *codec)
codec->patch_ops = cx_auto_patch_ops;
if (spec->beep_amp)
snd_hda_attach_beep_device(codec, spec->beep_amp);
+
+ /* Some laptops with Conexant chips show stalls in S3 resume,
+ * which falls into the single-cmd mode.
+ * Better to make reset, then.
+ */
+ if (!codec->bus->sync_write) {
+ snd_printd("hda_codec: "
+ "Enable sync_write for stable communication\n");
+ codec->bus->sync_write = 1;
+ codec->bus->allow_bus_reset = 1;
+ }
+
return 0;
}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 1168ebd3fb5c..540cd13f7f15 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1912,6 +1912,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
{ .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi },
{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
+{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi },
{ .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi },
{} /* terminator */
};
@@ -1958,6 +1959,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862803");
MODULE_ALIAS("snd-hda-codec-id:80862804");
MODULE_ALIAS("snd-hda-codec-id:80862805");
MODULE_ALIAS("snd-hda-codec-id:80862806");
+MODULE_ALIAS("snd-hda-codec-id:80862880");
MODULE_ALIAS("snd-hda-codec-id:808629fb");
MODULE_LICENSE("GPL");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5e82acf77c5a..8ea2fd654327 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -80,6 +80,8 @@ enum {
ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */
};
+#define MAX_VOL_NIDS 0x40
+
struct alc_spec {
/* codec parameterization */
const struct snd_kcontrol_new *mixers[5]; /* mixer arrays */
@@ -118,8 +120,8 @@ struct alc_spec {
const hda_nid_t *capsrc_nids;
hda_nid_t dig_in_nid; /* digital-in NID; optional */
hda_nid_t mixer_nid; /* analog-mixer NID */
- DECLARE_BITMAP(vol_ctls, 0x20 << 1);
- DECLARE_BITMAP(sw_ctls, 0x20 << 1);
+ DECLARE_BITMAP(vol_ctls, MAX_VOL_NIDS << 1);
+ DECLARE_BITMAP(sw_ctls, MAX_VOL_NIDS << 1);
/* capture setup for dynamic dual-adc switch */
hda_nid_t cur_adc;
@@ -177,6 +179,7 @@ struct alc_spec {
unsigned int detect_lo:1; /* Line-out detection enabled */
unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */
unsigned int automute_lo_possible:1; /* there are line outs and HP */
+ unsigned int keep_vref_in_automute:1; /* Don't clear VREF in automute */
/* other flags */
unsigned int no_analog :1; /* digital I/O only */
@@ -185,7 +188,6 @@ struct alc_spec {
unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */
unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
- unsigned int use_jack_tbl:1; /* 1 for model=auto */
/* auto-mute control */
int automute_mode;
@@ -196,8 +198,11 @@ struct alc_spec {
/* for virtual master */
hda_nid_t vmaster_nid;
+ struct hda_vmaster_mute_hook vmaster_mute;
#ifdef CONFIG_SND_HDA_POWER_SAVE
struct hda_loopback_check loopback;
+ int num_loopbacks;
+ struct hda_amp_list loopback_list[8];
#endif
/* for PLL fix */
@@ -218,8 +223,6 @@ struct alc_spec {
struct snd_array bind_ctls;
};
-#define ALC_MODEL_AUTO 0 /* common for all chips */
-
static bool check_amp_caps(struct hda_codec *codec, hda_nid_t nid,
int dir, unsigned int bits)
{
@@ -298,6 +301,9 @@ static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
int i, type, num_conns;
hda_nid_t nid;
+ if (!spec->input_mux)
+ return 0;
+
mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
imux = &spec->input_mux[mux_idx];
if (!imux->num_items && mux_idx > 0)
@@ -496,13 +502,24 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
for (i = 0; i < num_pins; i++) {
hda_nid_t nid = pins[i];
+ unsigned int val;
if (!nid)
break;
switch (spec->automute_mode) {
case ALC_AUTOMUTE_PIN:
+ /* don't reset VREF value in case it's controlling
+ * the amp (see alc861_fixup_asus_amp_vref_0f())
+ */
+ if (spec->keep_vref_in_automute) {
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ val &= ~PIN_HP;
+ } else
+ val = 0;
+ val |= pin_bits;
snd_hda_codec_write(codec, nid, 0,
AC_VERB_SET_PIN_WIDGET_CONTROL,
- pin_bits);
+ val);
break;
case ALC_AUTOMUTE_AMP:
snd_hda_codec_amp_stereo(codec, nid, HDA_OUTPUT, 0,
@@ -621,17 +638,10 @@ static void alc_mic_automute(struct hda_codec *codec)
alc_mux_select(codec, 0, spec->int_mic_idx, false);
}
-/* unsolicited event for HP jack sensing */
-static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+/* handle the specified unsol action (ALC_XXX_EVENT) */
+static void alc_exec_unsol_event(struct hda_codec *codec, int action)
{
- struct alc_spec *spec = codec->spec;
- if (codec->vendor_id == 0x10ec0880)
- res >>= 28;
- else
- res >>= 26;
- if (spec->use_jack_tbl)
- res = snd_hda_jack_get_action(codec, res);
- switch (res) {
+ switch (action) {
case ALC_HP_EVENT:
alc_hp_automute(codec);
break;
@@ -645,6 +655,53 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
snd_hda_jack_report_sync(codec);
}
+/* update the master volume per volume-knob's unsol event */
+static void alc_update_knob_master(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int val;
+ struct snd_kcontrol *kctl;
+ struct snd_ctl_elem_value *uctl;
+
+ kctl = snd_hda_find_mixer_ctl(codec, "Master Playback Volume");
+ if (!kctl)
+ return;
+ uctl = kzalloc(sizeof(*uctl), GFP_KERNEL);
+ if (!uctl)
+ return;
+ val = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_VOLUME_KNOB_CONTROL, 0);
+ val &= HDA_AMP_VOLMASK;
+ uctl->value.integer.value[0] = val;
+ uctl->value.integer.value[1] = val;
+ kctl->put(kctl, uctl);
+ kfree(uctl);
+}
+
+/* unsolicited event for HP jack sensing */
+static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ int action;
+
+ if (codec->vendor_id == 0x10ec0880)
+ res >>= 28;
+ else
+ res >>= 26;
+ action = snd_hda_jack_get_action(codec, res);
+ if (action == ALC_DCVOL_EVENT) {
+ /* Execute the dc-vol event here as it requires the NID
+ * but we don't pass NID to alc_exec_unsol_event().
+ * Once when we convert all static quirks to the auto-parser,
+ * this can be integerated into there.
+ */
+ struct hda_jack_tbl *jack;
+ jack = snd_hda_jack_tbl_get_from_tag(codec, res);
+ if (jack)
+ alc_update_knob_master(codec, jack->nid);
+ return;
+ }
+ alc_exec_unsol_event(codec, action);
+}
+
/* call init functions of standard auto-mute helpers */
static void alc_inithook(struct hda_codec *codec)
{
@@ -785,7 +842,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol,
"Disabled", "Enabled"
};
static const char * const texts3[] = {
- "Disabled", "Speaker Only", "Line-Out+Speaker"
+ "Disabled", "Speaker Only", "Line Out+Speaker"
};
const char * const *texts;
@@ -1016,45 +1073,6 @@ static bool alc_check_dyn_adc_switch(struct hda_codec *codec)
return true;
}
-/* rebuild imux for matching with the given auto-mic pins (if not yet) */
-static bool alc_rebuild_imux_for_auto_mic(struct hda_codec *codec)
-{
- struct alc_spec *spec = codec->spec;
- struct hda_input_mux *imux;
- static char * const texts[3] = {
- "Mic", "Internal Mic", "Dock Mic"
- };
- int i;
-
- if (!spec->auto_mic)
- return false;
- imux = &spec->private_imux[0];
- if (spec->input_mux == imux)
- return true;
- spec->imux_pins[0] = spec->ext_mic_pin;
- spec->imux_pins[1] = spec->int_mic_pin;
- spec->imux_pins[2] = spec->dock_mic_pin;
- for (i = 0; i < 3; i++) {
- strcpy(imux->items[i].label, texts[i]);
- if (spec->imux_pins[i]) {
- hda_nid_t pin = spec->imux_pins[i];
- int c;
- for (c = 0; c < spec->num_adc_nids; c++) {
- hda_nid_t cap = get_capsrc(spec, c);
- int idx = get_connection_index(codec, cap, pin);
- if (idx >= 0) {
- imux->items[i].index = idx;
- break;
- }
- }
- imux->num_items = i + 1;
- }
- }
- spec->num_mux_defs = 1;
- spec->input_mux = imux;
- return true;
-}
-
/* check whether all auto-mic pins are valid; setup indices if OK */
static bool alc_auto_mic_check_imux(struct hda_codec *codec)
{
@@ -1424,6 +1442,7 @@ enum {
ALC_FIXUP_ACT_PRE_PROBE,
ALC_FIXUP_ACT_PROBE,
ALC_FIXUP_ACT_INIT,
+ ALC_FIXUP_ACT_BUILD,
};
static void alc_apply_fixup(struct hda_codec *codec, int action)
@@ -1503,6 +1522,13 @@ static void alc_pick_fixup(struct hda_codec *codec,
int id = -1;
const char *name = NULL;
+ /* when model=nofixup is given, don't pick up any fixups */
+ if (codec->modelname && !strcmp(codec->modelname, "nofixup")) {
+ spec->fixup_list = NULL;
+ spec->fixup_id = -1;
+ return;
+ }
+
if (codec->modelname && models) {
while (models->name) {
if (!strcmp(codec->modelname, models->name)) {
@@ -1830,32 +1856,10 @@ DEFINE_CAPMIX_NOSRC(3);
/*
* slave controls for virtual master
*/
-static const char * const alc_slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "Headphone Playback Volume",
- "Speaker Playback Volume",
- "Mono Playback Volume",
- "Line-Out Playback Volume",
- "PCM Playback Volume",
- NULL,
-};
-
-static const char * const alc_slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "Headphone Playback Switch",
- "Speaker Playback Switch",
- "Mono Playback Switch",
- "IEC958 Playback Switch",
- "Line-Out Playback Switch",
- "PCM Playback Switch",
+static const char * const alc_slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side",
+ "Headphone", "Speaker", "Mono", "Line Out",
+ "CLFE", "Bass Speaker", "PCM",
NULL,
};
@@ -1883,7 +1887,7 @@ static const struct snd_kcontrol_new alc_beep_mixer[] = {
};
#endif
-static int alc_build_controls(struct hda_codec *codec)
+static int __alc_build_controls(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
struct snd_kcontrol *kctl = NULL;
@@ -1946,14 +1950,17 @@ static int alc_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv, alc_slave_vols);
+ vmaster_tlv, alc_slave_pfxs,
+ "Playback Volume");
if (err < 0)
return err;
}
if (!spec->no_analog &&
!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
- err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, alc_slave_sws);
+ err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, alc_slave_pfxs,
+ "Playback Switch",
+ true, &spec->vmaster_mute.sw_kctl);
if (err < 0)
return err;
}
@@ -2029,10 +2036,19 @@ static int alc_build_controls(struct hda_codec *codec)
alc_free_kctls(codec); /* no longer needed */
+ return 0;
+}
+
+static int alc_build_controls(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ int err = __alc_build_controls(codec);
+ if (err < 0)
+ return err;
err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
if (err < 0)
return err;
-
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_BUILD);
return 0;
}
@@ -2042,21 +2058,23 @@ static int alc_build_controls(struct hda_codec *codec)
*/
static void alc_init_special_input_src(struct hda_codec *codec);
+static void alc_auto_init_std(struct hda_codec *codec);
static int alc_init(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
unsigned int i;
+ if (spec->init_hook)
+ spec->init_hook(codec);
+
alc_fix_pll(codec);
alc_auto_init_amp(codec, spec->init_amp);
for (i = 0; i < spec->num_init_verbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
alc_init_special_input_src(codec);
-
- if (spec->init_hook)
- spec->init_hook(codec);
+ alc_auto_init_std(codec);
alc_apply_fixup(codec, ALC_FIXUP_ACT_INIT);
@@ -2298,7 +2316,7 @@ static int alc_build_pcms(struct hda_codec *codec)
"%s Analog", codec->chip_name);
info->name = spec->stream_name_analog;
- if (spec->multiout.dac_nids > 0) {
+ if (spec->multiout.num_dacs > 0) {
p = spec->stream_analog_playback;
if (!p)
p = &alc_pcm_analog_playback;
@@ -2645,6 +2663,25 @@ static const char *alc_get_line_out_pfx(struct alc_spec *spec, int ch,
return channel_name[ch];
}
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+/* add the powersave loopback-list entry */
+static void add_loopback_list(struct alc_spec *spec, hda_nid_t mix, int idx)
+{
+ struct hda_amp_list *list;
+
+ if (spec->num_loopbacks >= ARRAY_SIZE(spec->loopback_list) - 1)
+ return;
+ list = spec->loopback_list + spec->num_loopbacks;
+ list->nid = mix;
+ list->dir = HDA_INPUT;
+ list->idx = idx;
+ spec->num_loopbacks++;
+ spec->loopback.amplist = spec->loopback_list;
+}
+#else
+#define add_loopback_list(spec, mix, idx) /* NOP */
+#endif
+
/* create input playback/capture controls for the given pin */
static int new_analog_input(struct alc_spec *spec, hda_nid_t pin,
const char *ctlname, int ctlidx,
@@ -2660,6 +2697,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin,
HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT));
if (err < 0)
return err;
+ add_loopback_list(spec, mix_nid, idx);
return 0;
}
@@ -2924,10 +2962,27 @@ static int alc_auto_select_dac(struct hda_codec *codec, hda_nid_t pin,
return 0;
}
+static bool alc_is_dac_already_used(struct hda_codec *codec, hda_nid_t nid)
+{
+ struct alc_spec *spec = codec->spec;
+ int i;
+ if (found_in_nid_list(nid, spec->multiout.dac_nids,
+ ARRAY_SIZE(spec->private_dac_nids)) ||
+ found_in_nid_list(nid, spec->multiout.hp_out_nid,
+ ARRAY_SIZE(spec->multiout.hp_out_nid)) ||
+ found_in_nid_list(nid, spec->multiout.extra_out_nid,
+ ARRAY_SIZE(spec->multiout.extra_out_nid)))
+ return true;
+ for (i = 0; i < spec->multi_ios; i++) {
+ if (spec->multi_io[i].dac == nid)
+ return true;
+ }
+ return false;
+}
+
/* look for an empty DAC slot */
static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
{
- struct alc_spec *spec = codec->spec;
hda_nid_t srcs[5];
int i, num;
@@ -2937,16 +2992,8 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
hda_nid_t nid = alc_auto_mix_to_dac(codec, srcs[i]);
if (!nid)
continue;
- if (found_in_nid_list(nid, spec->multiout.dac_nids,
- ARRAY_SIZE(spec->private_dac_nids)))
- continue;
- if (found_in_nid_list(nid, spec->multiout.hp_out_nid,
- ARRAY_SIZE(spec->multiout.hp_out_nid)))
- continue;
- if (found_in_nid_list(nid, spec->multiout.extra_out_nid,
- ARRAY_SIZE(spec->multiout.extra_out_nid)))
- continue;
- return nid;
+ if (!alc_is_dac_already_used(codec, nid))
+ return nid;
}
return 0;
}
@@ -2958,6 +3005,8 @@ static bool alc_auto_is_dac_reachable(struct hda_codec *codec,
hda_nid_t srcs[5];
int i, num;
+ if (!pin || !dac)
+ return false;
pin = alc_go_down_to_selector(codec, pin);
num = snd_hda_get_connections(codec, pin, srcs, ARRAY_SIZE(srcs));
for (i = 0; i < num; i++) {
@@ -2970,83 +3019,260 @@ static bool alc_auto_is_dac_reachable(struct hda_codec *codec,
static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin)
{
+ struct alc_spec *spec = codec->spec;
hda_nid_t sel = alc_go_down_to_selector(codec, pin);
- if (snd_hda_get_conn_list(codec, sel, NULL) == 1)
+ hda_nid_t nid, nid_found, srcs[5];
+ int i, num = snd_hda_get_connections(codec, sel, srcs,
+ ARRAY_SIZE(srcs));
+ if (num == 1)
return alc_auto_look_for_dac(codec, pin);
- return 0;
+ nid_found = 0;
+ for (i = 0; i < num; i++) {
+ if (srcs[i] == spec->mixer_nid)
+ continue;
+ nid = alc_auto_mix_to_dac(codec, srcs[i]);
+ if (nid && !alc_is_dac_already_used(codec, nid)) {
+ if (nid_found)
+ return 0;
+ nid_found = nid;
+ }
+ }
+ return nid_found;
}
-/* return 0 if no possible DAC is found, 1 if one or more found */
-static int alc_auto_fill_extra_dacs(struct hda_codec *codec, int num_outs,
- const hda_nid_t *pins, hda_nid_t *dacs)
+/* mark up volume and mute control NIDs: used during badness parsing and
+ * at creating actual controls
+ */
+static inline unsigned int get_ctl_pos(unsigned int data)
{
- int i;
+ hda_nid_t nid = get_amp_nid_(data);
+ unsigned int dir;
+ if (snd_BUG_ON(nid >= MAX_VOL_NIDS))
+ return 0;
+ dir = get_amp_direction_(data);
+ return (nid << 1) | dir;
+}
- if (num_outs && !dacs[0]) {
- dacs[0] = alc_auto_look_for_dac(codec, pins[0]);
- if (!dacs[0])
- return 0;
- }
+#define is_ctl_used(bits, data) \
+ test_bit(get_ctl_pos(data), bits)
+#define mark_ctl_usage(bits, data) \
+ set_bit(get_ctl_pos(data), bits)
- for (i = 1; i < num_outs; i++)
- dacs[i] = get_dac_if_single(codec, pins[i]);
- for (i = 1; i < num_outs; i++) {
+static void clear_vol_marks(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ memset(spec->vol_ctls, 0, sizeof(spec->vol_ctls));
+ memset(spec->sw_ctls, 0, sizeof(spec->sw_ctls));
+}
+
+/* badness definition */
+enum {
+ /* No primary DAC is found for the main output */
+ BAD_NO_PRIMARY_DAC = 0x10000,
+ /* No DAC is found for the extra output */
+ BAD_NO_DAC = 0x4000,
+ /* No possible multi-ios */
+ BAD_MULTI_IO = 0x103,
+ /* No individual DAC for extra output */
+ BAD_NO_EXTRA_DAC = 0x102,
+ /* No individual DAC for extra surrounds */
+ BAD_NO_EXTRA_SURR_DAC = 0x101,
+ /* Primary DAC shared with main surrounds */
+ BAD_SHARED_SURROUND = 0x100,
+ /* Primary DAC shared with main CLFE */
+ BAD_SHARED_CLFE = 0x10,
+ /* Primary DAC shared with extra surrounds */
+ BAD_SHARED_EXTRA_SURROUND = 0x10,
+ /* Volume widget is shared */
+ BAD_SHARED_VOL = 0x10,
+};
+
+static hda_nid_t alc_look_for_out_mute_nid(struct hda_codec *codec,
+ hda_nid_t pin, hda_nid_t dac);
+static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec,
+ hda_nid_t pin, hda_nid_t dac);
+
+static int eval_shared_vol_badness(struct hda_codec *codec, hda_nid_t pin,
+ hda_nid_t dac)
+{
+ struct alc_spec *spec = codec->spec;
+ hda_nid_t nid;
+ unsigned int val;
+ int badness = 0;
+
+ nid = alc_look_for_out_vol_nid(codec, pin, dac);
+ if (nid) {
+ val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+ if (is_ctl_used(spec->vol_ctls, nid))
+ badness += BAD_SHARED_VOL;
+ else
+ mark_ctl_usage(spec->vol_ctls, val);
+ } else
+ badness += BAD_SHARED_VOL;
+ nid = alc_look_for_out_mute_nid(codec, pin, dac);
+ if (nid) {
+ unsigned int wid_type = get_wcaps_type(get_wcaps(codec, nid));
+ if (wid_type == AC_WID_PIN || wid_type == AC_WID_AUD_OUT)
+ val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+ else
+ val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT);
+ if (is_ctl_used(spec->sw_ctls, val))
+ badness += BAD_SHARED_VOL;
+ else
+ mark_ctl_usage(spec->sw_ctls, val);
+ } else
+ badness += BAD_SHARED_VOL;
+ return badness;
+}
+
+struct badness_table {
+ int no_primary_dac; /* no primary DAC */
+ int no_dac; /* no secondary DACs */
+ int shared_primary; /* primary DAC is shared with main output */
+ int shared_surr; /* secondary DAC shared with main or primary */
+ int shared_clfe; /* third DAC shared with main or primary */
+ int shared_surr_main; /* secondary DAC sahred with main/DAC0 */
+};
+
+static struct badness_table main_out_badness = {
+ .no_primary_dac = BAD_NO_PRIMARY_DAC,
+ .no_dac = BAD_NO_DAC,
+ .shared_primary = BAD_NO_PRIMARY_DAC,
+ .shared_surr = BAD_SHARED_SURROUND,
+ .shared_clfe = BAD_SHARED_CLFE,
+ .shared_surr_main = BAD_SHARED_SURROUND,
+};
+
+static struct badness_table extra_out_badness = {
+ .no_primary_dac = BAD_NO_DAC,
+ .no_dac = BAD_NO_DAC,
+ .shared_primary = BAD_NO_EXTRA_DAC,
+ .shared_surr = BAD_SHARED_EXTRA_SURROUND,
+ .shared_clfe = BAD_SHARED_EXTRA_SURROUND,
+ .shared_surr_main = BAD_NO_EXTRA_SURR_DAC,
+};
+
+/* try to assign DACs to pins and return the resultant badness */
+static int alc_auto_fill_dacs(struct hda_codec *codec, int num_outs,
+ const hda_nid_t *pins, hda_nid_t *dacs,
+ const struct badness_table *bad)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, j;
+ int badness = 0;
+ hda_nid_t dac;
+
+ if (!num_outs)
+ return 0;
+
+ for (i = 0; i < num_outs; i++) {
+ hda_nid_t pin = pins[i];
if (!dacs[i])
- dacs[i] = alc_auto_look_for_dac(codec, pins[i]);
+ dacs[i] = alc_auto_look_for_dac(codec, pin);
+ if (!dacs[i] && !i) {
+ for (j = 1; j < num_outs; j++) {
+ if (alc_auto_is_dac_reachable(codec, pin, dacs[j])) {
+ dacs[0] = dacs[j];
+ dacs[j] = 0;
+ break;
+ }
+ }
+ }
+ dac = dacs[i];
+ if (!dac) {
+ if (alc_auto_is_dac_reachable(codec, pin, dacs[0]))
+ dac = dacs[0];
+ else if (cfg->line_outs > i &&
+ alc_auto_is_dac_reachable(codec, pin,
+ spec->private_dac_nids[i]))
+ dac = spec->private_dac_nids[i];
+ if (dac) {
+ if (!i)
+ badness += bad->shared_primary;
+ else if (i == 1)
+ badness += bad->shared_surr;
+ else
+ badness += bad->shared_clfe;
+ } else if (alc_auto_is_dac_reachable(codec, pin,
+ spec->private_dac_nids[0])) {
+ dac = spec->private_dac_nids[0];
+ badness += bad->shared_surr_main;
+ } else if (!i)
+ badness += bad->no_primary_dac;
+ else
+ badness += bad->no_dac;
+ }
+ if (dac)
+ badness += eval_shared_vol_badness(codec, pin, dac);
}
- return 1;
+
+ return badness;
}
static int alc_auto_fill_multi_ios(struct hda_codec *codec,
- unsigned int location, int offset);
-static hda_nid_t alc_look_for_out_vol_nid(struct hda_codec *codec,
- hda_nid_t pin, hda_nid_t dac);
+ hda_nid_t reference_pin,
+ bool hardwired, int offset);
+
+static bool alc_map_singles(struct hda_codec *codec, int outs,
+ const hda_nid_t *pins, hda_nid_t *dacs)
+{
+ int i;
+ bool found = false;
+ for (i = 0; i < outs; i++) {
+ if (dacs[i])
+ continue;
+ dacs[i] = get_dac_if_single(codec, pins[i]);
+ if (dacs[i])
+ found = true;
+ }
+ return found;
+}
/* fill in the dac_nids table from the parsed pin configuration */
-static int alc_auto_fill_dac_nids(struct hda_codec *codec)
+static int fill_and_eval_dacs(struct hda_codec *codec,
+ bool fill_hardwired,
+ bool fill_mio_first)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
- unsigned int location, defcfg;
- int num_pins;
- bool redone = false;
- int i;
+ int i, err, badness;
- again:
/* set num_dacs once to full for alc_auto_look_for_dac() */
spec->multiout.num_dacs = cfg->line_outs;
- spec->multiout.hp_out_nid[0] = 0;
- spec->multiout.extra_out_nid[0] = 0;
- memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
spec->multiout.dac_nids = spec->private_dac_nids;
+ memset(spec->private_dac_nids, 0, sizeof(spec->private_dac_nids));
+ memset(spec->multiout.hp_out_nid, 0, sizeof(spec->multiout.hp_out_nid));
+ memset(spec->multiout.extra_out_nid, 0, sizeof(spec->multiout.extra_out_nid));
spec->multi_ios = 0;
+ clear_vol_marks(codec);
+ badness = 0;
/* fill hard-wired DACs first */
- if (!redone) {
- for (i = 0; i < cfg->line_outs; i++)
- spec->private_dac_nids[i] =
- get_dac_if_single(codec, cfg->line_out_pins[i]);
- if (cfg->hp_outs)
- spec->multiout.hp_out_nid[0] =
- get_dac_if_single(codec, cfg->hp_pins[0]);
- if (cfg->speaker_outs)
- spec->multiout.extra_out_nid[0] =
- get_dac_if_single(codec, cfg->speaker_pins[0]);
+ if (fill_hardwired) {
+ bool mapped;
+ do {
+ mapped = alc_map_singles(codec, cfg->line_outs,
+ cfg->line_out_pins,
+ spec->private_dac_nids);
+ mapped |= alc_map_singles(codec, cfg->hp_outs,
+ cfg->hp_pins,
+ spec->multiout.hp_out_nid);
+ mapped |= alc_map_singles(codec, cfg->speaker_outs,
+ cfg->speaker_pins,
+ spec->multiout.extra_out_nid);
+ if (fill_mio_first && cfg->line_outs == 1 &&
+ cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+ err = alc_auto_fill_multi_ios(codec, cfg->line_out_pins[0], true, 0);
+ if (!err)
+ mapped = true;
+ }
+ } while (mapped);
}
- for (i = 0; i < cfg->line_outs; i++) {
- hda_nid_t pin = cfg->line_out_pins[i];
- if (spec->private_dac_nids[i])
- continue;
- spec->private_dac_nids[i] = alc_auto_look_for_dac(codec, pin);
- if (!spec->private_dac_nids[i] && !redone) {
- /* if we can't find primary DACs, re-probe without
- * checking the hard-wired DACs
- */
- redone = true;
- goto again;
- }
- }
+ badness += alc_auto_fill_dacs(codec, cfg->line_outs, cfg->line_out_pins,
+ spec->private_dac_nids,
+ &main_out_badness);
/* re-count num_dacs and squash invalid entries */
spec->multiout.num_dacs = 0;
@@ -3061,30 +3287,144 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
}
}
- if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+ if (fill_mio_first &&
+ cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
/* try to fill multi-io first */
- defcfg = snd_hda_codec_get_pincfg(codec, cfg->line_out_pins[0]);
- location = get_defcfg_location(defcfg);
-
- num_pins = alc_auto_fill_multi_ios(codec, location, 0);
- if (num_pins > 0) {
- spec->multi_ios = num_pins;
- spec->ext_channel_count = 2;
- spec->multiout.num_dacs = num_pins + 1;
- }
+ err = alc_auto_fill_multi_ios(codec, cfg->line_out_pins[0], false, 0);
+ if (err < 0)
+ return err;
+ /* we don't count badness at this stage yet */
}
- if (cfg->line_out_type != AUTO_PIN_HP_OUT)
- alc_auto_fill_extra_dacs(codec, cfg->hp_outs, cfg->hp_pins,
- spec->multiout.hp_out_nid);
+ if (cfg->line_out_type != AUTO_PIN_HP_OUT) {
+ err = alc_auto_fill_dacs(codec, cfg->hp_outs, cfg->hp_pins,
+ spec->multiout.hp_out_nid,
+ &extra_out_badness);
+ if (err < 0)
+ return err;
+ badness += err;
+ }
if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
- int err = alc_auto_fill_extra_dacs(codec, cfg->speaker_outs,
- cfg->speaker_pins,
- spec->multiout.extra_out_nid);
- /* if no speaker volume is assigned, try again as the primary
- * output
- */
- if (!err && cfg->speaker_outs > 0 &&
+ err = alc_auto_fill_dacs(codec, cfg->speaker_outs,
+ cfg->speaker_pins,
+ spec->multiout.extra_out_nid,
+ &extra_out_badness);
+ if (err < 0)
+ return err;
+ badness += err;
+ }
+ if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+ err = alc_auto_fill_multi_ios(codec, cfg->line_out_pins[0], false, 0);
+ if (err < 0)
+ return err;
+ badness += err;
+ }
+ if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
+ /* try multi-ios with HP + inputs */
+ int offset = 0;
+ if (cfg->line_outs >= 3)
+ offset = 1;
+ err = alc_auto_fill_multi_ios(codec, cfg->hp_pins[0], false,
+ offset);
+ if (err < 0)
+ return err;
+ badness += err;
+ }
+
+ if (spec->multi_ios == 2) {
+ for (i = 0; i < 2; i++)
+ spec->private_dac_nids[spec->multiout.num_dacs++] =
+ spec->multi_io[i].dac;
+ spec->ext_channel_count = 2;
+ } else if (spec->multi_ios) {
+ spec->multi_ios = 0;
+ badness += BAD_MULTI_IO;
+ }
+
+ return badness;
+}
+
+#define DEBUG_BADNESS
+
+#ifdef DEBUG_BADNESS
+#define debug_badness snd_printdd
+#else
+#define debug_badness(...)
+#endif
+
+static void debug_show_configs(struct alc_spec *spec, struct auto_pin_cfg *cfg)
+{
+ debug_badness("multi_outs = %x/%x/%x/%x : %x/%x/%x/%x\n",
+ cfg->line_out_pins[0], cfg->line_out_pins[1],
+ cfg->line_out_pins[2], cfg->line_out_pins[2],
+ spec->multiout.dac_nids[0],
+ spec->multiout.dac_nids[1],
+ spec->multiout.dac_nids[2],
+ spec->multiout.dac_nids[3]);
+ if (spec->multi_ios > 0)
+ debug_badness("multi_ios(%d) = %x/%x : %x/%x\n",
+ spec->multi_ios,
+ spec->multi_io[0].pin, spec->multi_io[1].pin,
+ spec->multi_io[0].dac, spec->multi_io[1].dac);
+ debug_badness("hp_outs = %x/%x/%x/%x : %x/%x/%x/%x\n",
+ cfg->hp_pins[0], cfg->hp_pins[1],
+ cfg->hp_pins[2], cfg->hp_pins[2],
+ spec->multiout.hp_out_nid[0],
+ spec->multiout.hp_out_nid[1],
+ spec->multiout.hp_out_nid[2],
+ spec->multiout.hp_out_nid[3]);
+ debug_badness("spk_outs = %x/%x/%x/%x : %x/%x/%x/%x\n",
+ cfg->speaker_pins[0], cfg->speaker_pins[1],
+ cfg->speaker_pins[2], cfg->speaker_pins[3],
+ spec->multiout.extra_out_nid[0],
+ spec->multiout.extra_out_nid[1],
+ spec->multiout.extra_out_nid[2],
+ spec->multiout.extra_out_nid[3]);
+}
+
+static int alc_auto_fill_dac_nids(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ struct auto_pin_cfg *best_cfg;
+ int best_badness = INT_MAX;
+ int badness;
+ bool fill_hardwired = true, fill_mio_first = true;
+ bool best_wired = true, best_mio = true;
+ bool hp_spk_swapped = false;
+
+ best_cfg = kmalloc(sizeof(*best_cfg), GFP_KERNEL);
+ if (!best_cfg)
+ return -ENOMEM;
+ *best_cfg = *cfg;
+
+ for (;;) {
+ badness = fill_and_eval_dacs(codec, fill_hardwired,
+ fill_mio_first);
+ if (badness < 0)
+ return badness;
+ debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n",
+ cfg->line_out_type, fill_hardwired, fill_mio_first,
+ badness);
+ debug_show_configs(spec, cfg);
+ if (badness < best_badness) {
+ best_badness = badness;
+ *best_cfg = *cfg;
+ best_wired = fill_hardwired;
+ best_mio = fill_mio_first;
+ }
+ if (!badness)
+ break;
+ fill_mio_first = !fill_mio_first;
+ if (!fill_mio_first)
+ continue;
+ fill_hardwired = !fill_hardwired;
+ if (!fill_hardwired)
+ continue;
+ if (hp_spk_swapped)
+ break;
+ hp_spk_swapped = true;
+ if (cfg->speaker_outs > 0 &&
cfg->line_out_type == AUTO_PIN_HP_OUT) {
cfg->hp_outs = cfg->line_outs;
memcpy(cfg->hp_pins, cfg->line_out_pins,
@@ -3095,45 +3435,45 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec)
cfg->speaker_outs = 0;
memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins));
cfg->line_out_type = AUTO_PIN_SPEAKER_OUT;
- redone = false;
- goto again;
- }
+ fill_hardwired = true;
+ continue;
+ }
+ if (cfg->hp_outs > 0 &&
+ cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
+ cfg->speaker_outs = cfg->line_outs;
+ memcpy(cfg->speaker_pins, cfg->line_out_pins,
+ sizeof(cfg->speaker_pins));
+ cfg->line_outs = cfg->hp_outs;
+ memcpy(cfg->line_out_pins, cfg->hp_pins,
+ sizeof(cfg->hp_pins));
+ cfg->hp_outs = 0;
+ memset(cfg->hp_pins, 0, sizeof(cfg->hp_pins));
+ cfg->line_out_type = AUTO_PIN_HP_OUT;
+ fill_hardwired = true;
+ continue;
+ }
+ break;
}
- if (!spec->multi_ios &&
- cfg->line_out_type == AUTO_PIN_SPEAKER_OUT &&
- cfg->hp_outs) {
- /* try multi-ios with HP + inputs */
- defcfg = snd_hda_codec_get_pincfg(codec, cfg->hp_pins[0]);
- location = get_defcfg_location(defcfg);
-
- num_pins = alc_auto_fill_multi_ios(codec, location, 1);
- if (num_pins > 0) {
- spec->multi_ios = num_pins;
- spec->ext_channel_count = 2;
- spec->multiout.num_dacs = num_pins + 1;
- }
+ if (badness) {
+ *cfg = *best_cfg;
+ fill_and_eval_dacs(codec, best_wired, best_mio);
}
+ debug_badness("==> Best config: lo_type=%d, wired=%d, mio=%d\n",
+ cfg->line_out_type, best_wired, best_mio);
+ debug_show_configs(spec, cfg);
if (cfg->line_out_pins[0])
spec->vmaster_nid =
alc_look_for_out_vol_nid(codec, cfg->line_out_pins[0],
spec->multiout.dac_nids[0]);
- return 0;
-}
-static inline unsigned int get_ctl_pos(unsigned int data)
-{
- hda_nid_t nid = get_amp_nid_(data);
- unsigned int dir = get_amp_direction_(data);
- return (nid << 1) | dir;
+ /* clear the bitmap flags for creating controls */
+ clear_vol_marks(codec);
+ kfree(best_cfg);
+ return 0;
}
-#define is_ctl_used(bits, data) \
- test_bit(get_ctl_pos(data), bits)
-#define mark_ctl_usage(bits, data) \
- set_bit(get_ctl_pos(data), bits)
-
static int alc_auto_add_vol_ctl(struct hda_codec *codec,
const char *pfx, int cidx,
hda_nid_t nid, unsigned int chs)
@@ -3233,7 +3573,7 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
int i, err, noutputs;
noutputs = cfg->line_outs;
- if (spec->multi_ios > 0)
+ if (spec->multi_ios > 0 && cfg->line_outs < 3)
noutputs += spec->multi_ios;
for (i = 0; i < noutputs; i++) {
@@ -3245,14 +3585,17 @@ static int alc_auto_create_multi_out_ctls(struct hda_codec *codec,
dac = spec->multiout.dac_nids[i];
if (!dac)
continue;
- if (i >= cfg->line_outs)
+ if (i >= cfg->line_outs) {
pin = spec->multi_io[i - 1].pin;
- else
+ index = 0;
+ name = channel_name[i];
+ } else {
pin = cfg->line_out_pins[i];
+ name = alc_get_line_out_pfx(spec, i, true, &index);
+ }
sw = alc_look_for_out_mute_nid(codec, pin, dac);
vol = alc_look_for_out_vol_nid(codec, pin, dac);
- name = alc_get_line_out_pfx(spec, i, true, &index);
if (!name || !strcmp(name, "CLFE")) {
/* Center/LFE */
err = alc_auto_add_vol_ctl(codec, "Center", 0, vol, 1);
@@ -3349,41 +3692,31 @@ static int alc_auto_create_extra_outs(struct hda_codec *codec, int num_pins,
return alc_auto_create_extra_out(codec, *pins, dac, pfx, 0);
}
- if (dacs[num_pins - 1]) {
- /* OK, we have a multi-output system with individual volumes */
- for (i = 0; i < num_pins; i++) {
- if (num_pins >= 3) {
- snprintf(name, sizeof(name), "%s %s",
- pfx, channel_name[i]);
- err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
- name, 0);
- } else {
- err = alc_auto_create_extra_out(codec, pins[i], dacs[i],
- pfx, i);
- }
- if (err < 0)
- return err;
- }
- return 0;
- }
-
- /* Let's create a bind-controls */
- ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_sw);
- if (!ctl)
- return -ENOMEM;
- n = 0;
for (i = 0; i < num_pins; i++) {
- if (get_wcaps(codec, pins[i]) & AC_WCAP_OUT_AMP)
- ctl->values[n++] =
- HDA_COMPOSE_AMP_VAL(pins[i], 3, 0, HDA_OUTPUT);
- }
- if (n) {
- snprintf(name, sizeof(name), "%s Playback Switch", pfx);
- err = add_control(spec, ALC_CTL_BIND_SW, name, 0, (long)ctl);
+ hda_nid_t dac;
+ if (dacs[num_pins - 1])
+ dac = dacs[i]; /* with individual volumes */
+ else
+ dac = 0;
+ if (num_pins == 2 && i == 1 && !strcmp(pfx, "Speaker")) {
+ err = alc_auto_create_extra_out(codec, pins[i], dac,
+ "Bass Speaker", 0);
+ } else if (num_pins >= 3) {
+ snprintf(name, sizeof(name), "%s %s",
+ pfx, channel_name[i]);
+ err = alc_auto_create_extra_out(codec, pins[i], dac,
+ name, 0);
+ } else {
+ err = alc_auto_create_extra_out(codec, pins[i], dac,
+ pfx, i);
+ }
if (err < 0)
return err;
}
+ if (dacs[num_pins - 1])
+ return 0;
+ /* Let's create a bind-controls for volumes */
ctl = new_bind_ctl(codec, num_pins, &snd_hda_bind_vol);
if (!ctl)
return -ENOMEM;
@@ -3519,58 +3852,111 @@ static void alc_auto_init_extra_out(struct hda_codec *codec)
}
}
+/* check whether the given pin can be a multi-io pin */
+static bool can_be_multiio_pin(struct hda_codec *codec,
+ unsigned int location, hda_nid_t nid)
+{
+ unsigned int defcfg, caps;
+
+ defcfg = snd_hda_codec_get_pincfg(codec, nid);
+ if (get_defcfg_connect(defcfg) != AC_JACK_PORT_COMPLEX)
+ return false;
+ if (location && get_defcfg_location(defcfg) != location)
+ return false;
+ caps = snd_hda_query_pin_caps(codec, nid);
+ if (!(caps & AC_PINCAP_OUT))
+ return false;
+ return true;
+}
+
/*
* multi-io helper
+ *
+ * When hardwired is set, try to fill ony hardwired pins, and returns
+ * zero if any pins are filled, non-zero if nothing found.
+ * When hardwired is off, try to fill possible input pins, and returns
+ * the badness value.
*/
static int alc_auto_fill_multi_ios(struct hda_codec *codec,
- unsigned int location,
- int offset)
+ hda_nid_t reference_pin,
+ bool hardwired, int offset)
{
struct alc_spec *spec = codec->spec;
struct auto_pin_cfg *cfg = &spec->autocfg;
- hda_nid_t prime_dac = spec->private_dac_nids[0];
- int type, i, dacs, num_pins = 0;
+ int type, i, j, dacs, num_pins, old_pins;
+ unsigned int defcfg = snd_hda_codec_get_pincfg(codec, reference_pin);
+ unsigned int location = get_defcfg_location(defcfg);
+ int badness = 0;
+
+ old_pins = spec->multi_ios;
+ if (old_pins >= 2)
+ goto end_fill;
+
+ num_pins = 0;
+ for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) {
+ for (i = 0; i < cfg->num_inputs; i++) {
+ if (cfg->inputs[i].type != type)
+ continue;
+ if (can_be_multiio_pin(codec, location,
+ cfg->inputs[i].pin))
+ num_pins++;
+ }
+ }
+ if (num_pins < 2)
+ goto end_fill;
dacs = spec->multiout.num_dacs;
for (type = AUTO_PIN_LINE_IN; type >= AUTO_PIN_MIC; type--) {
for (i = 0; i < cfg->num_inputs; i++) {
hda_nid_t nid = cfg->inputs[i].pin;
hda_nid_t dac = 0;
- unsigned int defcfg, caps;
+
if (cfg->inputs[i].type != type)
continue;
- defcfg = snd_hda_codec_get_pincfg(codec, nid);
- if (get_defcfg_connect(defcfg) != AC_JACK_PORT_COMPLEX)
- continue;
- if (location && get_defcfg_location(defcfg) != location)
+ if (!can_be_multiio_pin(codec, location, nid))
continue;
- caps = snd_hda_query_pin_caps(codec, nid);
- if (!(caps & AC_PINCAP_OUT))
+ for (j = 0; j < spec->multi_ios; j++) {
+ if (nid == spec->multi_io[j].pin)
+ break;
+ }
+ if (j < spec->multi_ios)
continue;
- if (offset && offset + num_pins < dacs) {
- dac = spec->private_dac_nids[offset + num_pins];
+
+ if (offset && offset + spec->multi_ios < dacs) {
+ dac = spec->private_dac_nids[offset + spec->multi_ios];
if (!alc_auto_is_dac_reachable(codec, nid, dac))
dac = 0;
}
- if (!dac)
+ if (hardwired)
+ dac = get_dac_if_single(codec, nid);
+ else if (!dac)
dac = alc_auto_look_for_dac(codec, nid);
- if (!dac)
+ if (!dac) {
+ badness++;
continue;
- spec->multi_io[num_pins].pin = nid;
- spec->multi_io[num_pins].dac = dac;
- num_pins++;
- spec->private_dac_nids[spec->multiout.num_dacs++] = dac;
+ }
+ spec->multi_io[spec->multi_ios].pin = nid;
+ spec->multi_io[spec->multi_ios].dac = dac;
+ spec->multi_ios++;
+ if (spec->multi_ios >= 2)
+ break;
}
}
- spec->multiout.num_dacs = dacs;
- if (num_pins < 2) {
- /* clear up again */
- memset(spec->private_dac_nids + dacs, 0,
- sizeof(hda_nid_t) * (AUTO_CFG_MAX_OUTS - dacs));
- spec->private_dac_nids[0] = prime_dac;
- return 0;
+ end_fill:
+ if (badness)
+ badness = BAD_MULTI_IO;
+ if (old_pins == spec->multi_ios) {
+ if (hardwired)
+ return 1; /* nothing found */
+ else
+ return badness; /* no badness if nothing found */
}
- return num_pins;
+ if (!hardwired && spec->multi_ios < 2) {
+ spec->multi_ios = old_pins;
+ return badness;
+ }
+
+ return 0;
}
static int alc_auto_ch_mode_info(struct snd_kcontrol *kcontrol,
@@ -3768,7 +4154,7 @@ static void alc_auto_init_input_src(struct hda_codec *codec)
else
nums = spec->num_adc_nids;
for (c = 0; c < nums; c++)
- alc_mux_select(codec, 0, spec->cur_mux[c], true);
+ alc_mux_select(codec, c, spec->cur_mux[c], true);
}
/* add mic boosts if needed */
@@ -3904,7 +4290,6 @@ static void set_capture_mixer(struct hda_codec *codec)
static void alc_auto_init_std(struct hda_codec *codec)
{
struct alc_spec *spec = codec->spec;
- spec->use_jack_tbl = 1;
alc_auto_init_multi_out(codec);
alc_auto_init_extra_out(codec);
alc_auto_init_analog_input(codec);
@@ -3926,6 +4311,7 @@ static const struct snd_pci_quirk beep_white_list[] = {
SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1),
SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1),
SND_PCI_QUIRK(0x1043, 0x834a, "EeePC", 1),
+ SND_PCI_QUIRK(0x1458, 0xa002, "GA-MA790X", 1),
SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1),
{}
};
@@ -4025,6 +4411,9 @@ static int alc_parse_auto_config(struct hda_codec *codec,
if (spec->kctls.list)
add_mixer(spec, spec->kctls.list);
+ if (!spec->no_analog && !spec->cap_mixer)
+ set_capture_mixer(codec);
+
return 1;
}
@@ -4035,26 +4424,47 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids);
}
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list alc880_loopbacks[] = {
- { 0x0b, HDA_INPUT, 0 },
- { 0x0b, HDA_INPUT, 1 },
- { 0x0b, HDA_INPUT, 2 },
- { 0x0b, HDA_INPUT, 3 },
- { 0x0b, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
-
/*
* ALC880 fix-ups
*/
enum {
+ ALC880_FIXUP_GPIO1,
ALC880_FIXUP_GPIO2,
ALC880_FIXUP_MEDION_RIM,
+ ALC880_FIXUP_LG,
+ ALC880_FIXUP_W810,
+ ALC880_FIXUP_EAPD_COEF,
+ ALC880_FIXUP_TCL_S700,
+ ALC880_FIXUP_VOL_KNOB,
+ ALC880_FIXUP_FUJITSU,
+ ALC880_FIXUP_F1734,
+ ALC880_FIXUP_UNIWILL,
+ ALC880_FIXUP_UNIWILL_DIG,
+ ALC880_FIXUP_Z71V,
+ ALC880_FIXUP_3ST_BASE,
+ ALC880_FIXUP_3ST,
+ ALC880_FIXUP_3ST_DIG,
+ ALC880_FIXUP_5ST_BASE,
+ ALC880_FIXUP_5ST,
+ ALC880_FIXUP_5ST_DIG,
+ ALC880_FIXUP_6ST_BASE,
+ ALC880_FIXUP_6ST,
+ ALC880_FIXUP_6ST_DIG,
};
+/* enable the volume-knob widget support on NID 0x21 */
+static void alc880_fixup_vol_knob(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ if (action == ALC_FIXUP_ACT_PROBE)
+ snd_hda_jack_detect_enable(codec, 0x21, ALC_DCVOL_EVENT);
+}
+
static const struct alc_fixup alc880_fixups[] = {
+ [ALC880_FIXUP_GPIO1] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = alc_gpio1_init_verbs,
+ },
[ALC880_FIXUP_GPIO2] = {
.type = ALC_FIXUP_VERBS,
.v.verbs = alc_gpio2_init_verbs,
@@ -4069,40 +4479,323 @@ static const struct alc_fixup alc880_fixups[] = {
.chained = true,
.chain_id = ALC880_FIXUP_GPIO2,
},
+ [ALC880_FIXUP_LG] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ /* disable bogus unused pins */
+ { 0x16, 0x411111f0 },
+ { 0x18, 0x411111f0 },
+ { 0x1a, 0x411111f0 },
+ { }
+ }
+ },
+ [ALC880_FIXUP_W810] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ /* disable bogus unused pins */
+ { 0x17, 0x411111f0 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_GPIO2,
+ },
+ [ALC880_FIXUP_EAPD_COEF] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* change to EAPD mode */
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x3060 },
+ {}
+ },
+ },
+ [ALC880_FIXUP_TCL_S700] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ /* change to EAPD mode */
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x3070 },
+ {}
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_GPIO2,
+ },
+ [ALC880_FIXUP_VOL_KNOB] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc880_fixup_vol_knob,
+ },
+ [ALC880_FIXUP_FUJITSU] = {
+ /* override all pins as BIOS on old Amilo is broken */
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x14, 0x0121411f }, /* HP */
+ { 0x15, 0x99030120 }, /* speaker */
+ { 0x16, 0x99030130 }, /* bass speaker */
+ { 0x17, 0x411111f0 }, /* N/A */
+ { 0x18, 0x411111f0 }, /* N/A */
+ { 0x19, 0x01a19950 }, /* mic-in */
+ { 0x1a, 0x411111f0 }, /* N/A */
+ { 0x1b, 0x411111f0 }, /* N/A */
+ { 0x1c, 0x411111f0 }, /* N/A */
+ { 0x1d, 0x411111f0 }, /* N/A */
+ { 0x1e, 0x01454140 }, /* SPDIF out */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_VOL_KNOB,
+ },
+ [ALC880_FIXUP_F1734] = {
+ /* almost compatible with FUJITSU, but no bass and SPDIF */
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x14, 0x0121411f }, /* HP */
+ { 0x15, 0x99030120 }, /* speaker */
+ { 0x16, 0x411111f0 }, /* N/A */
+ { 0x17, 0x411111f0 }, /* N/A */
+ { 0x18, 0x411111f0 }, /* N/A */
+ { 0x19, 0x01a19950 }, /* mic-in */
+ { 0x1a, 0x411111f0 }, /* N/A */
+ { 0x1b, 0x411111f0 }, /* N/A */
+ { 0x1c, 0x411111f0 }, /* N/A */
+ { 0x1d, 0x411111f0 }, /* N/A */
+ { 0x1e, 0x411111f0 }, /* N/A */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_VOL_KNOB,
+ },
+ [ALC880_FIXUP_UNIWILL] = {
+ /* need to fix HP and speaker pins to be parsed correctly */
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x14, 0x0121411f }, /* HP */
+ { 0x15, 0x99030120 }, /* speaker */
+ { 0x16, 0x99030130 }, /* bass speaker */
+ { }
+ },
+ },
+ [ALC880_FIXUP_UNIWILL_DIG] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ /* disable bogus unused pins */
+ { 0x17, 0x411111f0 },
+ { 0x19, 0x411111f0 },
+ { 0x1b, 0x411111f0 },
+ { 0x1f, 0x411111f0 },
+ { }
+ }
+ },
+ [ALC880_FIXUP_Z71V] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ /* set up the whole pins as BIOS is utterly broken */
+ { 0x14, 0x99030120 }, /* speaker */
+ { 0x15, 0x0121411f }, /* HP */
+ { 0x16, 0x411111f0 }, /* N/A */
+ { 0x17, 0x411111f0 }, /* N/A */
+ { 0x18, 0x01a19950 }, /* mic-in */
+ { 0x19, 0x411111f0 }, /* N/A */
+ { 0x1a, 0x01813031 }, /* line-in */
+ { 0x1b, 0x411111f0 }, /* N/A */
+ { 0x1c, 0x411111f0 }, /* N/A */
+ { 0x1d, 0x411111f0 }, /* N/A */
+ { 0x1e, 0x0144111e }, /* SPDIF */
+ { }
+ }
+ },
+ [ALC880_FIXUP_3ST_BASE] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x14, 0x01014010 }, /* line-out */
+ { 0x15, 0x411111f0 }, /* N/A */
+ { 0x16, 0x411111f0 }, /* N/A */
+ { 0x17, 0x411111f0 }, /* N/A */
+ { 0x18, 0x01a19c30 }, /* mic-in */
+ { 0x19, 0x0121411f }, /* HP */
+ { 0x1a, 0x01813031 }, /* line-in */
+ { 0x1b, 0x02a19c40 }, /* front-mic */
+ { 0x1c, 0x411111f0 }, /* N/A */
+ { 0x1d, 0x411111f0 }, /* N/A */
+ /* 0x1e is filled in below */
+ { 0x1f, 0x411111f0 }, /* N/A */
+ { }
+ }
+ },
+ [ALC880_FIXUP_3ST] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x1e, 0x411111f0 }, /* N/A */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_3ST_BASE,
+ },
+ [ALC880_FIXUP_3ST_DIG] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x1e, 0x0144111e }, /* SPDIF */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_3ST_BASE,
+ },
+ [ALC880_FIXUP_5ST_BASE] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x14, 0x01014010 }, /* front */
+ { 0x15, 0x411111f0 }, /* N/A */
+ { 0x16, 0x01011411 }, /* CLFE */
+ { 0x17, 0x01016412 }, /* surr */
+ { 0x18, 0x01a19c30 }, /* mic-in */
+ { 0x19, 0x0121411f }, /* HP */
+ { 0x1a, 0x01813031 }, /* line-in */
+ { 0x1b, 0x02a19c40 }, /* front-mic */
+ { 0x1c, 0x411111f0 }, /* N/A */
+ { 0x1d, 0x411111f0 }, /* N/A */
+ /* 0x1e is filled in below */
+ { 0x1f, 0x411111f0 }, /* N/A */
+ { }
+ }
+ },
+ [ALC880_FIXUP_5ST] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x1e, 0x411111f0 }, /* N/A */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_5ST_BASE,
+ },
+ [ALC880_FIXUP_5ST_DIG] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x1e, 0x0144111e }, /* SPDIF */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_5ST_BASE,
+ },
+ [ALC880_FIXUP_6ST_BASE] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x14, 0x01014010 }, /* front */
+ { 0x15, 0x01016412 }, /* surr */
+ { 0x16, 0x01011411 }, /* CLFE */
+ { 0x17, 0x01012414 }, /* side */
+ { 0x18, 0x01a19c30 }, /* mic-in */
+ { 0x19, 0x02a19c40 }, /* front-mic */
+ { 0x1a, 0x01813031 }, /* line-in */
+ { 0x1b, 0x0121411f }, /* HP */
+ { 0x1c, 0x411111f0 }, /* N/A */
+ { 0x1d, 0x411111f0 }, /* N/A */
+ /* 0x1e is filled in below */
+ { 0x1f, 0x411111f0 }, /* N/A */
+ { }
+ }
+ },
+ [ALC880_FIXUP_6ST] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x1e, 0x411111f0 }, /* N/A */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_6ST_BASE,
+ },
+ [ALC880_FIXUP_6ST_DIG] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x1e, 0x0144111e }, /* SPDIF */
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC880_FIXUP_6ST_BASE,
+ },
};
static const struct snd_pci_quirk alc880_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x1019, 0x0f69, "Coeus G610P", ALC880_FIXUP_W810),
+ SND_PCI_QUIRK(0x1043, 0x1964, "ASUS Z71V", ALC880_FIXUP_Z71V),
+ SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_FIXUP_GPIO1),
+ SND_PCI_QUIRK(0x1558, 0x5401, "Clevo GPIO2", ALC880_FIXUP_GPIO2),
+ SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", ALC880_FIXUP_EAPD_COEF),
+ SND_PCI_QUIRK(0x1584, 0x9050, "Uniwill", ALC880_FIXUP_UNIWILL_DIG),
+ SND_PCI_QUIRK(0x1584, 0x9054, "Uniwill", ALC880_FIXUP_F1734),
+ SND_PCI_QUIRK(0x1584, 0x9070, "Uniwill", ALC880_FIXUP_UNIWILL),
+ SND_PCI_QUIRK(0x1584, 0x9077, "Uniwill P53", ALC880_FIXUP_VOL_KNOB),
+ SND_PCI_QUIRK(0x161f, 0x203d, "W810", ALC880_FIXUP_W810),
SND_PCI_QUIRK(0x161f, 0x205d, "Medion Rim 2150", ALC880_FIXUP_MEDION_RIM),
+ SND_PCI_QUIRK(0x1734, 0x107c, "FSC F1734", ALC880_FIXUP_F1734),
+ SND_PCI_QUIRK(0x1734, 0x1094, "FSC Amilo M1451G", ALC880_FIXUP_FUJITSU),
+ SND_PCI_QUIRK(0x1734, 0x10ac, "FSC AMILO Xi 1526", ALC880_FIXUP_F1734),
+ SND_PCI_QUIRK(0x1734, 0x10b0, "FSC Amilo Pi1556", ALC880_FIXUP_FUJITSU),
+ SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG),
+ SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG),
+ SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG),
+ SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700),
+
+ /* Below is the copied entries from alc880_quirks.c.
+ * It's not quite sure whether BIOS sets the correct pin-config table
+ * on these machines, thus they are kept to be compatible with
+ * the old static quirks. Once when it's confirmed to work without
+ * these overrides, it'd be better to remove.
+ */
+ SND_PCI_QUIRK(0x1019, 0xa880, "ECS", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x1019, 0xa884, "Acer APFV", ALC880_FIXUP_6ST),
+ SND_PCI_QUIRK(0x1025, 0x0070, "ULI", ALC880_FIXUP_3ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0x0077, "ULI", ALC880_FIXUP_6ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0x0078, "ULI", ALC880_FIXUP_6ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0x0087, "ULI", ALC880_FIXUP_6ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_FIXUP_3ST_DIG),
+ SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_FIXUP_3ST),
+ SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_FIXUP_6ST_DIG),
+ SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_FIXUP_3ST),
+ SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_FIXUP_3ST),
+ SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_FIXUP_5ST),
+ SND_PCI_QUIRK(0x107b, 0x3033, "Gateway", ALC880_FIXUP_5ST),
+ SND_PCI_QUIRK(0x107b, 0x4039, "Gateway", ALC880_FIXUP_5ST),
+ SND_PCI_QUIRK(0x1297, 0xc790, "Shuttle ST20G5", ALC880_FIXUP_6ST_DIG),
+ SND_PCI_QUIRK(0x1458, 0xa102, "Gigabyte K8", ALC880_FIXUP_6ST_DIG),
+ SND_PCI_QUIRK(0x1462, 0x1150, "MSI", ALC880_FIXUP_6ST_DIG),
+ SND_PCI_QUIRK(0x1509, 0x925d, "FIC P4M", ALC880_FIXUP_6ST_DIG),
+ SND_PCI_QUIRK(0x1565, 0x8202, "Biostar", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x1695, 0x400d, "EPoX", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x1695, 0x4012, "EPox EP-5LDA", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x2668, 0x8086, NULL, ALC880_FIXUP_6ST_DIG), /* broken BIOS */
+ SND_PCI_QUIRK(0x8086, 0x2668, NULL, ALC880_FIXUP_6ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xa100, "Intel mobo", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xd400, "Intel mobo", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xd401, "Intel mobo", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xd402, "Intel mobo", ALC880_FIXUP_3ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe224, "Intel mobo", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe305, "Intel mobo", ALC880_FIXUP_3ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe308, "Intel mobo", ALC880_FIXUP_3ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_FIXUP_5ST_DIG),
+ /* default Intel */
+ SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_FIXUP_3ST),
+ SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_FIXUP_5ST_DIG),
+ SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_FIXUP_6ST_DIG),
{}
};
+static const struct alc_model_fixup alc880_fixup_models[] = {
+ {.id = ALC880_FIXUP_3ST, .name = "3stack"},
+ {.id = ALC880_FIXUP_3ST_DIG, .name = "3stack-digout"},
+ {.id = ALC880_FIXUP_5ST, .name = "5stack"},
+ {.id = ALC880_FIXUP_5ST_DIG, .name = "5stack-digout"},
+ {.id = ALC880_FIXUP_6ST, .name = "6stack"},
+ {.id = ALC880_FIXUP_6ST_DIG, .name = "6stack-digout"},
+ {}
+};
-/*
- * board setups
- */
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#define alc_board_config \
- snd_hda_check_board_config
-#define alc_board_codec_sid_config \
- snd_hda_check_board_codec_sid_config
-#include "alc_quirks.c"
-#else
-#define alc_board_config(codec, nums, models, tbl) -1
-#define alc_board_codec_sid_config(codec, nums, models, tbl) -1
-#define setup_preset(codec, x) /* NOP */
-#endif
/*
* OK, here we have finally the patch for ALC880
*/
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc880_quirks.c"
-#endif
-
static int patch_alc880(struct hda_codec *codec)
{
struct alc_spec *spec;
- int board_config;
int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
@@ -4114,47 +4807,14 @@ static int patch_alc880(struct hda_codec *codec)
spec->mixer_nid = 0x0b;
spec->need_dac_fix = 1;
- board_config = alc_board_config(codec, ALC880_MODEL_LAST,
- alc880_models, alc880_cfg_tbl);
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC_MODEL_AUTO;
- }
-
- if (board_config == ALC_MODEL_AUTO) {
- alc_pick_fixup(codec, NULL, alc880_fixup_tbl, alc880_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
-
- if (board_config == ALC_MODEL_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc880_parse_auto_config(codec);
- if (err < 0)
- goto error;
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
- else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using 3-stack mode...\n");
- board_config = ALC880_3ST;
- }
-#endif
- }
-
- if (board_config != ALC_MODEL_AUTO) {
- spec->vmaster_nid = 0x0c;
- setup_preset(codec, &alc880_presets[board_config]);
- }
-
- if (!spec->no_analog && !spec->adc_nids) {
- alc_auto_fill_adc_caps(codec);
- alc_rebuild_imux_for_auto_mic(codec);
- alc_remove_invalid_adc_nids(codec);
- }
+ alc_pick_fixup(codec, alc880_fixup_models, alc880_fixup_tbl,
+ alc880_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
+ /* automatic parse from the BIOS config */
+ err = alc880_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x1);
@@ -4163,15 +4823,9 @@ static int patch_alc880(struct hda_codec *codec)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC_MODEL_AUTO)
- spec->init_hook = alc_auto_init_std;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc880_loopbacks;
-#endif
+
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
return 0;
@@ -4191,49 +4845,115 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
return alc_parse_auto_config(codec, alc260_ignore, alc260_ssids);
}
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list alc260_loopbacks[] = {
- { 0x07, HDA_INPUT, 0 },
- { 0x07, HDA_INPUT, 1 },
- { 0x07, HDA_INPUT, 2 },
- { 0x07, HDA_INPUT, 3 },
- { 0x07, HDA_INPUT, 4 },
- { } /* end */
-};
-#endif
-
/*
* Pin config fixes
*/
enum {
- PINFIX_HP_DC5750,
+ ALC260_FIXUP_HP_DC5750,
+ ALC260_FIXUP_HP_PIN_0F,
+ ALC260_FIXUP_COEF,
+ ALC260_FIXUP_GPIO1,
+ ALC260_FIXUP_GPIO1_TOGGLE,
+ ALC260_FIXUP_REPLACER,
+ ALC260_FIXUP_HP_B1900,
};
+static void alc260_gpio1_automute(struct hda_codec *codec)
+{
+ struct alc_spec *spec = codec->spec;
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ spec->hp_jack_present);
+}
+
+static void alc260_fixup_gpio1_toggle(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ if (action == ALC_FIXUP_ACT_PROBE) {
+ /* although the machine has only one output pin, we need to
+ * toggle GPIO1 according to the jack state
+ */
+ spec->automute_hook = alc260_gpio1_automute;
+ spec->detect_hp = 1;
+ spec->automute_speaker = 1;
+ spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */
+ snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT);
+ spec->unsol_event = alc_sku_unsol_event;
+ add_verb(codec->spec, alc_gpio1_init_verbs);
+ }
+}
+
static const struct alc_fixup alc260_fixups[] = {
- [PINFIX_HP_DC5750] = {
+ [ALC260_FIXUP_HP_DC5750] = {
.type = ALC_FIXUP_PINS,
.v.pins = (const struct alc_pincfg[]) {
{ 0x11, 0x90130110 }, /* speaker */
{ }
}
},
+ [ALC260_FIXUP_HP_PIN_0F] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x0f, 0x01214000 }, /* HP */
+ { }
+ }
+ },
+ [ALC260_FIXUP_COEF] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x3040 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC260_FIXUP_HP_PIN_0F,
+ },
+ [ALC260_FIXUP_GPIO1] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = alc_gpio1_init_verbs,
+ },
+ [ALC260_FIXUP_GPIO1_TOGGLE] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc260_fixup_gpio1_toggle,
+ .chained = true,
+ .chain_id = ALC260_FIXUP_HP_PIN_0F,
+ },
+ [ALC260_FIXUP_REPLACER] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = (const struct hda_verb[]) {
+ { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 },
+ { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 },
+ { }
+ },
+ .chained = true,
+ .chain_id = ALC260_FIXUP_GPIO1_TOGGLE,
+ },
+ [ALC260_FIXUP_HP_B1900] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc260_fixup_gpio1_toggle,
+ .chained = true,
+ .chain_id = ALC260_FIXUP_COEF,
+ }
};
static const struct snd_pci_quirk alc260_fixup_tbl[] = {
- SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", PINFIX_HP_DC5750),
+ SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_FIXUP_GPIO1),
+ SND_PCI_QUIRK(0x1025, 0x007f, "Acer Aspire 9500", ALC260_FIXUP_COEF),
+ SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_FIXUP_GPIO1),
+ SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750),
+ SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900),
+ SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1),
+ SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER),
+ SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF),
{}
};
/*
*/
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc260_quirks.c"
-#endif
-
static int patch_alc260(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err, board_config;
+ int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -4243,47 +4963,13 @@ static int patch_alc260(struct hda_codec *codec)
spec->mixer_nid = 0x07;
- board_config = alc_board_config(codec, ALC260_MODEL_LAST,
- alc260_models, alc260_cfg_tbl);
- if (board_config < 0) {
- snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC_MODEL_AUTO;
- }
-
- if (board_config == ALC_MODEL_AUTO) {
- alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
-
- if (board_config == ALC_MODEL_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc260_parse_auto_config(codec);
- if (err < 0)
- goto error;
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
- else if (!err) {
- printk(KERN_INFO
- "hda_codec: Cannot set up configuration "
- "from BIOS. Using base mode...\n");
- board_config = ALC260_BASIC;
- }
-#endif
- }
-
- if (board_config != ALC_MODEL_AUTO) {
- setup_preset(codec, &alc260_presets[board_config]);
- spec->vmaster_nid = 0x08;
- }
-
- if (!spec->no_analog && !spec->adc_nids) {
- alc_auto_fill_adc_caps(codec);
- alc_rebuild_imux_for_auto_mic(codec);
- alc_remove_invalid_adc_nids(codec);
- }
+ alc_pick_fixup(codec, NULL, alc260_fixup_tbl, alc260_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
+ /* automatic parse from the BIOS config */
+ err = alc260_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x1);
@@ -4292,16 +4978,10 @@ static int patch_alc260(struct hda_codec *codec)
set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
}
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC_MODEL_AUTO)
- spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc260_loopbacks;
-#endif
+
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
return 0;
@@ -4322,9 +5002,6 @@ static int patch_alc260(struct hda_codec *codec)
* In addition, an independent DAC for the multi-playback (not used in this
* driver yet).
*/
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc882_loopbacks alc880_loopbacks
-#endif
/*
* Pin config fixes
@@ -4335,11 +5012,14 @@ enum {
ALC882_FIXUP_PB_M5210,
ALC882_FIXUP_ACER_ASPIRE_7736,
ALC882_FIXUP_ASUS_W90V,
+ ALC889_FIXUP_CD,
ALC889_FIXUP_VAIO_TT,
ALC888_FIXUP_EEE1601,
ALC882_FIXUP_EAPD,
ALC883_FIXUP_EAPD,
ALC883_FIXUP_ACER_EAPD,
+ ALC882_FIXUP_GPIO1,
+ ALC882_FIXUP_GPIO2,
ALC882_FIXUP_GPIO3,
ALC889_FIXUP_COEF,
ALC882_FIXUP_ASUS_W2JC,
@@ -4347,6 +5027,9 @@ enum {
ALC882_FIXUP_ACER_ASPIRE_8930G,
ALC882_FIXUP_ASPIRE_8930G_VERBS,
ALC885_FIXUP_MACPRO_GPIO,
+ ALC889_FIXUP_DAC_ROUTE,
+ ALC889_FIXUP_MBP_VREF,
+ ALC889_FIXUP_IMAC91_VREF,
};
static void alc889_fixup_coef(struct hda_codec *codec,
@@ -4400,6 +5083,76 @@ static void alc885_fixup_macpro_gpio(struct hda_codec *codec,
alc882_gpio_mute(codec, 1, 0);
}
+/* Fix the connection of some pins for ALC889:
+ * At least, Acer Aspire 5935 shows the connections to DAC3/4 don't
+ * work correctly (bko#42740)
+ */
+static void alc889_fixup_dac_route(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ if (action == ALC_FIXUP_ACT_PRE_PROBE) {
+ /* fake the connections during parsing the tree */
+ hda_nid_t conn1[2] = { 0x0c, 0x0d };
+ hda_nid_t conn2[2] = { 0x0e, 0x0f };
+ snd_hda_override_conn_list(codec, 0x14, 2, conn1);
+ snd_hda_override_conn_list(codec, 0x15, 2, conn1);
+ snd_hda_override_conn_list(codec, 0x18, 2, conn2);
+ snd_hda_override_conn_list(codec, 0x1a, 2, conn2);
+ } else if (action == ALC_FIXUP_ACT_PROBE) {
+ /* restore the connections */
+ hda_nid_t conn[5] = { 0x0c, 0x0d, 0x0e, 0x0f, 0x26 };
+ snd_hda_override_conn_list(codec, 0x14, 5, conn);
+ snd_hda_override_conn_list(codec, 0x15, 5, conn);
+ snd_hda_override_conn_list(codec, 0x18, 5, conn);
+ snd_hda_override_conn_list(codec, 0x1a, 5, conn);
+ }
+}
+
+/* Set VREF on HP pin */
+static void alc889_fixup_mbp_vref(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ static hda_nid_t nids[2] = { 0x14, 0x15 };
+ int i;
+
+ if (action != ALC_FIXUP_ACT_INIT)
+ return;
+ for (i = 0; i < ARRAY_SIZE(nids); i++) {
+ unsigned int val = snd_hda_codec_get_pincfg(codec, nids[i]);
+ if (get_defcfg_device(val) != AC_JACK_HP_OUT)
+ continue;
+ val = snd_hda_codec_read(codec, nids[i], 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ val |= AC_PINCTL_VREF_80;
+ snd_hda_codec_write(codec, nids[i], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ spec->keep_vref_in_automute = 1;
+ break;
+ }
+}
+
+/* Set VREF on speaker pins on imac91 */
+static void alc889_fixup_imac91_vref(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ static hda_nid_t nids[2] = { 0x18, 0x1a };
+ int i;
+
+ if (action != ALC_FIXUP_ACT_INIT)
+ return;
+ for (i = 0; i < ARRAY_SIZE(nids); i++) {
+ unsigned int val;
+ val = snd_hda_codec_read(codec, nids[i], 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ val |= AC_PINCTL_VREF_50;
+ snd_hda_codec_write(codec, nids[i], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ }
+ spec->keep_vref_in_automute = 1;
+}
+
static const struct alc_fixup alc882_fixups[] = {
[ALC882_FIXUP_ABIT_AW9D_MAX] = {
.type = ALC_FIXUP_PINS,
@@ -4436,6 +5189,13 @@ static const struct alc_fixup alc882_fixups[] = {
{ }
}
},
+ [ALC889_FIXUP_CD] = {
+ .type = ALC_FIXUP_PINS,
+ .v.pins = (const struct alc_pincfg[]) {
+ { 0x1c, 0x993301f0 }, /* CD */
+ { }
+ }
+ },
[ALC889_FIXUP_VAIO_TT] = {
.type = ALC_FIXUP_PINS,
.v.pins = (const struct alc_pincfg[]) {
@@ -4478,6 +5238,14 @@ static const struct alc_fixup alc882_fixups[] = {
{ }
}
},
+ [ALC882_FIXUP_GPIO1] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = alc_gpio1_init_verbs,
+ },
+ [ALC882_FIXUP_GPIO2] = {
+ .type = ALC_FIXUP_VERBS,
+ .v.verbs = alc_gpio2_init_verbs,
+ },
[ALC882_FIXUP_GPIO3] = {
.type = ALC_FIXUP_VERBS,
.v.verbs = alc_gpio3_init_verbs,
@@ -4547,6 +5315,22 @@ static const struct alc_fixup alc882_fixups[] = {
.type = ALC_FIXUP_FUNC,
.v.func = alc885_fixup_macpro_gpio,
},
+ [ALC889_FIXUP_DAC_ROUTE] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc889_fixup_dac_route,
+ },
+ [ALC889_FIXUP_MBP_VREF] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc889_fixup_mbp_vref,
+ .chained = true,
+ .chain_id = ALC882_FIXUP_GPIO1,
+ },
+ [ALC889_FIXUP_IMAC91_VREF] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc889_fixup_imac91_vref,
+ .chained = true,
+ .chain_id = ALC882_FIXUP_GPIO1,
+ },
};
static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -4571,6 +5355,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
ALC882_FIXUP_ACER_ASPIRE_4930G),
SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210),
+ SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE),
SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736),
SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V),
@@ -4579,14 +5364,30 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
/* All Apple entries are in codec SSIDs */
+ SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF),
+ SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF),
+ SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF),
SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO),
SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO),
SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO),
+ SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF),
+ SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF),
SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD),
+ SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF),
+ SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF),
+ SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF),
+ SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF),
SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO),
+ SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF),
+ SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF),
+ SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF),
+ SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF),
+ SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF),
+ SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF),
SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3),
+ SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3", ALC889_FIXUP_CD),
SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX),
SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD),
SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD),
@@ -4608,14 +5409,10 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
/*
*/
-#ifdef CONFIG_SND_HDA_ENABLE_REALTEK_QUIRKS
-#include "alc882_quirks.c"
-#endif
-
static int patch_alc882(struct hda_codec *codec)
{
struct alc_spec *spec;
- int err, board_config;
+ int err;
spec = kzalloc(sizeof(*spec), GFP_KERNEL);
if (spec == NULL)
@@ -4639,45 +5436,15 @@ static int patch_alc882(struct hda_codec *codec)
if (err < 0)
goto error;
- board_config = alc_board_config(codec, ALC882_MODEL_LAST,
- alc882_models, NULL);
- if (board_config < 0)
- board_config = alc_board_codec_sid_config(codec,
- ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl);
-
- if (board_config < 0) {
- printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
- codec->chip_name);
- board_config = ALC_MODEL_AUTO;
- }
-
- if (board_config == ALC_MODEL_AUTO) {
- alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups);
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
- }
+ alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups);
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
alc_auto_parse_customize_define(codec);
- if (board_config == ALC_MODEL_AUTO) {
- /* automatic parse from the BIOS config */
- err = alc882_parse_auto_config(codec);
- if (err < 0)
- goto error;
- }
-
- if (board_config != ALC_MODEL_AUTO) {
- setup_preset(codec, &alc882_presets[board_config]);
- spec->vmaster_nid = 0x0c;
- }
-
- if (!spec->no_analog && !spec->adc_nids) {
- alc_auto_fill_adc_caps(codec);
- alc_rebuild_imux_for_auto_mic(codec);
- alc_remove_invalid_adc_nids(codec);
- }
-
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
+ /* automatic parse from the BIOS config */
+ err = alc882_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
@@ -4686,16 +5453,9 @@ static int patch_alc882(struct hda_codec *codec)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
codec->patch_ops = alc_patch_ops;
- if (board_config == ALC_MODEL_AUTO)
- spec->init_hook = alc_auto_init_std;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc882_loopbacks;
-#endif
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
return 0;
@@ -4722,7 +5482,6 @@ enum {
ALC262_FIXUP_FSC_H270,
ALC262_FIXUP_HP_Z200,
ALC262_FIXUP_TYAN,
- ALC262_FIXUP_TOSHIBA_RX1,
ALC262_FIXUP_LENOVO_3000,
ALC262_FIXUP_BENQ,
ALC262_FIXUP_BENQ_T31,
@@ -4752,16 +5511,6 @@ static const struct alc_fixup alc262_fixups[] = {
{ }
}
},
- [ALC262_FIXUP_TOSHIBA_RX1] = {
- .type = ALC_FIXUP_PINS,
- .v.pins = (const struct alc_pincfg[]) {
- { 0x14, 0x90170110 }, /* speaker */
- { 0x15, 0x0421101f }, /* HP */
- { 0x1a, 0x40f000f0 }, /* N/A */
- { 0x1b, 0x40f000f0 }, /* N/A */
- { 0x1e, 0x40f000f0 }, /* N/A */
- }
- },
[ALC262_FIXUP_LENOVO_3000] = {
.type = ALC_FIXUP_VERBS,
.v.verbs = (const struct hda_verb[]) {
@@ -4794,8 +5543,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = {
SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FIXUP_BENQ),
SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FIXUP_BENQ),
SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_FIXUP_TYAN),
- SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
- ALC262_FIXUP_TOSHIBA_RX1),
SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", ALC262_FIXUP_FSC_H270),
SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000", ALC262_FIXUP_LENOVO_3000),
SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_FIXUP_BENQ),
@@ -4804,10 +5551,6 @@ static const struct snd_pci_quirk alc262_fixup_tbl[] = {
};
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc262_loopbacks alc880_loopbacks
-#endif
-
/*
*/
static int patch_alc262(struct hda_codec *codec)
@@ -4847,15 +5590,6 @@ static int patch_alc262(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->no_analog && !spec->adc_nids) {
- alc_auto_fill_adc_caps(codec);
- alc_rebuild_imux_for_auto_mic(codec);
- alc_remove_invalid_adc_nids(codec);
- }
-
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
-
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0)
@@ -4863,16 +5597,10 @@ static int patch_alc262(struct hda_codec *codec)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc262_loopbacks;
-#endif
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
return 0;
@@ -4966,17 +5694,7 @@ static int patch_alc268(struct hda_codec *codec)
(0 << AC_AMPCAP_MUTE_SHIFT));
}
- if (!spec->no_analog && !spec->adc_nids) {
- alc_auto_fill_adc_caps(codec);
- alc_rebuild_imux_for_auto_mic(codec);
- alc_remove_invalid_adc_nids(codec);
- }
-
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
-
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
return 0;
@@ -4989,10 +5707,6 @@ static int patch_alc268(struct hda_codec *codec)
/*
* ALC269
*/
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc269_loopbacks alc880_loopbacks
-#endif
-
static const struct hda_pcm_stream alc269_44k_pcm_analog_playback = {
.substreams = 1,
.channels_min = 2,
@@ -5014,35 +5728,6 @@ static const struct hda_pcm_stream alc269_44k_pcm_analog_capture = {
/* NID is set in alc_build_pcms */
};
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static int alc269_mic2_for_mute_led(struct hda_codec *codec)
-{
- switch (codec->subsystem_id) {
- case 0x103c1586:
- return 1;
- }
- return 0;
-}
-
-static int alc269_mic2_mute_check_ps(struct hda_codec *codec, hda_nid_t nid)
-{
- /* update mute-LED according to the speaker mute state */
- if (nid == 0x01 || nid == 0x14) {
- int pinval;
- if (snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0) &
- HDA_AMP_MUTE)
- pinval = 0x24;
- else
- pinval = 0x20;
- /* mic2 vref pin is used for mute LED control */
- snd_hda_codec_update_cache(codec, 0x19, 0,
- AC_VERB_SET_PIN_WIDGET_CONTROL,
- pinval);
- }
- return alc_check_power_status(codec, nid);
-}
-#endif /* CONFIG_SND_HDA_POWER_SAVE */
-
/* different alc269-variants */
enum {
ALC269_TYPE_ALC269VA,
@@ -5193,6 +5878,31 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec,
spec->automute_hook = alc269_quanta_automute;
}
+/* update mute-LED according to the speaker mute state via mic2 VREF pin */
+static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled)
+{
+ struct hda_codec *codec = private_data;
+ unsigned int pinval = enabled ? 0x20 : 0x24;
+ snd_hda_codec_update_cache(codec, 0x19, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ pinval);
+}
+
+static void alc269_fixup_mic2_mute(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ switch (action) {
+ case ALC_FIXUP_ACT_BUILD:
+ spec->vmaster_mute.hook = alc269_fixup_mic2_mute_hook;
+ snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true);
+ /* fallthru */
+ case ALC_FIXUP_ACT_INIT:
+ snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
+ break;
+ }
+}
+
enum {
ALC269_FIXUP_SONY_VAIO,
ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -5210,6 +5920,7 @@ enum {
ALC269_FIXUP_DMIC,
ALC269VB_FIXUP_AMIC,
ALC269VB_FIXUP_DMIC,
+ ALC269_FIXUP_MIC2_MUTE_LED,
};
static const struct alc_fixup alc269_fixups[] = {
@@ -5330,9 +6041,14 @@ static const struct alc_fixup alc269_fixups[] = {
{ }
},
},
+ [ALC269_FIXUP_MIC2_MUTE_LED] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc269_fixup_mic2_mute,
+ },
};
static const struct snd_pci_quirk alc269_fixup_tbl[] = {
+ SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
@@ -5355,7 +6071,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
-#if 1
+#if 0
/* Below is a quirk table taken from the old code.
* Basically the device should work as is without the fixup table.
* If BIOS doesn't give a proper info, enable the corresponding
@@ -5364,7 +6080,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A",
ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC),
- SND_PCI_QUIRK(0x1043, 0x1113, "ASUS N63Jn", ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1143, "ASUS B53f", ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1133, "ASUS UJ20ft", ALC269_FIXUP_AMIC),
SND_PCI_QUIRK(0x1043, 0x1183, "ASUS K72DR", ALC269_FIXUP_AMIC),
@@ -5414,10 +6129,14 @@ static const struct alc_model_fixup alc269_fixup_models[] = {
};
-static int alc269_fill_coef(struct hda_codec *codec)
+static void alc269_fill_coef(struct hda_codec *codec)
{
+ struct alc_spec *spec = codec->spec;
int val;
+ if (spec->codec_variant != ALC269_TYPE_ALC269VB)
+ return;
+
if ((alc_get_coef0(codec) & 0x00ff) < 0x015) {
alc_write_coef_idx(codec, 0xf, 0x960b);
alc_write_coef_idx(codec, 0xe, 0x8817);
@@ -5452,8 +6171,6 @@ static int alc269_fill_coef(struct hda_codec *codec)
val = alc_read_coef_idx(codec, 0x4); /* HP */
alc_write_coef_idx(codec, 0x4, val | (1<<11));
-
- return 0;
}
/*
@@ -5497,6 +6214,7 @@ static int patch_alc269(struct hda_codec *codec)
}
if (err < 0)
goto error;
+ spec->init_hook = alc269_fill_coef;
alc269_fill_coef(codec);
}
@@ -5509,15 +6227,6 @@ static int patch_alc269(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->no_analog && !spec->adc_nids) {
- alc_auto_fill_adc_caps(codec);
- alc_rebuild_imux_for_auto_mic(codec);
- alc_remove_invalid_adc_nids(codec);
- }
-
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
-
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0)
@@ -5525,21 +6234,13 @@ static int patch_alc269(struct hda_codec *codec)
set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
}
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
codec->patch_ops = alc_patch_ops;
#ifdef CONFIG_PM
codec->patch_ops.resume = alc269_resume;
#endif
- spec->init_hook = alc_auto_init_std;
spec->shutup = alc269_shutup;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc269_loopbacks;
- if (alc269_mic2_for_mute_led(codec))
- codec->patch_ops.check_power_status = alc269_mic2_mute_check_ps;
-#endif
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
return 0;
@@ -5559,24 +6260,43 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
return alc_parse_auto_config(codec, alc861_ignore, alc861_ssids);
}
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-static const struct hda_amp_list alc861_loopbacks[] = {
- { 0x15, HDA_INPUT, 0 },
- { 0x15, HDA_INPUT, 1 },
- { 0x15, HDA_INPUT, 2 },
- { 0x15, HDA_INPUT, 3 },
- { } /* end */
-};
-#endif
-
-
/* Pin config fixes */
enum {
- PINFIX_FSC_AMILO_PI1505,
+ ALC861_FIXUP_FSC_AMILO_PI1505,
+ ALC861_FIXUP_AMP_VREF_0F,
+ ALC861_FIXUP_NO_JACK_DETECT,
+ ALC861_FIXUP_ASUS_A6RP,
};
+/* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */
+static void alc861_fixup_asus_amp_vref_0f(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ struct alc_spec *spec = codec->spec;
+ unsigned int val;
+
+ if (action != ALC_FIXUP_ACT_INIT)
+ return;
+ val = snd_hda_codec_read(codec, 0x0f, 0,
+ AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+ if (!(val & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)))
+ val |= AC_PINCTL_IN_EN;
+ val |= AC_PINCTL_VREF_50;
+ snd_hda_codec_write(codec, 0x0f, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+ spec->keep_vref_in_automute = 1;
+}
+
+/* suppress the jack-detection */
+static void alc_fixup_no_jack_detect(struct hda_codec *codec,
+ const struct alc_fixup *fix, int action)
+{
+ if (action == ALC_FIXUP_ACT_PRE_PROBE)
+ codec->no_jack_detect = 1;
+}
+
static const struct alc_fixup alc861_fixups[] = {
- [PINFIX_FSC_AMILO_PI1505] = {
+ [ALC861_FIXUP_FSC_AMILO_PI1505] = {
.type = ALC_FIXUP_PINS,
.v.pins = (const struct alc_pincfg[]) {
{ 0x0b, 0x0221101f }, /* HP */
@@ -5584,10 +6304,29 @@ static const struct alc_fixup alc861_fixups[] = {
{ }
}
},
+ [ALC861_FIXUP_AMP_VREF_0F] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc861_fixup_asus_amp_vref_0f,
+ },
+ [ALC861_FIXUP_NO_JACK_DETECT] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_no_jack_detect,
+ },
+ [ALC861_FIXUP_ASUS_A6RP] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc861_fixup_asus_amp_vref_0f,
+ .chained = true,
+ .chain_id = ALC861_FIXUP_NO_JACK_DETECT,
+ }
};
static const struct snd_pci_quirk alc861_fixup_tbl[] = {
- SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505),
+ SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP),
+ SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F),
+ SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT),
+ SND_PCI_QUIRK(0x1584, 0x2b01, "Haier W18", ALC861_FIXUP_AMP_VREF_0F),
+ SND_PCI_QUIRK(0x1584, 0x0000, "Uniwill ECS M31EI", ALC861_FIXUP_AMP_VREF_0F),
+ SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", ALC861_FIXUP_FSC_AMILO_PI1505),
{}
};
@@ -5614,15 +6353,6 @@ static int patch_alc861(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->no_analog && !spec->adc_nids) {
- alc_auto_fill_adc_caps(codec);
- alc_rebuild_imux_for_auto_mic(codec);
- alc_remove_invalid_adc_nids(codec);
- }
-
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
-
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x23);
if (err < 0)
@@ -5630,16 +6360,13 @@ static int patch_alc861(struct hda_codec *codec)
set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
}
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
#ifdef CONFIG_SND_HDA_POWER_SAVE
spec->power_hook = alc_power_eapd;
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc861_loopbacks;
#endif
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
return 0;
error:
@@ -5654,10 +6381,6 @@ static int patch_alc861(struct hda_codec *codec)
*
* In addition, an independent DAC
*/
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc861vd_loopbacks alc880_loopbacks
-#endif
-
static int alc861vd_parse_auto_config(struct hda_codec *codec)
{
static const hda_nid_t alc861vd_ignore[] = { 0x1d, 0 };
@@ -5738,15 +6461,6 @@ static int patch_alc861vd(struct hda_codec *codec)
add_verb(spec, alc660vd_eapd_verbs);
}
- if (!spec->no_analog && !spec->adc_nids) {
- alc_auto_fill_adc_caps(codec);
- alc_rebuild_imux_for_auto_mic(codec);
- alc_remove_invalid_adc_nids(codec);
- }
-
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
-
if (!spec->no_analog) {
err = snd_hda_attach_beep_device(codec, 0x23);
if (err < 0)
@@ -5754,16 +6468,11 @@ static int patch_alc861vd(struct hda_codec *codec)
set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
}
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc861vd_loopbacks;
-#endif
+
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
return 0;
@@ -5783,9 +6492,6 @@ static int patch_alc861vd(struct hda_codec *codec)
* In addition, an independent DAC for the multi-playback (not used in this
* driver yet).
*/
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc662_loopbacks alc880_loopbacks
-#endif
/*
* BIOS auto configuration
@@ -5835,6 +6541,7 @@ enum {
ALC662_FIXUP_ASUS_MODE6,
ALC662_FIXUP_ASUS_MODE7,
ALC662_FIXUP_ASUS_MODE8,
+ ALC662_FIXUP_NO_JACK_DETECT,
};
static const struct alc_fixup alc662_fixups[] = {
@@ -5980,6 +6687,10 @@ static const struct alc_fixup alc662_fixups[] = {
.chained = true,
.chain_id = ALC662_FIXUP_SKU_IGNORE
},
+ [ALC662_FIXUP_NO_JACK_DETECT] = {
+ .type = ALC_FIXUP_FUNC,
+ .v.func = alc_fixup_no_jack_detect,
+ },
};
static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -5988,6 +6699,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
+ SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT),
SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2),
SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD),
SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD),
@@ -6109,15 +6821,6 @@ static int patch_alc662(struct hda_codec *codec)
if (err < 0)
goto error;
- if (!spec->no_analog && !spec->adc_nids) {
- alc_auto_fill_adc_caps(codec);
- alc_rebuild_imux_for_auto_mic(codec);
- alc_remove_invalid_adc_nids(codec);
- }
-
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
-
if (!spec->no_analog && has_cdefine_beep(codec)) {
err = snd_hda_attach_beep_device(codec, 0x1);
if (err < 0)
@@ -6137,16 +6840,10 @@ static int patch_alc662(struct hda_codec *codec)
}
}
- alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
-
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
spec->shutup = alc_eapd_shutup;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
- if (!spec->loopback.amplist)
- spec->loopback.amplist = alc662_loopbacks;
-#endif
+ alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
return 0;
@@ -6186,11 +6883,7 @@ static int patch_alc680(struct hda_codec *codec)
return err;
}
- if (!spec->no_analog && !spec->cap_mixer)
- set_capture_mixer(codec);
-
codec->patch_ops = alc_patch_ops;
- spec->init_hook = alc_auto_init_std;
return 0;
}
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 3556408d6ece..33a9946b492c 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -99,6 +99,7 @@ enum {
STAC_DELL_VOSTRO_3500,
STAC_92HD83XXX_HP_cNB11_INTQUAD,
STAC_HP_DV7_4000,
+ STAC_HP_ZEPHYR,
STAC_92HD83XXX_MODELS
};
@@ -309,6 +310,8 @@ struct sigmatel_spec {
unsigned long auto_capvols[MAX_ADCS_NUM];
unsigned auto_dmic_cnt;
hda_nid_t auto_dmic_nids[MAX_DMICS_NUM];
+
+ struct hda_vmaster_mute_hook vmaster_mute;
};
static const hda_nid_t stac9200_adc_nids[1] = {
@@ -662,7 +665,6 @@ static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol,
return 0;
}
-#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac_vrefout_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
{
@@ -686,7 +688,6 @@ static int stac_vrefout_set(struct hda_codec *codec,
return 1;
}
-#endif
static unsigned int stac92xx_vref_set(struct hda_codec *codec,
hda_nid_t nid, unsigned int new_vref)
@@ -894,6 +895,13 @@ static const struct hda_verb stac92hd83xxx_core_init[] = {
{}
};
+static const struct hda_verb stac92hd83xxx_hp_zephyr_init[] = {
+ { 0x22, 0x785, 0x43 },
+ { 0x22, 0x782, 0xe0 },
+ { 0x22, 0x795, 0x00 },
+ {}
+};
+
static const struct hda_verb stac92hd71bxx_core_init[] = {
/* set master volume and direct control */
{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
@@ -999,8 +1007,8 @@ static const struct hda_verb stac9205_core_init[] = {
}
static const struct snd_kcontrol_new stac9200_mixer[] = {
- HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xb, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0xb, 0, HDA_OUTPUT),
HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
{ } /* end */
@@ -1027,8 +1035,8 @@ static const struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
};
static const struct snd_kcontrol_new stac925x_mixer[] = {
- HDA_CODEC_VOLUME_MIN_MUTE("Master Playback Volume", 0xe, 0, HDA_OUTPUT),
- HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
+ HDA_CODEC_VOLUME_MIN_MUTE("PCM Playback Volume", 0xe, 0, HDA_OUTPUT),
+ HDA_CODEC_MUTE("PCM Playback Switch", 0x0e, 0, HDA_OUTPUT),
{ } /* end */
};
@@ -1060,34 +1068,25 @@ static struct snd_kcontrol_new stac_smux_mixer = {
.put = stac92xx_smux_enum_put,
};
-static const char * const slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "Headphone Playback Volume",
- "Speaker Playback Volume",
+static const char * const slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side",
+ "Headphone", "Speaker", "IEC958",
NULL
};
-static const char * const slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "Headphone Playback Switch",
- "Speaker Playback Switch",
- "IEC958 Playback Switch",
- NULL
-};
+static void stac92xx_update_led_status(struct hda_codec *codec, int enabled);
+
+static void stac92xx_vmaster_hook(void *private_data, int val)
+{
+ stac92xx_update_led_status(private_data, val);
+}
static void stac92xx_free_kctls(struct hda_codec *codec);
static int stac92xx_build_controls(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
+ unsigned int vmaster_tlv[4];
int err;
int i;
@@ -1144,22 +1143,28 @@ static int stac92xx_build_controls(struct hda_codec *codec)
}
/* if we have no master control, let's create it */
- if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
- unsigned int vmaster_tlv[4];
- snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
- HDA_OUTPUT, vmaster_tlv);
- /* correct volume offset */
- vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset;
- /* minimum value is actually mute */
- vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
- err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv, slave_vols);
- if (err < 0)
- return err;
- }
- if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
- err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, slave_sws);
+ snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
+ HDA_OUTPUT, vmaster_tlv);
+ /* correct volume offset */
+ vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset;
+ /* minimum value is actually mute */
+ vmaster_tlv[3] |= TLV_DB_SCALE_MUTE;
+ err = snd_hda_add_vmaster(codec, "Master Playback Volume",
+ vmaster_tlv, slave_pfxs,
+ "Playback Volume");
+ if (err < 0)
+ return err;
+
+ err = __snd_hda_add_vmaster(codec, "Master Playback Switch",
+ NULL, slave_pfxs,
+ "Playback Switch", true,
+ &spec->vmaster_mute.sw_kctl);
+ if (err < 0)
+ return err;
+
+ if (spec->gpio_led) {
+ spec->vmaster_mute.hook = stac92xx_vmaster_hook;
+ err = snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true);
if (err < 0)
return err;
}
@@ -1608,7 +1613,7 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a,
"Alienware M17x", STAC_ALIENWARE_M17X),
SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490,
- "Alienware M17x", STAC_ALIENWARE_M17X),
+ "Alienware M17x R3", STAC_DELL_EQ),
{} /* terminator */
};
@@ -1636,6 +1641,12 @@ static const unsigned int hp_dv7_4000_pin_configs[10] = {
0x40f000f0, 0x40f000f0,
};
+static const unsigned int hp_zephyr_pin_configs[10] = {
+ 0x01813050, 0x0421201f, 0x04a1205e, 0x96130310,
+ 0x96130310, 0x0101401f, 0x1111611f, 0xd5a30130,
+ 0, 0,
+};
+
static const unsigned int hp_cNB11_intquad_pin_configs[10] = {
0x40f000f0, 0x0221101f, 0x02a11020, 0x92170110,
0x40f000f0, 0x92170110, 0x40f000f0, 0xd5a30130,
@@ -1649,6 +1660,7 @@ static const unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
[STAC_DELL_VOSTRO_3500] = dell_vostro_3500_pin_configs,
[STAC_92HD83XXX_HP_cNB11_INTQUAD] = hp_cNB11_intquad_pin_configs,
[STAC_HP_DV7_4000] = hp_dv7_4000_pin_configs,
+ [STAC_HP_ZEPHYR] = hp_zephyr_pin_configs,
};
static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
@@ -1659,6 +1671,7 @@ static const char * const stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
[STAC_DELL_VOSTRO_3500] = "dell-vostro-3500",
[STAC_92HD83XXX_HP_cNB11_INTQUAD] = "hp_cNB11_intquad",
[STAC_HP_DV7_4000] = "hp-dv7-4000",
+ [STAC_HP_ZEPHYR] = "hp-zephyr",
};
static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
@@ -1711,6 +1724,14 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
"HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3593,
"HP", STAC_92HD83XXX_HP_cNB11_INTQUAD),
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561,
+ "HP", STAC_HP_ZEPHYR),
+ {} /* terminator */
+};
+
+static const struct snd_pci_quirk stac92hd83xxx_codec_id_cfg_tbl[] = {
+ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3561,
+ "HP", STAC_HP_ZEPHYR),
{} /* terminator */
};
@@ -4163,13 +4184,15 @@ static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
return 1;
}
-static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
+static int is_nid_out_jack_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
{
int i;
for (i = 0; i < cfg->hp_outs; i++)
if (cfg->hp_pins[i] == nid)
return 1; /* nid is a HP-Out */
-
+ for (i = 0; i < cfg->line_outs; i++)
+ if (cfg->line_out_pins[i] == nid)
+ return 1; /* nid is a line-Out */
return 0; /* nid is not a HP-Out */
};
@@ -4375,7 +4398,7 @@ static int stac92xx_init(struct hda_codec *codec)
continue;
}
- if (is_nid_hp_pin(cfg, nid))
+ if (is_nid_out_jack_pin(cfg, nid))
continue; /* already has an unsol event */
pinctl = snd_hda_codec_read(codec, nid, 0,
@@ -4408,8 +4431,7 @@ static int stac92xx_init(struct hda_codec *codec)
snd_hda_jack_report_sync(codec);
/* sync mute LED */
- if (spec->gpio_led)
- hda_call_check_power_status(codec, 0x01);
+ snd_hda_sync_vmaster_hook(&spec->vmaster_mute);
if (spec->dac_list)
stac92xx_power_down(codec);
return 0;
@@ -4627,7 +4649,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
if (no_hp_sensing(spec, i))
continue;
- if (presence)
+ if (1 /*presence*/)
stac92xx_set_pinctl(codec, cfg->hp_pins[i], val);
#if 0 /* FIXME */
/* Resetting the pinctl like below may lead to (a sort of) regressions
@@ -4868,7 +4890,14 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity)
/* BIOS bug: unfilled OEM string */
if (strstr(dev->name, "HP_Mute_LED_P_G")) {
set_hp_led_gpio(codec);
- spec->gpio_led_polarity = 1;
+ switch (codec->subsystem_id) {
+ case 0x103c148a:
+ spec->gpio_led_polarity = 0;
+ break;
+ default:
+ spec->gpio_led_polarity = 1;
+ break;
+ }
return 1;
}
}
@@ -4980,7 +5009,6 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
return 0;
}
-#ifdef CONFIG_SND_HDA_POWER_SAVE
static int stac92xx_pre_resume(struct hda_codec *codec)
{
struct sigmatel_spec *spec = codec->spec;
@@ -5015,83 +5043,41 @@ static void stac92xx_set_power_state(struct hda_codec *codec, hda_nid_t fg,
afg_power_state);
snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
}
+#else
+#define stac92xx_suspend NULL
+#define stac92xx_resume NULL
+#define stac92xx_pre_resume NULL
+#define stac92xx_set_power_state NULL
+#endif /* CONFIG_PM */
-/*
- * For this feature CONFIG_SND_HDA_POWER_SAVE is needed
- * as mute LED state is updated in check_power_status hook
- */
-static int stac92xx_update_led_status(struct hda_codec *codec)
+/* update mute-LED accoring to the master switch */
+static void stac92xx_update_led_status(struct hda_codec *codec, int enabled)
{
struct sigmatel_spec *spec = codec->spec;
- int i, num_ext_dacs, muted = 1;
- unsigned int muted_lvl, notmtd_lvl;
- hda_nid_t nid;
+ int muted = !enabled;
if (!spec->gpio_led)
- return 0;
+ return;
+
+ /* LED state is inverted on these systems */
+ if (spec->gpio_led_polarity)
+ muted = !muted;
- for (i = 0; i < spec->multiout.num_dacs; i++) {
- nid = spec->multiout.dac_nids[i];
- if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
- HDA_AMP_MUTE)) {
- muted = 0; /* something heard */
- break;
- }
- }
- if (muted && spec->multiout.hp_nid)
- if (!(snd_hda_codec_amp_read(codec,
- spec->multiout.hp_nid, 0, HDA_OUTPUT, 0) &
- HDA_AMP_MUTE)) {
- muted = 0; /* HP is not muted */
- }
- num_ext_dacs = ARRAY_SIZE(spec->multiout.extra_out_nid);
- for (i = 0; muted && i < num_ext_dacs; i++) {
- nid = spec->multiout.extra_out_nid[i];
- if (nid == 0)
- break;
- if (!(snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
- HDA_AMP_MUTE)) {
- muted = 0; /* extra output is not muted */
- }
- }
/*polarity defines *not* muted state level*/
if (!spec->vref_mute_led_nid) {
if (muted)
spec->gpio_data &= ~spec->gpio_led; /* orange */
else
spec->gpio_data |= spec->gpio_led; /* white */
-
- if (!spec->gpio_led_polarity) {
- /* LED state is inverted on these systems */
- spec->gpio_data ^= spec->gpio_led;
- }
stac_gpio_set(codec, spec->gpio_mask,
spec->gpio_dir, spec->gpio_data);
} else {
- notmtd_lvl = spec->gpio_led_polarity ?
- AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_GRD;
- muted_lvl = spec->gpio_led_polarity ?
- AC_PINCTL_VREF_GRD : AC_PINCTL_VREF_HIZ;
- spec->vref_led = muted ? muted_lvl : notmtd_lvl;
+ spec->vref_led = muted ? AC_PINCTL_VREF_50 : AC_PINCTL_VREF_GRD;
stac_vrefout_set(codec, spec->vref_mute_led_nid,
spec->vref_led);
}
- return 0;
}
-/*
- * use power check for controlling mute led of HP notebooks
- */
-static int stac92xx_check_power_status(struct hda_codec *codec,
- hda_nid_t nid)
-{
- stac92xx_update_led_status(codec);
-
- return 0;
-}
-#endif /* CONFIG_SND_HDA_POWER_SAVE */
-#endif /* CONFIG_PM */
-
static const struct hda_codec_ops stac92xx_patch_ops = {
.build_controls = stac92xx_build_controls,
.build_pcms = stac92xx_build_pcms,
@@ -5571,6 +5557,12 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
STAC_92HD83XXX_MODELS,
stac92hd83xxx_models,
stac92hd83xxx_cfg_tbl);
+ /* check codec subsystem id if not found */
+ if (spec->board_config < 0)
+ spec->board_config =
+ snd_hda_check_board_codec_sid_config(codec,
+ STAC_92HD83XXX_MODELS, stac92hd83xxx_models,
+ stac92hd83xxx_codec_id_cfg_tbl);
again:
if (spec->board_config < 0)
snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
@@ -5581,12 +5573,17 @@ again:
codec->patch_ops = stac92xx_patch_ops;
+ switch (spec->board_config) {
+ case STAC_HP_ZEPHYR:
+ spec->init = stac92hd83xxx_hp_zephyr_init;
+ break;
+ }
+
if (find_mute_led_cfg(codec, -1/*no default cfg*/))
snd_printd("mute LED gpio %d polarity %d\n",
spec->gpio_led,
spec->gpio_led_polarity);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
@@ -5596,11 +5593,10 @@ again:
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
+#ifdef CONFIG_PM
codec->patch_ops.pre_resume = stac92xx_pre_resume;
- codec->patch_ops.check_power_status =
- stac92xx_check_power_status;
+#endif
}
-#endif
err = stac92xx_parse_auto_config(codec);
if (!err) {
@@ -5897,7 +5893,6 @@ again:
spec->gpio_led,
spec->gpio_led_polarity);
-#ifdef CONFIG_SND_HDA_POWER_SAVE
if (spec->gpio_led) {
if (!spec->vref_mute_led_nid) {
spec->gpio_mask |= spec->gpio_led;
@@ -5907,11 +5902,10 @@ again:
codec->patch_ops.set_power_state =
stac92xx_set_power_state;
}
+#ifdef CONFIG_PM
codec->patch_ops.pre_resume = stac92xx_pre_resume;
- codec->patch_ops.check_power_status =
- stac92xx_check_power_status;
+#endif
}
-#endif
spec->multiout.dac_nids = spec->dac_nids;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 03e63fed9caf..06214fdc9486 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -199,6 +199,9 @@ struct via_spec {
unsigned int no_pin_power_ctl;
enum VIA_HDA_CODEC codec_type;
+ /* analog low-power control */
+ bool alc_mode;
+
/* smart51 setup */
unsigned int smart51_nums;
hda_nid_t smart51_pins[2];
@@ -547,7 +550,10 @@ static void via_auto_init_output(struct hda_codec *codec,
pin = path->path[path->depth - 1];
init_output_pin(codec, pin, pin_type);
- caps = query_amp_caps(codec, pin, HDA_OUTPUT);
+ if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
+ caps = query_amp_caps(codec, pin, HDA_OUTPUT);
+ else
+ caps = 0;
if (caps & AC_AMPCAP_MUTE) {
unsigned int val;
val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
@@ -642,6 +648,10 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
/* init ADCs */
for (i = 0; i < spec->num_adc_nids; i++) {
+ hda_nid_t nid = spec->adc_nids[i];
+ if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP) ||
+ !(query_amp_caps(codec, nid, HDA_INPUT) & AC_AMPCAP_MUTE))
+ continue;
snd_hda_codec_write(codec, spec->adc_nids[i], 0,
AC_VERB_SET_AMP_GAIN_MUTE,
AMP_IN_UNMUTE(0));
@@ -663,6 +673,9 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
/* init input-src */
for (i = 0; i < spec->num_adc_nids; i++) {
int adc_idx = spec->inputs[spec->cur_mux[i]].adc_idx;
+ /* secondary ADCs must have the unique MUX */
+ if (i > 0 && !spec->mux_nids[i])
+ break;
if (spec->mux_nids[adc_idx]) {
int mux_idx = spec->inputs[spec->cur_mux[i]].mux_idx;
snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0,
@@ -687,6 +700,15 @@ static void via_auto_init_analog_input(struct hda_codec *codec)
}
}
+static void update_power_state(struct hda_codec *codec, hda_nid_t nid,
+ unsigned int parm)
+{
+ if (snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_POWER_STATE, 0) == parm)
+ return;
+ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm);
+}
+
static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
unsigned int *affected_parm)
{
@@ -709,7 +731,7 @@ static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid,
} else
parm = AC_PWRST_D3;
- snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, nid, parm);
}
static int via_pin_power_ctl_info(struct snd_kcontrol *kcontrol,
@@ -749,6 +771,7 @@ static int via_pin_power_ctl_put(struct snd_kcontrol *kcontrol,
return 0;
spec->no_pin_power_ctl = val;
set_widgets_power_state(codec);
+ analog_low_current_mode(codec);
return 1;
}
@@ -1036,13 +1059,19 @@ static bool is_aa_path_mute(struct hda_codec *codec)
}
/* enter/exit analog low-current mode */
-static void analog_low_current_mode(struct hda_codec *codec)
+static void __analog_low_current_mode(struct hda_codec *codec, bool force)
{
struct via_spec *spec = codec->spec;
bool enable;
unsigned int verb, parm;
- enable = is_aa_path_mute(codec) && (spec->opened_streams != 0);
+ if (spec->no_pin_power_ctl)
+ enable = false;
+ else
+ enable = is_aa_path_mute(codec) && !spec->opened_streams;
+ if (enable == spec->alc_mode && !force)
+ return;
+ spec->alc_mode = enable;
/* decide low current mode's verb & parameter */
switch (spec->codec_type) {
@@ -1074,6 +1103,11 @@ static void analog_low_current_mode(struct hda_codec *codec)
snd_hda_codec_write(codec, codec->afg, 0, verb, parm);
}
+static void analog_low_current_mode(struct hda_codec *codec)
+{
+ return __analog_low_current_mode(codec, false);
+}
+
/*
* generic initialization of ADC, input mixers and output mixers
*/
@@ -1418,25 +1452,9 @@ static const struct hda_pcm_stream via_pcm_digital_capture = {
/*
* slave controls for virtual master
*/
-static const char * const via_slave_vols[] = {
- "Front Playback Volume",
- "Surround Playback Volume",
- "Center Playback Volume",
- "LFE Playback Volume",
- "Side Playback Volume",
- "Headphone Playback Volume",
- "Speaker Playback Volume",
- NULL,
-};
-
-static const char * const via_slave_sws[] = {
- "Front Playback Switch",
- "Surround Playback Switch",
- "Center Playback Switch",
- "LFE Playback Switch",
- "Side Playback Switch",
- "Headphone Playback Switch",
- "Speaker Playback Switch",
+static const char * const via_slave_pfxs[] = {
+ "Front", "Surround", "Center", "LFE", "Side",
+ "Headphone", "Speaker",
NULL,
};
@@ -1446,6 +1464,7 @@ static int via_build_controls(struct hda_codec *codec)
struct snd_kcontrol *kctl;
int err, i;
+ spec->no_pin_power_ctl = 1;
if (spec->set_widgets_power_state)
if (!via_clone_control(spec, &via_pin_power_ctl_enum))
return -ENOMEM;
@@ -1480,13 +1499,15 @@ static int via_build_controls(struct hda_codec *codec)
snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
HDA_OUTPUT, vmaster_tlv);
err = snd_hda_add_vmaster(codec, "Master Playback Volume",
- vmaster_tlv, via_slave_vols);
+ vmaster_tlv, via_slave_pfxs,
+ "Playback Volume");
if (err < 0)
return err;
}
if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
err = snd_hda_add_vmaster(codec, "Master Playback Switch",
- NULL, via_slave_sws);
+ NULL, via_slave_pfxs,
+ "Playback Switch");
if (err < 0)
return err;
}
@@ -1494,15 +1515,13 @@ static int via_build_controls(struct hda_codec *codec)
/* assign Capture Source enums to NID */
kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
for (i = 0; kctl && i < kctl->count; i++) {
+ if (!spec->mux_nids[i])
+ continue;
err = snd_hda_add_nid(codec, kctl, i, spec->mux_nids[i]);
if (err < 0)
return err;
}
- /* init power states */
- set_widgets_power_state(codec);
- analog_low_current_mode(codec);
-
via_free_kctls(codec); /* no longer needed */
err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
@@ -2295,10 +2314,7 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol,
if (mux) {
/* switch to D0 beofre change index */
- if (snd_hda_codec_read(codec, mux, 0,
- AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0)
- snd_hda_codec_write(codec, mux, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, mux, AC_PWRST_D0);
snd_hda_codec_write(codec, mux, 0,
AC_VERB_SET_CONNECT_SEL,
spec->inputs[cur].mux_idx);
@@ -2467,6 +2483,8 @@ static int create_mic_boost_ctls(struct hda_codec *codec)
{
struct via_spec *spec = codec->spec;
const struct auto_pin_cfg *cfg = &spec->autocfg;
+ const char *prev_label = NULL;
+ int type_idx = 0;
int i, err;
for (i = 0; i < cfg->num_inputs; i++) {
@@ -2481,8 +2499,13 @@ static int create_mic_boost_ctls(struct hda_codec *codec)
if (caps == -1 || !(caps & AC_AMPCAP_NUM_STEPS))
continue;
label = hda_get_autocfg_input_label(codec, cfg, i);
+ if (prev_label && !strcmp(label, prev_label))
+ type_idx++;
+ else
+ type_idx = 0;
+ prev_label = label;
snprintf(name, sizeof(name), "%s Boost Volume", label);
- err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name,
+ err = __via_add_control(spec, VIA_CTL_WIDGET_VOL, name, type_idx,
HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT));
if (err < 0)
return err;
@@ -2776,6 +2799,10 @@ static int via_init(struct hda_codec *codec)
for (i = 0; i < spec->num_iverbs; i++)
snd_hda_sequence_write(codec, spec->init_verbs[i]);
+ /* init power states */
+ set_widgets_power_state(codec);
+ __analog_low_current_mode(codec, true);
+
via_auto_init_multi_out(codec);
via_auto_init_hp_out(codec);
via_auto_init_speaker_out(codec);
@@ -2922,9 +2949,9 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0;
/* SW0 (17h), AIW 0/1 (13h/14h) */
- snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x17, parm);
+ update_power_state(codec, 0x13, parm);
+ update_power_state(codec, 0x14, parm);
/* outputs */
/* PW0 (19h), SW1 (18h), AOW1 (11h) */
@@ -2932,8 +2959,8 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
set_pin_power_state(codec, 0x19, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1b, &parm);
- snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x11, parm);
/* PW6 (22h), SW2 (26h), AOW2 (24h) */
if (is_8ch) {
@@ -2941,20 +2968,16 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
set_pin_power_state(codec, 0x22, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1a, &parm);
- snd_hda_codec_write(codec, 0x26, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x24, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x26, parm);
+ update_power_state(codec, 0x24, parm);
} else if (codec->vendor_id == 0x11064397) {
/* PW7(23h), SW2(27h), AOW2(25h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x23, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1a, &parm);
- snd_hda_codec_write(codec, 0x27, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x27, parm);
+ update_power_state(codec, 0x25, parm);
}
/* PW 3/4/7 (1ch/1dh/23h) */
@@ -2966,17 +2989,13 @@ static void set_widgets_power_state_vt1708B(struct hda_codec *codec)
set_pin_power_state(codec, 0x23, &parm);
/* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x10, parm);
if (is_8ch) {
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x27, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
+ update_power_state(codec, 0x27, parm);
} else if (codec->vendor_id == 0x11064397 && spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
}
static int patch_vt1708S(struct hda_codec *codec);
@@ -3149,10 +3168,10 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0; /* SW0 (13h) = stereo mixer (idx 3) */
/* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */
- snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x13, parm);
+ update_power_state(codec, 0x12, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x20, parm);
/* outputs */
/* PW 3/4 (16h/17h) */
@@ -3160,10 +3179,9 @@ static void set_widgets_power_state_vt1702(struct hda_codec *codec)
set_pin_power_state(codec, 0x17, &parm);
set_pin_power_state(codec, 0x16, &parm);
/* MW0 (1ah), AOW 0/1 (10h/1dh) */
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1a, imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x1d, parm);
}
static int patch_vt1702(struct hda_codec *codec)
@@ -3228,52 +3246,48 @@ static void set_widgets_power_state_vt1718S(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0;
/* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1e, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x11, parm);
/* outputs */
/* PW3 (27h), MW2 (1ah), AOW3 (bh) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x27, &parm);
- snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1a, parm);
+ update_power_state(codec, 0xb, parm);
/* PW2 (26h), AOW2 (ah) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x26, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x2b, &parm);
- snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0xa, parm);
/* PW0 (24h), AOW0 (8h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x24, &parm);
if (!spec->hp_independent_mode) /* check for redirected HP */
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x8, parm);
/* MW9 (21h), Mw2 (1ah), AOW0 (8h) */
- snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x21, imux_is_smixer ? AC_PWRST_D0 : parm);
/* PW1 (25h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x2a, &parm);
- snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x9, parm);
if (spec->hp_independent_mode) {
/* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x1b, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x34, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0xc, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1b, parm);
+ update_power_state(codec, 0x34, parm);
+ update_power_state(codec, 0xc, parm);
}
}
@@ -3433,8 +3447,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
if (imux_is_smixer)
parm = AC_PWRST_D0;
/* SW0 (17h), AIW0(13h) */
- snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x17, parm);
+ update_power_state(codec, 0x13, parm);
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x1e, &parm);
@@ -3442,12 +3456,11 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
if (spec->dmic_enabled)
set_pin_power_state(codec, 0x22, &parm);
else
- snd_hda_codec_write(codec, 0x22, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x22, AC_PWRST_D3);
/* SW2(26h), AIW1(14h) */
- snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x26, parm);
+ update_power_state(codec, 0x14, parm);
/* outputs */
/* PW0 (19h), SW1 (18h), AOW1 (11h) */
@@ -3456,8 +3469,8 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
/* Smart 5.1 PW2(1bh) */
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1b, &parm);
- snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x11, parm);
/* PW7 (23h), SW3 (27h), AOW3 (25h) */
parm = AC_PWRST_D3;
@@ -3465,12 +3478,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
/* Smart 5.1 PW1(1ah) */
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1a, &parm);
- snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x27, parm);
/* Smart 5.1 PW5(1eh) */
if (spec->smart51_enabled)
set_pin_power_state(codec, 0x1e, &parm);
- snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
/* Mono out */
/* SW4(28h)->MW1(29h)-> PW12 (2ah)*/
@@ -3486,9 +3499,9 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
mono_out = 1;
}
parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3;
- snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x28, parm);
+ update_power_state(codec, 0x29, parm);
+ update_power_state(codec, 0x2a, parm);
/* PW 3/4 (1ch/1dh) */
parm = AC_PWRST_D3;
@@ -3496,15 +3509,12 @@ static void set_widgets_power_state_vt1716S(struct hda_codec *codec)
set_pin_power_state(codec, 0x1d, &parm);
/* HP Independent Mode, power on AOW3 */
if (spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x25, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x25, parm);
/* force to D0 for internal Speaker */
/* MW0 (16h), AOW0 (10h) */
- snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE,
- imux_is_smixer ? AC_PWRST_D0 : parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE,
- mono_out ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x16, imux_is_smixer ? AC_PWRST_D0 : parm);
+ update_power_state(codec, 0x10, mono_out ? AC_PWRST_D0 : parm);
}
static int patch_vt1716S(struct hda_codec *codec)
@@ -3580,54 +3590,45 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
set_pin_power_state(codec, 0x2b, &parm);
parm = AC_PWRST_D0;
/* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1e, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x11, parm);
/* outputs */
/* AOW0 (8h)*/
- snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x8, parm);
if (spec->codec_type == VT1802) {
/* PW4 (28h), MW4 (18h), MUX4(38h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x18, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x38, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x38, parm);
} else {
/* PW4 (26h), MW4 (1ch), MUX4(37h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x26, &parm);
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x37, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1c, parm);
+ update_power_state(codec, 0x37, parm);
}
if (spec->codec_type == VT1802) {
/* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
- snd_hda_codec_write(codec, 0x15, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x35, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x15, parm);
+ update_power_state(codec, 0x35, parm);
} else {
/* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
- snd_hda_codec_write(codec, 0x19, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x35, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x19, parm);
+ update_power_state(codec, 0x35, parm);
}
if (spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x9, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x9, AC_PWRST_D0);
/* Class-D */
/* PW0 (24h), MW0(18h/14h), MUX0(34h) */
@@ -3637,12 +3638,10 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
set_pin_power_state(codec, 0x24, &parm);
parm = present ? AC_PWRST_D3 : AC_PWRST_D0;
if (spec->codec_type == VT1802)
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x14, parm);
else
- snd_hda_codec_write(codec, 0x18, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x34, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x34, parm);
/* Mono Out */
present = snd_hda_jack_detect(codec, 0x26);
@@ -3650,28 +3649,20 @@ static void set_widgets_power_state_vt2002P(struct hda_codec *codec)
parm = present ? AC_PWRST_D3 : AC_PWRST_D0;
if (spec->codec_type == VT1802) {
/* PW15 (33h), MW8(1ch), MUX8(3ch) */
- snd_hda_codec_write(codec, 0x33, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x3c, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x33, parm);
+ update_power_state(codec, 0x1c, parm);
+ update_power_state(codec, 0x3c, parm);
} else {
/* PW15 (31h), MW8(17h), MUX8(3bh) */
- snd_hda_codec_write(codec, 0x31, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x17, 0,
- AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x3b, 0,
- AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x31, parm);
+ update_power_state(codec, 0x17, parm);
+ update_power_state(codec, 0x3b, parm);
}
/* MW9 (21h) */
if (imux_is_smixer || !is_aa_path_mute(codec))
- snd_hda_codec_write(codec, 0x21, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x21, AC_PWRST_D0);
else
- snd_hda_codec_write(codec, 0x21, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x21, AC_PWRST_D3);
}
/* patch for vt2002P */
@@ -3731,30 +3722,28 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec)
set_pin_power_state(codec, 0x2b, &parm);
parm = AC_PWRST_D0;
/* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */
- snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1e, parm);
+ update_power_state(codec, 0x1f, parm);
+ update_power_state(codec, 0x10, parm);
+ update_power_state(codec, 0x11, parm);
/* outputs */
/* AOW0 (8h)*/
- snd_hda_codec_write(codec, 0x8, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x8, AC_PWRST_D0);
/* PW4 (28h), MW4 (18h), MUX4(38h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x28, &parm);
- snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x38, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x18, parm);
+ update_power_state(codec, 0x38, parm);
/* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x25, &parm);
- snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x35, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x15, parm);
+ update_power_state(codec, 0x35, parm);
if (spec->hp_independent_mode)
- snd_hda_codec_write(codec, 0x9, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x9, AC_PWRST_D0);
/* Internal Speaker */
/* PW0 (24h), MW0(14h), MUX0(34h) */
@@ -3763,15 +3752,11 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec)
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x24, &parm);
if (present) {
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- snd_hda_codec_write(codec, 0x34, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x14, AC_PWRST_D3);
+ update_power_state(codec, 0x34, AC_PWRST_D3);
} else {
- snd_hda_codec_write(codec, 0x14, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- snd_hda_codec_write(codec, 0x34, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x14, AC_PWRST_D0);
+ update_power_state(codec, 0x34, AC_PWRST_D0);
}
@@ -3782,26 +3767,20 @@ static void set_widgets_power_state_vt1812(struct hda_codec *codec)
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x31, &parm);
if (present) {
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- snd_hda_codec_write(codec, 0x3c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
- snd_hda_codec_write(codec, 0x3e, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
+ update_power_state(codec, 0x1c, AC_PWRST_D3);
+ update_power_state(codec, 0x3c, AC_PWRST_D3);
+ update_power_state(codec, 0x3e, AC_PWRST_D3);
} else {
- snd_hda_codec_write(codec, 0x1c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- snd_hda_codec_write(codec, 0x3c, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
- snd_hda_codec_write(codec, 0x3e, 0,
- AC_VERB_SET_POWER_STATE, AC_PWRST_D0);
+ update_power_state(codec, 0x1c, AC_PWRST_D0);
+ update_power_state(codec, 0x3c, AC_PWRST_D0);
+ update_power_state(codec, 0x3e, AC_PWRST_D0);
}
/* PW15 (33h), MW15 (1dh), MUX15(3dh) */
parm = AC_PWRST_D3;
set_pin_power_state(codec, 0x33, &parm);
- snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, parm);
- snd_hda_codec_write(codec, 0x3d, 0, AC_VERB_SET_POWER_STATE, parm);
+ update_power_state(codec, 0x1d, parm);
+ update_power_state(codec, 0x3d, parm);
}
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index 92362973764d..812d10e43ae0 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -1013,6 +1013,25 @@ static int set_rate_constraints(struct snd_ice1712 *ice,
ice->hw_rates);
}
+/* if the card has the internal rate locked (is_pro_locked), limit runtime
+ hw rates to the current internal rate only.
+*/
+static void constrain_rate_if_locked(struct snd_pcm_substream *substream)
+{
+ struct snd_ice1712 *ice = snd_pcm_substream_chip(substream);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned int rate;
+ if (is_pro_rate_locked(ice)) {
+ rate = ice->get_rate(ice);
+ if (rate >= runtime->hw.rate_min
+ && rate <= runtime->hw.rate_max) {
+ runtime->hw.rate_min = rate;
+ runtime->hw.rate_max = rate;
+ }
+ }
+}
+
+
/* multi-channel playback needs alignment 8x32bit regardless of the channels
* actually used
*/
@@ -1046,6 +1065,7 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
+ constrain_rate_if_locked(substream);
if (ice->pro_open)
ice->pro_open(ice, substream);
return 0;
@@ -1066,6 +1086,7 @@ static int snd_vt1724_capture_pro_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
+ constrain_rate_if_locked(substream);
if (ice->pro_open)
ice->pro_open(ice, substream);
return 0;
@@ -1215,6 +1236,7 @@ static int snd_vt1724_playback_spdif_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
+ constrain_rate_if_locked(substream);
if (ice->spdif.ops.open)
ice->spdif.ops.open(ice, substream);
return 0;
@@ -1251,6 +1273,7 @@ static int snd_vt1724_capture_spdif_open(struct snd_pcm_substream *substream)
VT1724_BUFFER_ALIGN);
snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
VT1724_BUFFER_ALIGN);
+ constrain_rate_if_locked(substream);
if (ice->spdif.ops.open)
ice->spdif.ops.open(ice, substream);
return 0;
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index 9f3b01bb72c8..e0a4263baa20 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2102,6 +2102,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = {
},
{
.subvendor = 0x161f,
+ .subdevice = 0x202f,
+ .name = "Gateway M520",
+ .type = AC97_TUNE_INV_EAPD
+ },
+ {
+ .subvendor = 0x161f,
.subdevice = 0x203a,
.name = "Gateway 4525GZ", /* AD1981B */
.type = AC97_TUNE_INV_EAPD
diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c
index 26c7e8bcb229..c0dbb52d45be 100644
--- a/sound/pci/oxygen/oxygen_mixer.c
+++ b/sound/pci/oxygen/oxygen_mixer.c
@@ -618,9 +618,12 @@ static int ac97_volume_get(struct snd_kcontrol *ctl,
mutex_lock(&chip->mutex);
reg = oxygen_read_ac97(chip, codec, index);
mutex_unlock(&chip->mutex);
- value->value.integer.value[0] = 31 - (reg & 0x1f);
- if (stereo)
- value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f);
+ if (!stereo) {
+ value->value.integer.value[0] = 31 - (reg & 0x1f);
+ } else {
+ value->value.integer.value[0] = 31 - ((reg >> 8) & 0x1f);
+ value->value.integer.value[1] = 31 - (reg & 0x1f);
+ }
return 0;
}
@@ -636,14 +639,14 @@ static int ac97_volume_put(struct snd_kcontrol *ctl,
mutex_lock(&chip->mutex);
oldreg = oxygen_read_ac97(chip, codec, index);
- newreg = oldreg;
- newreg = (newreg & ~0x1f) |
- (31 - (value->value.integer.value[0] & 0x1f));
- if (stereo)
- newreg = (newreg & ~0x1f00) |
- ((31 - (value->value.integer.value[1] & 0x1f)) << 8);
- else
- newreg = (newreg & ~0x1f00) | ((newreg & 0x1f) << 8);
+ if (!stereo) {
+ newreg = oldreg & ~0x1f;
+ newreg |= 31 - (value->value.integer.value[0] & 0x1f);
+ } else {
+ newreg = oldreg & ~0x1f1f;
+ newreg |= (31 - (value->value.integer.value[0] & 0x1f)) << 8;
+ newreg |= 31 - (value->value.integer.value[1] & 0x1f);
+ }
change = newreg != oldreg;
if (change)
oxygen_write_ac97(chip, codec, index, newreg);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index cc9f6c83d661..bc030a2088da 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6333,6 +6333,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card,
hw->ops.open = snd_hdspm_hwdep_dummy_op;
hw->ops.ioctl = snd_hdspm_hwdep_ioctl;
+ hw->ops.ioctl_compat = snd_hdspm_hwdep_ioctl;
hw->ops.release = snd_hdspm_hwdep_dummy_op;
return 0;
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index e57b89e8aa89..94ab728f5ca8 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -286,17 +286,22 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
snd_card_free(card);
return err;
}
- if ((err = snd_ymfpci_pcm_4ch(chip, 2, NULL)) < 0) {
+ err = snd_ymfpci_mixer(chip, rear_switch[dev]);
+ if (err < 0) {
snd_card_free(card);
return err;
}
- if ((err = snd_ymfpci_pcm2(chip, 3, NULL)) < 0) {
- snd_card_free(card);
- return err;
- }
- if ((err = snd_ymfpci_mixer(chip, rear_switch[dev])) < 0) {
- snd_card_free(card);
- return err;
+ if (chip->ac97->ext_id & AC97_EI_SDAC) {
+ err = snd_ymfpci_pcm_4ch(chip, 2, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ err = snd_ymfpci_pcm2(chip, 3, NULL);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
}
if ((err = snd_ymfpci_timer(chip, 0)) < 0) {
snd_card_free(card);
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 03ee4e365311..a8159b81e9c4 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -1614,6 +1614,14 @@ static int snd_ymfpci_put_dup4ch(struct snd_kcontrol *kcontrol, struct snd_ctl_e
return change;
}
+static struct snd_kcontrol_new snd_ymfpci_dup4ch __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "4ch Duplication",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = snd_ymfpci_info_dup4ch,
+ .get = snd_ymfpci_get_dup4ch,
+ .put = snd_ymfpci_put_dup4ch,
+};
static struct snd_kcontrol_new snd_ymfpci_controls[] __devinitdata = {
{
@@ -1642,13 +1650,6 @@ YMFPCI_DOUBLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,VOLUME), 1, YDSXGR_SPDIFLOOPVOL),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, YDSXGR_SPDIFOUTCTRL, 0),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH), 0, YDSXGR_SPDIFINCTRL, 0),
YMFPCI_SINGLE(SNDRV_CTL_NAME_IEC958("Loop",NONE,NONE), 0, YDSXGR_SPDIFINCTRL, 4),
-{
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "4ch Duplication",
- .info = snd_ymfpci_info_dup4ch,
- .get = snd_ymfpci_get_dup4ch,
- .put = snd_ymfpci_put_dup4ch,
-},
};
@@ -1838,6 +1839,12 @@ int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch)
if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_ymfpci_controls[idx], chip))) < 0)
return err;
}
+ if (chip->ac97->ext_id & AC97_EI_SDAC) {
+ kctl = snd_ctl_new1(&snd_ymfpci_dup4ch, chip);
+ err = snd_ctl_add(chip->card, kctl);
+ if (err < 0)
+ return err;
+ }
/* add S/PDIF control */
if (snd_BUG_ON(!chip->pcm_spdif))
@@ -2310,6 +2317,10 @@ int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state)
for (i = 0; i < YDSXGR_NUM_SAVED_REGS; i++)
chip->saved_regs[i] = snd_ymfpci_readl(chip, saved_regs_index[i]);
chip->saved_ydsxgr_mode = snd_ymfpci_readl(chip, YDSXGR_MODE);
+ pci_read_config_word(chip->pci, PCIR_DSXG_LEGACY,
+ &chip->saved_dsxg_legacy);
+ pci_read_config_word(chip->pci, PCIR_DSXG_ELEGACY,
+ &chip->saved_dsxg_elegacy);
snd_ymfpci_writel(chip, YDSXGR_NATIVEDACOUTVOL, 0);
snd_ymfpci_writel(chip, YDSXGR_BUF441OUTVOL, 0);
snd_ymfpci_disable_dsp(chip);
@@ -2344,6 +2355,11 @@ int snd_ymfpci_resume(struct pci_dev *pci)
snd_ac97_resume(chip->ac97);
+ pci_write_config_word(chip->pci, PCIR_DSXG_LEGACY,
+ chip->saved_dsxg_legacy);
+ pci_write_config_word(chip->pci, PCIR_DSXG_ELEGACY,
+ chip->saved_dsxg_elegacy);
+
/* start hw again */
if (chip->start_count > 0) {
spin_lock_irq(&chip->reg_lock);
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 35e662d270e6..91c985599d32 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -25,6 +25,9 @@ if SND_SOC
config SND_SOC_AC97_BUS
bool
+config SND_SOC_DMAENGINE_PCM
+ bool
+
# All the supported SoCs
source "sound/soc/atmel/Kconfig"
source "sound/soc/au1x/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 9ea8ac827adc..2feaf376e94b 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,6 +1,9 @@
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
snd-soc-core-objs += soc-pcm.o soc-io.o
+snd-soc-dmaengine-pcm-objs := soc-dmaengine-pcm.o
+obj-$(CONFIG_SND_SOC_DMAENGINE_PCM) += snd-soc-dmaengine-pcm.o
+
obj-$(CONFIG_SND_SOC) += snd-soc-core.o
obj-$(CONFIG_SND_SOC) += codecs/
obj-$(CONFIG_SND_SOC) += atmel/
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index a21ff459e5d3..9b84f985770e 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -362,7 +362,7 @@ static struct snd_pcm_ops atmel_pcm_ops = {
/*--------------------------------------------------------------------------*\
* ASoC platform driver
\*--------------------------------------------------------------------------*/
-static u64 atmel_pcm_dmamask = 0xffffffff;
+static u64 atmel_pcm_dmamask = DMA_BIT_MASK(32);
static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd)
{
@@ -373,7 +373,7 @@ static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd)
if (!card->dev->dma_mask)
card->dev->dma_mask = &atmel_pcm_dmamask;
if (!card->dev->coherent_dma_mask)
- card->dev->coherent_dma_mask = 0xffffffff;
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
ret = atmel_pcm_preallocate_dma_buffer(pcm,
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
index 4ca667d477f9..f65f08beac31 100644
--- a/sound/soc/atmel/snd-soc-afeb9260.c
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -46,29 +46,8 @@ static int afeb9260_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
int err;
- /* Set codec DAI configuration */
- err = snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_I2S|
- SND_SOC_DAIFMT_NB_IF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (err < 0) {
- printk(KERN_ERR "can't set codec DAI configuration\n");
- return err;
- }
-
- /* Set cpu DAI configuration */
- err = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_IF |
- SND_SOC_DAIFMT_CBM_CFM);
- if (err < 0) {
- printk(KERN_ERR "can't set cpu DAI configuration\n");
- return err;
- }
-
/* Set the codec system clock for DAC and ADC */
err =
snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
@@ -91,7 +70,7 @@ static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Mic Jack", NULL),
};
-static const struct snd_soc_dapm_route audio_map[] = {
+static const struct snd_soc_dapm_route afeb9260_audio_map[] = {
{"Headphone Jack", NULL, "LHPOUT"},
{"Headphone Jack", NULL, "RHPOUT"},
@@ -106,13 +85,6 @@ static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
- /* Add afeb9260 specific widgets */
- snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets,
- ARRAY_SIZE(tlv320aic23_dapm_widgets));
-
- /* Set up afeb9260 specific audio path audio_map */
- snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Line In");
snd_soc_dapm_enable_pin(dapm, "Mic Jack");
@@ -129,6 +101,8 @@ static struct snd_soc_dai_link afeb9260_dai = {
.platform_name = "atmel_pcm-audio",
.codec_name = "tlv320aic23-codec.0-001a",
.init = afeb9260_tlv320aic23_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM,
.ops = &afeb9260_ops,
};
@@ -138,6 +112,11 @@ static struct snd_soc_card snd_soc_machine_afeb9260 = {
.owner = THIS_MODULE,
.dai_link = &afeb9260_dai,
.num_links = 1,
+
+ .dapm_widgets = tlv320aic23_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
+ .dapm_routes = afeb9260_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map),
};
static struct platform_device *afeb9260_snd_device;
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index 60962ce6cd4d..d542d4063771 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -40,20 +40,8 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- struct snd_soc_dai *codec_dai = rtd->codec_dai;
unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7};
int ret = 0;
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
- SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
- SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
/* set cpu DAI channel mapping */
ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map),
@@ -68,6 +56,9 @@ static struct snd_soc_ops bf5xx_ad1836_ops = {
.hw_params = bf5xx_ad1836_hw_params,
};
+#define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \
+ SND_SOC_DAIFMT_CBM_CFM)
+
static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
{
.name = "ad1836",
@@ -77,6 +68,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
.platform_name = "bfin-tdm-pcm-audio",
.codec_name = "spi0.4",
.ops = &bf5xx_ad1836_ops,
+ .dai_fmt = BF5XX_AD1836_DAIFMT,
},
{
.name = "ad1836",
@@ -86,6 +78,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = {
.platform_name = "bfin-tdm-pcm-audio",
.codec_name = "spi0.4",
.ops = &bf5xx_ad1836_ops,
+ .dai_fmt = BF5XX_AD1836_DAIFMT,
},
};
diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c
index 2d8d82dbc159..0e55e9f2a514 100644
--- a/sound/soc/blackfin/bf5xx-ad193x.c
+++ b/sound/soc/blackfin/bf5xx-ad193x.c
@@ -60,18 +60,6 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream,
break;
}
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
- SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
- SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk,
SND_SOC_CLOCK_IN);
@@ -92,6 +80,9 @@ static int bf5xx_ad193x_hw_params(struct snd_pcm_substream *substream,
return 0;
}
+#define BF5XX_AD193X_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \
+ SND_SOC_DAIFMT_CBM_CFM)
+
static struct snd_soc_ops bf5xx_ad193x_ops = {
.hw_params = bf5xx_ad193x_hw_params,
};
@@ -105,6 +96,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
.platform_name = "bfin-tdm-pcm-audio",
.codec_name = "spi0.5",
.ops = &bf5xx_ad193x_ops,
+ .dai_fmt = BF5XX_AD193X_DAIFMT,
},
{
.name = "ad193x",
@@ -114,6 +106,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai[] = {
.platform_name = "bfin-tdm-pcm-audio",
.codec_name = "spi0.5",
.ops = &bf5xx_ad193x_ops,
+ .dai_fmt = BF5XX_AD193X_DAIFMT,
},
};
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index 8e49508596da..61cc91d4a028 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -145,29 +145,8 @@ static int bf5xx_probe(struct snd_soc_card *card)
return 0;
}
-static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
- int ret = 0;
-
- pr_debug("%s rate %d format %x\n", __func__, params_rate(params),
- params_format(params));
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-
-static struct snd_soc_ops bf5xx_ad73311_ops = {
- .hw_params = bf5xx_ad73311_hw_params,
-};
+#define BF5XX_AD7311_DAI_FMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBM_CFM)
static struct snd_soc_dai_link bf5xx_ad73311_dai[] = {
{
@@ -177,7 +156,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = {
.codec_dai_name = "ad73311-hifi",
.platform_name = "bfin-i2s-pcm-audio",
.codec_name = "ad73311",
- .ops = &bf5xx_ad73311_ops,
+ .dai_fmt = BF5XX_AD7311_DAI_FMT,
},
{
.name = "ad73311",
@@ -186,7 +165,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = {
.codec_dai_name = "ad73311-hifi",
.platform_name = "bfin-i2s-pcm-audio",
.codec_name = "ad73311",
- .ops = &bf5xx_ad73311_ops,
+ .dai_fmt = BF5XX_AD7311_DAI_FMT,
},
};
diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c
index 030303238042..df3ac73f8778 100644
--- a/sound/soc/blackfin/bf5xx-ssm2602.c
+++ b/sound/soc/blackfin/bf5xx-ssm2602.c
@@ -49,7 +49,6 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
unsigned int clk = 0;
int ret = 0;
@@ -75,21 +74,6 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
break;
}
- /*
- * CODEC is master for BCLK and LRC in this configuration.
- */
-
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret < 0)
- return ret;
-
ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
@@ -102,6 +86,10 @@ static struct snd_soc_ops bf5xx_ssm2602_ops = {
.hw_params = bf5xx_ssm2602_hw_params,
};
+/* CODEC is master for BCLK and LRC in this configuration. */
+#define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \
+ SND_SOC_DAIFMT_CBM_CFM)
+
static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = {
{
.name = "ssm2602",
diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c
index 26b271c62efa..f3adbdbdd5e1 100644
--- a/sound/soc/blackfin/bfin-eval-adau1373.c
+++ b/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -67,21 +67,10 @@ static int bfin_eval_adau1373_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
int pll_rate;
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret)
- return ret;
-
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret)
- return ret;
-
switch (params_rate(params)) {
case 48000:
case 8000:
@@ -143,6 +132,8 @@ static struct snd_soc_dai_link bfin_eval_adau1373_dai = {
.codec_name = "adau1373.0-001a",
.ops = &bfin_eval_adau1373_ops,
.init = bfin_eval_adau1373_codec_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
};
static struct snd_soc_card bfin_eval_adau1373 = {
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
index c0064fa1dca6..b0531fc9d814 100644
--- a/sound/soc/blackfin/bfin-eval-adau1701.c
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -37,20 +37,9 @@ static int bfin_eval_adau1701_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret)
- return ret;
-
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret)
- return ret;
-
ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000,
SND_SOC_CLOCK_IN);
@@ -61,6 +50,9 @@ static struct snd_soc_ops bfin_eval_adau1701_ops = {
.hw_params = bfin_eval_adau1701_hw_params,
};
+#define BFIN_EVAL_ADAU1701_DAI_FMT (SND_SOC_DAIFMT_I2S | \
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM)
+
static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = {
{
.name = "adau1701",
@@ -70,6 +62,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = {
.platform_name = "bfin-i2s-pcm-audio",
.codec_name = "adau1701.0-0034",
.ops = &bfin_eval_adau1701_ops,
+ .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT,
},
{
.name = "adau1701",
@@ -79,6 +72,7 @@ static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = {
.platform_name = "bfin-i2s-pcm-audio",
.codec_name = "adau1701.0-0034",
.ops = &bfin_eval_adau1701_ops,
+ .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT,
},
};
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
index 4ef079f95e2e..84b09987b7f3 100644
--- a/sound/soc/blackfin/bfin-eval-adav80x.c
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -34,20 +34,9 @@ static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret)
- return ret;
-
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
- if (ret)
- return ret;
-
ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL,
27000000, params_rate(params) * 256);
if (ret)
@@ -88,6 +77,8 @@ static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = {
.platform_name = "bfin-i2s-pcm-audio",
.init = bfin_eval_adav80x_codec_init,
.ops = &bfin_eval_adav80x_ops,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
},
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7c205e77d83a..6508e8b790bb 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX98088 if I2C
select SND_SOC_MAX98095 if I2C
select SND_SOC_MAX9850 if I2C
+ select SND_SOC_MAX9768 if I2C
select SND_SOC_MAX9877 if I2C
select SND_SOC_PCM3008
select SND_SOC_RT5631 if I2C
@@ -62,6 +63,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_WL1273 if MFD_WL1273_CORE
select SND_SOC_WM1250_EV1 if I2C
select SND_SOC_WM2000 if I2C
+ select SND_SOC_WM2200 if I2C
select SND_SOC_WM5100 if I2C
select SND_SOC_WM8350 if MFD_WM8350
select SND_SOC_WM8400 if MFD_WM8400
@@ -292,6 +294,9 @@ config SND_SOC_WM1250_EV1
config SND_SOC_WM2000
tristate
+config SND_SOC_WM2200
+ tristate
+
config SND_SOC_WM5100
tristate
@@ -425,6 +430,9 @@ config SND_SOC_WM9713
config SND_SOC_LM4857
tristate
+config SND_SOC_MAX9768
+ tristate
+
config SND_SOC_MAX9877
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index de8078178f86..6662eb0cdcc0 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -25,6 +25,7 @@ snd-soc-dmic-objs := dmic.o
snd-soc-jz4740-codec-objs := jz4740.o
snd-soc-l3-objs := l3.o
snd-soc-lm4857-objs := lm4857.o
+snd-soc-max9768-objs := max9768.o
snd-soc-max98088-objs := max98088.o
snd-soc-max98095-objs := max98095.o
snd-soc-max9850-objs := max9850.o
@@ -51,6 +52,7 @@ snd-soc-uda1380-objs := uda1380.o
snd-soc-wl1273-objs := wl1273.o
snd-soc-wm1250-ev1-objs := wm1250-ev1.o
snd-soc-wm2000-objs := wm2000.o
+snd-soc-wm2200-objs := wm2200.o
snd-soc-wm5100-objs := wm5100.o wm5100-tables.o
snd-soc-wm8350-objs := wm8350.o
snd-soc-wm8400-objs := wm8400.o
@@ -129,6 +131,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o
obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o
+obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o
obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o
obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
@@ -153,6 +156,7 @@ obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o
obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o
obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o
+obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o
obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o
obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o
obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o
diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c
index 982d201c2e86..12e3b4118557 100644
--- a/sound/soc/codecs/ad1836.c
+++ b/sound/soc/codecs/ad1836.c
@@ -277,7 +277,7 @@ static int ad1836_probe(struct snd_soc_codec *codec)
if (ad1836->type == AD1836) {
/* left/right diff:PGA/MUX */
snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
- ret = snd_soc_add_controls(codec, ad1836_controls,
+ ret = snd_soc_add_codec_controls(codec, ad1836_controls,
ARRAY_SIZE(ad1836_controls));
if (ret)
return ret;
@@ -285,11 +285,11 @@ static int ad1836_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, AD1836_ADC_CTRL3, 0x00);
}
- ret = snd_soc_add_controls(codec, ad183x_dac_controls, num_dacs * 2);
+ ret = snd_soc_add_codec_controls(codec, ad183x_dac_controls, num_dacs * 2);
if (ret)
return ret;
- ret = snd_soc_add_controls(codec, ad183x_adc_controls, num_adcs);
+ ret = snd_soc_add_codec_controls(codec, ad183x_adc_controls, num_adcs);
if (ret)
return ret;
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 9bba7f849464..8c39dddd7d00 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -228,7 +228,7 @@ static int ad1980_soc_probe(struct snd_soc_codec *codec)
ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
- snd_soc_add_controls(codec, ad1980_snd_ac97_controls,
+ snd_soc_add_codec_controls(codec, ad1980_snd_ac97_controls,
ARRAY_SIZE(ad1980_snd_ac97_controls));
return 0;
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 971ba4529171..44f59064d8de 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -1244,8 +1244,6 @@ static int adau1373_probe(struct snd_soc_codec *codec)
return ret;
}
- codec->dapm.idle_bias_off = true;
-
if (pdata) {
if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting))
return -EINVAL;
@@ -1259,7 +1257,7 @@ static int adau1373_probe(struct snd_soc_codec *codec)
pdata->drc_setting[i]);
}
- snd_soc_add_controls(codec, adau1373_drc_controls,
+ snd_soc_add_codec_controls(codec, adau1373_drc_controls,
pdata->num_drc);
val = 0;
@@ -1284,7 +1282,7 @@ static int adau1373_probe(struct snd_soc_codec *codec)
}
if (!lineout_differential) {
- snd_soc_add_controls(codec, adau1373_lineout2_controls,
+ snd_soc_add_codec_controls(codec, adau1373_lineout2_controls,
ARRAY_SIZE(adau1373_lineout2_controls));
}
@@ -1340,6 +1338,7 @@ static struct snd_soc_codec_driver adau1373_codec_driver = {
.suspend = adau1373_suspend,
.resume = adau1373_resume,
.set_bias_level = adau1373_set_bias_level,
+ .idle_bias_off = true,
.reg_cache_size = ARRAY_SIZE(adau1373_default_regs),
.reg_cache_default = adau1373_default_regs,
.reg_word_size = sizeof(uint8_t),
diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c
index 6b325ea03869..78e9ce48bb99 100644
--- a/sound/soc/codecs/adau1701.c
+++ b/sound/soc/codecs/adau1701.c
@@ -457,7 +457,6 @@ static int adau1701_probe(struct snd_soc_codec *codec)
{
int ret;
- codec->dapm.idle_bias_off = 1;
codec->control_data = to_i2c_client(codec->dev);
ret = adau1701_load_firmware(codec);
@@ -473,6 +472,7 @@ static int adau1701_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver adau1701_codec_drv = {
.probe = adau1701_probe,
.set_bias_level = adau1701_set_bias_level,
+ .idle_bias_off = true,
.reg_cache_size = ADAU1701_NUM_REGS,
.reg_word_size = sizeof(u16),
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index d27b5e4cce99..ceb96ecf5588 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -46,75 +46,15 @@
#define DRV_NAME "ak4104-codec"
struct ak4104_private {
- enum snd_soc_control_type control_type;
- void *control_data;
+ struct regmap *regmap;
};
-static int ak4104_fill_cache(struct snd_soc_codec *codec)
-{
- int i;
- u8 *reg_cache = codec->reg_cache;
- struct spi_device *spi = codec->control_data;
-
- for (i = 0; i < codec->driver->reg_cache_size; i++) {
- int ret = spi_w8r8(spi, i | AK4104_READ);
- if (ret < 0) {
- dev_err(&spi->dev, "SPI write failure\n");
- return ret;
- }
-
- reg_cache[i] = ret;
- }
-
- return 0;
-}
-
-static unsigned int ak4104_read_reg_cache(struct snd_soc_codec *codec,
- unsigned int reg)
-{
- u8 *reg_cache = codec->reg_cache;
-
- if (reg >= codec->driver->reg_cache_size)
- return -EINVAL;
-
- return reg_cache[reg];
-}
-
-static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg,
- unsigned int value)
-{
- u8 *cache = codec->reg_cache;
- struct spi_device *spi = codec->control_data;
-
- if (reg >= codec->driver->reg_cache_size)
- return -EINVAL;
-
- /* only write to the hardware if value has changed */
- if (cache[reg] != value) {
- u8 tmp[2] = { (reg & AK4104_REG_MASK) | AK4104_WRITE, value };
-
- if (spi_write(spi, tmp, sizeof(tmp))) {
- dev_err(&spi->dev, "SPI write failed\n");
- return -EIO;
- }
-
- cache[reg] = value;
- }
-
- return 0;
-}
-
static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int format)
{
struct snd_soc_codec *codec = codec_dai->codec;
int val = 0;
-
- val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
- if (val < 0)
- return val;
-
- val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1);
+ int ret;
/* set DAI format */
switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
@@ -135,7 +75,13 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
return -EINVAL;
- return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+ ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
+ AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1,
+ val);
+ if (ret < 0)
+ return ret;
+
+ return 0;
}
static int ak4104_hw_params(struct snd_pcm_substream *substream,
@@ -148,7 +94,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
/* set the IEC958 bits: consumer mode, no copyright bit */
val |= IEC958_AES0_CON_NOT_COPYRIGHT;
- ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val);
+ snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val);
val = 0;
@@ -167,7 +113,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
return -EINVAL;
}
- return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val);
+ return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val);
}
static const struct snd_soc_dai_ops ak4101_dai_ops = {
@@ -192,67 +138,57 @@ static struct snd_soc_dai_driver ak4104_dai = {
static int ak4104_probe(struct snd_soc_codec *codec)
{
struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
- int ret, val;
-
- codec->control_data = ak4104->control_data;
+ int ret;
- /* read all regs and fill the cache */
- ret = ak4104_fill_cache(codec);
- if (ret < 0) {
- dev_err(codec->dev, "failed to fill register cache\n");
+ codec->control_data = ak4104->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+ if (ret != 0)
return ret;
- }
-
- /* read the 'reserved' register - according to the datasheet, it
- * should contain 0x5b. Not a good way to verify the presence of
- * the device, but there is no hardware ID register. */
- if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) !=
- AK4104_RESERVED_VAL)
- return -ENODEV;
/* set power-up and non-reset bits */
- val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
- val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN;
- ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+ ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
+ AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN,
+ AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
if (ret < 0)
return ret;
/* enable transmitter */
- val = ak4104_read_reg_cache(codec, AK4104_REG_TX);
- val |= AK4104_TX_TXE;
- ret = ak4104_spi_write(codec, AK4104_REG_TX, val);
+ ret = snd_soc_update_bits(codec, AK4104_REG_TX,
+ AK4104_TX_TXE, AK4104_TX_TXE);
if (ret < 0)
return ret;
- dev_info(codec->dev, "SPI device initialized\n");
return 0;
}
static int ak4104_remove(struct snd_soc_codec *codec)
{
- int val, ret;
-
- val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
- if (val < 0)
- return val;
-
- /* clear power-up and non-reset bits */
- val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
- ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+ snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
+ AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0);
- return ret;
+ return 0;
}
static struct snd_soc_codec_driver soc_codec_device_ak4104 = {
.probe = ak4104_probe,
.remove = ak4104_remove,
- .reg_cache_size = AK4104_NUM_REGS,
- .reg_word_size = sizeof(u8),
+};
+
+static const struct regmap_config ak4104_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = AK4104_NUM_REGS - 1,
+ .read_flag_mask = AK4104_READ,
+ .write_flag_mask = AK4104_WRITE,
+
+ .cache_type = REGCACHE_RBTREE,
};
static int ak4104_spi_probe(struct spi_device *spi)
{
struct ak4104_private *ak4104;
+ unsigned int val;
int ret;
spi->bits_per_word = 8;
@@ -266,17 +202,41 @@ static int ak4104_spi_probe(struct spi_device *spi)
if (ak4104 == NULL)
return -ENOMEM;
- ak4104->control_data = spi;
- ak4104->control_type = SND_SOC_SPI;
+ ak4104->regmap = regmap_init_spi(spi, &ak4104_regmap);
+ if (IS_ERR(ak4104->regmap)) {
+ ret = PTR_ERR(ak4104->regmap);
+ return ret;
+ }
+
+ /* read the 'reserved' register - according to the datasheet, it
+ * should contain 0x5b. Not a good way to verify the presence of
+ * the device, but there is no hardware ID register. */
+ ret = regmap_read(ak4104->regmap, AK4104_REG_RESERVED, &val);
+ if (ret != 0)
+ goto err;
+ if (val != AK4104_RESERVED_VAL) {
+ ret = -ENODEV;
+ goto err;
+ }
+
spi_set_drvdata(spi, ak4104);
ret = snd_soc_register_codec(&spi->dev,
&soc_codec_device_ak4104, &ak4104_dai, 1);
+ if (ret != 0)
+ goto err;
+
+ return 0;
+
+err:
+ regmap_exit(ak4104->regmap);
return ret;
}
static int __devexit ak4104_spi_remove(struct spi_device *spi)
{
+ struct ak4104_private *ak4101 = spi_get_drvdata(spi);
+ regmap_exit(ak4101->regmap);
snd_soc_unregister_codec(&spi->dev);
return 0;
}
@@ -290,17 +250,7 @@ static struct spi_driver ak4104_spi_driver = {
.remove = __devexit_p(ak4104_spi_remove),
};
-static int __init ak4104_init(void)
-{
- return spi_register_driver(&ak4104_spi_driver);
-}
-module_init(ak4104_init);
-
-static void __exit ak4104_exit(void)
-{
- spi_unregister_driver(&ak4104_spi_driver);
-}
-module_exit(ak4104_exit);
+module_spi_driver(ak4104_spi_driver);
MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver");
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 9e809e05d066..838ae8b22b50 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -18,6 +18,7 @@
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -27,24 +28,43 @@
#include "ak4535.h"
-#define AK4535_VERSION "0.3"
-
/* codec private data */
struct ak4535_priv {
+ struct regmap *regmap;
unsigned int sysclk;
- enum snd_soc_control_type control_type;
};
/*
* ak4535 register cache
*/
-static const u8 ak4535_reg[AK4535_CACHEREGNUM] = {
- 0x00, 0x80, 0x00, 0x03,
- 0x02, 0x00, 0x11, 0x01,
- 0x00, 0x40, 0x36, 0x10,
- 0x00, 0x00, 0x57, 0x00,
+static const struct reg_default ak4535_reg_defaults[] = {
+ { 0, 0x00 },
+ { 1, 0x80 },
+ { 2, 0x00 },
+ { 3, 0x03 },
+ { 4, 0x02 },
+ { 5, 0x00 },
+ { 6, 0x11 },
+ { 7, 0x01 },
+ { 8, 0x00 },
+ { 9, 0x40 },
+ { 10, 0x36 },
+ { 11, 0x10 },
+ { 12, 0x00 },
+ { 13, 0x00 },
+ { 14, 0x57 },
};
+static bool ak4535_volatile(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case AK4535_STATUS:
+ return true;
+ default:
+ return false;
+ }
+}
+
static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"};
static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"};
static const char *ak4535_hp_out[] = {"Stereo", "Mono"};
@@ -372,9 +392,8 @@ static int ak4535_probe(struct snd_soc_codec *codec)
struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
int ret;
- printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION);
-
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4535->control_type);
+ codec->control_data = ak4535->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
@@ -382,7 +401,7 @@ static int ak4535_probe(struct snd_soc_codec *codec)
/* power on device */
ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- snd_soc_add_controls(codec, ak4535_snd_controls,
+ snd_soc_add_codec_controls(codec, ak4535_snd_controls,
ARRAY_SIZE(ak4535_snd_controls));
return 0;
}
@@ -394,22 +413,30 @@ static int ak4535_remove(struct snd_soc_codec *codec)
return 0;
}
+static const struct regmap_config ak4535_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = AK4535_STATUS,
+ .volatile_reg = ak4535_volatile,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults = ak4535_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(ak4535_reg_defaults),
+};
+
static struct snd_soc_codec_driver soc_codec_dev_ak4535 = {
.probe = ak4535_probe,
.remove = ak4535_remove,
.suspend = ak4535_suspend,
.resume = ak4535_resume,
.set_bias_level = ak4535_set_bias_level,
- .reg_cache_size = ARRAY_SIZE(ak4535_reg),
- .reg_word_size = sizeof(u8),
- .reg_cache_default = ak4535_reg,
.dapm_widgets = ak4535_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak4535_dapm_widgets),
.dapm_routes = ak4535_audio_map,
.num_dapm_routes = ARRAY_SIZE(ak4535_audio_map),
};
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static __devinit int ak4535_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
@@ -421,17 +448,29 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c,
if (ak4535 == NULL)
return -ENOMEM;
+ ak4535->regmap = regmap_init_i2c(i2c, &ak4535_regmap);
+ if (IS_ERR(ak4535->regmap)) {
+ ret = PTR_ERR(ak4535->regmap);
+ dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret);
+ return ret;
+ }
+
i2c_set_clientdata(i2c, ak4535);
- ak4535->control_type = SND_SOC_I2C;
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_ak4535, &ak4535_dai, 1);
+ if (ret != 0)
+ regmap_exit(ak4535->regmap);
+
return ret;
}
static __devexit int ak4535_i2c_remove(struct i2c_client *client)
{
+ struct ak4535_priv *ak4535 = i2c_get_clientdata(client);
+
snd_soc_unregister_codec(&client->dev);
+ regmap_exit(ak4535->regmap);
return 0;
}
@@ -443,36 +482,15 @@ MODULE_DEVICE_TABLE(i2c, ak4535_i2c_id);
static struct i2c_driver ak4535_i2c_driver = {
.driver = {
- .name = "ak4535-codec",
+ .name = "ak4535",
.owner = THIS_MODULE,
},
.probe = ak4535_i2c_probe,
.remove = __devexit_p(ak4535_i2c_remove),
.id_table = ak4535_i2c_id,
};
-#endif
-static int __init ak4535_modinit(void)
-{
- int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- ret = i2c_add_driver(&ak4535_i2c_driver);
- if (ret != 0) {
- printk(KERN_ERR "Failed to register AK4535 I2C driver: %d\n",
- ret);
- }
-#endif
- return ret;
-}
-module_init(ak4535_modinit);
-
-static void __exit ak4535_exit(void)
-{
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- i2c_del_driver(&ak4535_i2c_driver);
-#endif
-}
-module_exit(ak4535_exit);
+module_i2c_driver(ak4535_i2c_driver);
MODULE_DESCRIPTION("Soc AK4535 driver");
MODULE_AUTHOR("Richard Purdie");
diff --git a/sound/soc/codecs/ak4535.h b/sound/soc/codecs/ak4535.h
index 0431e5f634a2..402de1d274bf 100644
--- a/sound/soc/codecs/ak4535.h
+++ b/sound/soc/codecs/ak4535.h
@@ -34,6 +34,4 @@
#define AK4535_VOL 0xe
#define AK4535_STATUS 0xf
-#define AK4535_CACHEREGNUM 0x10
-
#endif
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 5ef70b5d27e4..f8e10ced244a 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -146,13 +146,10 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
0, 0xFF, 1, out_tlv),
-
- SOC_SINGLE("Headphone Switch", PW_MGMT2, 6, 1, 0),
};
-static const struct snd_kcontrol_new ak4642_hpout_mixer_controls[] = {
- SOC_DAPM_SINGLE("DACH", MD_CTL4, 0, 1, 0),
-};
+static const struct snd_kcontrol_new ak4642_headphone_control =
+ SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
@@ -165,13 +162,12 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LINEOUT"),
- SND_SOC_DAPM_MIXER("HPOUTL Mixer", PW_MGMT2, 5, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
+ &ak4642_headphone_control),
- SND_SOC_DAPM_MIXER("HPOUTR Mixer", PW_MGMT2, 4, 0,
- &ak4642_hpout_mixer_controls[0],
- ARRAY_SIZE(ak4642_hpout_mixer_controls)),
+ SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
&ak4642_lout_mixer_controls[0],
@@ -184,12 +180,17 @@ static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
static const struct snd_soc_dapm_route ak4642_intercon[] = {
/* Outputs */
- {"HPOUTL", NULL, "HPOUTL Mixer"},
- {"HPOUTR", NULL, "HPOUTR Mixer"},
+ {"HPOUTL", NULL, "HPL Out"},
+ {"HPOUTR", NULL, "HPR Out"},
{"LINEOUT", NULL, "LINEOUT Mixer"},
- {"HPOUTL Mixer", "DACH", "DAC"},
- {"HPOUTR Mixer", "DACH", "DAC"},
+ {"HPL Out", NULL, "Headphone Enable"},
+ {"HPR Out", NULL, "Headphone Enable"},
+
+ {"Headphone Enable", "Switch", "DACH"},
+
+ {"DACH", NULL, "DAC"},
+
{"LINEOUT Mixer", "DACL", "DAC"},
};
@@ -476,7 +477,7 @@ static int ak4642_probe(struct snd_soc_codec *codec)
return ret;
}
- snd_soc_add_controls(codec, ak4642_snd_controls,
+ snd_soc_add_codec_controls(codec, ak4642_snd_controls,
ARRAY_SIZE(ak4642_snd_controls));
ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c
index a53b152e6a07..5fb7c2a80e6d 100644
--- a/sound/soc/codecs/ak4671.c
+++ b/sound/soc/codecs/ak4671.c
@@ -628,7 +628,7 @@ static int ak4671_probe(struct snd_soc_codec *codec)
return ret;
}
- snd_soc_add_controls(codec, ak4671_snd_controls,
+ snd_soc_add_codec_controls(codec, ak4671_snd_controls,
ARRAY_SIZE(ak4671_snd_controls));
ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
index 3feee569ceea..d47b62ddb210 100644
--- a/sound/soc/codecs/alc5623.c
+++ b/sound/soc/codecs/alc5623.c
@@ -925,22 +925,22 @@ static int alc5623_probe(struct snd_soc_codec *codec)
switch (alc5623->id) {
case 0x21:
- snd_soc_add_controls(codec, alc5621_vol_snd_controls,
+ snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls,
ARRAY_SIZE(alc5621_vol_snd_controls));
break;
case 0x22:
- snd_soc_add_controls(codec, alc5622_vol_snd_controls,
+ snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls,
ARRAY_SIZE(alc5622_vol_snd_controls));
break;
case 0x23:
- snd_soc_add_controls(codec, alc5623_vol_snd_controls,
+ snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls,
ARRAY_SIZE(alc5623_vol_snd_controls));
break;
default:
return -EINVAL;
}
- snd_soc_add_controls(codec, alc5623_snd_controls,
+ snd_soc_add_codec_controls(codec, alc5623_snd_controls,
ARRAY_SIZE(alc5623_snd_controls));
snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
@@ -992,7 +992,7 @@ static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
* low = 0x1a
* high = 0x1b
*/
-static int alc5623_i2c_probe(struct i2c_client *client,
+static __devinit int alc5623_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
struct alc5623_platform_data *pdata;
@@ -1059,7 +1059,7 @@ static int alc5623_i2c_probe(struct i2c_client *client,
return ret;
}
-static int alc5623_i2c_remove(struct i2c_client *client)
+static __devexit int alc5623_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
return 0;
diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c
index 390e437d7c5e..e2111e0ccad7 100644
--- a/sound/soc/codecs/alc5632.c
+++ b/sound/soc/codecs/alc5632.c
@@ -145,15 +145,14 @@ static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
/* -16.5db min scale, 1.5db steps, no mute */
static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
static const unsigned int boost_tlv[] = {
- TLV_DB_RANGE_HEAD(3),
- 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
- 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
- 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
+ TLV_DB_RANGE_HEAD(2),
+ 0, 1, TLV_DB_SCALE_ITEM(0, 2000, 0),
+ 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0),
};
/* 0db min scale, 6 db steps, no mute */
static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
/* 0db min scalem 0.75db steps, no mute */
-static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 075, 0);
+static const DECLARE_TLV_DB_SCALE(vdac_tlv, -3525, 75, 0);
static const struct snd_kcontrol_new alc5632_vol_snd_controls[] = {
/* left starts at bit 8, right at bit 0 */
@@ -176,26 +175,32 @@ static const struct snd_kcontrol_new alc5632_snd_controls[] = {
ALC5632_AUX_OUT_VOL, 15, 7, 1, 1),
SOC_SINGLE_TLV("Voice DAC Playback Volume",
ALC5632_VOICE_DAC_VOL, 0, 63, 0, vdac_tlv),
- SOC_SINGLE_TLV("Phone Capture Volume",
+ SOC_SINGLE("Voice DAC Playback Switch",
+ ALC5632_VOICE_DAC_VOL, 12, 1, 1),
+ SOC_SINGLE_TLV("Phone Playback Volume",
ALC5632_PHONE_IN_VOL, 8, 31, 1, vol_tlv),
- SOC_DOUBLE_TLV("LineIn Capture Volume",
+ SOC_DOUBLE_TLV("LineIn Playback Volume",
ALC5632_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
SOC_DOUBLE_TLV("Master Playback Volume",
ALC5632_STEREO_DAC_IN_VOL, 8, 0, 63, 1, vdac_tlv),
SOC_DOUBLE("Master Playback Switch",
ALC5632_STEREO_DAC_IN_VOL, 15, 7, 1, 1),
- SOC_SINGLE_TLV("Mic1 Capture Volume",
+ SOC_SINGLE_TLV("Mic1 Playback Volume",
ALC5632_MIC_VOL, 8, 31, 1, vol_tlv),
- SOC_SINGLE_TLV("Mic2 Capture Volume",
+ SOC_SINGLE_TLV("Mic2 Playback Volume",
ALC5632_MIC_VOL, 0, 31, 1, vol_tlv),
SOC_DOUBLE_TLV("Rec Capture Volume",
ALC5632_ADC_REC_GAIN, 8, 0, 31, 0, adc_rec_tlv),
SOC_SINGLE_TLV("Mic 1 Boost Volume",
- ALC5632_MIC_CTRL, 10, 2, 0, boost_tlv),
+ ALC5632_MIC_CTRL, 10, 3, 0, boost_tlv),
SOC_SINGLE_TLV("Mic 2 Boost Volume",
- ALC5632_MIC_CTRL, 8, 2, 0, boost_tlv),
- SOC_SINGLE_TLV("Digital Boost Volume",
+ ALC5632_MIC_CTRL, 8, 3, 0, boost_tlv),
+ SOC_SINGLE_TLV("DMIC Boost Capture Volume",
ALC5632_DIGI_BOOST_CTRL, 0, 7, 0, dig_tlv),
+ SOC_SINGLE("DMIC En Capture Switch",
+ ALC5632_DIGI_BOOST_CTRL, 15, 1, 0),
+ SOC_SINGLE("DMIC PreFilter Capture Switch",
+ ALC5632_DIGI_BOOST_CTRL, 12, 1, 0),
};
/*
@@ -244,36 +249,48 @@ SOC_DAPM_SINGLE("VOICE2SPK Playback Switch", ALC5632_VOICE_DAC_VOL, 14, 1, 1),
/* Left Record Mixer */
static const struct snd_kcontrol_new alc5632_captureL_mixer_controls[] = {
-SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1),
-SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1),
-SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1),
-SOC_DAPM_SINGLE("Left Phone Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1),
-SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1),
-SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1),
-SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1),
+SOC_DAPM_SINGLE("MIC12REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 14, 1, 1),
+SOC_DAPM_SINGLE("MIC22REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 13, 1, 1),
+SOC_DAPM_SINGLE("LIL2REC Capture Switch", ALC5632_ADC_REC_MIXER, 12, 1, 1),
+SOC_DAPM_SINGLE("PH2REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 11, 1, 1),
+SOC_DAPM_SINGLE("HPL2REC Capture Switch", ALC5632_ADC_REC_MIXER, 10, 1, 1),
+SOC_DAPM_SINGLE("SPK2REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 9, 1, 1),
+SOC_DAPM_SINGLE("MONO2REC_L Capture Switch", ALC5632_ADC_REC_MIXER, 8, 1, 1),
};
/* Right Record Mixer */
static const struct snd_kcontrol_new alc5632_captureR_mixer_controls[] = {
-SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1),
-SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1),
-SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1),
-SOC_DAPM_SINGLE("Right Phone Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1),
-SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1),
-SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1),
-SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1),
+SOC_DAPM_SINGLE("MIC12REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 6, 1, 1),
+SOC_DAPM_SINGLE("MIC22REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 5, 1, 1),
+SOC_DAPM_SINGLE("LIR2REC Capture Switch", ALC5632_ADC_REC_MIXER, 4, 1, 1),
+SOC_DAPM_SINGLE("PH2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 3, 1, 1),
+SOC_DAPM_SINGLE("HPR2REC Capture Switch", ALC5632_ADC_REC_MIXER, 2, 1, 1),
+SOC_DAPM_SINGLE("SPK2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 1, 1, 1),
+SOC_DAPM_SINGLE("MONO2REC_R Capture Switch", ALC5632_ADC_REC_MIXER, 0, 1, 1),
};
-static const char *alc5632_spk_n_sour_sel[] = {
+/* Dmic Mixer */
+static const struct snd_kcontrol_new alc5632_dmicl_mixer_controls[] = {
+SOC_DAPM_SINGLE("DMICL2ADC Capture Switch", ALC5632_DIGI_BOOST_CTRL, 7, 1, 1),
+};
+static const struct snd_kcontrol_new alc5632_dmicr_mixer_controls[] = {
+SOC_DAPM_SINGLE("DMICR2ADC Capture Switch", ALC5632_DIGI_BOOST_CTRL, 6, 1, 1),
+};
+
+static const char * const alc5632_spk_n_sour_sel[] = {
"RN/-R", "RP/+R", "LN/-R", "Mute"};
-static const char *alc5632_hpl_out_input_sel[] = {
+static const char * const alc5632_hpl_out_input_sel[] = {
"Vmid", "HP Left Mix"};
-static const char *alc5632_hpr_out_input_sel[] = {
+static const char * const alc5632_hpr_out_input_sel[] = {
"Vmid", "HP Right Mix"};
-static const char *alc5632_spkout_input_sel[] = {
+static const char * const alc5632_spkout_input_sel[] = {
"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
-static const char *alc5632_aux_out_input_sel[] = {
+static const char * const alc5632_aux_out_input_sel[] = {
"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
+static const char * const alc5632_adcr_func_sel[] = {
+ "Stereo ADC", "Voice ADC"};
+static const char * const alc5632_i2s_out_sel[] = {
+ "ADC LR", "Voice Stereo Digital"};
/* auxout output mux */
static const struct soc_enum alc5632_aux_out_input_enum =
@@ -312,6 +329,17 @@ static const struct soc_enum alc5632_amp_enum =
static const struct snd_kcontrol_new alc5632_amp_mux_controls =
SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum);
+/* ADC output select */
+static const struct soc_enum alc5632_adcr_func_enum =
+ SOC_ENUM_SINGLE(ALC5632_DAC_FUNC_SELECT, 5, 2, alc5632_adcr_func_sel);
+static const struct snd_kcontrol_new alc5632_adcr_func_controls =
+ SOC_DAPM_ENUM("ADCR Mux", alc5632_adcr_func_enum);
+
+/* I2S out select */
+static const struct soc_enum alc5632_i2s_out_enum =
+ SOC_ENUM_SINGLE(ALC5632_I2S_OUT_CTL, 5, 2, alc5632_i2s_out_sel);
+static const struct snd_kcontrol_new alc5632_i2s_out_controls =
+ SOC_DAPM_ENUM("I2SOut Mux", alc5632_i2s_out_enum);
static const struct snd_soc_dapm_widget alc5632_dapm_widgets[] = {
/* Muxes */
@@ -325,6 +353,10 @@ SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
&alc5632_hpr_out_mux_controls),
SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
&alc5632_spkoutn_mux_controls),
+SND_SOC_DAPM_MUX("ADCR Mux", SND_SOC_NOPM, 0, 0,
+ &alc5632_adcr_func_controls),
+SND_SOC_DAPM_MUX("I2SOut Mux", ALC5632_PWR_MANAG_ADD1, 11, 0,
+ &alc5632_i2s_out_controls),
/* output mixers */
SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
@@ -343,6 +375,12 @@ SND_SOC_DAPM_MIXER("Mono Mix", ALC5632_PWR_MANAG_ADD2, 2, 0,
SND_SOC_DAPM_MIXER("Speaker Mix", ALC5632_PWR_MANAG_ADD2, 3, 0,
&alc5632_speaker_mixer_controls[0],
ARRAY_SIZE(alc5632_speaker_mixer_controls)),
+SND_SOC_DAPM_MIXER("DMICL Mix", SND_SOC_NOPM, 0, 0,
+ &alc5632_dmicl_mixer_controls[0],
+ ARRAY_SIZE(alc5632_dmicl_mixer_controls)),
+SND_SOC_DAPM_MIXER("DMICR Mix", SND_SOC_NOPM, 0, 0,
+ &alc5632_dmicr_mixer_controls[0],
+ ARRAY_SIZE(alc5632_dmicr_mixer_controls)),
/* input mixers */
SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5632_PWR_MANAG_ADD2, 1, 0,
@@ -352,20 +390,28 @@ SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5632_PWR_MANAG_ADD2, 0, 0,
&alc5632_captureR_mixer_controls[0],
ARRAY_SIZE(alc5632_captureR_mixer_controls)),
-SND_SOC_DAPM_DAC("Left DAC", "HiFi Playback",
- ALC5632_PWR_MANAG_ADD2, 9, 0),
-SND_SOC_DAPM_DAC("Right DAC", "HiFi Playback",
- ALC5632_PWR_MANAG_ADD2, 8, 0),
+SND_SOC_DAPM_AIF_IN("AIFRXL", "Left HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("AIFRXR", "Right HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIFTXL", "Left HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIFTXR", "Right HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("VAIFRX", "Voice Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("VAIFTX", "Voice Capture", 0, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_DAC("Voice DAC", NULL, ALC5632_PWR_MANAG_ADD2, 10, 0),
+SND_SOC_DAPM_DAC("Left DAC", NULL, ALC5632_PWR_MANAG_ADD2, 9, 0),
+SND_SOC_DAPM_DAC("Right DAC", NULL, ALC5632_PWR_MANAG_ADD2, 8, 0),
+SND_SOC_DAPM_ADC("Left ADC", NULL, ALC5632_PWR_MANAG_ADD2, 7, 0),
+SND_SOC_DAPM_ADC("Right ADC", NULL, ALC5632_PWR_MANAG_ADD2, 6, 0),
+
SND_SOC_DAPM_MIXER("DAC Left Channel", ALC5632_PWR_MANAG_ADD1, 15, 0, NULL, 0),
SND_SOC_DAPM_MIXER("DAC Right Channel",
ALC5632_PWR_MANAG_ADD1, 14, 0, NULL, 0),
SND_SOC_DAPM_MIXER("I2S Mix", ALC5632_PWR_MANAG_ADD1, 11, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Phone Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
-SND_SOC_DAPM_ADC("Left ADC", "HiFi Capture",
- ALC5632_PWR_MANAG_ADD2, 7, 0),
-SND_SOC_DAPM_ADC("Right ADC", "HiFi Capture",
- ALC5632_PWR_MANAG_ADD2, 6, 0),
+SND_SOC_DAPM_MIXER("Voice Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_MIXER("ADCLR", SND_SOC_NOPM, 0, 0, NULL, 0),
+
SND_SOC_DAPM_PGA("Left Headphone", ALC5632_PWR_MANAG_ADD3, 11, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Headphone", ALC5632_PWR_MANAG_ADD3, 10, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left Speaker", ALC5632_PWR_MANAG_ADD3, 13, 0, NULL, 0),
@@ -393,10 +439,12 @@ SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("HPR"),
SND_SOC_DAPM_OUTPUT("SPKOUT"),
SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+
SND_SOC_DAPM_INPUT("LINEINL"),
SND_SOC_DAPM_INPUT("LINEINR"),
SND_SOC_DAPM_INPUT("PHONEP"),
SND_SOC_DAPM_INPUT("PHONEN"),
+SND_SOC_DAPM_INPUT("DMICDAT"),
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_VMID("Vmid"),
@@ -404,6 +452,10 @@ SND_SOC_DAPM_VMID("Vmid"),
static const struct snd_soc_dapm_route alc5632_dapm_routes[] = {
+ /* Playback streams */
+ {"Left DAC", NULL, "AIFRXL"},
+ {"Right DAC", NULL, "AIFRXR"},
+
/* virtual mixer - mixes left & right channels */
{"I2S Mix", NULL, "Left DAC"},
{"I2S Mix", NULL, "Right DAC"},
@@ -426,9 +478,12 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = {
{"HP Mix", "PHONE2HP Playback Switch", "Phone Mix"},
{"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"},
{"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"},
-
+ {"HP Mix", "VOICE2HP Playback Switch", "Voice Mix"},
{"HPR Mix", "DACR2HP Playback Switch", "DAC Right Channel"},
{"HPL Mix", "DACL2HP Playback Switch", "DAC Left Channel"},
+ {"HPOut Mix", NULL, "HP Mix"},
+ {"HPOut Mix", NULL, "HPR Mix"},
+ {"HPOut Mix", NULL, "HPL Mix"},
/* speaker mixer */
{"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
@@ -436,35 +491,34 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = {
{"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"},
{"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"},
{"Speaker Mix", "DAC2SPK Playback Switch", "DAC Left Channel"},
-
-
+ {"Speaker Mix", "VOICE2SPK Playback Switch", "Voice Mix"},
/* mono mixer */
{"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"},
{"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"},
{"Mono Mix", "LI2MONO Playback Switch", "Line Mix"},
- {"Mono Mix", "VOICE2MONO Playback Switch", "Phone Mix"},
{"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"},
{"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"},
{"Mono Mix", "DAC2MONO Playback Switch", "DAC Left Channel"},
+ {"Mono Mix", "VOICE2MONO Playback Switch", "Voice Mix"},
/* Left record mixer */
- {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"},
- {"Left Capture Mix", "Left Phone Capture Switch", "PHONEN"},
- {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
- {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
- {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
- {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
- {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
+ {"Left Capture Mix", "LIL2REC Capture Switch", "LINEINL"},
+ {"Left Capture Mix", "PH2REC_L Capture Switch", "PHONEN"},
+ {"Left Capture Mix", "MIC12REC_L Capture Switch", "MIC1 Pre Amp"},
+ {"Left Capture Mix", "MIC22REC_L Capture Switch", "MIC2 Pre Amp"},
+ {"Left Capture Mix", "HPL2REC Capture Switch", "HPL Mix"},
+ {"Left Capture Mix", "SPK2REC_L Capture Switch", "Speaker Mix"},
+ {"Left Capture Mix", "MONO2REC_L Capture Switch", "Mono Mix"},
/*Right record mixer */
- {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"},
- {"Right Capture Mix", "Right Phone Capture Switch", "PHONEP"},
- {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
- {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
- {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
- {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
- {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
+ {"Right Capture Mix", "LIR2REC Capture Switch", "LINEINR"},
+ {"Right Capture Mix", "PH2REC_R Capture Switch", "PHONEP"},
+ {"Right Capture Mix", "MIC12REC_R Capture Switch", "MIC1 Pre Amp"},
+ {"Right Capture Mix", "MIC22REC_R Capture Switch", "MIC2 Pre Amp"},
+ {"Right Capture Mix", "HPR2REC Capture Switch", "HPR Mix"},
+ {"Right Capture Mix", "SPK2REC_R Capture Switch", "Speaker Mix"},
+ {"Right Capture Mix", "MONO2REC_R Capture Switch", "Mono Mix"},
/* headphone left mux */
{"Left Headphone Mux", "HP Left Mix", "HPL Mix"},
@@ -504,10 +558,30 @@ static const struct snd_soc_dapm_route alc5632_dapm_routes[] = {
/* left ADC */
{"Left ADC", NULL, "Left Capture Mix"},
+ {"DMICL Mix", "DMICL2ADC Capture Switch", "DMICDAT"},
+ {"Left ADC", NULL, "DMICL Mix"},
+ {"ADCLR", NULL, "Left ADC"},
/* right ADC */
- {"Right ADC", NULL, "Right Capture Mix"},
-
+ {"Right ADC", NULL, "Right Capture Mix"},
+ {"DMICR Mix", "DMICR2ADC Capture Switch", "DMICDAT"},
+ {"Right ADC", NULL, "DMICR Mix"},
+ {"ADCR Mux", "Stereo ADC", "Right ADC"},
+ {"ADCR Mux", "Voice ADC", "Right ADC"},
+ {"ADCLR", NULL, "ADCR Mux"},
+ {"VAIFTX", NULL, "ADCR Mux"},
+
+ /* Digital I2S out */
+ {"I2SOut Mux", "ADC LR", "ADCLR"},
+ {"I2SOut Mux", "Voice Stereo Digital", "VAIFRX"},
+ {"AIFTXL", NULL, "I2SOut Mux"},
+ {"AIFTXR", NULL, "I2SOut Mux"},
+
+ /* Voice Mix */
+ {"Voice DAC", NULL, "VAIFRX"},
+ {"Voice Mix", NULL, "Voice DAC"},
+
+ /* Speaker Output */
{"SpeakerOut N Mux", "RN/-R", "Left Speaker"},
{"SpeakerOut N Mux", "RP/+R", "Left Speaker"},
{"SpeakerOut N Mux", "LN/-R", "Left Speaker"},
@@ -714,6 +788,7 @@ static int alc5632_set_dai_sysclk(struct snd_soc_dai *codec_dai,
struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec);
switch (freq) {
+ case 4096000:
case 8192000:
case 11289600:
case 12288000:
@@ -994,7 +1069,7 @@ static int alc5632_probe(struct snd_soc_codec *codec)
switch (alc5632->id) {
case 0x5c:
- snd_soc_add_controls(codec, alc5632_vol_snd_controls,
+ snd_soc_add_codec_controls(codec, alc5632_vol_snd_controls,
ARRAY_SIZE(alc5632_vol_snd_controls));
break;
default:
@@ -1109,7 +1184,7 @@ static __devinit int alc5632_i2c_probe(struct i2c_client *client,
return ret;
}
-static int alc5632_i2c_remove(struct i2c_client *client)
+static __devexit int alc5632_i2c_remove(struct i2c_client *client)
{
struct alc5632_priv *alc5632 = i2c_get_clientdata(client);
snd_soc_unregister_codec(&client->dev);
diff --git a/sound/soc/codecs/alc5632.h b/sound/soc/codecs/alc5632.h
index 357651ec074e..1b5bda594ea3 100644
--- a/sound/soc/codecs/alc5632.h
+++ b/sound/soc/codecs/alc5632.h
@@ -51,6 +51,7 @@
#define ALC5632_ADC_REC_MONOMIX (1 << 0)
#define ALC5632_VOICE_DAC_VOL 0x18 /* voice dac vol */
+#define ALC5632_I2S_OUT_CTL 0x1A /* undocumented reg. found in path scheme */
/* ALC5632_OUTPUT_MIXER_CTRL : */
/* same remark as for reg 2 line vs speaker */
#define ALC5632_OUTPUT_MIXER_CTRL 0x1C /* out mix ctrl */
diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c
index 4854b472d5fd..064cd6a93516 100644
--- a/sound/soc/codecs/cq93vc.c
+++ b/sound/soc/codecs/cq93vc.c
@@ -38,8 +38,6 @@
#include <sound/soc.h>
#include <sound/initval.h>
-#include <mach/dm365.h>
-
static inline unsigned int cq93vc_read(struct snd_soc_codec *codec,
unsigned int reg)
{
@@ -159,7 +157,7 @@ static int cq93vc_probe(struct snd_soc_codec *codec)
codec->control_data = davinci_vc;
/* Set controls */
- snd_soc_add_controls(codec, cq93vc_snd_controls,
+ snd_soc_add_codec_controls(codec, cq93vc_snd_controls,
ARRAY_SIZE(cq93vc_snd_controls));
/* Off, with power on */
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 055536645da9..1d672f528662 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -521,7 +521,7 @@ static int cs4270_probe(struct snd_soc_codec *codec)
}
/* Add the non-DAPM controls */
- ret = snd_soc_add_controls(codec, cs4270_snd_controls,
+ ret = snd_soc_add_codec_controls(codec, cs4270_snd_controls,
ARRAY_SIZE(cs4270_snd_controls));
if (ret < 0) {
dev_err(codec->dev, "failed to add controls\n");
@@ -715,7 +715,7 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id);
*/
static struct i2c_driver cs4270_i2c_driver = {
.driver = {
- .name = "cs4270-codec",
+ .name = "cs4270",
.owner = THIS_MODULE,
},
.id_table = cs4270_id,
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index f6fe846b6a6c..bf7141280a74 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -513,7 +513,7 @@ static int cs4271_probe(struct snd_soc_codec *codec)
/* Power-up sequence requires 85 uS */
udelay(85);
- return snd_soc_add_controls(codec, cs4271_snd_controls,
+ return snd_soc_add_codec_controls(codec, cs4271_snd_controls,
ARRAY_SIZE(cs4271_snd_controls));
}
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index 9d38db8f1919..78979b3e0e95 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1113,7 +1113,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream,
priv->config[id].mmcc &= 0xC0;
priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc;
priv->config[id].spc &= 0xFC;
- priv->config[id].spc &= MCK_SCLK_64FS;
+ priv->config[id].spc |= MCK_SCLK_MCLK;
} else {
/* CS42L73 Slave */
priv->config[id].spc &= 0xFC;
diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c
index ab38e93c3543..7843711729bc 100644
--- a/sound/soc/codecs/da7210.c
+++ b/sound/soc/codecs/da7210.c
@@ -17,6 +17,7 @@
#include <linux/delay.h>
#include <linux/i2c.h>
+#include <linux/regmap.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <sound/pcm.h>
@@ -626,41 +627,82 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = {
/* Codec private data */
struct da7210_priv {
- enum snd_soc_control_type control_type;
+ struct regmap *regmap;
};
-/*
- * Register cache
- */
-static const u8 da7210_reg[] = {
- 0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R0 - R7 */
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08, /* R8 - RF */
- 0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54, /* R10 - R17 */
- 0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R18 - R1F */
- 0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00, /* R20 - R27 */
- 0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00, /* R28 - R2F */
- 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, /* R30 - R37 */
- 0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00, /* R38 - R3F */
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R40 - R4F */
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R48 - R4F */
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R50 - R57 */
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R58 - R5F */
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R60 - R67 */
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R68 - R6F */
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* R70 - R77 */
- 0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00, /* R78 - R7F */
- 0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00, /* R80 - R87 */
- 0x00, /* R88 */
+static struct reg_default da7210_reg_defaults[] = {
+ { 0x01, 0x11 },
+ { 0x03, 0x00 },
+ { 0x04, 0x00 },
+ { 0x05, 0x00 },
+ { 0x06, 0x00 },
+ { 0x07, 0x00 },
+ { 0x08, 0x00 },
+ { 0x09, 0x00 },
+ { 0x0a, 0x00 },
+ { 0x0b, 0x00 },
+ { 0x0c, 0x00 },
+ { 0x0d, 0x00 },
+ { 0x0e, 0x00 },
+ { 0x0f, 0x08 },
+ { 0x10, 0x00 },
+ { 0x11, 0x00 },
+ { 0x12, 0x00 },
+ { 0x13, 0x00 },
+ { 0x14, 0x08 },
+ { 0x15, 0x10 },
+ { 0x16, 0x10 },
+ { 0x17, 0x54 },
+ { 0x18, 0x40 },
+ { 0x19, 0x00 },
+ { 0x1a, 0x00 },
+ { 0x1b, 0x00 },
+ { 0x1c, 0x00 },
+ { 0x1d, 0x00 },
+ { 0x1e, 0x00 },
+ { 0x1f, 0x00 },
+ { 0x20, 0x00 },
+ { 0x21, 0x00 },
+ { 0x22, 0x00 },
+ { 0x23, 0x02 },
+ { 0x24, 0x00 },
+ { 0x25, 0x76 },
+ { 0x26, 0x00 },
+ { 0x27, 0x00 },
+ { 0x28, 0x04 },
+ { 0x29, 0x00 },
+ { 0x2a, 0x00 },
+ { 0x2b, 0x30 },
+ { 0x2c, 0x2A },
+ { 0x83, 0x00 },
+ { 0x84, 0x00 },
+ { 0x85, 0x00 },
+ { 0x86, 0x00 },
+ { 0x87, 0x00 },
+ { 0x88, 0x00 },
};
-static int da7210_volatile_register(struct snd_soc_codec *codec,
+static bool da7210_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case DA7210_A_HID_UNLOCK:
+ case DA7210_A_TEST_UNLOCK:
+ case DA7210_A_PLL1:
+ case DA7210_A_CP_MODE:
+ return false;
+ default:
+ return true;
+ }
+}
+
+static bool da7210_volatile_register(struct device *dev,
unsigned int reg)
{
switch (reg) {
case DA7210_STATUS:
- return 1;
+ return true;
default:
- return 0;
+ return false;
}
}
@@ -866,7 +908,8 @@ static int da7210_probe(struct snd_soc_codec *codec)
dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type);
+ codec->control_data = da7210->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
if (ret < 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
@@ -983,12 +1026,14 @@ static int da7210_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
/* As suggested by Dialog */
- snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */
- snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4);
- snd_soc_write(codec, DA7210_A_PLL1, 0x01);
- snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C);
- snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */
- snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00);
+ /* unlock */
+ regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x8B);
+ regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0xB4);
+ regmap_write(da7210->regmap, DA7210_A_PLL1, 0x01);
+ regmap_write(da7210->regmap, DA7210_A_CP_MODE, 0x7C);
+ /* re-lock */
+ regmap_write(da7210->regmap, DA7210_A_HID_UNLOCK, 0x00);
+ regmap_write(da7210->regmap, DA7210_A_TEST_UNLOCK, 0x00);
/* Activate all enabled subsystem */
snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN);
@@ -1000,10 +1045,6 @@ static int da7210_probe(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_da7210 = {
.probe = da7210_probe,
- .reg_cache_size = ARRAY_SIZE(da7210_reg),
- .reg_word_size = sizeof(u8),
- .reg_cache_default = da7210_reg,
- .volatile_register = da7210_volatile_register,
.controls = da7210_snd_controls,
.num_controls = ARRAY_SIZE(da7210_snd_controls),
@@ -1014,6 +1055,17 @@ static struct snd_soc_codec_driver soc_codec_dev_da7210 = {
.num_dapm_routes = ARRAY_SIZE(da7210_audio_map),
};
+static struct regmap_config da7210_regmap = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .reg_defaults = da7210_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(da7210_reg_defaults),
+ .volatile_reg = da7210_volatile_register,
+ .readable_reg = da7210_readable_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
@@ -1027,16 +1079,34 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
return -ENOMEM;
i2c_set_clientdata(i2c, da7210);
- da7210->control_type = SND_SOC_I2C;
+
+ da7210->regmap = regmap_init_i2c(i2c, &da7210_regmap);
+ if (IS_ERR(da7210->regmap)) {
+ ret = PTR_ERR(da7210->regmap);
+ dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
+ return ret;
+ }
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_da7210, &da7210_dai, 1);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+ goto err_regmap;
+ }
+ return ret;
+
+err_regmap:
+ regmap_exit(da7210->regmap);
+
return ret;
}
static int __devexit da7210_i2c_remove(struct i2c_client *client)
{
+ struct da7210_priv *da7210 = i2c_get_clientdata(client);
+
snd_soc_unregister_codec(&client->dev);
+ regmap_exit(da7210->regmap);
return 0;
}
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
index 319039240e0f..ba4fafb93e56 100644
--- a/sound/soc/codecs/lm4857.c
+++ b/sound/soc/codecs/lm4857.c
@@ -179,7 +179,7 @@ static int lm4857_probe(struct snd_soc_codec *codec)
codec->control_data = lm4857->i2c;
- ret = snd_soc_add_controls(codec, lm4857_controls,
+ ret = snd_soc_add_codec_controls(codec, lm4857_controls,
ARRAY_SIZE(lm4857_controls));
if (ret)
return ret;
diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c
new file mode 100644
index 000000000000..17b3ec2d05cb
--- /dev/null
+++ b/sound/soc/codecs/max9768.c
@@ -0,0 +1,247 @@
+/*
+ * MAX9768 AMP driver
+ *
+ * Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; version 2 of the License.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+#include <linux/regmap.h>
+
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/max9768.h>
+
+/* "Registers" */
+#define MAX9768_VOL 0
+#define MAX9768_CTRL 3
+
+/* Commands */
+#define MAX9768_CTRL_PWM 0x15
+#define MAX9768_CTRL_FILTERLESS 0x16
+
+struct max9768 {
+ struct regmap *regmap;
+ int mute_gpio;
+ int shdn_gpio;
+ u32 flags;
+};
+
+static struct reg_default max9768_default_regs[] = {
+ { 0, 0 },
+ { 3, MAX9768_CTRL_FILTERLESS},
+};
+
+static int max9768_get_gpio(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec);
+ int val = gpio_get_value_cansleep(max9768->mute_gpio);
+
+ ucontrol->value.integer.value[0] = !val;
+
+ return 0;
+}
+
+static int max9768_set_gpio(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec);
+
+ gpio_set_value_cansleep(max9768->mute_gpio, !ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static const unsigned int volume_tlv[] = {
+ TLV_DB_RANGE_HEAD(43),
+ 0, 0, TLV_DB_SCALE_ITEM(-16150, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(-9280, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(-9030, 0, 0),
+ 3, 3, TLV_DB_SCALE_ITEM(-8680, 0, 0),
+ 4, 4, TLV_DB_SCALE_ITEM(-8430, 0, 0),
+ 5, 5, TLV_DB_SCALE_ITEM(-8080, 0, 0),
+ 6, 6, TLV_DB_SCALE_ITEM(-7830, 0, 0),
+ 7, 7, TLV_DB_SCALE_ITEM(-7470, 0, 0),
+ 8, 8, TLV_DB_SCALE_ITEM(-7220, 0, 0),
+ 9, 9, TLV_DB_SCALE_ITEM(-6870, 0, 0),
+ 10, 10, TLV_DB_SCALE_ITEM(-6620, 0, 0),
+ 11, 11, TLV_DB_SCALE_ITEM(-6270, 0, 0),
+ 12, 12, TLV_DB_SCALE_ITEM(-6020, 0, 0),
+ 13, 13, TLV_DB_SCALE_ITEM(-5670, 0, 0),
+ 14, 14, TLV_DB_SCALE_ITEM(-5420, 0, 0),
+ 15, 17, TLV_DB_SCALE_ITEM(-5060, 250, 0),
+ 18, 18, TLV_DB_SCALE_ITEM(-4370, 0, 0),
+ 19, 19, TLV_DB_SCALE_ITEM(-4210, 0, 0),
+ 20, 20, TLV_DB_SCALE_ITEM(-3960, 0, 0),
+ 21, 21, TLV_DB_SCALE_ITEM(-3760, 0, 0),
+ 22, 22, TLV_DB_SCALE_ITEM(-3600, 0, 0),
+ 23, 23, TLV_DB_SCALE_ITEM(-3340, 0, 0),
+ 24, 24, TLV_DB_SCALE_ITEM(-3150, 0, 0),
+ 25, 25, TLV_DB_SCALE_ITEM(-2980, 0, 0),
+ 26, 26, TLV_DB_SCALE_ITEM(-2720, 0, 0),
+ 27, 27, TLV_DB_SCALE_ITEM(-2520, 0, 0),
+ 28, 30, TLV_DB_SCALE_ITEM(-2350, 190, 0),
+ 31, 31, TLV_DB_SCALE_ITEM(-1750, 0, 0),
+ 32, 34, TLV_DB_SCALE_ITEM(-1640, 100, 0),
+ 35, 37, TLV_DB_SCALE_ITEM(-1310, 110, 0),
+ 38, 39, TLV_DB_SCALE_ITEM(-990, 100, 0),
+ 40, 40, TLV_DB_SCALE_ITEM(-710, 0, 0),
+ 41, 41, TLV_DB_SCALE_ITEM(-600, 0, 0),
+ 42, 42, TLV_DB_SCALE_ITEM(-500, 0, 0),
+ 43, 43, TLV_DB_SCALE_ITEM(-340, 0, 0),
+ 44, 44, TLV_DB_SCALE_ITEM(-190, 0, 0),
+ 45, 45, TLV_DB_SCALE_ITEM(-50, 0, 0),
+ 46, 46, TLV_DB_SCALE_ITEM(50, 0, 0),
+ 47, 50, TLV_DB_SCALE_ITEM(120, 40, 0),
+ 51, 57, TLV_DB_SCALE_ITEM(290, 50, 0),
+ 58, 58, TLV_DB_SCALE_ITEM(650, 0, 0),
+ 59, 62, TLV_DB_SCALE_ITEM(700, 60, 0),
+ 63, 63, TLV_DB_SCALE_ITEM(950, 0, 0),
+};
+
+static const struct snd_kcontrol_new max9768_volume[] = {
+ SOC_SINGLE_TLV("Playback Volume", MAX9768_VOL, 0, 63, 0, volume_tlv),
+};
+
+static const struct snd_kcontrol_new max9768_mute[] = {
+ SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio),
+};
+
+static int max9768_probe(struct snd_soc_codec *codec)
+{
+ struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ codec->control_data = max9768->regmap;
+ ret = snd_soc_codec_set_cache_io(codec, 2, 6, SND_SOC_REGMAP);
+ if (ret)
+ return ret;
+
+ if (max9768->flags & MAX9768_FLAG_CLASSIC_PWM) {
+ ret = snd_soc_write(codec, MAX9768_CTRL, MAX9768_CTRL_PWM);
+ if (ret)
+ return ret;
+ }
+
+ if (gpio_is_valid(max9768->mute_gpio)) {
+ ret = snd_soc_add_codec_controls(codec, max9768_mute,
+ ARRAY_SIZE(max9768_mute));
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver max9768_codec_driver = {
+ .probe = max9768_probe,
+ .controls = max9768_volume,
+ .num_controls = ARRAY_SIZE(max9768_volume),
+};
+
+static const struct regmap_config max9768_i2c_regmap_config = {
+ .reg_bits = 2,
+ .val_bits = 6,
+ .max_register = 3,
+ .reg_defaults = max9768_default_regs,
+ .num_reg_defaults = ARRAY_SIZE(max9768_default_regs),
+ .cache_type = REGCACHE_RBTREE,
+};
+
+static int __devinit max9768_i2c_probe(struct i2c_client *client,
+ const struct i2c_device_id *id)
+{
+ struct max9768 *max9768;
+ struct max9768_pdata *pdata = client->dev.platform_data;
+ int err;
+
+ max9768 = devm_kzalloc(&client->dev, sizeof(*max9768), GFP_KERNEL);
+ if (!max9768)
+ return -ENOMEM;
+
+ if (pdata) {
+ /* Mute on powerup to avoid clicks */
+ err = gpio_request_one(pdata->mute_gpio, GPIOF_INIT_HIGH, "MAX9768 Mute");
+ max9768->mute_gpio = err ?: pdata->mute_gpio;
+
+ /* Activate chip by releasing shutdown, enables I2C */
+ err = gpio_request_one(pdata->shdn_gpio, GPIOF_INIT_HIGH, "MAX9768 Shutdown");
+ max9768->shdn_gpio = err ?: pdata->shdn_gpio;
+
+ max9768->flags = pdata->flags;
+ } else {
+ max9768->shdn_gpio = -EINVAL;
+ max9768->mute_gpio = -EINVAL;
+ }
+
+ i2c_set_clientdata(client, max9768);
+
+ max9768->regmap = regmap_init_i2c(client, &max9768_i2c_regmap_config);
+ if (IS_ERR(max9768->regmap)) {
+ err = PTR_ERR(max9768->regmap);
+ goto err_gpio_free;
+ }
+
+ err = snd_soc_register_codec(&client->dev, &max9768_codec_driver, NULL, 0);
+ if (err)
+ goto err_regmap_free;
+
+ return 0;
+
+ err_regmap_free:
+ regmap_exit(max9768->regmap);
+ err_gpio_free:
+ if (gpio_is_valid(max9768->shdn_gpio))
+ gpio_free(max9768->shdn_gpio);
+ if (gpio_is_valid(max9768->mute_gpio))
+ gpio_free(max9768->mute_gpio);
+
+ return err;
+}
+
+static int __devexit max9768_i2c_remove(struct i2c_client *client)
+{
+ struct max9768 *max9768 = i2c_get_clientdata(client);
+
+ snd_soc_unregister_codec(&client->dev);
+ regmap_exit(max9768->regmap);
+
+ if (gpio_is_valid(max9768->shdn_gpio))
+ gpio_free(max9768->shdn_gpio);
+ if (gpio_is_valid(max9768->mute_gpio))
+ gpio_free(max9768->mute_gpio);
+
+ return 0;
+}
+
+static const struct i2c_device_id max9768_i2c_id[] = {
+ { "max9768", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, max9768_i2c_id);
+
+static struct i2c_driver max9768_i2c_driver = {
+ .driver = {
+ .name = "max9768",
+ .owner = THIS_MODULE,
+ },
+ .probe = max9768_i2c_probe,
+ .remove = __devexit_p(max9768_i2c_remove),
+ .id_table = max9768_i2c_id,
+};
+module_i2c_driver(max9768_i2c_driver);
+
+MODULE_AUTHOR("Wolfram Sang <w.sang@pengutronix.de>");
+MODULE_DESCRIPTION("ASoC MAX9768 amplifier driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c
index 006efcfe6dda..af7324b79dd0 100644
--- a/sound/soc/codecs/max98088.c
+++ b/sound/soc/codecs/max98088.c
@@ -1908,7 +1908,7 @@ static void max98088_handle_eq_pdata(struct snd_soc_codec *codec)
max98088->eq_enum.texts = max98088->eq_texts;
max98088->eq_enum.max = max98088->eq_textcnt;
- ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls));
+ ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
if (ret != 0)
dev_err(codec->dev, "Failed to add EQ control: %d\n", ret);
}
@@ -2030,7 +2030,7 @@ static int max98088_probe(struct snd_soc_codec *codec)
max98088_handle_pdata(codec);
- snd_soc_add_controls(codec, max98088_snd_controls,
+ snd_soc_add_codec_controls(codec, max98088_snd_controls,
ARRAY_SIZE(max98088_snd_controls));
err_access:
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index fcfa7497d7b7..0bb511a0388d 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -1284,7 +1284,7 @@ static const struct snd_soc_dapm_route max98095_audio_map[] = {
static int max98095_add_widgets(struct snd_soc_codec *codec)
{
- snd_soc_add_controls(codec, max98095_snd_controls,
+ snd_soc_add_codec_controls(codec, max98095_snd_controls,
ARRAY_SIZE(max98095_snd_controls));
return 0;
@@ -1984,7 +1984,7 @@ static void max98095_handle_eq_pdata(struct snd_soc_codec *codec)
max98095->eq_enum.texts = max98095->eq_texts;
max98095->eq_enum.max = max98095->eq_textcnt;
- ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls));
+ ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
if (ret != 0)
dev_err(codec->dev, "Failed to add EQ control: %d\n", ret);
}
@@ -2139,7 +2139,7 @@ static void max98095_handle_bq_pdata(struct snd_soc_codec *codec)
max98095->bq_enum.texts = max98095->bq_texts;
max98095->bq_enum.max = max98095->bq_textcnt;
- ret = snd_soc_add_controls(codec, controls, ARRAY_SIZE(controls));
+ ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls));
if (ret != 0)
dev_err(codec->dev, "Failed to add Biquad control: %d\n", ret);
}
diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c
index dcf6f2a1600a..3a2ba3d8fd6d 100644
--- a/sound/soc/codecs/max9877.c
+++ b/sound/soc/codecs/max9877.c
@@ -253,7 +253,7 @@ static const struct snd_kcontrol_new max9877_controls[] = {
/* This function is called from ASoC machine driver */
int max9877_add_controls(struct snd_soc_codec *codec)
{
- return snd_soc_add_controls(codec, max9877_controls,
+ return snd_soc_add_codec_controls(codec, max9877_controls,
ARRAY_SIZE(max9877_controls));
}
EXPORT_SYMBOL_GPL(max9877_add_controls);
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index f8863ebb4304..d1926266fe00 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -227,7 +227,7 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
};
/* routes for sgtl5000 */
-static const struct snd_soc_dapm_route audio_map[] = {
+static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = {
{"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */
{"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */
@@ -987,12 +987,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
/* restore regular registers */
for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) {
- /* this regs depends on the others */
+ /* These regs should restore in particular order */
if (reg == SGTL5000_CHIP_ANA_POWER ||
reg == SGTL5000_CHIP_CLK_CTRL ||
reg == SGTL5000_CHIP_LINREG_CTRL ||
reg == SGTL5000_CHIP_LINE_OUT_CTRL ||
- reg == SGTL5000_CHIP_CLK_CTRL)
+ reg == SGTL5000_CHIP_REF_CTRL)
continue;
snd_soc_write(codec, reg, cache[reg]);
@@ -1003,8 +1003,17 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
snd_soc_write(codec, reg, cache[reg]);
/*
- * restore power and other regs according
- * to set_power() and set_clock()
+ * restore these regs according to the power setting sequence in
+ * sgtl5000_set_power_regs() and clock setting sequence in
+ * sgtl5000_set_clock().
+ *
+ * The order of restore is:
+ * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after
+ * SGTL5000_CHIP_ANA_POWER PLL bits set
+ * 2. SGTL5000_CHIP_LINREG_CTRL should be set before
+ * SGTL5000_CHIP_ANA_POWER LINREG_D restored
+ * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage,
+ * prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored
*/
snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
cache[SGTL5000_CHIP_LINREG_CTRL]);
@@ -1239,7 +1248,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
}
rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT;
- dev_info(codec->dev, "sgtl5000 revision %d\n", rev);
+ dev_info(codec->dev, "sgtl5000 revision 0x%x\n", rev);
/*
* workaround for revision 0x11 and later,
@@ -1344,15 +1353,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
if (ret)
goto err;
- snd_soc_add_controls(codec, sgtl5000_snd_controls,
- ARRAY_SIZE(sgtl5000_snd_controls));
-
- snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets,
- ARRAY_SIZE(sgtl5000_dapm_widgets));
-
- snd_soc_dapm_add_routes(&codec->dapm, audio_map,
- ARRAY_SIZE(audio_map));
-
snd_soc_dapm_new_widgets(&codec->dapm);
return 0;
@@ -1393,6 +1393,12 @@ static struct snd_soc_codec_driver sgtl5000_driver = {
.reg_cache_step = 2,
.reg_cache_default = sgtl5000_regs,
.volatile_register = sgtl5000_volatile_register,
+ .controls = sgtl5000_snd_controls,
+ .num_controls = ARRAY_SIZE(sgtl5000_snd_controls),
+ .dapm_widgets = sgtl5000_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(sgtl5000_dapm_widgets),
+ .dapm_routes = sgtl5000_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(sgtl5000_dapm_routes),
};
static __devinit int sgtl5000_i2c_probe(struct i2c_client *client,
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index f99baa0b8c39..50dbdb9357ea 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -827,8 +827,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
{
pr_debug("codec_probe called\n");
- codec->dapm.idle_bias_off = 1;
-
/* PCM interface config
* This sets the pcm rx slot conguration to max 6 slots
* for max 4 dais (2 stereo and 2 mono)
@@ -871,7 +869,7 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, SN95031_SSR2, 0x10);
snd_soc_write(codec, SN95031_SSR3, 0x40);
- snd_soc_add_controls(codec, sn95031_snd_controls,
+ snd_soc_add_codec_controls(codec, sn95031_snd_controls,
ARRAY_SIZE(sn95031_snd_controls));
return 0;
@@ -891,6 +889,7 @@ struct snd_soc_codec_driver sn95031_codec = {
.read = sn95031_read,
.write = sn95031_write,
.set_bias_level = sn95031_set_vaud_bias,
+ .idle_bias_off = true,
.dapm_widgets = sn95031_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(sn95031_dapm_widgets),
.dapm_routes = sn95031_audio_map,
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 333dd98af39c..de2b20544ceb 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -548,7 +548,7 @@ static int ssm2602_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, SSM2602_ROUT1V,
ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
- ret = snd_soc_add_controls(codec, ssm2602_snd_controls,
+ ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls,
ARRAY_SIZE(ssm2602_snd_controls));
if (ret)
return ret;
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
index cc0566c22ec1..982e437799a8 100644
--- a/sound/soc/codecs/stac9766.c
+++ b/sound/soc/codecs/stac9766.c
@@ -355,7 +355,7 @@ static int stac9766_codec_probe(struct snd_soc_codec *codec)
stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
+ snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls,
ARRAY_SIZE(stac9766_snd_ac97_controls));
return 0;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index dfa41a96599b..16d55f91a653 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -593,7 +593,7 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec)
snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1);
- snd_soc_add_controls(codec, tlv320aic23_snd_controls,
+ snd_soc_add_codec_controls(codec, tlv320aic23_snd_controls,
ARRAY_SIZE(tlv320aic23_snd_controls));
return 0;
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index a038daec682b..802064b5030d 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -389,7 +389,7 @@ static int aic26_probe(struct snd_soc_codec *codec)
/* register controls */
dev_dbg(codec->dev, "Registering controls\n");
- err = snd_soc_add_controls(codec, aic26_snd_controls,
+ err = snd_soc_add_codec_controls(codec, aic26_snd_controls,
ARRAY_SIZE(aic26_snd_controls));
WARN_ON(err < 0);
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index eb401ef021fb..b0a73d37ed52 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -60,7 +60,6 @@ struct aic32x4_rate_divs {
struct aic32x4_priv {
u32 sysclk;
- s32 master;
u8 page_no;
void *control_data;
u32 power_cfg;
@@ -369,7 +368,6 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
u8 iface_reg_1;
u8 iface_reg_2;
u8 iface_reg_3;
@@ -384,11 +382,9 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
- aic32x4->master = 1;
iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
- aic32x4->master = 0;
break;
default:
printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n");
@@ -526,64 +522,58 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
- struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
-
switch (level) {
case SND_SOC_BIAS_ON:
- if (aic32x4->master) {
- /* Switch on PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, AIC32X4_PLLEN);
-
- /* Switch on NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, AIC32X4_NDACEN);
-
- /* Switch on MDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_MDAC,
- AIC32X4_MDACEN, AIC32X4_MDACEN);
-
- /* Switch on NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, AIC32X4_NADCEN);
-
- /* Switch on MADC Divider */
- snd_soc_update_bits(codec, AIC32X4_MADC,
- AIC32X4_MADCEN, AIC32X4_MADCEN);
-
- /* Switch on BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, AIC32X4_BCLKEN);
- }
+ /* Switch on PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, AIC32X4_PLLEN);
+
+ /* Switch on NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, AIC32X4_NDACEN);
+
+ /* Switch on MDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MDAC,
+ AIC32X4_MDACEN, AIC32X4_MDACEN);
+
+ /* Switch on NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, AIC32X4_NADCEN);
+
+ /* Switch on MADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MADC,
+ AIC32X4_MADCEN, AIC32X4_MADCEN);
+
+ /* Switch on BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, AIC32X4_BCLKEN);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
- if (aic32x4->master) {
- /* Switch off PLL */
- snd_soc_update_bits(codec, AIC32X4_PLLPR,
- AIC32X4_PLLEN, 0);
-
- /* Switch off NDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_NDAC,
- AIC32X4_NDACEN, 0);
-
- /* Switch off MDAC Divider */
- snd_soc_update_bits(codec, AIC32X4_MDAC,
- AIC32X4_MDACEN, 0);
-
- /* Switch off NADC Divider */
- snd_soc_update_bits(codec, AIC32X4_NADC,
- AIC32X4_NADCEN, 0);
-
- /* Switch off MADC Divider */
- snd_soc_update_bits(codec, AIC32X4_MADC,
- AIC32X4_MADCEN, 0);
-
- /* Switch off BCLK_N Divider */
- snd_soc_update_bits(codec, AIC32X4_BCLKN,
- AIC32X4_BCLKEN, 0);
- }
+ /* Switch off PLL */
+ snd_soc_update_bits(codec, AIC32X4_PLLPR,
+ AIC32X4_PLLEN, 0);
+
+ /* Switch off NDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NDAC,
+ AIC32X4_NDACEN, 0);
+
+ /* Switch off MDAC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MDAC,
+ AIC32X4_MDACEN, 0);
+
+ /* Switch off NADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_NADC,
+ AIC32X4_NADCEN, 0);
+
+ /* Switch off MADC Divider */
+ snd_soc_update_bits(codec, AIC32X4_MADC,
+ AIC32X4_MADCEN, 0);
+
+ /* Switch off BCLK_N Divider */
+ snd_soc_update_bits(codec, AIC32X4_BCLKN,
+ AIC32X4_BCLKEN, 0);
break;
case SND_SOC_BIAS_OFF:
break;
@@ -651,9 +641,11 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) {
snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
}
- if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) {
- snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN);
- }
+
+ tmp_reg = (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) ?
+ AIC32X4_LDOCTLEN : 0;
+ snd_soc_write(codec, AIC32X4_LDOCTL, tmp_reg);
+
tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE);
if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) {
tmp_reg |= AIC32X4_LDOIN_18_36;
@@ -679,7 +671,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
}
aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
- snd_soc_add_controls(codec, aic32x4_snd_controls,
+ snd_soc_add_codec_controls(codec, aic32x4_snd_controls,
ARRAY_SIZE(aic32x4_snd_controls));
aic32x4_add_widgets(codec);
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 492f22f8a4d7..8d20f6ec20f3 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -121,30 +121,6 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = {
0x00, 0x00, 0x02, /* 100 */
};
-/*
- * read from the aic3x register space. Only use for this function is if
- * wanting to read volatile bits from those registers that has both read-only
- * and read/write bits. All other cases should use snd_soc_read.
- */
-static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
- u8 *value)
-{
- u8 *cache = codec->reg_cache;
-
- if (codec->cache_only)
- return -EINVAL;
- if (reg >= AIC3X_CACHEREGNUM)
- return -1;
-
- codec->cache_bypass = 1;
- *value = snd_soc_read(codec, reg);
- codec->cache_bypass = 0;
-
- cache[reg] = *value;
-
- return 0;
-}
-
#define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
@@ -1185,25 +1161,6 @@ static int aic3x_set_bias_level(struct snd_soc_codec *codec,
return 0;
}
-void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state)
-{
- u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
- u8 bit = gpio ? 3: 0;
- u8 val = snd_soc_read(codec, reg) & ~(1 << bit);
- snd_soc_write(codec, reg, val | (!!state << bit));
-}
-EXPORT_SYMBOL_GPL(aic3x_set_gpio);
-
-int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio)
-{
- u8 reg = gpio ? AIC3X_GPIO2_REG : AIC3X_GPIO1_REG;
- u8 val = 0, bit = gpio ? 2 : 1;
-
- aic3x_read(codec, reg, &val);
- return (val >> bit) & 1;
-}
-EXPORT_SYMBOL_GPL(aic3x_get_gpio);
-
void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
int headset_debounce, int button_debounce)
{
@@ -1221,23 +1178,6 @@ void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val);
}
-EXPORT_SYMBOL_GPL(aic3x_set_headset_detection);
-
-int aic3x_headset_detected(struct snd_soc_codec *codec)
-{
- u8 val = 0;
- aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
- return (val >> 4) & 1;
-}
-EXPORT_SYMBOL_GPL(aic3x_headset_detected);
-
-int aic3x_button_pressed(struct snd_soc_codec *codec)
-{
- u8 val = 0;
- aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val);
- return (val >> 5) & 1;
-}
-EXPORT_SYMBOL_GPL(aic3x_button_pressed);
#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
@@ -1377,7 +1317,6 @@ static int aic3x_probe(struct snd_soc_codec *codec)
INIT_LIST_HEAD(&aic3x->list);
aic3x->codec = codec;
- codec->dapm.idle_bias_off = 1;
ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type);
if (ret != 0) {
@@ -1426,10 +1365,10 @@ static int aic3x_probe(struct snd_soc_codec *codec)
(aic3x->setup->gpio_func[1] & 0xf) << 4);
}
- snd_soc_add_controls(codec, aic3x_snd_controls,
+ snd_soc_add_codec_controls(codec, aic3x_snd_controls,
ARRAY_SIZE(aic3x_snd_controls));
if (aic3x->model == AIC3X_MODEL_3007)
- snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1);
+ snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1);
aic3x_add_widgets(codec);
list_add(&aic3x->list, &reset_list);
@@ -1471,6 +1410,7 @@ static int aic3x_remove(struct snd_soc_codec *codec)
static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
.set_bias_level = aic3x_set_bias_level,
+ .idle_bias_off = true,
.reg_cache_size = ARRAY_SIZE(aic3x_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = aic3x_reg,
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 06a19784b162..6f097fb60683 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -212,9 +212,6 @@
/* Default input volume */
#define DEFAULT_GAIN 0x20
-void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state);
-int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio);
-
/* headset detection / button API */
/* The AIC3x supports detection of stereo headsets (GND + left + right signal)
@@ -252,10 +249,4 @@ enum {
#define AIC3X_BUTTON_DEBOUNCE_SHIFT 0
#define AIC3X_BUTTON_DEBOUNCE_MASK 3
-/* see the enums above for valid parameters to this function */
-void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect,
- int headset_debounce, int button_debounce);
-int aic3x_headset_detected(struct snd_soc_codec *codec);
-int aic3x_button_pressed(struct snd_soc_codec *codec);
-
#endif /* _AIC3X_H */
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index f0aad26cdb31..4587ddd0fbf8 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -806,8 +806,6 @@ static int dac33_startup(struct snd_pcm_substream *substream,
/* Stream started, save the substream pointer */
dac33->substream = substream;
- snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24);
-
return 0;
}
@@ -1397,7 +1395,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
codec->control_data = dac33->control_data;
codec->hw_write = (hw_write_t) i2c_master_send;
- codec->dapm.idle_bias_off = 1;
dac33->codec = codec;
/* Read the tlv320dac33 ID registers */
@@ -1440,7 +1437,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
/* Only add the FIFO controls, if we have valid IRQ number */
if (dac33->irq >= 0)
- snd_soc_add_controls(codec, dac33_mode_snd_controls,
+ snd_soc_add_codec_controls(codec, dac33_mode_snd_controls,
ARRAY_SIZE(dac33_mode_snd_controls));
err_power:
@@ -1478,6 +1475,7 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = {
.read = dac33_read_reg_cache,
.write = dac33_write_locked,
.set_bias_level = dac33_set_bias_level,
+ .idle_bias_off = true,
.reg_cache_size = ARRAY_SIZE(dac33_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = dac33_reg,
@@ -1515,7 +1513,9 @@ static struct snd_soc_dai_driver dac33_dai = {
.channels_min = 2,
.channels_max = 2,
.rates = DAC33_RATES,
- .formats = DAC33_FORMATS,},
+ .formats = DAC33_FORMATS,
+ .sig_bits = 24,
+ },
.ops = &dac33_dai_ops,
};
diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c
index 363b99dad8e9..6fe4aa3ac544 100644
--- a/sound/soc/codecs/tpa6130a2.c
+++ b/sound/soc/codecs/tpa6130a2.c
@@ -351,10 +351,10 @@ int tpa6130a2_add_controls(struct snd_soc_codec *codec)
data = i2c_get_clientdata(tpa6130a2_client);
if (data->id == TPA6140A2)
- return snd_soc_add_controls(codec, tpa6140a2_controls,
+ return snd_soc_add_codec_controls(codec, tpa6140a2_controls,
ARRAY_SIZE(tpa6140a2_controls));
else
- return snd_soc_add_controls(codec, tpa6130a2_controls,
+ return snd_soc_add_codec_controls(codec, tpa6130a2_controls,
ARRAY_SIZE(tpa6130a2_controls));
}
EXPORT_SYMBOL_GPL(tpa6130a2_add_controls);
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index 18e71014cc2e..170cf9a8fc79 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -1002,8 +1002,8 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
unsigned short mask, bitmask;
if (twl4030->configured) {
- printk(KERN_ERR "twl4030 operation mode cannot be "
- "changed on-the-fly\n");
+ dev_err(codec->dev,
+ "operation mode cannot be changed on-the-fly\n");
return -EBUSY;
}
@@ -1689,7 +1689,6 @@ static int twl4030_startup(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = rtd->codec;
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
- snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24);
if (twl4030->master_substream) {
twl4030->slave_substream = substream;
/* The DAI has one configuration for playback and capture, so
@@ -1801,7 +1800,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
mode |= TWL4030_APLL_RATE_96000;
break;
default:
- printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
+ dev_err(codec->dev, "%s: unknown rate %d\n", __func__,
params_rate(params));
return -EINVAL;
}
@@ -1818,7 +1817,7 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
format |= TWL4030_DATA_WIDTH_32S_24W;
break;
default:
- printk(KERN_ERR "TWL4030 hw params: unknown format %d\n",
+ dev_err(codec->dev, "%s: unknown format %d\n", __func__,
params_format(params));
return -EINVAL;
}
@@ -1868,13 +1867,13 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai,
case 38400000:
break;
default:
- dev_err(codec->dev, "Unsupported APLL mclk: %u\n", freq);
+ dev_err(codec->dev, "Unsupported HFCLKIN: %u\n", freq);
return -EINVAL;
}
if ((freq / 1000) != twl4030->sysclk) {
dev_err(codec->dev,
- "Mismatch in APLL mclk: %u (configured: %u)\n",
+ "Mismatch in HFCLKIN: %u (configured: %u)\n",
freq, twl4030->sysclk * 1000);
return -EINVAL;
}
@@ -1984,9 +1983,9 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
* not available.
*/
if (twl4030->sysclk != 26000) {
- dev_err(codec->dev, "The board is configured for %u Hz, while"
- "the Voice interface needs 26MHz APLL mclk\n",
- twl4030->sysclk * 1000);
+ dev_err(codec->dev,
+ "%s: HFCLKIN is %u KHz, voice interface needs 26MHz\n",
+ __func__, twl4030->sysclk);
return -EINVAL;
}
@@ -1997,8 +1996,8 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream,
& TWL4030_OPT_MODE;
if (mode != TWL4030_OPTION_2) {
- printk(KERN_ERR "TWL4030 voice startup: "
- "the codec mode is not option2\n");
+ dev_err(codec->dev, "%s: the codec mode is not option2\n",
+ __func__);
return -EINVAL;
}
@@ -2039,7 +2038,7 @@ static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
mode |= TWL4030_SEL_16K;
break;
default:
- printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n",
+ dev_err(codec->dev, "%s: unknown rate %d\n", __func__,
params_rate(params));
return -EINVAL;
}
@@ -2068,13 +2067,14 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai,
struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec);
if (freq != 26000000) {
- dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice"
- "interface needs 26MHz APLL mclk\n", freq);
+ dev_err(codec->dev,
+ "%s: HFCLKIN is %u KHz, voice interface needs 26MHz\n",
+ __func__, freq / 1000);
return -EINVAL;
}
if ((freq / 1000) != twl4030->sysclk) {
dev_err(codec->dev,
- "Mismatch in APLL mclk: %u (configured: %u)\n",
+ "Mismatch in HFCLKIN: %u (configured: %u)\n",
freq, twl4030->sysclk * 1000);
return -EINVAL;
}
@@ -2175,13 +2175,15 @@ static struct snd_soc_dai_driver twl4030_dai[] = {
.channels_min = 2,
.channels_max = 4,
.rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
- .formats = TWL4030_FORMATS,},
+ .formats = TWL4030_FORMATS,
+ .sig_bits = 24,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 4,
.rates = TWL4030_RATES,
- .formats = TWL4030_FORMATS,},
+ .formats = TWL4030_FORMATS,
+ .sig_bits = 24,},
.ops = &twl4030_dai_hifi_ops,
},
{
@@ -2220,13 +2222,12 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec)
twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL);
if (twl4030 == NULL) {
- printk("Can not allocate memroy\n");
+ dev_err(codec->dev, "Can not allocate memory\n");
return -ENOMEM;
}
snd_soc_codec_set_drvdata(codec, twl4030);
/* Set the defaults, and power up the codec */
twl4030->sysclk = twl4030_audio_get_mclk() / 1000;
- codec->dapm.idle_bias_off = 1;
twl4030_init_chip(codec);
@@ -2252,6 +2253,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
.read = twl4030_read_reg_cache,
.write = twl4030_write,
.set_bias_level = twl4030_set_bias_level,
+ .idle_bias_off = true,
.reg_cache_size = sizeof(twl4030_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = twl4030_reg,
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index 5b9c79b6f65e..2d8c6b825e57 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -1052,6 +1052,19 @@ int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim)
}
EXPORT_SYMBOL_GPL(twl6040_get_trim_value);
+int twl6040_get_hs_step_size(struct snd_soc_codec *codec)
+{
+ struct twl6040 *twl6040 = codec->control_data;
+
+ if (twl6040_get_revid(twl6040) < TWL6040_REV_ES1_2)
+ /* For ES under ES_1.3 HS step is 2 mV */
+ return 2;
+ else
+ /* For ES_1.3 HS step is 1 mV */
+ return 1;
+}
+EXPORT_SYMBOL_GPL(twl6040_get_hs_step_size);
+
static const struct snd_kcontrol_new twl6040_snd_controls[] = {
/* Capture gains */
SOC_DOUBLE_TLV("Capture Preamplifier Volume",
@@ -1125,14 +1138,14 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
TWL6040_REG_MICRCTL, 2, 0),
/* Microphone bias */
- SND_SOC_DAPM_MICBIAS("Headset Mic Bias",
- TWL6040_REG_AMICBCTL, 0, 0),
- SND_SOC_DAPM_MICBIAS("Main Mic Bias",
- TWL6040_REG_AMICBCTL, 4, 0),
- SND_SOC_DAPM_MICBIAS("Digital Mic1 Bias",
- TWL6040_REG_DMICBCTL, 0, 0),
- SND_SOC_DAPM_MICBIAS("Digital Mic2 Bias",
- TWL6040_REG_DMICBCTL, 4, 0),
+ SND_SOC_DAPM_SUPPLY("Headset Mic Bias",
+ TWL6040_REG_AMICBCTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Main Mic Bias",
+ TWL6040_REG_AMICBCTL, 4, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias",
+ TWL6040_REG_DMICBCTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias",
+ TWL6040_REG_DMICBCTL, 4, 0, NULL, 0),
/* DACs */
SND_SOC_DAPM_DAC("HSDAC Left", "Headset Playback", SND_SOC_NOPM, 0, 0),
@@ -1527,7 +1540,6 @@ static int twl6040_probe(struct snd_soc_codec *codec)
priv->codec = codec;
codec->control_data = dev_get_drvdata(codec->dev->parent);
- codec->ignore_pmdown_time = 1;
if (pdata && pdata->hs_left_step && pdata->hs_right_step) {
priv->hs_left_step = pdata->hs_left_step;
@@ -1613,6 +1625,7 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = {
.reg_cache_size = ARRAY_SIZE(twl6040_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = twl6040_reg,
+ .ignore_pmdown_time = true,
.controls = twl6040_snd_controls,
.num_controls = ARRAY_SIZE(twl6040_snd_controls),
diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h
index ef273f1fac2f..0611406ca7c0 100644
--- a/sound/soc/codecs/twl6040.h
+++ b/sound/soc/codecs/twl6040.h
@@ -39,5 +39,6 @@ void twl6040_hs_jack_detect(struct snd_soc_codec *codec,
struct snd_soc_jack *jack, int report);
int twl6040_get_clk_id(struct snd_soc_codec *codec);
int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim);
+int twl6040_get_hs_step_size(struct snd_soc_codec *codec);
#endif /* End of __TWL6040_H__ */
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index 8f4f469d6411..797b0dde2c68 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -531,15 +531,15 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec)
switch (pd->model) {
case UDA134X_UDA1340:
case UDA134X_UDA1344:
- ret = snd_soc_add_controls(codec, uda1340_snd_controls,
+ ret = snd_soc_add_codec_controls(codec, uda1340_snd_controls,
ARRAY_SIZE(uda1340_snd_controls));
break;
case UDA134X_UDA1341:
- ret = snd_soc_add_controls(codec, uda1341_snd_controls,
+ ret = snd_soc_add_codec_controls(codec, uda1341_snd_controls,
ARRAY_SIZE(uda1341_snd_controls));
break;
case UDA134X_UDA1345:
- ret = snd_soc_add_controls(codec, uda1345_snd_controls,
+ ret = snd_soc_add_codec_controls(codec, uda1345_snd_controls,
ARRAY_SIZE(uda1345_snd_controls));
break;
default:
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
index 44aacf927ba9..3d868dc40092 100644
--- a/sound/soc/codecs/wl1273.c
+++ b/sound/soc/codecs/wl1273.c
@@ -464,7 +464,7 @@ static int wl1273_probe(struct snd_soc_codec *codec)
snd_soc_codec_set_drvdata(codec, wl1273);
- r = snd_soc_add_controls(codec, wl1273_controls,
+ r = snd_soc_add_codec_controls(codec, wl1273_controls,
ARRAY_SIZE(wl1273_controls));
if (r)
kfree(wl1273);
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index c2880907fced..a75c3766aede 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -733,8 +733,9 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
struct wm2000_priv *wm2000;
struct wm2000_platform_data *pdata;
const char *filename;
- const struct firmware *fw;
- int reg, ret;
+ const struct firmware *fw = NULL;
+ int ret;
+ int reg;
u16 id;
wm2000 = devm_kzalloc(&i2c->dev, sizeof(struct wm2000_priv),
@@ -751,7 +752,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
ret = PTR_ERR(wm2000->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
ret);
- goto err;
+ goto out;
}
/* Verify that this is a WM2000 */
@@ -763,7 +764,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
if (id != 0x2000) {
dev_err(&i2c->dev, "Device is not a WM2000 - ID %x\n", id);
ret = -ENODEV;
- goto err_regmap;
+ goto out_regmap_exit;
}
reg = wm2000_read(i2c, WM2000_REG_REVISON);
@@ -782,7 +783,7 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
ret = request_firmware(&fw, filename, &i2c->dev);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to acquire ANC data: %d\n", ret);
- goto err_regmap;
+ goto out_regmap_exit;
}
/* Pre-cook the concatenation of the register address onto the image */
@@ -793,15 +794,13 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
if (wm2000->anc_download == NULL) {
dev_err(&i2c->dev, "Out of memory\n");
ret = -ENOMEM;
- goto err_fw;
+ goto out_regmap_exit;
}
wm2000->anc_download[0] = 0x80;
wm2000->anc_download[1] = 0x00;
memcpy(wm2000->anc_download + 2, fw->data, fw->size);
- release_firmware(fw);
-
wm2000->anc_eng_ena = 1;
wm2000->anc_active = 1;
wm2000->spk_ena = 1;
@@ -809,18 +808,14 @@ static int __devinit wm2000_i2c_probe(struct i2c_client *i2c,
wm2000_reset(wm2000);
- ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000,
- NULL, 0);
- if (ret != 0)
- goto err_fw;
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm2000, NULL, 0);
+ if (!ret)
+ goto out;
- return 0;
-
-err_fw:
- release_firmware(fw);
-err_regmap:
+out_regmap_exit:
regmap_exit(wm2000->regmap);
-err:
+out:
+ release_firmware(fw);
return ret;
}
diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c
new file mode 100644
index 000000000000..acbdc5fde923
--- /dev/null
+++ b/sound/soc/codecs/wm2200.c
@@ -0,0 +1,2286 @@
+/*
+ * wm2200.c -- WM2200 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/gcd.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+#include <linux/pm_runtime.h>
+#include <linux/regulator/consumer.h>
+#include <linux/regulator/fixed.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/wm2200.h>
+
+#include "wm2200.h"
+
+/* The code assumes DCVDD is generated internally */
+#define WM2200_NUM_CORE_SUPPLIES 2
+static const char *wm2200_core_supply_names[WM2200_NUM_CORE_SUPPLIES] = {
+ "DBVDD",
+ "LDOVDD",
+};
+
+struct wm2200_fll {
+ int fref;
+ int fout;
+ int src;
+ struct completion lock;
+};
+
+/* codec private data */
+struct wm2200_priv {
+ struct regmap *regmap;
+ struct device *dev;
+ struct snd_soc_codec *codec;
+ struct wm2200_pdata pdata;
+ struct regulator_bulk_data core_supplies[WM2200_NUM_CORE_SUPPLIES];
+
+ struct completion fll_lock;
+ int fll_fout;
+ int fll_fref;
+ int fll_src;
+
+ int rev;
+ int sysclk;
+};
+
+static struct reg_default wm2200_reg_defaults[] = {
+ { 0x000B, 0x0000 }, /* R11 - Tone Generator 1 */
+ { 0x0102, 0x0000 }, /* R258 - Clocking 3 */
+ { 0x0103, 0x0011 }, /* R259 - Clocking 4 */
+ { 0x0111, 0x0000 }, /* R273 - FLL Control 1 */
+ { 0x0112, 0x0000 }, /* R274 - FLL Control 2 */
+ { 0x0113, 0x0000 }, /* R275 - FLL Control 3 */
+ { 0x0114, 0x0000 }, /* R276 - FLL Control 4 */
+ { 0x0116, 0x0177 }, /* R278 - FLL Control 6 */
+ { 0x0117, 0x0004 }, /* R279 - FLL Control 7 */
+ { 0x0119, 0x0000 }, /* R281 - FLL EFS 1 */
+ { 0x011A, 0x0002 }, /* R282 - FLL EFS 2 */
+ { 0x0200, 0x0000 }, /* R512 - Mic Charge Pump 1 */
+ { 0x0201, 0x03FF }, /* R513 - Mic Charge Pump 2 */
+ { 0x0202, 0x9BDE }, /* R514 - DM Charge Pump 1 */
+ { 0x020C, 0x0000 }, /* R524 - Mic Bias Ctrl 1 */
+ { 0x020D, 0x0000 }, /* R525 - Mic Bias Ctrl 2 */
+ { 0x020F, 0x0000 }, /* R527 - Ear Piece Ctrl 1 */
+ { 0x0210, 0x0000 }, /* R528 - Ear Piece Ctrl 2 */
+ { 0x0301, 0x0000 }, /* R769 - Input Enables */
+ { 0x0302, 0x2240 }, /* R770 - IN1L Control */
+ { 0x0303, 0x0040 }, /* R771 - IN1R Control */
+ { 0x0304, 0x2240 }, /* R772 - IN2L Control */
+ { 0x0305, 0x0040 }, /* R773 - IN2R Control */
+ { 0x0306, 0x2240 }, /* R774 - IN3L Control */
+ { 0x0307, 0x0040 }, /* R775 - IN3R Control */
+ { 0x030A, 0x0000 }, /* R778 - RXANC_SRC */
+ { 0x030B, 0x0022 }, /* R779 - Input Volume Ramp */
+ { 0x030C, 0x0180 }, /* R780 - ADC Digital Volume 1L */
+ { 0x030D, 0x0180 }, /* R781 - ADC Digital Volume 1R */
+ { 0x030E, 0x0180 }, /* R782 - ADC Digital Volume 2L */
+ { 0x030F, 0x0180 }, /* R783 - ADC Digital Volume 2R */
+ { 0x0310, 0x0180 }, /* R784 - ADC Digital Volume 3L */
+ { 0x0311, 0x0180 }, /* R785 - ADC Digital Volume 3R */
+ { 0x0400, 0x0000 }, /* R1024 - Output Enables */
+ { 0x0401, 0x0000 }, /* R1025 - DAC Volume Limit 1L */
+ { 0x0402, 0x0000 }, /* R1026 - DAC Volume Limit 1R */
+ { 0x0403, 0x0000 }, /* R1027 - DAC Volume Limit 2L */
+ { 0x0404, 0x0000 }, /* R1028 - DAC Volume Limit 2R */
+ { 0x0409, 0x0000 }, /* R1033 - DAC AEC Control 1 */
+ { 0x040A, 0x0022 }, /* R1034 - Output Volume Ramp */
+ { 0x040B, 0x0180 }, /* R1035 - DAC Digital Volume 1L */
+ { 0x040C, 0x0180 }, /* R1036 - DAC Digital Volume 1R */
+ { 0x040D, 0x0180 }, /* R1037 - DAC Digital Volume 2L */
+ { 0x040E, 0x0180 }, /* R1038 - DAC Digital Volume 2R */
+ { 0x0417, 0x0069 }, /* R1047 - PDM 1 */
+ { 0x0418, 0x0000 }, /* R1048 - PDM 2 */
+ { 0x0500, 0x0000 }, /* R1280 - Audio IF 1_1 */
+ { 0x0501, 0x0008 }, /* R1281 - Audio IF 1_2 */
+ { 0x0502, 0x0000 }, /* R1282 - Audio IF 1_3 */
+ { 0x0503, 0x0000 }, /* R1283 - Audio IF 1_4 */
+ { 0x0504, 0x0000 }, /* R1284 - Audio IF 1_5 */
+ { 0x0505, 0x0001 }, /* R1285 - Audio IF 1_6 */
+ { 0x0506, 0x0001 }, /* R1286 - Audio IF 1_7 */
+ { 0x0507, 0x0000 }, /* R1287 - Audio IF 1_8 */
+ { 0x0508, 0x0000 }, /* R1288 - Audio IF 1_9 */
+ { 0x0509, 0x0000 }, /* R1289 - Audio IF 1_10 */
+ { 0x050A, 0x0000 }, /* R1290 - Audio IF 1_11 */
+ { 0x050B, 0x0000 }, /* R1291 - Audio IF 1_12 */
+ { 0x050C, 0x0000 }, /* R1292 - Audio IF 1_13 */
+ { 0x050D, 0x0000 }, /* R1293 - Audio IF 1_14 */
+ { 0x050E, 0x0000 }, /* R1294 - Audio IF 1_15 */
+ { 0x050F, 0x0000 }, /* R1295 - Audio IF 1_16 */
+ { 0x0510, 0x0000 }, /* R1296 - Audio IF 1_17 */
+ { 0x0511, 0x0000 }, /* R1297 - Audio IF 1_18 */
+ { 0x0512, 0x0000 }, /* R1298 - Audio IF 1_19 */
+ { 0x0513, 0x0000 }, /* R1299 - Audio IF 1_20 */
+ { 0x0514, 0x0000 }, /* R1300 - Audio IF 1_21 */
+ { 0x0515, 0x0001 }, /* R1301 - Audio IF 1_22 */
+ { 0x0600, 0x0000 }, /* R1536 - OUT1LMIX Input 1 Source */
+ { 0x0601, 0x0080 }, /* R1537 - OUT1LMIX Input 1 Volume */
+ { 0x0602, 0x0000 }, /* R1538 - OUT1LMIX Input 2 Source */
+ { 0x0603, 0x0080 }, /* R1539 - OUT1LMIX Input 2 Volume */
+ { 0x0604, 0x0000 }, /* R1540 - OUT1LMIX Input 3 Source */
+ { 0x0605, 0x0080 }, /* R1541 - OUT1LMIX Input 3 Volume */
+ { 0x0606, 0x0000 }, /* R1542 - OUT1LMIX Input 4 Source */
+ { 0x0607, 0x0080 }, /* R1543 - OUT1LMIX Input 4 Volume */
+ { 0x0608, 0x0000 }, /* R1544 - OUT1RMIX Input 1 Source */
+ { 0x0609, 0x0080 }, /* R1545 - OUT1RMIX Input 1 Volume */
+ { 0x060A, 0x0000 }, /* R1546 - OUT1RMIX Input 2 Source */
+ { 0x060B, 0x0080 }, /* R1547 - OUT1RMIX Input 2 Volume */
+ { 0x060C, 0x0000 }, /* R1548 - OUT1RMIX Input 3 Source */
+ { 0x060D, 0x0080 }, /* R1549 - OUT1RMIX Input 3 Volume */
+ { 0x060E, 0x0000 }, /* R1550 - OUT1RMIX Input 4 Source */
+ { 0x060F, 0x0080 }, /* R1551 - OUT1RMIX Input 4 Volume */
+ { 0x0610, 0x0000 }, /* R1552 - OUT2LMIX Input 1 Source */
+ { 0x0611, 0x0080 }, /* R1553 - OUT2LMIX Input 1 Volume */
+ { 0x0612, 0x0000 }, /* R1554 - OUT2LMIX Input 2 Source */
+ { 0x0613, 0x0080 }, /* R1555 - OUT2LMIX Input 2 Volume */
+ { 0x0614, 0x0000 }, /* R1556 - OUT2LMIX Input 3 Source */
+ { 0x0615, 0x0080 }, /* R1557 - OUT2LMIX Input 3 Volume */
+ { 0x0616, 0x0000 }, /* R1558 - OUT2LMIX Input 4 Source */
+ { 0x0617, 0x0080 }, /* R1559 - OUT2LMIX Input 4 Volume */
+ { 0x0618, 0x0000 }, /* R1560 - OUT2RMIX Input 1 Source */
+ { 0x0619, 0x0080 }, /* R1561 - OUT2RMIX Input 1 Volume */
+ { 0x061A, 0x0000 }, /* R1562 - OUT2RMIX Input 2 Source */
+ { 0x061B, 0x0080 }, /* R1563 - OUT2RMIX Input 2 Volume */
+ { 0x061C, 0x0000 }, /* R1564 - OUT2RMIX Input 3 Source */
+ { 0x061D, 0x0080 }, /* R1565 - OUT2RMIX Input 3 Volume */
+ { 0x061E, 0x0000 }, /* R1566 - OUT2RMIX Input 4 Source */
+ { 0x061F, 0x0080 }, /* R1567 - OUT2RMIX Input 4 Volume */
+ { 0x0620, 0x0000 }, /* R1568 - AIF1TX1MIX Input 1 Source */
+ { 0x0621, 0x0080 }, /* R1569 - AIF1TX1MIX Input 1 Volume */
+ { 0x0622, 0x0000 }, /* R1570 - AIF1TX1MIX Input 2 Source */
+ { 0x0623, 0x0080 }, /* R1571 - AIF1TX1MIX Input 2 Volume */
+ { 0x0624, 0x0000 }, /* R1572 - AIF1TX1MIX Input 3 Source */
+ { 0x0625, 0x0080 }, /* R1573 - AIF1TX1MIX Input 3 Volume */
+ { 0x0626, 0x0000 }, /* R1574 - AIF1TX1MIX Input 4 Source */
+ { 0x0627, 0x0080 }, /* R1575 - AIF1TX1MIX Input 4 Volume */
+ { 0x0628, 0x0000 }, /* R1576 - AIF1TX2MIX Input 1 Source */
+ { 0x0629, 0x0080 }, /* R1577 - AIF1TX2MIX Input 1 Volume */
+ { 0x062A, 0x0000 }, /* R1578 - AIF1TX2MIX Input 2 Source */
+ { 0x062B, 0x0080 }, /* R1579 - AIF1TX2MIX Input 2 Volume */
+ { 0x062C, 0x0000 }, /* R1580 - AIF1TX2MIX Input 3 Source */
+ { 0x062D, 0x0080 }, /* R1581 - AIF1TX2MIX Input 3 Volume */
+ { 0x062E, 0x0000 }, /* R1582 - AIF1TX2MIX Input 4 Source */
+ { 0x062F, 0x0080 }, /* R1583 - AIF1TX2MIX Input 4 Volume */
+ { 0x0630, 0x0000 }, /* R1584 - AIF1TX3MIX Input 1 Source */
+ { 0x0631, 0x0080 }, /* R1585 - AIF1TX3MIX Input 1 Volume */
+ { 0x0632, 0x0000 }, /* R1586 - AIF1TX3MIX Input 2 Source */
+ { 0x0633, 0x0080 }, /* R1587 - AIF1TX3MIX Input 2 Volume */
+ { 0x0634, 0x0000 }, /* R1588 - AIF1TX3MIX Input 3 Source */
+ { 0x0635, 0x0080 }, /* R1589 - AIF1TX3MIX Input 3 Volume */
+ { 0x0636, 0x0000 }, /* R1590 - AIF1TX3MIX Input 4 Source */
+ { 0x0637, 0x0080 }, /* R1591 - AIF1TX3MIX Input 4 Volume */
+ { 0x0638, 0x0000 }, /* R1592 - AIF1TX4MIX Input 1 Source */
+ { 0x0639, 0x0080 }, /* R1593 - AIF1TX4MIX Input 1 Volume */
+ { 0x063A, 0x0000 }, /* R1594 - AIF1TX4MIX Input 2 Source */
+ { 0x063B, 0x0080 }, /* R1595 - AIF1TX4MIX Input 2 Volume */
+ { 0x063C, 0x0000 }, /* R1596 - AIF1TX4MIX Input 3 Source */
+ { 0x063D, 0x0080 }, /* R1597 - AIF1TX4MIX Input 3 Volume */
+ { 0x063E, 0x0000 }, /* R1598 - AIF1TX4MIX Input 4 Source */
+ { 0x063F, 0x0080 }, /* R1599 - AIF1TX4MIX Input 4 Volume */
+ { 0x0640, 0x0000 }, /* R1600 - AIF1TX5MIX Input 1 Source */
+ { 0x0641, 0x0080 }, /* R1601 - AIF1TX5MIX Input 1 Volume */
+ { 0x0642, 0x0000 }, /* R1602 - AIF1TX5MIX Input 2 Source */
+ { 0x0643, 0x0080 }, /* R1603 - AIF1TX5MIX Input 2 Volume */
+ { 0x0644, 0x0000 }, /* R1604 - AIF1TX5MIX Input 3 Source */
+ { 0x0645, 0x0080 }, /* R1605 - AIF1TX5MIX Input 3 Volume */
+ { 0x0646, 0x0000 }, /* R1606 - AIF1TX5MIX Input 4 Source */
+ { 0x0647, 0x0080 }, /* R1607 - AIF1TX5MIX Input 4 Volume */
+ { 0x0648, 0x0000 }, /* R1608 - AIF1TX6MIX Input 1 Source */
+ { 0x0649, 0x0080 }, /* R1609 - AIF1TX6MIX Input 1 Volume */
+ { 0x064A, 0x0000 }, /* R1610 - AIF1TX6MIX Input 2 Source */
+ { 0x064B, 0x0080 }, /* R1611 - AIF1TX6MIX Input 2 Volume */
+ { 0x064C, 0x0000 }, /* R1612 - AIF1TX6MIX Input 3 Source */
+ { 0x064D, 0x0080 }, /* R1613 - AIF1TX6MIX Input 3 Volume */
+ { 0x064E, 0x0000 }, /* R1614 - AIF1TX6MIX Input 4 Source */
+ { 0x064F, 0x0080 }, /* R1615 - AIF1TX6MIX Input 4 Volume */
+ { 0x0650, 0x0000 }, /* R1616 - EQLMIX Input 1 Source */
+ { 0x0651, 0x0080 }, /* R1617 - EQLMIX Input 1 Volume */
+ { 0x0652, 0x0000 }, /* R1618 - EQLMIX Input 2 Source */
+ { 0x0653, 0x0080 }, /* R1619 - EQLMIX Input 2 Volume */
+ { 0x0654, 0x0000 }, /* R1620 - EQLMIX Input 3 Source */
+ { 0x0655, 0x0080 }, /* R1621 - EQLMIX Input 3 Volume */
+ { 0x0656, 0x0000 }, /* R1622 - EQLMIX Input 4 Source */
+ { 0x0657, 0x0080 }, /* R1623 - EQLMIX Input 4 Volume */
+ { 0x0658, 0x0000 }, /* R1624 - EQRMIX Input 1 Source */
+ { 0x0659, 0x0080 }, /* R1625 - EQRMIX Input 1 Volume */
+ { 0x065A, 0x0000 }, /* R1626 - EQRMIX Input 2 Source */
+ { 0x065B, 0x0080 }, /* R1627 - EQRMIX Input 2 Volume */
+ { 0x065C, 0x0000 }, /* R1628 - EQRMIX Input 3 Source */
+ { 0x065D, 0x0080 }, /* R1629 - EQRMIX Input 3 Volume */
+ { 0x065E, 0x0000 }, /* R1630 - EQRMIX Input 4 Source */
+ { 0x065F, 0x0080 }, /* R1631 - EQRMIX Input 4 Volume */
+ { 0x0660, 0x0000 }, /* R1632 - LHPF1MIX Input 1 Source */
+ { 0x0661, 0x0080 }, /* R1633 - LHPF1MIX Input 1 Volume */
+ { 0x0662, 0x0000 }, /* R1634 - LHPF1MIX Input 2 Source */
+ { 0x0663, 0x0080 }, /* R1635 - LHPF1MIX Input 2 Volume */
+ { 0x0664, 0x0000 }, /* R1636 - LHPF1MIX Input 3 Source */
+ { 0x0665, 0x0080 }, /* R1637 - LHPF1MIX Input 3 Volume */
+ { 0x0666, 0x0000 }, /* R1638 - LHPF1MIX Input 4 Source */
+ { 0x0667, 0x0080 }, /* R1639 - LHPF1MIX Input 4 Volume */
+ { 0x0668, 0x0000 }, /* R1640 - LHPF2MIX Input 1 Source */
+ { 0x0669, 0x0080 }, /* R1641 - LHPF2MIX Input 1 Volume */
+ { 0x066A, 0x0000 }, /* R1642 - LHPF2MIX Input 2 Source */
+ { 0x066B, 0x0080 }, /* R1643 - LHPF2MIX Input 2 Volume */
+ { 0x066C, 0x0000 }, /* R1644 - LHPF2MIX Input 3 Source */
+ { 0x066D, 0x0080 }, /* R1645 - LHPF2MIX Input 3 Volume */
+ { 0x066E, 0x0000 }, /* R1646 - LHPF2MIX Input 4 Source */
+ { 0x066F, 0x0080 }, /* R1647 - LHPF2MIX Input 4 Volume */
+ { 0x0670, 0x0000 }, /* R1648 - DSP1LMIX Input 1 Source */
+ { 0x0671, 0x0080 }, /* R1649 - DSP1LMIX Input 1 Volume */
+ { 0x0672, 0x0000 }, /* R1650 - DSP1LMIX Input 2 Source */
+ { 0x0673, 0x0080 }, /* R1651 - DSP1LMIX Input 2 Volume */
+ { 0x0674, 0x0000 }, /* R1652 - DSP1LMIX Input 3 Source */
+ { 0x0675, 0x0080 }, /* R1653 - DSP1LMIX Input 3 Volume */
+ { 0x0676, 0x0000 }, /* R1654 - DSP1LMIX Input 4 Source */
+ { 0x0677, 0x0080 }, /* R1655 - DSP1LMIX Input 4 Volume */
+ { 0x0678, 0x0000 }, /* R1656 - DSP1RMIX Input 1 Source */
+ { 0x0679, 0x0080 }, /* R1657 - DSP1RMIX Input 1 Volume */
+ { 0x067A, 0x0000 }, /* R1658 - DSP1RMIX Input 2 Source */
+ { 0x067B, 0x0080 }, /* R1659 - DSP1RMIX Input 2 Volume */
+ { 0x067C, 0x0000 }, /* R1660 - DSP1RMIX Input 3 Source */
+ { 0x067D, 0x0080 }, /* R1661 - DSP1RMIX Input 3 Volume */
+ { 0x067E, 0x0000 }, /* R1662 - DSP1RMIX Input 4 Source */
+ { 0x067F, 0x0080 }, /* R1663 - DSP1RMIX Input 4 Volume */
+ { 0x0680, 0x0000 }, /* R1664 - DSP1AUX1MIX Input 1 Source */
+ { 0x0681, 0x0000 }, /* R1665 - DSP1AUX2MIX Input 1 Source */
+ { 0x0682, 0x0000 }, /* R1666 - DSP1AUX3MIX Input 1 Source */
+ { 0x0683, 0x0000 }, /* R1667 - DSP1AUX4MIX Input 1 Source */
+ { 0x0684, 0x0000 }, /* R1668 - DSP1AUX5MIX Input 1 Source */
+ { 0x0685, 0x0000 }, /* R1669 - DSP1AUX6MIX Input 1 Source */
+ { 0x0686, 0x0000 }, /* R1670 - DSP2LMIX Input 1 Source */
+ { 0x0687, 0x0080 }, /* R1671 - DSP2LMIX Input 1 Volume */
+ { 0x0688, 0x0000 }, /* R1672 - DSP2LMIX Input 2 Source */
+ { 0x0689, 0x0080 }, /* R1673 - DSP2LMIX Input 2 Volume */
+ { 0x068A, 0x0000 }, /* R1674 - DSP2LMIX Input 3 Source */
+ { 0x068B, 0x0080 }, /* R1675 - DSP2LMIX Input 3 Volume */
+ { 0x068C, 0x0000 }, /* R1676 - DSP2LMIX Input 4 Source */
+ { 0x068D, 0x0080 }, /* R1677 - DSP2LMIX Input 4 Volume */
+ { 0x068E, 0x0000 }, /* R1678 - DSP2RMIX Input 1 Source */
+ { 0x068F, 0x0080 }, /* R1679 - DSP2RMIX Input 1 Volume */
+ { 0x0690, 0x0000 }, /* R1680 - DSP2RMIX Input 2 Source */
+ { 0x0691, 0x0080 }, /* R1681 - DSP2RMIX Input 2 Volume */
+ { 0x0692, 0x0000 }, /* R1682 - DSP2RMIX Input 3 Source */
+ { 0x0693, 0x0080 }, /* R1683 - DSP2RMIX Input 3 Volume */
+ { 0x0694, 0x0000 }, /* R1684 - DSP2RMIX Input 4 Source */
+ { 0x0695, 0x0080 }, /* R1685 - DSP2RMIX Input 4 Volume */
+ { 0x0696, 0x0000 }, /* R1686 - DSP2AUX1MIX Input 1 Source */
+ { 0x0697, 0x0000 }, /* R1687 - DSP2AUX2MIX Input 1 Source */
+ { 0x0698, 0x0000 }, /* R1688 - DSP2AUX3MIX Input 1 Source */
+ { 0x0699, 0x0000 }, /* R1689 - DSP2AUX4MIX Input 1 Source */
+ { 0x069A, 0x0000 }, /* R1690 - DSP2AUX5MIX Input 1 Source */
+ { 0x069B, 0x0000 }, /* R1691 - DSP2AUX6MIX Input 1 Source */
+ { 0x0700, 0xA101 }, /* R1792 - GPIO CTRL 1 */
+ { 0x0701, 0xA101 }, /* R1793 - GPIO CTRL 2 */
+ { 0x0702, 0xA101 }, /* R1794 - GPIO CTRL 3 */
+ { 0x0703, 0xA101 }, /* R1795 - GPIO CTRL 4 */
+ { 0x0709, 0x0000 }, /* R1801 - Misc Pad Ctrl 1 */
+ { 0x0801, 0x00FF }, /* R2049 - Interrupt Status 1 Mask */
+ { 0x0804, 0xFFFF }, /* R2052 - Interrupt Status 2 Mask */
+ { 0x0808, 0x0000 }, /* R2056 - Interrupt Control */
+ { 0x0900, 0x0000 }, /* R2304 - EQL_1 */
+ { 0x0901, 0x0000 }, /* R2305 - EQL_2 */
+ { 0x0902, 0x0000 }, /* R2306 - EQL_3 */
+ { 0x0903, 0x0000 }, /* R2307 - EQL_4 */
+ { 0x0904, 0x0000 }, /* R2308 - EQL_5 */
+ { 0x0905, 0x0000 }, /* R2309 - EQL_6 */
+ { 0x0906, 0x0000 }, /* R2310 - EQL_7 */
+ { 0x0907, 0x0000 }, /* R2311 - EQL_8 */
+ { 0x0908, 0x0000 }, /* R2312 - EQL_9 */
+ { 0x0909, 0x0000 }, /* R2313 - EQL_10 */
+ { 0x090A, 0x0000 }, /* R2314 - EQL_11 */
+ { 0x090B, 0x0000 }, /* R2315 - EQL_12 */
+ { 0x090C, 0x0000 }, /* R2316 - EQL_13 */
+ { 0x090D, 0x0000 }, /* R2317 - EQL_14 */
+ { 0x090E, 0x0000 }, /* R2318 - EQL_15 */
+ { 0x090F, 0x0000 }, /* R2319 - EQL_16 */
+ { 0x0910, 0x0000 }, /* R2320 - EQL_17 */
+ { 0x0911, 0x0000 }, /* R2321 - EQL_18 */
+ { 0x0912, 0x0000 }, /* R2322 - EQL_19 */
+ { 0x0913, 0x0000 }, /* R2323 - EQL_20 */
+ { 0x0916, 0x0000 }, /* R2326 - EQR_1 */
+ { 0x0917, 0x0000 }, /* R2327 - EQR_2 */
+ { 0x0918, 0x0000 }, /* R2328 - EQR_3 */
+ { 0x0919, 0x0000 }, /* R2329 - EQR_4 */
+ { 0x091A, 0x0000 }, /* R2330 - EQR_5 */
+ { 0x091B, 0x0000 }, /* R2331 - EQR_6 */
+ { 0x091C, 0x0000 }, /* R2332 - EQR_7 */
+ { 0x091D, 0x0000 }, /* R2333 - EQR_8 */
+ { 0x091E, 0x0000 }, /* R2334 - EQR_9 */
+ { 0x091F, 0x0000 }, /* R2335 - EQR_10 */
+ { 0x0920, 0x0000 }, /* R2336 - EQR_11 */
+ { 0x0921, 0x0000 }, /* R2337 - EQR_12 */
+ { 0x0922, 0x0000 }, /* R2338 - EQR_13 */
+ { 0x0923, 0x0000 }, /* R2339 - EQR_14 */
+ { 0x0924, 0x0000 }, /* R2340 - EQR_15 */
+ { 0x0925, 0x0000 }, /* R2341 - EQR_16 */
+ { 0x0926, 0x0000 }, /* R2342 - EQR_17 */
+ { 0x0927, 0x0000 }, /* R2343 - EQR_18 */
+ { 0x0928, 0x0000 }, /* R2344 - EQR_19 */
+ { 0x0929, 0x0000 }, /* R2345 - EQR_20 */
+ { 0x093E, 0x0000 }, /* R2366 - HPLPF1_1 */
+ { 0x093F, 0x0000 }, /* R2367 - HPLPF1_2 */
+ { 0x0942, 0x0000 }, /* R2370 - HPLPF2_1 */
+ { 0x0943, 0x0000 }, /* R2371 - HPLPF2_2 */
+ { 0x0A00, 0x0000 }, /* R2560 - DSP1 Control 1 */
+ { 0x0A02, 0x0000 }, /* R2562 - DSP1 Control 2 */
+ { 0x0A03, 0x0000 }, /* R2563 - DSP1 Control 3 */
+ { 0x0A04, 0x0000 }, /* R2564 - DSP1 Control 4 */
+ { 0x0A06, 0x0000 }, /* R2566 - DSP1 Control 5 */
+ { 0x0A07, 0x0000 }, /* R2567 - DSP1 Control 6 */
+ { 0x0A08, 0x0000 }, /* R2568 - DSP1 Control 7 */
+ { 0x0A09, 0x0000 }, /* R2569 - DSP1 Control 8 */
+ { 0x0A0A, 0x0000 }, /* R2570 - DSP1 Control 9 */
+ { 0x0A0B, 0x0000 }, /* R2571 - DSP1 Control 10 */
+ { 0x0A0C, 0x0000 }, /* R2572 - DSP1 Control 11 */
+ { 0x0A0D, 0x0000 }, /* R2573 - DSP1 Control 12 */
+ { 0x0A0F, 0x0000 }, /* R2575 - DSP1 Control 13 */
+ { 0x0A10, 0x0000 }, /* R2576 - DSP1 Control 14 */
+ { 0x0A11, 0x0000 }, /* R2577 - DSP1 Control 15 */
+ { 0x0A12, 0x0000 }, /* R2578 - DSP1 Control 16 */
+ { 0x0A13, 0x0000 }, /* R2579 - DSP1 Control 17 */
+ { 0x0A14, 0x0000 }, /* R2580 - DSP1 Control 18 */
+ { 0x0A16, 0x0000 }, /* R2582 - DSP1 Control 19 */
+ { 0x0A17, 0x0000 }, /* R2583 - DSP1 Control 20 */
+ { 0x0A18, 0x0000 }, /* R2584 - DSP1 Control 21 */
+ { 0x0A1A, 0x1800 }, /* R2586 - DSP1 Control 22 */
+ { 0x0A1B, 0x1000 }, /* R2587 - DSP1 Control 23 */
+ { 0x0A1C, 0x0400 }, /* R2588 - DSP1 Control 24 */
+ { 0x0A1E, 0x0000 }, /* R2590 - DSP1 Control 25 */
+ { 0x0A20, 0x0000 }, /* R2592 - DSP1 Control 26 */
+ { 0x0A21, 0x0000 }, /* R2593 - DSP1 Control 27 */
+ { 0x0A22, 0x0000 }, /* R2594 - DSP1 Control 28 */
+ { 0x0A23, 0x0000 }, /* R2595 - DSP1 Control 29 */
+ { 0x0A24, 0x0000 }, /* R2596 - DSP1 Control 30 */
+ { 0x0A26, 0x0000 }, /* R2598 - DSP1 Control 31 */
+ { 0x0B00, 0x0000 }, /* R2816 - DSP2 Control 1 */
+ { 0x0B02, 0x0000 }, /* R2818 - DSP2 Control 2 */
+ { 0x0B03, 0x0000 }, /* R2819 - DSP2 Control 3 */
+ { 0x0B04, 0x0000 }, /* R2820 - DSP2 Control 4 */
+ { 0x0B06, 0x0000 }, /* R2822 - DSP2 Control 5 */
+ { 0x0B07, 0x0000 }, /* R2823 - DSP2 Control 6 */
+ { 0x0B08, 0x0000 }, /* R2824 - DSP2 Control 7 */
+ { 0x0B09, 0x0000 }, /* R2825 - DSP2 Control 8 */
+ { 0x0B0A, 0x0000 }, /* R2826 - DSP2 Control 9 */
+ { 0x0B0B, 0x0000 }, /* R2827 - DSP2 Control 10 */
+ { 0x0B0C, 0x0000 }, /* R2828 - DSP2 Control 11 */
+ { 0x0B0D, 0x0000 }, /* R2829 - DSP2 Control 12 */
+ { 0x0B0F, 0x0000 }, /* R2831 - DSP2 Control 13 */
+ { 0x0B10, 0x0000 }, /* R2832 - DSP2 Control 14 */
+ { 0x0B11, 0x0000 }, /* R2833 - DSP2 Control 15 */
+ { 0x0B12, 0x0000 }, /* R2834 - DSP2 Control 16 */
+ { 0x0B13, 0x0000 }, /* R2835 - DSP2 Control 17 */
+ { 0x0B14, 0x0000 }, /* R2836 - DSP2 Control 18 */
+ { 0x0B16, 0x0000 }, /* R2838 - DSP2 Control 19 */
+ { 0x0B17, 0x0000 }, /* R2839 - DSP2 Control 20 */
+ { 0x0B18, 0x0000 }, /* R2840 - DSP2 Control 21 */
+ { 0x0B1A, 0x0800 }, /* R2842 - DSP2 Control 22 */
+ { 0x0B1B, 0x1000 }, /* R2843 - DSP2 Control 23 */
+ { 0x0B1C, 0x0400 }, /* R2844 - DSP2 Control 24 */
+ { 0x0B1E, 0x0000 }, /* R2846 - DSP2 Control 25 */
+ { 0x0B20, 0x0000 }, /* R2848 - DSP2 Control 26 */
+ { 0x0B21, 0x0000 }, /* R2849 - DSP2 Control 27 */
+ { 0x0B22, 0x0000 }, /* R2850 - DSP2 Control 28 */
+ { 0x0B23, 0x0000 }, /* R2851 - DSP2 Control 29 */
+ { 0x0B24, 0x0000 }, /* R2852 - DSP2 Control 30 */
+ { 0x0B26, 0x0000 }, /* R2854 - DSP2 Control 31 */
+};
+
+static bool wm2200_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case WM2200_SOFTWARE_RESET:
+ case WM2200_DEVICE_REVISION:
+ case WM2200_ADPS1_IRQ0:
+ case WM2200_ADPS1_IRQ1:
+ case WM2200_INTERRUPT_STATUS_1:
+ case WM2200_INTERRUPT_STATUS_2:
+ case WM2200_INTERRUPT_RAW_STATUS_2:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static bool wm2200_readable_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case WM2200_SOFTWARE_RESET:
+ case WM2200_DEVICE_REVISION:
+ case WM2200_TONE_GENERATOR_1:
+ case WM2200_CLOCKING_3:
+ case WM2200_CLOCKING_4:
+ case WM2200_FLL_CONTROL_1:
+ case WM2200_FLL_CONTROL_2:
+ case WM2200_FLL_CONTROL_3:
+ case WM2200_FLL_CONTROL_4:
+ case WM2200_FLL_CONTROL_6:
+ case WM2200_FLL_CONTROL_7:
+ case WM2200_FLL_EFS_1:
+ case WM2200_FLL_EFS_2:
+ case WM2200_MIC_CHARGE_PUMP_1:
+ case WM2200_MIC_CHARGE_PUMP_2:
+ case WM2200_DM_CHARGE_PUMP_1:
+ case WM2200_MIC_BIAS_CTRL_1:
+ case WM2200_MIC_BIAS_CTRL_2:
+ case WM2200_EAR_PIECE_CTRL_1:
+ case WM2200_EAR_PIECE_CTRL_2:
+ case WM2200_INPUT_ENABLES:
+ case WM2200_IN1L_CONTROL:
+ case WM2200_IN1R_CONTROL:
+ case WM2200_IN2L_CONTROL:
+ case WM2200_IN2R_CONTROL:
+ case WM2200_IN3L_CONTROL:
+ case WM2200_IN3R_CONTROL:
+ case WM2200_RXANC_SRC:
+ case WM2200_INPUT_VOLUME_RAMP:
+ case WM2200_ADC_DIGITAL_VOLUME_1L:
+ case WM2200_ADC_DIGITAL_VOLUME_1R:
+ case WM2200_ADC_DIGITAL_VOLUME_2L:
+ case WM2200_ADC_DIGITAL_VOLUME_2R:
+ case WM2200_ADC_DIGITAL_VOLUME_3L:
+ case WM2200_ADC_DIGITAL_VOLUME_3R:
+ case WM2200_OUTPUT_ENABLES:
+ case WM2200_DAC_VOLUME_LIMIT_1L:
+ case WM2200_DAC_VOLUME_LIMIT_1R:
+ case WM2200_DAC_VOLUME_LIMIT_2L:
+ case WM2200_DAC_VOLUME_LIMIT_2R:
+ case WM2200_DAC_AEC_CONTROL_1:
+ case WM2200_OUTPUT_VOLUME_RAMP:
+ case WM2200_DAC_DIGITAL_VOLUME_1L:
+ case WM2200_DAC_DIGITAL_VOLUME_1R:
+ case WM2200_DAC_DIGITAL_VOLUME_2L:
+ case WM2200_DAC_DIGITAL_VOLUME_2R:
+ case WM2200_PDM_1:
+ case WM2200_PDM_2:
+ case WM2200_AUDIO_IF_1_1:
+ case WM2200_AUDIO_IF_1_2:
+ case WM2200_AUDIO_IF_1_3:
+ case WM2200_AUDIO_IF_1_4:
+ case WM2200_AUDIO_IF_1_5:
+ case WM2200_AUDIO_IF_1_6:
+ case WM2200_AUDIO_IF_1_7:
+ case WM2200_AUDIO_IF_1_8:
+ case WM2200_AUDIO_IF_1_9:
+ case WM2200_AUDIO_IF_1_10:
+ case WM2200_AUDIO_IF_1_11:
+ case WM2200_AUDIO_IF_1_12:
+ case WM2200_AUDIO_IF_1_13:
+ case WM2200_AUDIO_IF_1_14:
+ case WM2200_AUDIO_IF_1_15:
+ case WM2200_AUDIO_IF_1_16:
+ case WM2200_AUDIO_IF_1_17:
+ case WM2200_AUDIO_IF_1_18:
+ case WM2200_AUDIO_IF_1_19:
+ case WM2200_AUDIO_IF_1_20:
+ case WM2200_AUDIO_IF_1_21:
+ case WM2200_AUDIO_IF_1_22:
+ case WM2200_OUT1LMIX_INPUT_1_SOURCE:
+ case WM2200_OUT1LMIX_INPUT_1_VOLUME:
+ case WM2200_OUT1LMIX_INPUT_2_SOURCE:
+ case WM2200_OUT1LMIX_INPUT_2_VOLUME:
+ case WM2200_OUT1LMIX_INPUT_3_SOURCE:
+ case WM2200_OUT1LMIX_INPUT_3_VOLUME:
+ case WM2200_OUT1LMIX_INPUT_4_SOURCE:
+ case WM2200_OUT1LMIX_INPUT_4_VOLUME:
+ case WM2200_OUT1RMIX_INPUT_1_SOURCE:
+ case WM2200_OUT1RMIX_INPUT_1_VOLUME:
+ case WM2200_OUT1RMIX_INPUT_2_SOURCE:
+ case WM2200_OUT1RMIX_INPUT_2_VOLUME:
+ case WM2200_OUT1RMIX_INPUT_3_SOURCE:
+ case WM2200_OUT1RMIX_INPUT_3_VOLUME:
+ case WM2200_OUT1RMIX_INPUT_4_SOURCE:
+ case WM2200_OUT1RMIX_INPUT_4_VOLUME:
+ case WM2200_OUT2LMIX_INPUT_1_SOURCE:
+ case WM2200_OUT2LMIX_INPUT_1_VOLUME:
+ case WM2200_OUT2LMIX_INPUT_2_SOURCE:
+ case WM2200_OUT2LMIX_INPUT_2_VOLUME:
+ case WM2200_OUT2LMIX_INPUT_3_SOURCE:
+ case WM2200_OUT2LMIX_INPUT_3_VOLUME:
+ case WM2200_OUT2LMIX_INPUT_4_SOURCE:
+ case WM2200_OUT2LMIX_INPUT_4_VOLUME:
+ case WM2200_OUT2RMIX_INPUT_1_SOURCE:
+ case WM2200_OUT2RMIX_INPUT_1_VOLUME:
+ case WM2200_OUT2RMIX_INPUT_2_SOURCE:
+ case WM2200_OUT2RMIX_INPUT_2_VOLUME:
+ case WM2200_OUT2RMIX_INPUT_3_SOURCE:
+ case WM2200_OUT2RMIX_INPUT_3_VOLUME:
+ case WM2200_OUT2RMIX_INPUT_4_SOURCE:
+ case WM2200_OUT2RMIX_INPUT_4_VOLUME:
+ case WM2200_AIF1TX1MIX_INPUT_1_SOURCE:
+ case WM2200_AIF1TX1MIX_INPUT_1_VOLUME:
+ case WM2200_AIF1TX1MIX_INPUT_2_SOURCE:
+ case WM2200_AIF1TX1MIX_INPUT_2_VOLUME:
+ case WM2200_AIF1TX1MIX_INPUT_3_SOURCE:
+ case WM2200_AIF1TX1MIX_INPUT_3_VOLUME:
+ case WM2200_AIF1TX1MIX_INPUT_4_SOURCE:
+ case WM2200_AIF1TX1MIX_INPUT_4_VOLUME:
+ case WM2200_AIF1TX2MIX_INPUT_1_SOURCE:
+ case WM2200_AIF1TX2MIX_INPUT_1_VOLUME:
+ case WM2200_AIF1TX2MIX_INPUT_2_SOURCE:
+ case WM2200_AIF1TX2MIX_INPUT_2_VOLUME:
+ case WM2200_AIF1TX2MIX_INPUT_3_SOURCE:
+ case WM2200_AIF1TX2MIX_INPUT_3_VOLUME:
+ case WM2200_AIF1TX2MIX_INPUT_4_SOURCE:
+ case WM2200_AIF1TX2MIX_INPUT_4_VOLUME:
+ case WM2200_AIF1TX3MIX_INPUT_1_SOURCE:
+ case WM2200_AIF1TX3MIX_INPUT_1_VOLUME:
+ case WM2200_AIF1TX3MIX_INPUT_2_SOURCE:
+ case WM2200_AIF1TX3MIX_INPUT_2_VOLUME:
+ case WM2200_AIF1TX3MIX_INPUT_3_SOURCE:
+ case WM2200_AIF1TX3MIX_INPUT_3_VOLUME:
+ case WM2200_AIF1TX3MIX_INPUT_4_SOURCE:
+ case WM2200_AIF1TX3MIX_INPUT_4_VOLUME:
+ case WM2200_AIF1TX4MIX_INPUT_1_SOURCE:
+ case WM2200_AIF1TX4MIX_INPUT_1_VOLUME:
+ case WM2200_AIF1TX4MIX_INPUT_2_SOURCE:
+ case WM2200_AIF1TX4MIX_INPUT_2_VOLUME:
+ case WM2200_AIF1TX4MIX_INPUT_3_SOURCE:
+ case WM2200_AIF1TX4MIX_INPUT_3_VOLUME:
+ case WM2200_AIF1TX4MIX_INPUT_4_SOURCE:
+ case WM2200_AIF1TX4MIX_INPUT_4_VOLUME:
+ case WM2200_AIF1TX5MIX_INPUT_1_SOURCE:
+ case WM2200_AIF1TX5MIX_INPUT_1_VOLUME:
+ case WM2200_AIF1TX5MIX_INPUT_2_SOURCE:
+ case WM2200_AIF1TX5MIX_INPUT_2_VOLUME:
+ case WM2200_AIF1TX5MIX_INPUT_3_SOURCE:
+ case WM2200_AIF1TX5MIX_INPUT_3_VOLUME:
+ case WM2200_AIF1TX5MIX_INPUT_4_SOURCE:
+ case WM2200_AIF1TX5MIX_INPUT_4_VOLUME:
+ case WM2200_AIF1TX6MIX_INPUT_1_SOURCE:
+ case WM2200_AIF1TX6MIX_INPUT_1_VOLUME:
+ case WM2200_AIF1TX6MIX_INPUT_2_SOURCE:
+ case WM2200_AIF1TX6MIX_INPUT_2_VOLUME:
+ case WM2200_AIF1TX6MIX_INPUT_3_SOURCE:
+ case WM2200_AIF1TX6MIX_INPUT_3_VOLUME:
+ case WM2200_AIF1TX6MIX_INPUT_4_SOURCE:
+ case WM2200_AIF1TX6MIX_INPUT_4_VOLUME:
+ case WM2200_EQLMIX_INPUT_1_SOURCE:
+ case WM2200_EQLMIX_INPUT_1_VOLUME:
+ case WM2200_EQLMIX_INPUT_2_SOURCE:
+ case WM2200_EQLMIX_INPUT_2_VOLUME:
+ case WM2200_EQLMIX_INPUT_3_SOURCE:
+ case WM2200_EQLMIX_INPUT_3_VOLUME:
+ case WM2200_EQLMIX_INPUT_4_SOURCE:
+ case WM2200_EQLMIX_INPUT_4_VOLUME:
+ case WM2200_EQRMIX_INPUT_1_SOURCE:
+ case WM2200_EQRMIX_INPUT_1_VOLUME:
+ case WM2200_EQRMIX_INPUT_2_SOURCE:
+ case WM2200_EQRMIX_INPUT_2_VOLUME:
+ case WM2200_EQRMIX_INPUT_3_SOURCE:
+ case WM2200_EQRMIX_INPUT_3_VOLUME:
+ case WM2200_EQRMIX_INPUT_4_SOURCE:
+ case WM2200_EQRMIX_INPUT_4_VOLUME:
+ case WM2200_LHPF1MIX_INPUT_1_SOURCE:
+ case WM2200_LHPF1MIX_INPUT_1_VOLUME:
+ case WM2200_LHPF1MIX_INPUT_2_SOURCE:
+ case WM2200_LHPF1MIX_INPUT_2_VOLUME:
+ case WM2200_LHPF1MIX_INPUT_3_SOURCE:
+ case WM2200_LHPF1MIX_INPUT_3_VOLUME:
+ case WM2200_LHPF1MIX_INPUT_4_SOURCE:
+ case WM2200_LHPF1MIX_INPUT_4_VOLUME:
+ case WM2200_LHPF2MIX_INPUT_1_SOURCE:
+ case WM2200_LHPF2MIX_INPUT_1_VOLUME:
+ case WM2200_LHPF2MIX_INPUT_2_SOURCE:
+ case WM2200_LHPF2MIX_INPUT_2_VOLUME:
+ case WM2200_LHPF2MIX_INPUT_3_SOURCE:
+ case WM2200_LHPF2MIX_INPUT_3_VOLUME:
+ case WM2200_LHPF2MIX_INPUT_4_SOURCE:
+ case WM2200_LHPF2MIX_INPUT_4_VOLUME:
+ case WM2200_DSP1LMIX_INPUT_1_SOURCE:
+ case WM2200_DSP1LMIX_INPUT_1_VOLUME:
+ case WM2200_DSP1LMIX_INPUT_2_SOURCE:
+ case WM2200_DSP1LMIX_INPUT_2_VOLUME:
+ case WM2200_DSP1LMIX_INPUT_3_SOURCE:
+ case WM2200_DSP1LMIX_INPUT_3_VOLUME:
+ case WM2200_DSP1LMIX_INPUT_4_SOURCE:
+ case WM2200_DSP1LMIX_INPUT_4_VOLUME:
+ case WM2200_DSP1RMIX_INPUT_1_SOURCE:
+ case WM2200_DSP1RMIX_INPUT_1_VOLUME:
+ case WM2200_DSP1RMIX_INPUT_2_SOURCE:
+ case WM2200_DSP1RMIX_INPUT_2_VOLUME:
+ case WM2200_DSP1RMIX_INPUT_3_SOURCE:
+ case WM2200_DSP1RMIX_INPUT_3_VOLUME:
+ case WM2200_DSP1RMIX_INPUT_4_SOURCE:
+ case WM2200_DSP1RMIX_INPUT_4_VOLUME:
+ case WM2200_DSP1AUX1MIX_INPUT_1_SOURCE:
+ case WM2200_DSP1AUX2MIX_INPUT_1_SOURCE:
+ case WM2200_DSP1AUX3MIX_INPUT_1_SOURCE:
+ case WM2200_DSP1AUX4MIX_INPUT_1_SOURCE:
+ case WM2200_DSP1AUX5MIX_INPUT_1_SOURCE:
+ case WM2200_DSP1AUX6MIX_INPUT_1_SOURCE:
+ case WM2200_DSP2LMIX_INPUT_1_SOURCE:
+ case WM2200_DSP2LMIX_INPUT_1_VOLUME:
+ case WM2200_DSP2LMIX_INPUT_2_SOURCE:
+ case WM2200_DSP2LMIX_INPUT_2_VOLUME:
+ case WM2200_DSP2LMIX_INPUT_3_SOURCE:
+ case WM2200_DSP2LMIX_INPUT_3_VOLUME:
+ case WM2200_DSP2LMIX_INPUT_4_SOURCE:
+ case WM2200_DSP2LMIX_INPUT_4_VOLUME:
+ case WM2200_DSP2RMIX_INPUT_1_SOURCE:
+ case WM2200_DSP2RMIX_INPUT_1_VOLUME:
+ case WM2200_DSP2RMIX_INPUT_2_SOURCE:
+ case WM2200_DSP2RMIX_INPUT_2_VOLUME:
+ case WM2200_DSP2RMIX_INPUT_3_SOURCE:
+ case WM2200_DSP2RMIX_INPUT_3_VOLUME:
+ case WM2200_DSP2RMIX_INPUT_4_SOURCE:
+ case WM2200_DSP2RMIX_INPUT_4_VOLUME:
+ case WM2200_DSP2AUX1MIX_INPUT_1_SOURCE:
+ case WM2200_DSP2AUX2MIX_INPUT_1_SOURCE:
+ case WM2200_DSP2AUX3MIX_INPUT_1_SOURCE:
+ case WM2200_DSP2AUX4MIX_INPUT_1_SOURCE:
+ case WM2200_DSP2AUX5MIX_INPUT_1_SOURCE:
+ case WM2200_DSP2AUX6MIX_INPUT_1_SOURCE:
+ case WM2200_GPIO_CTRL_1:
+ case WM2200_GPIO_CTRL_2:
+ case WM2200_GPIO_CTRL_3:
+ case WM2200_GPIO_CTRL_4:
+ case WM2200_ADPS1_IRQ0:
+ case WM2200_ADPS1_IRQ1:
+ case WM2200_MISC_PAD_CTRL_1:
+ case WM2200_INTERRUPT_STATUS_1:
+ case WM2200_INTERRUPT_STATUS_1_MASK:
+ case WM2200_INTERRUPT_STATUS_2:
+ case WM2200_INTERRUPT_RAW_STATUS_2:
+ case WM2200_INTERRUPT_STATUS_2_MASK:
+ case WM2200_INTERRUPT_CONTROL:
+ case WM2200_EQL_1:
+ case WM2200_EQL_2:
+ case WM2200_EQL_3:
+ case WM2200_EQL_4:
+ case WM2200_EQL_5:
+ case WM2200_EQL_6:
+ case WM2200_EQL_7:
+ case WM2200_EQL_8:
+ case WM2200_EQL_9:
+ case WM2200_EQL_10:
+ case WM2200_EQL_11:
+ case WM2200_EQL_12:
+ case WM2200_EQL_13:
+ case WM2200_EQL_14:
+ case WM2200_EQL_15:
+ case WM2200_EQL_16:
+ case WM2200_EQL_17:
+ case WM2200_EQL_18:
+ case WM2200_EQL_19:
+ case WM2200_EQL_20:
+ case WM2200_EQR_1:
+ case WM2200_EQR_2:
+ case WM2200_EQR_3:
+ case WM2200_EQR_4:
+ case WM2200_EQR_5:
+ case WM2200_EQR_6:
+ case WM2200_EQR_7:
+ case WM2200_EQR_8:
+ case WM2200_EQR_9:
+ case WM2200_EQR_10:
+ case WM2200_EQR_11:
+ case WM2200_EQR_12:
+ case WM2200_EQR_13:
+ case WM2200_EQR_14:
+ case WM2200_EQR_15:
+ case WM2200_EQR_16:
+ case WM2200_EQR_17:
+ case WM2200_EQR_18:
+ case WM2200_EQR_19:
+ case WM2200_EQR_20:
+ case WM2200_HPLPF1_1:
+ case WM2200_HPLPF1_2:
+ case WM2200_HPLPF2_1:
+ case WM2200_HPLPF2_2:
+ case WM2200_DSP1_CONTROL_1:
+ case WM2200_DSP1_CONTROL_2:
+ case WM2200_DSP1_CONTROL_3:
+ case WM2200_DSP1_CONTROL_4:
+ case WM2200_DSP1_CONTROL_5:
+ case WM2200_DSP1_CONTROL_6:
+ case WM2200_DSP1_CONTROL_7:
+ case WM2200_DSP1_CONTROL_8:
+ case WM2200_DSP1_CONTROL_9:
+ case WM2200_DSP1_CONTROL_10:
+ case WM2200_DSP1_CONTROL_11:
+ case WM2200_DSP1_CONTROL_12:
+ case WM2200_DSP1_CONTROL_13:
+ case WM2200_DSP1_CONTROL_14:
+ case WM2200_DSP1_CONTROL_15:
+ case WM2200_DSP1_CONTROL_16:
+ case WM2200_DSP1_CONTROL_17:
+ case WM2200_DSP1_CONTROL_18:
+ case WM2200_DSP1_CONTROL_19:
+ case WM2200_DSP1_CONTROL_20:
+ case WM2200_DSP1_CONTROL_21:
+ case WM2200_DSP1_CONTROL_22:
+ case WM2200_DSP1_CONTROL_23:
+ case WM2200_DSP1_CONTROL_24:
+ case WM2200_DSP1_CONTROL_25:
+ case WM2200_DSP1_CONTROL_26:
+ case WM2200_DSP1_CONTROL_27:
+ case WM2200_DSP1_CONTROL_28:
+ case WM2200_DSP1_CONTROL_29:
+ case WM2200_DSP1_CONTROL_30:
+ case WM2200_DSP1_CONTROL_31:
+ case WM2200_DSP2_CONTROL_1:
+ case WM2200_DSP2_CONTROL_2:
+ case WM2200_DSP2_CONTROL_3:
+ case WM2200_DSP2_CONTROL_4:
+ case WM2200_DSP2_CONTROL_5:
+ case WM2200_DSP2_CONTROL_6:
+ case WM2200_DSP2_CONTROL_7:
+ case WM2200_DSP2_CONTROL_8:
+ case WM2200_DSP2_CONTROL_9:
+ case WM2200_DSP2_CONTROL_10:
+ case WM2200_DSP2_CONTROL_11:
+ case WM2200_DSP2_CONTROL_12:
+ case WM2200_DSP2_CONTROL_13:
+ case WM2200_DSP2_CONTROL_14:
+ case WM2200_DSP2_CONTROL_15:
+ case WM2200_DSP2_CONTROL_16:
+ case WM2200_DSP2_CONTROL_17:
+ case WM2200_DSP2_CONTROL_18:
+ case WM2200_DSP2_CONTROL_19:
+ case WM2200_DSP2_CONTROL_20:
+ case WM2200_DSP2_CONTROL_21:
+ case WM2200_DSP2_CONTROL_22:
+ case WM2200_DSP2_CONTROL_23:
+ case WM2200_DSP2_CONTROL_24:
+ case WM2200_DSP2_CONTROL_25:
+ case WM2200_DSP2_CONTROL_26:
+ case WM2200_DSP2_CONTROL_27:
+ case WM2200_DSP2_CONTROL_28:
+ case WM2200_DSP2_CONTROL_29:
+ case WM2200_DSP2_CONTROL_30:
+ case WM2200_DSP2_CONTROL_31:
+ return true;
+ default:
+ return false;
+ }
+}
+
+static const struct reg_default wm2200_reva_patch[] = {
+ { 0x07, 0x0003 },
+ { 0x102, 0x0200 },
+ { 0x203, 0x0084 },
+ { 0x201, 0x83FF },
+ { 0x20C, 0x0062 },
+ { 0x20D, 0x0062 },
+ { 0x207, 0x2002 },
+ { 0x208, 0x20C0 },
+ { 0x21D, 0x01C0 },
+ { 0x50A, 0x0001 },
+ { 0x50B, 0x0002 },
+ { 0x50C, 0x0003 },
+ { 0x50D, 0x0004 },
+ { 0x50E, 0x0005 },
+ { 0x510, 0x0001 },
+ { 0x511, 0x0002 },
+ { 0x512, 0x0003 },
+ { 0x513, 0x0004 },
+ { 0x514, 0x0005 },
+ { 0x515, 0x0000 },
+ { 0x201, 0x8084 },
+ { 0x202, 0xBBDE },
+ { 0x203, 0x00EC },
+ { 0x500, 0x8000 },
+ { 0x507, 0x1820 },
+ { 0x508, 0x1820 },
+ { 0x505, 0x0300 },
+ { 0x506, 0x0300 },
+ { 0x302, 0x2280 },
+ { 0x303, 0x0080 },
+ { 0x304, 0x2280 },
+ { 0x305, 0x0080 },
+ { 0x306, 0x2280 },
+ { 0x307, 0x0080 },
+ { 0x401, 0x0080 },
+ { 0x402, 0x0080 },
+ { 0x417, 0x3069 },
+ { 0x900, 0x6318 },
+ { 0x901, 0x6300 },
+ { 0x902, 0x0FC8 },
+ { 0x903, 0x03FE },
+ { 0x904, 0x00E0 },
+ { 0x905, 0x1EC4 },
+ { 0x906, 0xF136 },
+ { 0x907, 0x0409 },
+ { 0x908, 0x04CC },
+ { 0x909, 0x1C9B },
+ { 0x90A, 0xF337 },
+ { 0x90B, 0x040B },
+ { 0x90C, 0x0CBB },
+ { 0x90D, 0x16F8 },
+ { 0x90E, 0xF7D9 },
+ { 0x90F, 0x040A },
+ { 0x910, 0x1F14 },
+ { 0x911, 0x058C },
+ { 0x912, 0x0563 },
+ { 0x913, 0x4000 },
+ { 0x916, 0x6318 },
+ { 0x917, 0x6300 },
+ { 0x918, 0x0FC8 },
+ { 0x919, 0x03FE },
+ { 0x91A, 0x00E0 },
+ { 0x91B, 0x1EC4 },
+ { 0x91C, 0xF136 },
+ { 0x91D, 0x0409 },
+ { 0x91E, 0x04CC },
+ { 0x91F, 0x1C9B },
+ { 0x920, 0xF337 },
+ { 0x921, 0x040B },
+ { 0x922, 0x0CBB },
+ { 0x923, 0x16F8 },
+ { 0x924, 0xF7D9 },
+ { 0x925, 0x040A },
+ { 0x926, 0x1F14 },
+ { 0x927, 0x058C },
+ { 0x928, 0x0563 },
+ { 0x929, 0x4000 },
+ { 0x709, 0x2000 },
+ { 0x207, 0x200E },
+ { 0x208, 0x20D4 },
+ { 0x20A, 0x0080 },
+ { 0x07, 0x0000 },
+};
+
+static int wm2200_reset(struct wm2200_priv *wm2200)
+{
+ if (wm2200->pdata.reset) {
+ gpio_set_value_cansleep(wm2200->pdata.reset, 0);
+ gpio_set_value_cansleep(wm2200->pdata.reset, 1);
+
+ return 0;
+ } else {
+ return regmap_write(wm2200->regmap, WM2200_SOFTWARE_RESET,
+ 0x2200);
+ }
+}
+
+static DECLARE_TLV_DB_SCALE(in_tlv, -6300, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(out_tlv, -6400, 100, 0);
+
+static const char *wm2200_mixer_texts[] = {
+ "None",
+ "Tone Generator",
+ "AEC loopback",
+ "IN1L",
+ "IN1R",
+ "IN2L",
+ "IN2R",
+ "IN3L",
+ "IN3R",
+ "AIF1RX1",
+ "AIF1RX2",
+ "AIF1RX3",
+ "AIF1RX4",
+ "AIF1RX5",
+ "AIF1RX6",
+ "EQL",
+ "EQR",
+ "LHPF1",
+ "LHPF2",
+ "LHPF3",
+ "LHPF4",
+ "DSP1.1",
+ "DSP1.2",
+ "DSP1.3",
+ "DSP1.4",
+ "DSP1.5",
+ "DSP1.6",
+ "DSP2.1",
+ "DSP2.2",
+ "DSP2.3",
+ "DSP2.4",
+ "DSP2.5",
+ "DSP2.6",
+};
+
+static int wm2200_mixer_values[] = {
+ 0x00,
+ 0x04, /* Tone */
+ 0x08, /* AEC */
+ 0x10, /* Input */
+ 0x11,
+ 0x12,
+ 0x13,
+ 0x14,
+ 0x15,
+ 0x20, /* AIF */
+ 0x21,
+ 0x22,
+ 0x23,
+ 0x24,
+ 0x25,
+ 0x50, /* EQ */
+ 0x51,
+ 0x52,
+ 0x60, /* LHPF1 */
+ 0x61, /* LHPF2 */
+ 0x68, /* DSP1 */
+ 0x69,
+ 0x6a,
+ 0x6b,
+ 0x6c,
+ 0x6d,
+ 0x70, /* DSP2 */
+ 0x71,
+ 0x72,
+ 0x73,
+ 0x74,
+ 0x75,
+};
+
+#define WM2200_MIXER_CONTROLS(name, base) \
+ SOC_SINGLE_TLV(name " Input 1 Volume", base + 1 , \
+ WM2200_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \
+ SOC_SINGLE_TLV(name " Input 2 Volume", base + 3 , \
+ WM2200_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \
+ SOC_SINGLE_TLV(name " Input 3 Volume", base + 5 , \
+ WM2200_MIXER_VOL_SHIFT, 80, 0, mixer_tlv), \
+ SOC_SINGLE_TLV(name " Input 4 Volume", base + 7 , \
+ WM2200_MIXER_VOL_SHIFT, 80, 0, mixer_tlv)
+
+#define WM2200_MUX_ENUM_DECL(name, reg) \
+ SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \
+ wm2200_mixer_texts, wm2200_mixer_values)
+
+#define WM2200_MUX_CTL_DECL(name) \
+ const struct snd_kcontrol_new name##_mux = \
+ SOC_DAPM_VALUE_ENUM("Route", name##_enum)
+
+#define WM2200_MIXER_ENUMS(name, base_reg) \
+ static WM2200_MUX_ENUM_DECL(name##_in1_enum, base_reg); \
+ static WM2200_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \
+ static WM2200_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \
+ static WM2200_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \
+ static WM2200_MUX_CTL_DECL(name##_in1); \
+ static WM2200_MUX_CTL_DECL(name##_in2); \
+ static WM2200_MUX_CTL_DECL(name##_in3); \
+ static WM2200_MUX_CTL_DECL(name##_in4)
+
+static const struct snd_kcontrol_new wm2200_snd_controls[] = {
+SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL,
+ WM2200_IN1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN2 High Performance Switch", WM2200_IN2L_CONTROL,
+ WM2200_IN2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN3 High Performance Switch", WM2200_IN3L_CONTROL,
+ WM2200_IN3_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R_TLV("IN1 Volume", WM2200_IN1L_CONTROL, WM2200_IN1R_CONTROL,
+ WM2200_IN1L_PGA_VOL_SHIFT, 0x5f, 0, in_tlv),
+SOC_DOUBLE_R_TLV("IN2 Volume", WM2200_IN2L_CONTROL, WM2200_IN2R_CONTROL,
+ WM2200_IN2L_PGA_VOL_SHIFT, 0x5f, 0, in_tlv),
+SOC_DOUBLE_R_TLV("IN3 Volume", WM2200_IN3L_CONTROL, WM2200_IN3R_CONTROL,
+ WM2200_IN3L_PGA_VOL_SHIFT, 0x5f, 0, in_tlv),
+
+SOC_DOUBLE_R("IN1 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L,
+ WM2200_ADC_DIGITAL_VOLUME_1R, WM2200_IN1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN2 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L,
+ WM2200_ADC_DIGITAL_VOLUME_2R, WM2200_IN2L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN3 Digital Switch", WM2200_ADC_DIGITAL_VOLUME_1L,
+ WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("IN1 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_1L,
+ WM2200_ADC_DIGITAL_VOLUME_1R, WM2200_IN1L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN2 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_2L,
+ WM2200_ADC_DIGITAL_VOLUME_2R, WM2200_IN2L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN3 Digital Volume", WM2200_ADC_DIGITAL_VOLUME_3L,
+ WM2200_ADC_DIGITAL_VOLUME_3R, WM2200_IN3L_DIG_VOL_SHIFT,
+ 0xbf, 0, digital_tlv),
+
+SOC_SINGLE("OUT1 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_1L,
+ WM2200_OUT1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("OUT2 High Performance Switch", WM2200_DAC_DIGITAL_VOLUME_2L,
+ WM2200_OUT2_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("OUT1 Digital Switch", WM2200_DAC_DIGITAL_VOLUME_1L,
+ WM2200_DAC_DIGITAL_VOLUME_1R, WM2200_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R_TLV("OUT1 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_1L,
+ WM2200_DAC_DIGITAL_VOLUME_1R, WM2200_OUT1L_VOL_SHIFT, 0x9f, 0,
+ digital_tlv),
+SOC_DOUBLE_R_TLV("OUT1 Volume", WM2200_DAC_VOLUME_LIMIT_1L,
+ WM2200_DAC_VOLUME_LIMIT_1R, WM2200_OUT1L_PGA_VOL_SHIFT,
+ 0x46, 0, out_tlv),
+
+SOC_DOUBLE_R("OUT2 Digital Switch", WM2200_DAC_DIGITAL_VOLUME_2L,
+ WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L,
+ WM2200_DAC_DIGITAL_VOLUME_2R, WM2200_OUT2L_VOL_SHIFT, 0x9f, 0,
+ digital_tlv),
+SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT,
+ WM2200_SPK1R_MUTE_SHIFT, 1, 0),
+};
+
+WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(OUT1R, WM2200_OUT1RMIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(OUT2L, WM2200_OUT2LMIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(OUT2R, WM2200_OUT2RMIX_INPUT_1_SOURCE);
+
+WM2200_MIXER_ENUMS(AIF1TX1, WM2200_AIF1TX1MIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(AIF1TX2, WM2200_AIF1TX2MIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(AIF1TX3, WM2200_AIF1TX3MIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(AIF1TX4, WM2200_AIF1TX4MIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(AIF1TX5, WM2200_AIF1TX5MIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(AIF1TX6, WM2200_AIF1TX6MIX_INPUT_1_SOURCE);
+
+WM2200_MIXER_ENUMS(EQL, WM2200_EQLMIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(EQR, WM2200_EQRMIX_INPUT_1_SOURCE);
+
+WM2200_MIXER_ENUMS(DSP1L, WM2200_DSP1LMIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(DSP1R, WM2200_DSP1RMIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(DSP2L, WM2200_DSP2LMIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(DSP2R, WM2200_DSP2RMIX_INPUT_1_SOURCE);
+
+WM2200_MIXER_ENUMS(LHPF1, WM2200_LHPF1MIX_INPUT_1_SOURCE);
+WM2200_MIXER_ENUMS(LHPF2, WM2200_LHPF2MIX_INPUT_1_SOURCE);
+
+#define WM2200_MUX(name, ctrl) \
+ SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl)
+
+#define WM2200_MIXER_WIDGETS(name, name_str) \
+ WM2200_MUX(name_str " Input 1", &name##_in1_mux), \
+ WM2200_MUX(name_str " Input 2", &name##_in2_mux), \
+ WM2200_MUX(name_str " Input 3", &name##_in3_mux), \
+ WM2200_MUX(name_str " Input 4", &name##_in4_mux), \
+ SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0)
+
+#define WM2200_MIXER_INPUT_ROUTES(name) \
+ { name, "Tone Generator", "Tone Generator" }, \
+ { name, "IN1L", "IN1L PGA" }, \
+ { name, "IN1R", "IN1R PGA" }, \
+ { name, "IN2L", "IN2L PGA" }, \
+ { name, "IN2R", "IN2R PGA" }, \
+ { name, "IN3L", "IN3L PGA" }, \
+ { name, "IN3R", "IN3R PGA" }, \
+ { name, "DSP1.1", "DSP1" }, \
+ { name, "DSP1.2", "DSP1" }, \
+ { name, "DSP1.3", "DSP1" }, \
+ { name, "DSP1.4", "DSP1" }, \
+ { name, "DSP1.5", "DSP1" }, \
+ { name, "DSP1.6", "DSP1" }, \
+ { name, "DSP2.1", "DSP2" }, \
+ { name, "DSP2.2", "DSP2" }, \
+ { name, "DSP2.3", "DSP2" }, \
+ { name, "DSP2.4", "DSP2" }, \
+ { name, "DSP2.5", "DSP2" }, \
+ { name, "DSP2.6", "DSP2" }, \
+ { name, "AIF1RX1", "AIF1RX1" }, \
+ { name, "AIF1RX2", "AIF1RX2" }, \
+ { name, "AIF1RX3", "AIF1RX3" }, \
+ { name, "AIF1RX4", "AIF1RX4" }, \
+ { name, "AIF1RX5", "AIF1RX5" }, \
+ { name, "AIF1RX6", "AIF1RX6" }, \
+ { name, "EQL", "EQL" }, \
+ { name, "EQR", "EQR" }, \
+ { name, "LHPF1", "LHPF1" }, \
+ { name, "LHPF2", "LHPF2" }
+
+#define WM2200_MIXER_ROUTES(widget, name) \
+ { widget, NULL, name " Mixer" }, \
+ { name " Mixer", NULL, name " Input 1" }, \
+ { name " Mixer", NULL, name " Input 2" }, \
+ { name " Mixer", NULL, name " Input 3" }, \
+ { name " Mixer", NULL, name " Input 4" }, \
+ WM2200_MIXER_INPUT_ROUTES(name " Input 1"), \
+ WM2200_MIXER_INPUT_ROUTES(name " Input 2"), \
+ WM2200_MIXER_INPUT_ROUTES(name " Input 3"), \
+ WM2200_MIXER_INPUT_ROUTES(name " Input 4")
+
+static const struct snd_soc_dapm_widget wm2200_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", WM2200_CLOCKING_3, WM2200_SYSCLK_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_SUPPLY("CP1", WM2200_DM_CHARGE_PUMP_1, WM2200_CPDM_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_SUPPLY("CP2", WM2200_MIC_CHARGE_PUMP_1, WM2200_CPMIC_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS1", WM2200_MIC_BIAS_CTRL_1, WM2200_MICB1_ENA_SHIFT,
+ 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", WM2200_MIC_BIAS_CTRL_2, WM2200_MICB2_ENA_SHIFT,
+ 0, NULL, 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20),
+SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 20),
+
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+SND_SOC_DAPM_INPUT("IN3L"),
+SND_SOC_DAPM_INPUT("IN3R"),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_PGA("Tone Generator", WM2200_TONE_GENERATOR_1,
+ WM2200_TONE_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("IN1L PGA", WM2200_INPUT_ENABLES, WM2200_IN1L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("IN1R PGA", WM2200_INPUT_ENABLES, WM2200_IN1R_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("IN2L PGA", WM2200_INPUT_ENABLES, WM2200_IN2L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("IN2R PGA", WM2200_INPUT_ENABLES, WM2200_IN2R_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("IN3L PGA", WM2200_INPUT_ENABLES, WM2200_IN3L_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("IN3R PGA", WM2200_INPUT_ENABLES, WM2200_IN3R_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", "Playback", 0,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", "Playback", 1,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", "Playback", 2,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", "Playback", 3,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", "Playback", 4,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", "Playback", 5,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1RX6_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA("EQL", WM2200_EQL_1, WM2200_EQL_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQR", WM2200_EQR_1, WM2200_EQR_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", WM2200_HPLPF1_1, WM2200_LHPF1_ENA_SHIFT, 0,
+ NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", WM2200_HPLPF2_1, WM2200_LHPF2_ENA_SHIFT, 0,
+ NULL, 0),
+
+SND_SOC_DAPM_PGA_E("DSP1", SND_SOC_NOPM, 0, 0, NULL, 0, NULL, 0),
+SND_SOC_DAPM_PGA_E("DSP2", SND_SOC_NOPM, 1, 0, NULL, 0, NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 1,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 2,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 3,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 4,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 5,
+ WM2200_AUDIO_IF_1_22, WM2200_AIF1TX6_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_S("OUT1L", 0, WM2200_OUTPUT_ENABLES,
+ WM2200_OUT1L_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("OUT1R", 0, WM2200_OUTPUT_ENABLES,
+ WM2200_OUT1R_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_S("EPD_LP", 1, WM2200_EAR_PIECE_CTRL_1,
+ WM2200_EPD_LP_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("EPD_OUTP_LP", 1, WM2200_EAR_PIECE_CTRL_1,
+ WM2200_EPD_OUTP_LP_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("EPD_RMV_SHRT_LP", 1, WM2200_EAR_PIECE_CTRL_1,
+ WM2200_EPD_RMV_SHRT_LP_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_S("EPD_LN", 1, WM2200_EAR_PIECE_CTRL_1,
+ WM2200_EPD_LN_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("EPD_OUTP_LN", 1, WM2200_EAR_PIECE_CTRL_1,
+ WM2200_EPD_OUTP_LN_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("EPD_RMV_SHRT_LN", 1, WM2200_EAR_PIECE_CTRL_1,
+ WM2200_EPD_RMV_SHRT_LN_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_S("EPD_RP", 1, WM2200_EAR_PIECE_CTRL_2,
+ WM2200_EPD_RP_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("EPD_OUTP_RP", 1, WM2200_EAR_PIECE_CTRL_2,
+ WM2200_EPD_OUTP_RP_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("EPD_RMV_SHRT_RP", 1, WM2200_EAR_PIECE_CTRL_2,
+ WM2200_EPD_RMV_SHRT_RP_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_S("EPD_RN", 1, WM2200_EAR_PIECE_CTRL_2,
+ WM2200_EPD_RN_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("EPD_OUTP_RN", 1, WM2200_EAR_PIECE_CTRL_2,
+ WM2200_EPD_OUTP_RN_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("EPD_RMV_SHRT_RN", 1, WM2200_EAR_PIECE_CTRL_2,
+ WM2200_EPD_RMV_SHRT_RN_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("OUT2L", WM2200_OUTPUT_ENABLES, WM2200_OUT2L_ENA_SHIFT,
+ 0, NULL, 0),
+SND_SOC_DAPM_PGA("OUT2R", WM2200_OUTPUT_ENABLES, WM2200_OUT2R_ENA_SHIFT,
+ 0, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("EPOUTLN"),
+SND_SOC_DAPM_OUTPUT("EPOUTLP"),
+SND_SOC_DAPM_OUTPUT("EPOUTRN"),
+SND_SOC_DAPM_OUTPUT("EPOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPK"),
+
+WM2200_MIXER_WIDGETS(EQL, "EQL"),
+WM2200_MIXER_WIDGETS(EQR, "EQR"),
+
+WM2200_MIXER_WIDGETS(LHPF1, "LHPF1"),
+WM2200_MIXER_WIDGETS(LHPF2, "LHPF2"),
+
+WM2200_MIXER_WIDGETS(DSP1L, "DSP1L"),
+WM2200_MIXER_WIDGETS(DSP1R, "DSP1R"),
+WM2200_MIXER_WIDGETS(DSP2L, "DSP2L"),
+WM2200_MIXER_WIDGETS(DSP2R, "DSP2R"),
+
+WM2200_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+WM2200_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+WM2200_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+WM2200_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+WM2200_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+WM2200_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+
+WM2200_MIXER_WIDGETS(OUT1L, "OUT1L"),
+WM2200_MIXER_WIDGETS(OUT1R, "OUT1R"),
+WM2200_MIXER_WIDGETS(OUT2L, "OUT2L"),
+WM2200_MIXER_WIDGETS(OUT2R, "OUT2R"),
+};
+
+static const struct snd_soc_dapm_route wm2200_dapm_routes[] = {
+ /* Everything needs SYSCLK but only hook up things on the edge
+ * of the chip */
+ { "IN1L", NULL, "SYSCLK" },
+ { "IN1R", NULL, "SYSCLK" },
+ { "IN2L", NULL, "SYSCLK" },
+ { "IN2R", NULL, "SYSCLK" },
+ { "IN3L", NULL, "SYSCLK" },
+ { "IN3R", NULL, "SYSCLK" },
+ { "OUT1L", NULL, "SYSCLK" },
+ { "OUT1R", NULL, "SYSCLK" },
+ { "OUT2L", NULL, "SYSCLK" },
+ { "OUT2R", NULL, "SYSCLK" },
+ { "AIF1RX1", NULL, "SYSCLK" },
+ { "AIF1RX2", NULL, "SYSCLK" },
+ { "AIF1RX3", NULL, "SYSCLK" },
+ { "AIF1RX4", NULL, "SYSCLK" },
+ { "AIF1RX5", NULL, "SYSCLK" },
+ { "AIF1RX6", NULL, "SYSCLK" },
+ { "AIF1TX1", NULL, "SYSCLK" },
+ { "AIF1TX2", NULL, "SYSCLK" },
+ { "AIF1TX3", NULL, "SYSCLK" },
+ { "AIF1TX4", NULL, "SYSCLK" },
+ { "AIF1TX5", NULL, "SYSCLK" },
+ { "AIF1TX6", NULL, "SYSCLK" },
+
+ { "IN1L", NULL, "AVDD" },
+ { "IN1R", NULL, "AVDD" },
+ { "IN2L", NULL, "AVDD" },
+ { "IN2R", NULL, "AVDD" },
+ { "IN3L", NULL, "AVDD" },
+ { "IN3R", NULL, "AVDD" },
+ { "OUT1L", NULL, "AVDD" },
+ { "OUT1R", NULL, "AVDD" },
+
+ { "IN1L PGA", NULL, "IN1L" },
+ { "IN1R PGA", NULL, "IN1R" },
+ { "IN2L PGA", NULL, "IN2L" },
+ { "IN2R PGA", NULL, "IN2R" },
+ { "IN3L PGA", NULL, "IN3L" },
+ { "IN3R PGA", NULL, "IN3R" },
+
+ { "Tone Generator", NULL, "TONE" },
+
+ { "CP2", NULL, "CPVDD" },
+ { "MICBIAS1", NULL, "CP2" },
+ { "MICBIAS2", NULL, "CP2" },
+
+ { "CP1", NULL, "CPVDD" },
+ { "EPD_LN", NULL, "CP1" },
+ { "EPD_LP", NULL, "CP1" },
+ { "EPD_RN", NULL, "CP1" },
+ { "EPD_RP", NULL, "CP1" },
+
+ { "EPD_LP", NULL, "OUT1L" },
+ { "EPD_OUTP_LP", NULL, "EPD_LP" },
+ { "EPD_RMV_SHRT_LP", NULL, "EPD_OUTP_LP" },
+ { "EPOUTLP", NULL, "EPD_RMV_SHRT_LP" },
+
+ { "EPD_LN", NULL, "OUT1L" },
+ { "EPD_OUTP_LN", NULL, "EPD_LN" },
+ { "EPD_RMV_SHRT_LN", NULL, "EPD_OUTP_LN" },
+ { "EPOUTLN", NULL, "EPD_RMV_SHRT_LN" },
+
+ { "EPD_RP", NULL, "OUT1R" },
+ { "EPD_OUTP_RP", NULL, "EPD_RP" },
+ { "EPD_RMV_SHRT_RP", NULL, "EPD_OUTP_RP" },
+ { "EPOUTRP", NULL, "EPD_RMV_SHRT_RP" },
+
+ { "EPD_RN", NULL, "OUT1R" },
+ { "EPD_OUTP_RN", NULL, "EPD_RN" },
+ { "EPD_RMV_SHRT_RN", NULL, "EPD_OUTP_RN" },
+ { "EPOUTRN", NULL, "EPD_RMV_SHRT_RN" },
+
+ { "SPK", NULL, "OUT2L" },
+ { "SPK", NULL, "OUT2R" },
+
+ WM2200_MIXER_ROUTES("DSP1", "DSP1L"),
+ WM2200_MIXER_ROUTES("DSP1", "DSP1R"),
+ WM2200_MIXER_ROUTES("DSP2", "DSP2L"),
+ WM2200_MIXER_ROUTES("DSP2", "DSP2R"),
+
+ WM2200_MIXER_ROUTES("OUT1L", "OUT1L"),
+ WM2200_MIXER_ROUTES("OUT1R", "OUT1R"),
+ WM2200_MIXER_ROUTES("OUT2L", "OUT2L"),
+ WM2200_MIXER_ROUTES("OUT2R", "OUT2R"),
+
+ WM2200_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+ WM2200_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+ WM2200_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+ WM2200_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+ WM2200_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+ WM2200_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+
+ WM2200_MIXER_ROUTES("EQL", "EQL"),
+ WM2200_MIXER_ROUTES("EQR", "EQR"),
+
+ WM2200_MIXER_ROUTES("LHPF1", "LHPF1"),
+ WM2200_MIXER_ROUTES("LHPF2", "LHPF2"),
+};
+
+static int wm2200_probe(struct snd_soc_codec *codec)
+{
+ struct wm2200_priv *wm2200 = dev_get_drvdata(codec->dev);
+ int ret;
+
+ wm2200->codec = codec;
+ codec->control_data = wm2200->regmap;
+ codec->dapm.bias_level = SND_SOC_BIAS_OFF;
+
+ ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+ return ret;
+ }
+
+ return ret;
+}
+
+static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ int lrclk, bclk, fmt_val;
+
+ lrclk = 0;
+ bclk = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_DSP_A:
+ fmt_val = 0;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ fmt_val = 1;
+ break;
+ case SND_SOC_DAIFMT_I2S:
+ fmt_val = 2;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ fmt_val = 3;
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported DAI format %d\n",
+ fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ case SND_SOC_DAIFMT_CBS_CFM:
+ lrclk |= WM2200_AIF1TX_LRCLK_MSTR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFS:
+ bclk |= WM2200_AIF1_BCLK_MSTR;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ lrclk |= WM2200_AIF1TX_LRCLK_MSTR;
+ bclk |= WM2200_AIF1_BCLK_MSTR;
+ break;
+ default:
+ dev_err(codec->dev, "Unsupported master mode %d\n",
+ fmt & SND_SOC_DAIFMT_MASTER_MASK);
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ bclk |= WM2200_AIF1_BCLK_INV;
+ lrclk |= WM2200_AIF1TX_LRCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ bclk |= WM2200_AIF1_BCLK_INV;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ lrclk |= WM2200_AIF1TX_LRCLK_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_1, WM2200_AIF1_BCLK_MSTR |
+ WM2200_AIF1_BCLK_INV, bclk);
+ snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_2,
+ WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV,
+ lrclk);
+ snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_3,
+ WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV,
+ lrclk);
+ snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5,
+ WM2200_AIF1_FMT_MASK << 1, fmt_val << 1);
+
+ return 0;
+}
+
+static int wm2200_sr_code[] = {
+ 0,
+ 12000,
+ 24000,
+ 48000,
+ 96000,
+ 192000,
+ 384000,
+ 768000,
+ 0,
+ 11025,
+ 22050,
+ 44100,
+ 88200,
+ 176400,
+ 352800,
+ 705600,
+ 4000,
+ 8000,
+ 16000,
+ 32000,
+ 64000,
+ 128000,
+ 256000,
+ 512000,
+};
+
+#define WM2200_NUM_BCLK_RATES 12
+
+static int wm2200_bclk_rates_dat[WM2200_NUM_BCLK_RATES] = {
+ 6144000,
+ 3072000,
+ 2048000,
+ 1536000,
+ 768000,
+ 512000,
+ 384000,
+ 256000,
+ 192000,
+ 128000,
+ 96000,
+ 64000,
+};
+
+static int wm2200_bclk_rates_cd[WM2200_NUM_BCLK_RATES] = {
+ 5644800,
+ 2882400,
+ 1881600,
+ 1411200,
+ 705600,
+ 470400,
+ 352800,
+ 176400,
+ 117600,
+ 88200,
+ 58800,
+};
+
+static int wm2200_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm2200_priv *wm2200 = snd_soc_codec_get_drvdata(codec);
+ int i, bclk, lrclk, wl, fl, sr_code;
+ int *bclk_rates;
+
+ /* Data sizes if not using TDM */
+ wl = snd_pcm_format_width(params_format(params));
+ if (wl < 0)
+ return wl;
+ fl = snd_soc_params_to_frame_size(params);
+ if (fl < 0)
+ return fl;
+
+ dev_dbg(codec->dev, "Word length %d bits, frame length %d bits\n",
+ wl, fl);
+
+ /* Target BCLK rate */
+ bclk = snd_soc_params_to_bclk(params);
+ if (bclk < 0)
+ return bclk;
+
+ if (!wm2200->sysclk) {
+ dev_err(codec->dev, "SYSCLK has no rate set\n");
+ return -EINVAL;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(wm2200_sr_code); i++)
+ if (wm2200_sr_code[i] == params_rate(params))
+ break;
+ if (i == ARRAY_SIZE(wm2200_sr_code)) {
+ dev_err(codec->dev, "Unsupported sample rate: %dHz\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+ sr_code = i;
+
+ dev_dbg(codec->dev, "Target BCLK is %dHz, using %dHz SYSCLK\n",
+ bclk, wm2200->sysclk);
+
+ if (wm2200->sysclk % 4000)
+ bclk_rates = wm2200_bclk_rates_cd;
+ else
+ bclk_rates = wm2200_bclk_rates_dat;
+
+ for (i = 0; i < WM2200_NUM_BCLK_RATES; i++)
+ if (bclk_rates[i] >= bclk && (bclk_rates[i] % bclk == 0))
+ break;
+ if (i == WM2200_NUM_BCLK_RATES) {
+ dev_err(codec->dev,
+ "No valid BCLK for %dHz found from %dHz SYSCLK\n",
+ bclk, wm2200->sysclk);
+ return -EINVAL;
+ }
+
+ bclk = i;
+ dev_dbg(codec->dev, "Setting %dHz BCLK\n", bclk_rates[bclk]);
+ snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_1,
+ WM2200_AIF1_BCLK_DIV_MASK, bclk);
+
+ lrclk = bclk_rates[bclk] / params_rate(params);
+ dev_dbg(codec->dev, "Setting %dHz LRCLK\n", bclk_rates[bclk] / lrclk);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
+ dai->symmetric_rates)
+ snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_7,
+ WM2200_AIF1RX_BCPF_MASK, lrclk);
+ else
+ snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_6,
+ WM2200_AIF1TX_BCPF_MASK, lrclk);
+
+ i = (wl << WM2200_AIF1TX_WL_SHIFT) | wl;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_9,
+ WM2200_AIF1RX_WL_MASK |
+ WM2200_AIF1RX_SLOT_LEN_MASK, i);
+ else
+ snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_8,
+ WM2200_AIF1TX_WL_MASK |
+ WM2200_AIF1TX_SLOT_LEN_MASK, i);
+
+ snd_soc_update_bits(codec, WM2200_CLOCKING_4,
+ WM2200_SAMPLE_RATE_1_MASK, sr_code);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops wm2200_dai_ops = {
+ .set_fmt = wm2200_set_fmt,
+ .hw_params = wm2200_hw_params,
+};
+
+static int wm2200_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+ int source, unsigned int freq, int dir)
+{
+ struct wm2200_priv *wm2200 = snd_soc_codec_get_drvdata(codec);
+ int fval;
+
+ switch (clk_id) {
+ case WM2200_CLK_SYSCLK:
+ break;
+
+ default:
+ dev_err(codec->dev, "Unknown clock %d\n", clk_id);
+ return -EINVAL;
+ }
+
+ switch (source) {
+ case WM2200_CLKSRC_MCLK1:
+ case WM2200_CLKSRC_MCLK2:
+ case WM2200_CLKSRC_FLL:
+ case WM2200_CLKSRC_BCLK1:
+ break;
+ default:
+ dev_err(codec->dev, "Invalid source %d\n", source);
+ return -EINVAL;
+ }
+
+