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-rw-r--r--Documentation/sound/alsa/HD-Audio-Models.txt4
-rw-r--r--sound/pci/hda/Kconfig13
-rw-r--r--sound/pci/hda/Makefile4
-rw-r--r--sound/pci/hda/hda_codec.c1
-rw-r--r--sound/pci/hda/patch_cirrus.c1194
5 files changed, 1216 insertions, 0 deletions
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 4c95a6c3f79f..775beea4506d 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -383,3 +383,7 @@ STAC9872
========
vaio VAIO laptop without SPDIF
auto BIOS setup (default)
+
+Cirrus Logic CS4206/4207
+========================
+ mbp55 MacBook Pro 5,5
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index b8a77f9b0827..55545e0818b5 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -148,6 +148,19 @@ config SND_HDA_ELD
def_bool y
depends on SND_HDA_CODEC_INTELHDMI
+config SND_HDA_CODEC_CIRRUS
+ bool "Build Cirrus Logic codec support"
+ depends on SND_HDA_INTEL
+ default y
+ help
+ Say Y here to include Cirrus Logic codec support in
+ snd-hda-intel driver, such as CS4206.
+
+ When the HD-audio driver is built as a module, the codec
+ support code is also built as another module,
+ snd-hda-codec-cirrus.
+ This module is automatically loaded at probing.
+
config SND_HDA_CODEC_CONEXANT
bool "Build Conexant HD-audio codec support"
default y
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index e3081d4586cc..315a1c4f8998 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -13,6 +13,7 @@ snd-hda-codec-analog-objs := patch_analog.o
snd-hda-codec-idt-objs := patch_sigmatel.o
snd-hda-codec-si3054-objs := patch_si3054.o
snd-hda-codec-atihdmi-objs := patch_atihdmi.o
+snd-hda-codec-cirrus-objs := patch_cirrus.o
snd-hda-codec-ca0110-objs := patch_ca0110.o
snd-hda-codec-conexant-objs := patch_conexant.o
snd-hda-codec-via-objs := patch_via.o
@@ -41,6 +42,9 @@ endif
ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o
endif
+ifdef CONFIG_SND_HDA_CODEC_CIRRUS
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cirrus.o
+endif
ifdef CONFIG_SND_HDA_CODEC_CA0110
obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
endif
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3a603cde8cc4..a23c27d2fb2f 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -44,6 +44,7 @@ struct hda_vendor_id {
/* codec vendor labels */
static struct hda_vendor_id hda_vendor_ids[] = {
{ 0x1002, "ATI" },
+ { 0x1013, "Cirrus Logic" },
{ 0x1057, "Motorola" },
{ 0x1095, "Silicon Image" },
{ 0x10de, "Nvidia" },
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
new file mode 100644
index 000000000000..8ba306856d38
--- /dev/null
+++ b/sound/pci/hda/patch_cirrus.c
@@ -0,0 +1,1194 @@
+/*
+ * HD audio interface patch for Cirrus Logic CS420x chip
+ *
+ * Copyright (c) 2009 Takashi Iwai <tiwai@suse.de>
+ *
+ * This driver 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; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This driver is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+/*
+ */
+
+struct cs_spec {
+ int board_config;
+ struct auto_pin_cfg autocfg;
+ struct hda_multi_out multiout;
+ struct snd_kcontrol *vmaster_sw;
+ struct snd_kcontrol *vmaster_vol;
+
+ hda_nid_t dac_nid[AUTO_CFG_MAX_OUTS];
+ hda_nid_t slave_dig_outs[2];
+
+ unsigned int input_idx[AUTO_PIN_LAST];
+ unsigned int capsrc_idx[AUTO_PIN_LAST];
+ hda_nid_t adc_nid[AUTO_PIN_LAST];
+ unsigned int adc_idx[AUTO_PIN_LAST];
+ unsigned int num_inputs;
+ unsigned int cur_input;
+ unsigned int automic_idx;
+ hda_nid_t cur_adc;
+ unsigned int cur_adc_stream_tag;
+ unsigned int cur_adc_format;
+ hda_nid_t dig_in;
+
+ struct hda_bind_ctls *capture_bind[2];
+
+ unsigned int gpio_mask;
+ unsigned int gpio_dir;
+ unsigned int gpio_data;
+
+ struct hda_pcm pcm_rec[2]; /* PCM information */
+
+ unsigned int hp_detect:1;
+ unsigned int mic_detect:1;
+};
+
+/* available models */
+enum {
+ CS420X_MBP55,
+ CS420X_AUTO,
+ CS420X_MODELS
+};
+
+/* Vendor-specific processing widget */
+#define CS420X_VENDOR_NID 0x11
+#define CS_DIG_OUT1_PIN_NID 0x10
+#define CS_DIG_OUT2_PIN_NID 0x15
+#define CS_DMIC1_PIN_NID 0x12
+#define CS_DMIC2_PIN_NID 0x0e
+
+/* coef indices */
+#define IDX_SPDIF_STAT 0x0000
+#define IDX_SPDIF_CTL 0x0001
+#define IDX_ADC_CFG 0x0002
+/* SZC bitmask, 4 modes below:
+ * 0 = immediate,
+ * 1 = digital immediate, analog zero-cross
+ * 2 = digtail & analog soft-ramp
+ * 3 = digital soft-ramp, analog zero-cross
+ */
+#define CS_COEF_ADC_SZC_MASK (3 << 0)
+#define CS_COEF_ADC_MIC_SZC_MODE (3 << 0) /* SZC setup for mic */
+#define CS_COEF_ADC_LI_SZC_MODE (3 << 0) /* SZC setup for line-in */
+/* PGA mode: 0 = differential, 1 = signle-ended */
+#define CS_COEF_ADC_MIC_PGA_MODE (1 << 5) /* PGA setup for mic */
+#define CS_COEF_ADC_LI_PGA_MODE (1 << 6) /* PGA setup for line-in */
+#define IDX_DAC_CFG 0x0003
+/* SZC bitmask, 4 modes below:
+ * 0 = Immediate
+ * 1 = zero-cross
+ * 2 = soft-ramp
+ * 3 = soft-ramp on zero-cross
+ */
+#define CS_COEF_DAC_HP_SZC_MODE (3 << 0) /* nid 0x02 */
+#define CS_COEF_DAC_LO_SZC_MODE (3 << 2) /* nid 0x03 */
+#define CS_COEF_DAC_SPK_SZC_MODE (3 << 4) /* nid 0x04 */
+
+#define IDX_BEEP_CFG 0x0004
+/* 0x0008 - test reg key */
+/* 0x0009 - 0x0014 -> 12 test regs */
+/* 0x0015 - visibility reg */
+
+
+static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx)
+{
+ snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+ AC_VERB_SET_COEF_INDEX, idx);
+ return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0,
+ AC_VERB_GET_PROC_COEF, 0);
+}
+
+static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx,
+ unsigned int coef)
+{
+ snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+ AC_VERB_SET_COEF_INDEX, idx);
+ snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+ AC_VERB_SET_PROC_COEF, coef);
+}
+
+
+#define HP_EVENT 1
+#define MIC_EVENT 2
+
+/*
+ * PCM callbacks
+ */
+static int cs_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+ hinfo);
+}
+
+static int cs_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+ stream_tag, format, substream);
+}
+
+static int cs_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int cs_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int cs_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int cs_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+ format, substream);
+}
+
+static int cs_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Analog capture
+ */
+static int cs_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ unsigned int stream_tag,
+ unsigned int format,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ spec->cur_adc = spec->adc_nid[spec->cur_input];
+ spec->cur_adc_stream_tag = stream_tag;
+ spec->cur_adc_format = format;
+ snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
+ return 0;
+}
+
+static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+ struct hda_codec *codec,
+ struct snd_pcm_substream *substream)
+{
+ struct cs_spec *spec = codec->spec;
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ spec->cur_adc = 0;
+ return 0;
+}
+
+/*
+ */
+static struct hda_pcm_stream cs_pcm_analog_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .open = cs_playback_pcm_open,
+ .prepare = cs_playback_pcm_prepare,
+ .cleanup = cs_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream cs_pcm_analog_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .prepare = cs_capture_pcm_prepare,
+ .cleanup = cs_capture_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream cs_pcm_digital_playback = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+ .ops = {
+ .open = cs_dig_playback_pcm_open,
+ .close = cs_dig_playback_pcm_close,
+ .prepare = cs_dig_playback_pcm_prepare,
+ .cleanup = cs_dig_playback_pcm_cleanup
+ },
+};
+
+static struct hda_pcm_stream cs_pcm_digital_capture = {
+ .substreams = 1,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static int cs_build_pcms(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct hda_pcm *info = spec->pcm_rec;
+
+ codec->pcm_info = info;
+ codec->num_pcms = 0;
+
+ info->name = "Cirrus Analog";
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cs_pcm_analog_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dac_nid[0];
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+ spec->multiout.max_channels;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] = cs_pcm_analog_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
+ spec->adc_nid[spec->cur_input];
+ codec->num_pcms++;
+
+ if (!spec->multiout.dig_out_nid && !spec->dig_in)
+ return 0;
+
+ info++;
+ info->name = "Cirrus Digital";
+ info->pcm_type = spec->autocfg.dig_out_type[0];
+ if (!info->pcm_type)
+ info->pcm_type = HDA_PCM_TYPE_SPDIF;
+ if (spec->multiout.dig_out_nid) {
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+ cs_pcm_digital_playback;
+ info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+ spec->multiout.dig_out_nid;
+ }
+ if (spec->dig_in) {
+ info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+ cs_pcm_digital_capture;
+ info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+ }
+ codec->num_pcms++;
+
+ return 0;
+}
+
+/*
+ * parse codec topology
+ */
+
+static hda_nid_t get_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+ hda_nid_t dac;
+ if (!pin)
+ return 0;
+ if (snd_hda_get_connections(codec, pin, &dac, 1) != 1)
+ return 0;
+ return dac;
+}
+
+static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t pin = cfg->input_pins[idx];
+ unsigned int val = snd_hda_query_pin_caps(codec, pin);
+ if (!(val & AC_PINCAP_PRES_DETECT))
+ return 0;
+ val = snd_hda_codec_get_pincfg(codec, pin);
+ return (get_defcfg_connect(val) == AC_JACK_PORT_COMPLEX);
+}
+
+static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
+ unsigned int *idxp)
+{
+ int i;
+ hda_nid_t nid;
+
+ nid = codec->start_nid;
+ for (i = 0; i < codec->num_nodes; i++, nid++) {
+ hda_nid_t pins[2];
+ unsigned int type;
+ int j, nums;
+ type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
+ >> AC_WCAP_TYPE_SHIFT;
+ if (type != AC_WID_AUD_IN)
+ continue;
+ nums = snd_hda_get_connections(codec, nid, pins,
+ ARRAY_SIZE(pins));
+ if (nums <= 0)
+ continue;
+ for (j = 0; j < nums; j++) {
+ if (pins[j] == pin) {
+ *idxp = j;
+ return nid;
+ }
+ }
+ }
+ return 0;
+}
+
+static int is_active_pin(struct hda_codec *codec, hda_nid_t nid)
+{
+ unsigned int val;
+ val = snd_hda_codec_get_pincfg(codec, nid);
+ return (get_defcfg_connect(val) != AC_JACK_PORT_NONE);
+}
+
+static int parse_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, extra_nids;
+ hda_nid_t dac;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ dac = get_dac(codec, cfg->line_out_pins[i]);
+ if (!dac)
+ break;
+ spec->dac_nid[i] = dac;
+ }
+ spec->multiout.num_dacs = i;
+ spec->multiout.dac_nids = spec->dac_nid;
+ spec->multiout.max_channels = i * 2;
+
+ /* add HP and speakers */
+ extra_nids = 0;
+ for (i = 0; i < cfg->hp_outs; i++) {
+ dac = get_dac(codec, cfg->hp_pins[i]);
+ if (!dac)
+ break;
+ if (!i)
+ spec->multiout.hp_nid = dac;
+ else
+ spec->multiout.extra_out_nid[extra_nids++] = dac;
+ }
+ for (i = 0; i < cfg->speaker_outs; i++) {
+ dac = get_dac(codec, cfg->speaker_pins[i]);
+ if (!dac)
+ break;
+ spec->multiout.extra_out_nid[extra_nids++] = dac;
+ }
+
+ if (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 = 0;
+ }
+
+ return 0;
+}
+
+static int parse_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ hda_nid_t pin = cfg->input_pins[i];
+ if (!pin)
+ continue;
+ spec->input_idx[spec->num_inputs] = i;
+ spec->capsrc_idx[i] = spec->num_inputs++;
+ spec->cur_input = i;
+ spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]);
+ }
+ if (!spec->num_inputs)
+ return 0;
+
+ /* check whether the automatic mic switch is available */
+ if (spec->num_inputs == 2 &&
+ spec->adc_nid[AUTO_PIN_MIC] && spec->adc_nid[AUTO_PIN_FRONT_MIC]) {
+ if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_FRONT_MIC])) {
+ if (!is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
+ spec->mic_detect = 1;
+ spec->automic_idx = AUTO_PIN_FRONT_MIC;
+ }
+ } else {
+ if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
+ spec->mic_detect = 1;
+ spec->automic_idx = AUTO_PIN_MIC;
+ }
+ }
+ }
+ return 0;
+}
+
+
+static int parse_digital_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t nid;
+
+ if (!cfg->dig_outs)
+ return 0;
+ if (snd_hda_get_connections(codec, cfg->dig_out_pins[0], &nid, 1) < 1)
+ return 0;
+ spec->multiout.dig_out_nid = nid;
+ spec->multiout.share_spdif = 1;
+ if (cfg->dig_outs > 1 &&
+ snd_hda_get_connections(codec, cfg->dig_out_pins[1], &nid, 1) > 0) {
+ spec->slave_dig_outs[0] = nid;
+ codec->slave_dig_outs = spec->slave_dig_outs;
+ }
+ return 0;
+}
+
+static int parse_digital_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int idx;
+
+ if (cfg->dig_in_pin)
+ spec->dig_in = get_adc(codec, cfg->dig_in_pin, &idx);
+ return 0;
+}
+
+/*
+ * create mixer controls
+ */
+
+static const char *dir_sfx[2] = { "Playback", "Capture" };
+
+static int add_mute(struct hda_codec *codec, const char *name, int index,
+ unsigned int pval, int dir, struct snd_kcontrol **kctlp)
+{
+ char tmp[44];
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_MUTE_IDX(tmp, index, 0, 0, HDA_OUTPUT);
+ knew.private_value = pval;
+ snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]);
+ *kctlp = snd_ctl_new1(&knew, codec);
+ return snd_hda_ctl_add(codec, *kctlp);
+}
+
+static int add_volume(struct hda_codec *codec, const char *name,
+ int index, unsigned int pval, int dir,
+ struct snd_kcontrol **kctlp)
+{
+ char tmp[32];
+ struct snd_kcontrol_new knew =
+ HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
+ knew.private_value = pval;
+ snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]);
+ *kctlp = snd_ctl_new1(&knew, codec);
+ return snd_hda_ctl_add(codec, *kctlp);
+}
+
+static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac)
+{
+ unsigned int caps;
+
+ /* set the upper-limit for mixer amp to 0dB */
+ caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+ caps &= ~(0x7f << AC_AMPCAP_NUM_STEPS_SHIFT);
+ caps |= ((caps >> AC_AMPCAP_OFFSET_SHIFT) & 0x7f)
+ << AC_AMPCAP_NUM_STEPS_SHIFT;
+ snd_hda_override_amp_caps(codec, dac, HDA_OUTPUT, caps);
+}
+
+static int add_vmaster(struct hda_codec *codec, hda_nid_t dac)
+{
+ struct cs_spec *spec = codec->spec;
+ unsigned int tlv[4];
+ int err;
+
+ spec->vmaster_sw =
+ snd_ctl_make_virtual_master("Master Playback Switch", NULL);
+ err = snd_hda_ctl_add(codec, spec->vmaster_sw);
+ if (err < 0)
+ return err;
+
+ snd_hda_set_vmaster_tlv(codec, dac, HDA_OUTPUT, tlv);
+ spec->vmaster_vol =
+ snd_ctl_make_virtual_master("Master Playback Volume", tlv);
+ err = snd_hda_ctl_add(codec, spec->vmaster_vol);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
+ int num_ctls, int type)
+{
+ struct cs_spec *spec = codec->spec;
+ const char *name;
+ int err, index;
+ struct snd_kcontrol *kctl;
+ static char *speakers[] = {
+ "Front Speaker", "Surround Speaker", "Bass Speaker"
+ };
+ static char *line_outs[] = {
+ "Front Line-Out", "Surround Line-Out", "Bass Line-Out"
+ };
+
+ fix_volume_caps(codec, dac);
+ if (!spec->vmaster_sw) {
+ err = add_vmaster(codec, dac);
+ if (err < 0)
+ return err;
+ }
+
+ index = 0;
+ switch (type) {
+ case AUTO_PIN_HP_OUT:
+ name = "Headphone";
+ index = idx;
+ break;
+ case AUTO_PIN_SPEAKER_OUT:
+ if (num_ctls > 1)
+ name = speakers[idx];
+ else
+ name = "Speaker";
+ break;
+ default:
+ if (num_ctls > 1)
+ name = line_outs[idx];
+ else
+ name = "Line-Out";
+ break;
+ }
+
+ err = add_mute(codec, name, index,
+ HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+ if (err < 0)
+ return err;
+ err = snd_ctl_add_slave(spec->vmaster_sw, kctl);
+ if (err < 0)
+ return err;
+
+ err = add_volume(codec, name, index,
+ HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+ if (err < 0)
+ return err;
+ err = snd_ctl_add_slave(spec->vmaster_vol, kctl);
+ if (err < 0)
+ return err;
+
+ return 0;
+}
+
+static int build_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i, err;
+
+ for (i = 0; i < cfg->line_outs; i++) {
+ err = add_output(codec, get_dac(codec, cfg->line_out_pins[i]),
+ i, cfg->line_outs, cfg->line_out_type);
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < cfg->hp_outs; i++) {
+ err = add_output(codec, get_dac(codec, cfg->hp_pins[i]),
+ i, cfg->hp_outs, AUTO_PIN_HP_OUT);
+ if (err < 0)
+ return err;
+ }
+ for (i = 0; i < cfg->speaker_outs; i++) {
+ err = add_output(codec, get_dac(codec, cfg->speaker_pins[i]),
+ i, cfg->speaker_outs, AUTO_PIN_SPEAKER_OUT);
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
+/*
+ */
+
+static struct snd_kcontrol_new cs_capture_ctls[] = {
+ HDA_BIND_SW("Capture Switch", 0),
+ HDA_BIND_VOL("Capture Volume", 0),
+};
+
+static int change_cur_input(struct hda_codec *codec, unsigned int idx,
+ int force)
+{
+ struct cs_spec *spec = codec->spec;
+
+ if (spec->cur_input == idx && !force)
+ return 0;
+ if (spec->cur_adc && spec->cur_adc != spec->adc_nid[idx]) {
+ /* stream is running, let's swap the current ADC */
+ snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+ spec->cur_adc = spec->adc_nid[idx];
+ snd_hda_codec_setup_stream(codec, spec->cur_adc,
+ spec->cur_adc_stream_tag, 0,
+ spec->cur_adc_format);
+ }
+ snd_hda_codec_write(codec, spec->cur_adc, 0,
+ AC_VERB_SET_CONNECT_SEL,
+ spec->adc_idx[idx]);
+ spec->cur_input = idx;
+ return 1;
+}
+
+static int cs_capture_source_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs_spec *spec = codec->spec;
+ unsigned int idx;
+
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = spec->num_inputs;
+ if (uinfo->value.enumerated.item >= spec->num_inputs)
+ uinfo->value.enumerated.item = spec->num_inputs - 1;
+ idx = spec->input_idx[uinfo->value.enumerated.item];
+ strcpy(uinfo->value.enumerated.name, auto_pin_cfg_labels[idx]);
+ return 0;
+}
+
+static int cs_capture_source_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs_spec *spec = codec->spec;
+ ucontrol->value.enumerated.item[0] = spec->capsrc_idx[spec->cur_input];
+ return 0;
+}
+
+static int cs_capture_source_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs_spec *spec = codec->spec;
+ unsigned int idx = ucontrol->value.enumerated.item[0];
+
+ if (idx >= spec->num_inputs)
+ return -EINVAL;
+ idx = spec->input_idx[idx];
+ return change_cur_input(codec, idx, 0);
+}
+
+static struct snd_kcontrol_new cs_capture_source = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "Capture Source",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = cs_capture_source_info,
+ .get = cs_capture_source_get,
+ .put = cs_capture_source_put,
+};
+
+static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec,
+ struct hda_ctl_ops *ops)
+{
+ struct cs_spec *spec = codec->spec;
+ struct hda_bind_ctls *bind;
+ int i, n;
+
+ bind = kzalloc(sizeof(*bind) + sizeof(long) * (spec->num_inputs + 1),
+ GFP_KERNEL);
+ if (!bind)
+ return NULL;
+ bind->ops = ops;
+ n = 0;
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ if (!spec->adc_nid[i])
+ continue;
+ bind->values[n++] =
+ HDA_COMPOSE_AMP_VAL(spec->adc_nid[i], 3,
+ spec->adc_idx[i], HDA_INPUT);
+ }
+ return bind;
+}
+
+static int build_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ int i, err;
+
+ if (!spec->num_inputs)
+ return 0;
+
+ /* make bind-capture */
+ spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw);
+ spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol);
+ for (i = 0; i < 2; i++) {
+ struct snd_kcontrol *kctl;
+ if (!spec->capture_bind[i])
+ return -ENOMEM;
+ kctl = snd_ctl_new1(&cs_capture_ctls[i], codec);
+ if (!kctl)
+ return -ENOMEM;
+ kctl->private_value = (long)spec->capture_bind[i];
+ err = snd_hda_ctl_add(codec, kctl);
+ if (err < 0)
+ return err;
+ }
+
+ if (spec->num_inputs > 1 && !spec->mic_detect) {
+ err = snd_hda_ctl_add(codec,
+ snd_ctl_new1(&cs_capture_source, codec));
+ if (err < 0)
+ return err;
+ }
+
+ return 0;
+}
+
+/*
+ */
+
+static int build_digital_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ int err;
+
+ if (!spec->multiout.dig_out_nid)
+ return 0;
+
+ err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+ if (err < 0)
+ return err;
+ err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static int build_digital_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ if (spec->dig_in)
+ return snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+ return 0;
+}
+
+/*
+ * auto-mute and auto-mic switching
+ */
+
+static void cs_automute(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int caps, present, hp_present;
+ hda_nid_t nid;
+ int i;
+
+ hp_present = 0;
+ for (i = 0; i < cfg->hp_outs; i++) {
+ nid = cfg->hp_pins[i];
+ caps = snd_hda_query_pin_caps(codec, nid);
+ if (!(caps & AC_PINCAP_PRES_DETECT))
+ continue;
+ if (caps & AC_PINCAP_TRIG_REQ)
+ snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ hp_present |= (present & AC_PINSENSE_PRESENCE) != 0;
+ if (hp_present)
+ break;
+ }
+ for (i = 0; i < cfg->speaker_outs; i++) {
+ nid = cfg->speaker_pins[i];
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL,
+ hp_present ? 0 : PIN_OUT);
+ }
+ if (spec->board_config == CS420X_MBP55) {
+ unsigned int gpio = hp_present ? 0x02 : 0x08;
+ snd_hda_codec_write(codec, 0x01, 0,
+ AC_VERB_SET_GPIO_DATA, gpio);
+ }
+}
+
+static void cs_automic(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ hda_nid_t nid;
+ unsigned int caps, present;
+
+ nid = cfg->input_pins[spec->automic_idx];
+ caps = snd_hda_query_pin_caps(codec, nid);
+ if (caps & AC_PINCAP_TRIG_REQ)
+ snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+ present = snd_hda_codec_read(codec, nid, 0,
+ AC_VERB_GET_PIN_SENSE, 0);
+ if (present & AC_PINSENSE_PRESENCE)
+ change_cur_input(codec, spec->automic_idx, 0);
+ else {
+ unsigned int imic = (spec->automic_idx == AUTO_PIN_MIC) ?
+ AUTO_PIN_FRONT_MIC : AUTO_PIN_MIC;
+ change_cur_input(codec, imic, 0);
+ }
+}
+
+/*
+ */
+
+static void init_output(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ int i;
+
+ /* mute first */
+ for (i = 0; i < spec->multiout.num_dacs; i++)
+ snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ if (spec->multiout.hp_nid)
+ snd_hda_codec_write(codec, spec->multiout.hp_nid, 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) {
+ if (!spec->multiout.extra_out_nid[i])
+ break;
+ snd_hda_codec_write(codec, spec->multiout.extra_out_nid[i], 0,
+ AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+ }
+
+ /* set appropriate pin controls */
+ for (i = 0; i < cfg->line_outs; i++)
+ snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ for (i = 0; i < cfg->hp_outs; i++) {
+ hda_nid_t nid = cfg->hp_pins[i];
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+ if (!cfg->speaker_outs)
+ continue;
+ if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
+ snd_hda_codec_write(codec, nid, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | HP_EVENT);
+ spec->hp_detect = 1;
+ }
+ }
+ for (i = 0; i < cfg->speaker_outs; i++)
+ snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+ if (spec->hp_detect)
+ cs_automute(codec);
+}
+
+static void init_input(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ struct auto_pin_cfg *cfg = &spec->autocfg;
+ unsigned int coef;
+ int i;
+
+ for (i = 0; i < AUTO_PIN_LAST; i++) {
+ unsigned int ctl;
+ hda_nid_t pin = cfg->input_pins[i];
+ if (!pin || !spec->adc_nid[i])
+ continue;
+ /* set appropriate pin control and mute first */
+ ctl = PIN_IN;
+ if (i <= AUTO_PIN_FRONT_MIC) {
+ unsigned int caps = snd_hda_query_pin_caps(codec, pin);
+ caps >>= AC_PINCAP_VREF_SHIFT;
+ if (caps & AC_PINCAP_VREF_80)
+ ctl = PIN_VREF80;
+ }
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
+ snd_hda_codec_write(codec, spec->adc_nid[i], 0,
+ AC_VERB_SET_AMP_GAIN_MUTE,
+ AMP_IN_MUTE(spec->adc_idx[i]));
+ if (spec->mic_detect && spec->automic_idx == i)
+ snd_hda_codec_write(codec, pin, 0,
+ AC_VERB_SET_UNSOLICITED_ENABLE,
+ AC_USRSP_EN | MIC_EVENT);
+ }
+ change_cur_input(codec, spec->cur_input, 1);
+ if (spec->mic_detect)
+ cs_automic(codec);
+
+ coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */
+ if (is_active_pin(codec, CS_DMIC2_PIN_NID))
+ coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */
+ if (is_active_pin(codec, CS_DMIC1_PIN_NID))
+ coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0
+ * No effect if SPDIF_OUT2 is slected in
+ * IDX_SPDIF_CTL.
+ */
+ cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
+}
+
+static struct hda_verb cs_coef_init_verbs[] = {
+ {0x11, AC_VERB_SET_PROC_STATE, 1},
+ {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG},
+ {0x11, AC_VERB_SET_PROC_COEF,
+ (0x002a /* DAC1/2/3 SZCMode Soft Ramp */
+ | 0x0040 /* Mute DACs on FIFO error */
+ | 0x1000 /* Enable DACs High Pass Filter */
+ | 0x0400 /* Disable Coefficient Auto increment */
+ )},
+ /* Beep */
+ {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG},
+ {0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */
+
+ {} /* terminator */
+};
+
+/* SPDIF setup */
+static void init_digital(struct hda_codec *codec)
+{
+ unsigned int coef;
+
+ coef = 0x0002; /* SRC_MUTE soft-mute on SPDIF (if no lock) */
+ coef |= 0x0008; /* Replace with mute on error */
+ if (is_active_pin(codec, CS_DIG_OUT2_PIN_NID))
+ coef |= 0x4000; /* RX to TX1 or TX2 Loopthru / SPDIF2
+ * SPDIF_OUT2 is shared with GPIO1 and
+ * DMIC_SDA2.
+ */
+ cs_vendor_coef_set(codec, IDX_SPDIF_CTL, coef);
+}
+
+static int cs_init(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+
+ snd_hda_sequence_write(codec, cs_coef_init_verbs);
+
+ if (spec->gpio_mask) {
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK,
+ spec->gpio_mask);
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION,
+ spec->gpio_dir);
+ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+ spec->gpio_data);
+ }
+
+ init_output(codec);
+ init_input(codec);
+ init_digital(codec);
+ return 0;
+}
+
+static int cs_build_controls(struct hda_codec *codec)
+{
+ int err;
+
+ err = build_output(codec);
+ if (err < 0)
+ return err;
+ err = build_input(codec);
+ if (err < 0)
+ return err;
+ err = build_digital_output(codec);
+ if (err < 0)
+ return err;
+ err = build_digital_input(codec);
+ if (err < 0)
+ return err;
+ return cs_init(codec);
+}
+
+static void cs_free(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ kfree(spec->capture_bind[0]);
+ kfree(spec->capture_bind[1]);
+ kfree(codec->spec);
+}
+
+static void cs_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+ switch ((res >> 26) & 0x7f) {
+ case HP_EVENT:
+ cs_automute(codec);
+ break;
+ case MIC_EVENT:
+ cs_automic(codec);
+ break;
+ }
+}
+
+static struct hda_codec_ops cs_patch_ops = {
+ .build_controls = cs_build_controls,
+ .build_pcms = cs_build_pcms,
+ .init = cs_init,
+ .free = cs_free,
+ .unsol_event = cs_unsol_event,
+};
+
+static int cs_parse_auto_config(struct hda_codec *codec)
+{
+ struct cs_spec *spec = codec->spec;
+ int err;
+
+ err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+ if (err < 0)
+ return err;
+
+ err = parse_output(codec);
+ if (err < 0)
+ return err;
+ err = parse_input(codec);
+ if (err < 0)
+ return err;
+ err = parse_digital_output(codec);
+ if (err < 0)
+ return err;
+ err = parse_digital_input(codec);
+ if (err < 0)
+ return err;
+ return 0;
+}
+
+static const char *cs420x_models[CS420X_MODELS] = {
+ [CS420X_MBP55] = "mbp55",
+ [CS420X_AUTO] = "auto",
+};
+
+
+static struct snd_pci_quirk cs420x_cfg_tbl[] = {
+ SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
+ {} /* terminator */
+};
+
+struct cs_pincfg {
+ hda_nid_t nid;
+ u32 val;
+};
+
+static struct cs_pincfg mbp55_pincfgs[] = {
+ { 0x09, 0x012b4030 },
+ { 0x0a, 0x90100121 },
+ { 0x0b, 0x90100120 },
+ { 0x0c, 0x400000f0 },
+ { 0x0d, 0x90a00110 },
+ { 0x0e, 0x400000f0 },
+ { 0x0f, 0x400000f0 },
+ { 0x10, 0x014be040 },
+ { 0x12, 0x400000f0 },
+ { 0x15, 0x400000f0 },
+ {} /* terminator */
+};
+
+static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
+ [CS420X_MBP55] = mbp55_pincfgs,
+};
+
+static void fix_pincfg(struct hda_codec *codec, int model)
+{
+ const struct cs_pincfg *cfg = cs_pincfgs[model];
+ if (!cfg)
+ return;
+ for (; cfg->nid; cfg++)
+ snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+}
+
+
+static int patch_cs420x(struct hda_codec *codec)
+{
+ struct cs_spec *spec;
+ int err;
+
+ spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+ if (!spec)
+ return -ENOMEM;
+ codec->spec = spec;
+
+ spec->board_config =
+ snd_hda_check_board_config(codec, CS420X_MODELS,
+ cs420x_models, cs420x_cfg_tbl);
+ if (spec->board_config >= 0)
+ fix_pincfg(codec, spec->board_config);
+
+ switch (spec->board_config) {
+ case CS420X_MBP55:
+ /* GPIO1 = headphones */
+ /* GPIO3 = speakers */
+ spec->gpio_mask = 0x0a;
+ spec->gpio_dir = 0x0a;
+ break;
+ }
+
+ err = cs_parse_auto_config(codec);
+ if (err < 0)
+ goto error;
+
+ codec->patch_ops = cs_patch_ops;
+
+ return 0;
+
+ error:
+ kfree(codec->spec);
+ codec->spec = NULL;
+ return err;
+}
+
+
+/*
+ * patch entries
+ */
+static struct hda_codec_preset snd_hda_preset_cirrus[] = {
+ { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x },
+ { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x },
+ {} /* terminator */
+};
+
+MODULE_ALIAS("snd-hda-codec-id:10134206");
+MODULE_ALIAS("snd-hda-codec-id:10134207");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Cirrus Logic HD-audio codec");
+
+static struct hda_codec_preset_list cirrus_list = {
+ .preset = snd_hda_preset_cirrus,
+ .owner = THIS_MODULE,
+};
+
+static int __init patch_cirrus_init(void)
+{
+ return snd_hda_add_codec_preset(&cirrus_list);
+}
+
+static void __exit patch_cirrus_exit(void)
+{
+ snd_hda_delete_codec_preset(&cirrus_list);
+}
+
+module_init(patch_cirrus_init)
+module_exit(patch_cirrus_exit)