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Diffstat (limited to 'sound/soc/codecs/ssm2602.c')
-rw-r--r--sound/soc/codecs/ssm2602.c651
1 files changed, 651 insertions, 0 deletions
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
new file mode 100644
index 000000000..314eaece1
--- /dev/null
+++ b/sound/soc/codecs/ssm2602.c
@@ -0,0 +1,651 @@
+/*
+ * File: sound/soc/codecs/ssm2602.c
+ * Author: Cliff Cai <Cliff.Cai@analog.com>
+ *
+ * Created: Tue June 06 2008
+ * Description: Driver for ssm2602 sound chip
+ *
+ * Modified:
+ * Copyright 2008 Analog Devices Inc.
+ *
+ * Bugs: Enter bugs at http://blackfin.uclinux.org/
+ *
+ * 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; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "ssm2602.h"
+
+/* codec private data */
+struct ssm2602_priv {
+ unsigned int sysclk;
+ const struct snd_pcm_hw_constraint_list *sysclk_constraints;
+
+ struct regmap *regmap;
+
+ enum ssm2602_type type;
+ unsigned int clk_out_pwr;
+};
+
+/*
+ * ssm2602 register cache
+ * We can't read the ssm2602 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ * There is no point in caching the reset register
+ */
+static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = {
+ 0x0097, 0x0097, 0x0079, 0x0079,
+ 0x000a, 0x0008, 0x009f, 0x000a,
+ 0x0000, 0x0000
+};
+
+
+/*Appending several "None"s just for OSS mixer use*/
+static const char *ssm2602_input_select[] = {
+ "Line", "Mic",
+};
+
+static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum ssm2602_enum[] = {
+ SOC_ENUM_SINGLE(SSM2602_APANA, 2, ARRAY_SIZE(ssm2602_input_select),
+ ssm2602_input_select),
+ SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, ARRAY_SIZE(ssm2602_deemph),
+ ssm2602_deemph),
+};
+
+static const unsigned int ssm260x_outmix_tlv[] = {
+ TLV_DB_RANGE_HEAD(2),
+ 0, 47, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0),
+ 48, 127, TLV_DB_SCALE_ITEM(-7400, 100, 0),
+};
+
+static const DECLARE_TLV_DB_SCALE(ssm260x_inpga_tlv, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(ssm260x_sidetone_tlv, -1500, 300, 0);
+
+static const struct snd_kcontrol_new ssm260x_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Capture Volume", SSM2602_LINVOL, SSM2602_RINVOL, 0, 45, 0,
+ ssm260x_inpga_tlv),
+SOC_DOUBLE_R("Capture Switch", SSM2602_LINVOL, SSM2602_RINVOL, 7, 1, 1),
+
+SOC_SINGLE("ADC High Pass Filter Switch", SSM2602_APDIGI, 0, 1, 1),
+SOC_SINGLE("Store DC Offset Switch", SSM2602_APDIGI, 4, 1, 0),
+
+SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
+};
+
+static const struct snd_kcontrol_new ssm2602_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Master Playback Volume", SSM2602_LOUT1V, SSM2602_ROUT1V,
+ 0, 127, 0, ssm260x_outmix_tlv),
+SOC_DOUBLE_R("Master Playback ZC Switch", SSM2602_LOUT1V, SSM2602_ROUT1V,
+ 7, 1, 0),
+SOC_SINGLE_TLV("Sidetone Playback Volume", SSM2602_APANA, 6, 3, 1,
+ ssm260x_sidetone_tlv),
+
+SOC_SINGLE("Mic Boost (+20dB)", SSM2602_APANA, 0, 1, 0),
+SOC_SINGLE("Mic Boost2 (+20dB)", SSM2602_APANA, 8, 1, 0),
+SOC_SINGLE("Mic Switch", SSM2602_APANA, 1, 1, 1),
+};
+
+/* Output Mixer */
+static const struct snd_kcontrol_new ssm260x_output_mixer_controls[] = {
+SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
+SOC_DAPM_SINGLE("HiFi Playback Switch", SSM2602_APANA, 4, 1, 0),
+SOC_DAPM_SINGLE("Mic Sidetone Switch", SSM2602_APANA, 5, 1, 0),
+};
+
+/* Input mux */
+static const struct snd_kcontrol_new ssm2602_input_mux_controls =
+SOC_DAPM_ENUM("Input Select", ssm2602_enum[0]);
+
+static const struct snd_soc_dapm_widget ssm260x_dapm_widgets[] = {
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
+SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
+SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
+
+SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_INPUT("RLINEIN"),
+SND_SOC_DAPM_INPUT("LLINEIN"),
+};
+
+static const struct snd_soc_dapm_widget ssm2602_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Output Mixer", SSM2602_PWR, 4, 1,
+ ssm260x_output_mixer_controls,
+ ARRAY_SIZE(ssm260x_output_mixer_controls)),
+
+SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &ssm2602_input_mux_controls),
+SND_SOC_DAPM_MICBIAS("Mic Bias", SSM2602_PWR, 1, 1),
+
+SND_SOC_DAPM_OUTPUT("LHPOUT"),
+SND_SOC_DAPM_OUTPUT("RHPOUT"),
+SND_SOC_DAPM_INPUT("MICIN"),
+};
+
+static const struct snd_soc_dapm_widget ssm2604_dapm_widgets[] = {
+SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0,
+ ssm260x_output_mixer_controls,
+ ARRAY_SIZE(ssm260x_output_mixer_controls) - 1), /* Last element is the mic */
+};
+
+static const struct snd_soc_dapm_route ssm260x_routes[] = {
+ {"DAC", NULL, "Digital Core Power"},
+ {"ADC", NULL, "Digital Core Power"},
+
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "HiFi Playback Switch", "DAC"},
+
+ {"ROUT", NULL, "Output Mixer"},
+ {"LOUT", NULL, "Output Mixer"},
+
+ {"Line Input", NULL, "LLINEIN"},
+ {"Line Input", NULL, "RLINEIN"},
+};
+
+static const struct snd_soc_dapm_route ssm2602_routes[] = {
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"},
+
+ {"RHPOUT", NULL, "Output Mixer"},
+ {"LHPOUT", NULL, "Output Mixer"},
+
+ {"Input Mux", "Line", "Line Input"},
+ {"Input Mux", "Mic", "Mic Bias"},
+ {"ADC", NULL, "Input Mux"},
+
+ {"Mic Bias", NULL, "MICIN"},
+};
+
+static const struct snd_soc_dapm_route ssm2604_routes[] = {
+ {"ADC", NULL, "Line Input"},
+};
+
+static const unsigned int ssm2602_rates_12288000[] = {
+ 8000, 16000, 32000, 48000, 96000,
+};
+
+static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = {
+ .list = ssm2602_rates_12288000,
+ .count = ARRAY_SIZE(ssm2602_rates_12288000),
+};
+
+static const unsigned int ssm2602_rates_11289600[] = {
+ 8000, 11025, 22050, 44100, 88200,
+};
+
+static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = {
+ .list = ssm2602_rates_11289600,
+ .count = ARRAY_SIZE(ssm2602_rates_11289600),
+};
+
+struct ssm2602_coeff {
+ u32 mclk;
+ u32 rate;
+ u8 srate;
+};
+
+#define SSM2602_COEFF_SRATE(sr, bosr, usb) (((sr) << 2) | ((bosr) << 1) | (usb))
+
+/* codec mclk clock coefficients */
+static const struct ssm2602_coeff ssm2602_coeff_table[] = {
+ /* 48k */
+ {12288000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x0)},
+ {18432000, 48000, SSM2602_COEFF_SRATE(0x0, 0x1, 0x0)},
+ {12000000, 48000, SSM2602_COEFF_SRATE(0x0, 0x0, 0x1)},
+
+ /* 32k */
+ {12288000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x0)},
+ {18432000, 32000, SSM2602_COEFF_SRATE(0x6, 0x1, 0x0)},
+ {12000000, 32000, SSM2602_COEFF_SRATE(0x6, 0x0, 0x1)},
+
+ /* 16k */
+ {12288000, 16000, SSM2602_COEFF_SRATE(0x5, 0x0, 0x0)},
+ {18432000, 16000, SSM2602_COEFF_SRATE(0x5, 0x1, 0x0)},
+ {12000000, 16000, SSM2602_COEFF_SRATE(0xa, 0x0, 0x1)},
+
+ /* 8k */
+ {12288000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x0)},
+ {18432000, 8000, SSM2602_COEFF_SRATE(0x3, 0x1, 0x0)},
+ {11289600, 8000, SSM2602_COEFF_SRATE(0xb, 0x0, 0x0)},
+ {16934400, 8000, SSM2602_COEFF_SRATE(0xb, 0x1, 0x0)},
+ {12000000, 8000, SSM2602_COEFF_SRATE(0x3, 0x0, 0x1)},
+
+ /* 96k */
+ {12288000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x0)},
+ {18432000, 96000, SSM2602_COEFF_SRATE(0x7, 0x1, 0x0)},
+ {12000000, 96000, SSM2602_COEFF_SRATE(0x7, 0x0, 0x1)},
+
+ /* 11.025k */
+ {11289600, 11025, SSM2602_COEFF_SRATE(0xc, 0x0, 0x0)},
+ {16934400, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x0)},
+ {12000000, 11025, SSM2602_COEFF_SRATE(0xc, 0x1, 0x1)},
+
+ /* 22.05k */
+ {11289600, 22050, SSM2602_COEFF_SRATE(0xd, 0x0, 0x0)},
+ {16934400, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x0)},
+ {12000000, 22050, SSM2602_COEFF_SRATE(0xd, 0x1, 0x1)},
+
+ /* 44.1k */
+ {11289600, 44100, SSM2602_COEFF_SRATE(0x8, 0x0, 0x0)},
+ {16934400, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x0)},
+ {12000000, 44100, SSM2602_COEFF_SRATE(0x8, 0x1, 0x1)},
+
+ /* 88.2k */
+ {11289600, 88200, SSM2602_COEFF_SRATE(0xf, 0x0, 0x0)},
+ {16934400, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x0)},
+ {12000000, 88200, SSM2602_COEFF_SRATE(0xf, 0x1, 0x1)},
+};
+
+static inline int ssm2602_get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(ssm2602_coeff_table); i++) {
+ if (ssm2602_coeff_table[i].rate == rate &&
+ ssm2602_coeff_table[i].mclk == mclk)
+ return ssm2602_coeff_table[i].srate;
+ }
+ return -EINVAL;
+}
+
+static int ssm2602_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 ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+ int srate = ssm2602_get_coeff(ssm2602->sysclk, params_rate(params));
+ unsigned int iface;
+
+ if (srate < 0)
+ return srate;
+
+ regmap_write(ssm2602->regmap, SSM2602_SRATE, srate);
+
+ /* bit size */
+ switch (params_width(params)) {
+ case 16:
+ iface = 0x0;
+ break;
+ case 20:
+ iface = 0x4;
+ break;
+ case 24:
+ iface = 0x8;
+ break;
+ case 32:
+ iface = 0xc;
+ break;
+ default:
+ return -EINVAL;
+ }
+ regmap_update_bits(ssm2602->regmap, SSM2602_IFACE,
+ IFACE_AUDIO_DATA_LEN, iface);
+ return 0;
+}
+
+static int ssm2602_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+
+ if (ssm2602->sysclk_constraints) {
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ ssm2602->sysclk_constraints);
+ }
+
+ return 0;
+}
+
+static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(dai->codec);
+
+ if (mute)
+ regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
+ APDIGI_ENABLE_DAC_MUTE,
+ APDIGI_ENABLE_DAC_MUTE);
+ else
+ regmap_update_bits(ssm2602->regmap, SSM2602_APDIGI,
+ APDIGI_ENABLE_DAC_MUTE, 0);
+ return 0;
+}
+
+static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+
+ if (dir == SND_SOC_CLOCK_IN) {
+ if (clk_id != SSM2602_SYSCLK)
+ return -EINVAL;
+
+ switch (freq) {
+ case 12288000:
+ case 18432000:
+ ssm2602->sysclk_constraints = &ssm2602_constraints_12288000;
+ break;
+ case 11289600:
+ case 16934400:
+ ssm2602->sysclk_constraints = &ssm2602_constraints_11289600;
+ break;
+ case 12000000:
+ ssm2602->sysclk_constraints = NULL;
+ break;
+ default:
+ return -EINVAL;
+ }
+ ssm2602->sysclk = freq;
+ } else {
+ unsigned int mask;
+
+ switch (clk_id) {
+ case SSM2602_CLK_CLKOUT:
+ mask = PWR_CLK_OUT_PDN;
+ break;
+ case SSM2602_CLK_XTO:
+ mask = PWR_OSC_PDN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (freq == 0)
+ ssm2602->clk_out_pwr |= mask;
+ else
+ ssm2602->clk_out_pwr &= ~mask;
+
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
+ PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr);
+ }
+
+ return 0;
+}
+
+static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec_dai->codec);
+ unsigned int iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface |= 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0013;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0003;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface */
+ regmap_write(ssm2602->regmap, SSM2602_IFACE, iface);
+ return 0;
+}
+
+static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* vref/mid on, osc and clkout on if enabled */
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
+ PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
+ ssm2602->clk_out_pwr);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* everything off except vref/vmid, */
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
+ PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
+ PWR_CLK_OUT_PDN | PWR_OSC_PDN);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* everything off */
+ regmap_update_bits(ssm2602->regmap, SSM2602_PWR,
+ PWR_POWER_OFF, PWR_POWER_OFF);
+ break;
+
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000)
+
+#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops ssm2602_dai_ops = {
+ .startup = ssm2602_startup,
+ .hw_params = ssm2602_hw_params,
+ .digital_mute = ssm2602_mute,
+ .set_sysclk = ssm2602_set_dai_sysclk,
+ .set_fmt = ssm2602_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver ssm2602_dai = {
+ .name = "ssm2602-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SSM2602_RATES,
+ .formats = SSM2602_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SSM2602_RATES,
+ .formats = SSM2602_FORMATS,},
+ .ops = &ssm2602_dai_ops,
+ .symmetric_rates = 1,
+ .symmetric_samplebits = 1,
+};
+
+static int ssm2602_resume(struct snd_soc_codec *codec)
+{
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+
+ regcache_sync(ssm2602->regmap);
+
+ return 0;
+}
+
+static int ssm2602_codec_probe(struct snd_soc_codec *codec)
+{
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int ret;
+
+ regmap_update_bits(ssm2602->regmap, SSM2602_LOUT1V,
+ LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH);
+ regmap_update_bits(ssm2602->regmap, SSM2602_ROUT1V,
+ ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
+
+ ret = snd_soc_add_codec_controls(codec, ssm2602_snd_controls,
+ ARRAY_SIZE(ssm2602_snd_controls));
+ if (ret)
+ return ret;
+
+ ret = snd_soc_dapm_new_controls(dapm, ssm2602_dapm_widgets,
+ ARRAY_SIZE(ssm2602_dapm_widgets));
+ if (ret)
+ return ret;
+
+ return snd_soc_dapm_add_routes(dapm, ssm2602_routes,
+ ARRAY_SIZE(ssm2602_routes));
+}
+
+static int ssm2604_codec_probe(struct snd_soc_codec *codec)
+{
+ struct snd_soc_dapm_context *dapm = &codec->dapm;
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(dapm, ssm2604_dapm_widgets,
+ ARRAY_SIZE(ssm2604_dapm_widgets));
+ if (ret)
+ return ret;
+
+ return snd_soc_dapm_add_routes(dapm, ssm2604_routes,
+ ARRAY_SIZE(ssm2604_routes));
+}
+
+static int ssm260x_codec_probe(struct snd_soc_codec *codec)
+{
+ struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
+ return ret;
+ }
+
+ /* set the update bits */
+ regmap_update_bits(ssm2602->regmap, SSM2602_LINVOL,
+ LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
+ regmap_update_bits(ssm2602->regmap, SSM2602_RINVOL,
+ RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH);
+ /*select Line in as default input*/
+ regmap_write(ssm2602->regmap, SSM2602_APANA, APANA_SELECT_DAC |
+ APANA_ENABLE_MIC_BOOST);
+
+ switch (ssm2602->type) {
+ case SSM2602:
+ ret = ssm2602_codec_probe(codec);
+ break;
+ case SSM2604:
+ ret = ssm2604_codec_probe(codec);
+ break;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = {
+ .probe = ssm260x_codec_probe,
+ .resume = ssm2602_resume,
+ .set_bias_level = ssm2602_set_bias_level,
+ .suspend_bias_off = true,
+
+ .controls = ssm260x_snd_controls,
+ .num_controls = ARRAY_SIZE(ssm260x_snd_controls),
+ .dapm_widgets = ssm260x_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ssm260x_dapm_widgets),
+ .dapm_routes = ssm260x_routes,
+ .num_dapm_routes = ARRAY_SIZE(ssm260x_routes),
+};
+
+static bool ssm2602_register_volatile(struct device *dev, unsigned int reg)
+{
+ return reg == SSM2602_RESET;
+}
+
+const struct regmap_config ssm2602_regmap_config = {
+ .val_bits = 9,
+ .reg_bits = 7,
+
+ .max_register = SSM2602_RESET,
+ .volatile_reg = ssm2602_register_volatile,
+
+ .cache_type = REGCACHE_RBTREE,
+ .reg_defaults_raw = ssm2602_reg,
+ .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg),
+};
+EXPORT_SYMBOL_GPL(ssm2602_regmap_config);
+
+int ssm2602_probe(struct device *dev, enum ssm2602_type type,
+ struct regmap *regmap)
+{
+ struct ssm2602_priv *ssm2602;
+
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ ssm2602 = devm_kzalloc(dev, sizeof(*ssm2602), GFP_KERNEL);
+ if (ssm2602 == NULL)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, ssm2602);
+ ssm2602->type = type;
+ ssm2602->regmap = regmap;
+
+ return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602,
+ &ssm2602_dai, 1);
+}
+EXPORT_SYMBOL_GPL(ssm2602_probe);
+
+MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver");
+MODULE_AUTHOR("Cliff Cai");
+MODULE_LICENSE("GPL");