--- /dev/null
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author: Arun KS, <arunks@mistralsolutions.com>
+ * Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * 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.
+ *
+ * Notes:
+ * The AIC23 is a driver for a low power stereo audio
+ * codec tlv320aic23
+ *
+ * The machine layer should disable unsupported inputs/outputs by
+ * snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+
+#include "tlv320aic23.h"
+
+/*
+ * AIC23 register cache
+ */
+static const struct reg_default tlv320aic23_reg[] = {
+ { 0, 0x0097 },
+ { 1, 0x0097 },
+ { 2, 0x00F9 },
+ { 3, 0x00F9 },
+ { 4, 0x001A },
+ { 5, 0x0004 },
+ { 6, 0x0007 },
+ { 7, 0x0001 },
+ { 8, 0x0020 },
+ { 9, 0x0000 },
+};
+
+const struct regmap_config tlv320aic23_regmap = {
+ .reg_bits = 7,
+ .val_bits = 9,
+
+ .max_register = TLV320AIC23_RESET,
+ .reg_defaults = tlv320aic23_reg,
+ .num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg),
+ .cache_type = REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL(tlv320aic23_regmap);
+
+static const char *rec_src_text[] = { "Line", "Mic" };
+static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static SOC_ENUM_SINGLE_DECL(rec_src_enum,
+ TLV320AIC23_ANLG, 2, rec_src_text);
+
+static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
+SOC_DAPM_ENUM("Input Select", rec_src_enum);
+
+static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src,
+ TLV320AIC23_ANLG, 2, rec_src_text);
+static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph,
+ TLV320AIC23_DIGT, 1, deemph_text);
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
+static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
+
+static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ u16 val, reg;
+
+ val = (ucontrol->value.integer.value[0] & 0x07);
+
+ /* linear conversion to userspace
+ * 000 = -6db
+ * 001 = -9db
+ * 010 = -12db
+ * 011 = -18db (Min)
+ * 100 = 0db (Max)
+ */
+ val = (val >= 4) ? 4 : (3 - val);
+
+ reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0);
+ snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
+
+ return 0;
+}
+
+static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ u16 val;
+
+ val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0);
+ val = val >> 6;
+ val = (val >= 4) ? 4 : (3 - val);
+ ucontrol->value.integer.value[0] = val;
+ return 0;
+
+}
+
+static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
+ TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
+ SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
+ SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
+ TLV320AIC23_RINVOL, 7, 1, 0),
+ SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
+ TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
+ SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
+ SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
+ SOC_SINGLE_EXT_TLV("Sidetone Volume", TLV320AIC23_ANLG, 6, 4, 0,
+ snd_soc_tlv320aic23_get_volsw,
+ snd_soc_tlv320aic23_put_volsw, sidetone_vol_tlv),
+ SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
+};
+
+/* PGA Mixer controls for Line and Mic switch */
+static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
+ SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+ SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
+ SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
+ SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
+ &tlv320aic23_rec_src_mux_controls),
+ SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
+ &tlv320aic23_output_mixer_controls[0],
+ ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
+ SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
+ SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
+
+ SND_SOC_DAPM_OUTPUT("LHPOUT"),
+ SND_SOC_DAPM_OUTPUT("RHPOUT"),
+ SND_SOC_DAPM_OUTPUT("LOUT"),
+ SND_SOC_DAPM_OUTPUT("ROUT"),
+
+ SND_SOC_DAPM_INPUT("LLINEIN"),
+ SND_SOC_DAPM_INPUT("RLINEIN"),
+
+ SND_SOC_DAPM_INPUT("MICIN"),
+};
+
+static const struct snd_soc_dapm_route tlv320aic23_intercon[] = {
+ /* Output Mixer */
+ {"Output Mixer", "Line Bypass Switch", "Line Input"},
+ {"Output Mixer", "Playback Switch", "DAC"},
+ {"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
+
+ /* Outputs */
+ {"RHPOUT", NULL, "Output Mixer"},
+ {"LHPOUT", NULL, "Output Mixer"},
+ {"LOUT", NULL, "Output Mixer"},
+ {"ROUT", NULL, "Output Mixer"},
+
+ /* Inputs */
+ {"Line Input", "NULL", "LLINEIN"},
+ {"Line Input", "NULL", "RLINEIN"},
+
+ {"Mic Input", "NULL", "MICIN"},
+
+ /* input mux */
+ {"Capture Source", "Line", "Line Input"},
+ {"Capture Source", "Mic", "Mic Input"},
+ {"ADC", NULL, "Capture Source"},
+
+};
+
+/* AIC23 driver data */
+struct aic23 {
+ struct regmap *regmap;
+ int mclk;
+ int requested_adc;
+ int requested_dac;
+};
+
+/*
+ * Common Crystals used
+ * 11.2896 Mhz /128 = *88.2k /192 = 58.8k
+ * 12.0000 Mhz /125 = *96k /136 = 88.235K
+ * 12.2880 Mhz /128 = *96k /192 = 64k
+ * 16.9344 Mhz /128 = 132.3k /192 = *88.2k
+ * 18.4320 Mhz /128 = 144k /192 = *96k
+ */
+
+/*
+ * Normal BOSR 0-256/2 = 128, 1-384/2 = 192
+ * USB BOSR 0-250/2 = 125, 1-272/2 = 136
+ */
+static const int bosr_usb_divisor_table[] = {
+ 128, 125, 192, 136
+};
+#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
+#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
+static const unsigned short sr_valid_mask[] = {
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
+ LOWER_GROUP, /* Usb, bosr - 0*/
+ LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
+ UPPER_GROUP, /* Usb, bosr - 1*/
+};
+/*
+ * Every divisor is a factor of 11*12
+ */
+#define SR_MULT (11*12)
+#define A(x) (SR_MULT/x)
+static const unsigned char sr_adc_mult_table[] = {
+ A(2), A(2), A(12), A(12), 0, 0, A(3), A(1),
+ A(2), A(2), A(11), A(11), 0, 0, 0, A(1)
+};
+static const unsigned char sr_dac_mult_table[] = {
+ A(2), A(12), A(2), A(12), 0, 0, A(3), A(1),
+ A(2), A(11), A(2), A(11), 0, 0, 0, A(1)
+};
+
+static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
+ int dac, int dac_l, int dac_h, int need_dac)
+{
+ if ((adc >= adc_l) && (adc <= adc_h) &&
+ (dac >= dac_l) && (dac <= dac_h)) {
+ int diff_adc = need_adc - adc;
+ int diff_dac = need_dac - dac;
+ return abs(diff_adc) + abs(diff_dac);
+ }
+ return UINT_MAX;
+}
+
+static int find_rate(int mclk, u32 need_adc, u32 need_dac)
+{
+ int i, j;
+ int best_i = -1;
+ int best_j = -1;
+ int best_div = 0;
+ unsigned best_score = UINT_MAX;
+ int adc_l, adc_h, dac_l, dac_h;
+
+ need_adc *= SR_MULT;
+ need_dac *= SR_MULT;
+ /*
+ * rates given are +/- 1/32
+ */
+ adc_l = need_adc - (need_adc >> 5);
+ adc_h = need_adc + (need_adc >> 5);
+ dac_l = need_dac - (need_dac >> 5);
+ dac_h = need_dac + (need_dac >> 5);
+ for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
+ int base = mclk / bosr_usb_divisor_table[i];
+ int mask = sr_valid_mask[i];
+ for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
+ j++, mask >>= 1) {
+ int adc;
+ int dac;
+ int score;
+ if ((mask & 1) == 0)
+ continue;
+ adc = base * sr_adc_mult_table[j];
+ dac = base * sr_dac_mult_table[j];
+ score = get_score(adc, adc_l, adc_h, need_adc,
+ dac, dac_l, dac_h, need_dac);
+ if (best_score > score) {
+ best_score = score;
+ best_i = i;
+ best_j = j;
+ best_div = 0;
+ }
+ score = get_score((adc >> 1), adc_l, adc_h, need_adc,
+ (dac >> 1), dac_l, dac_h, need_dac);
+ /* prefer to have a /2 */
+ if ((score != UINT_MAX) && (best_score >= score)) {
+ best_score = score;
+ best_i = i;
+ best_j = j;
+ best_div = 1;
+ }
+ }
+ }
+ return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
+}
+
+#ifdef DEBUG
+static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
+ u32 *sample_rate_adc, u32 *sample_rate_dac)
+{
+ int src = snd_soc_read(codec, TLV320AIC23_SRATE);
+ int sr = (src >> 2) & 0x0f;
+ int val = (mclk / bosr_usb_divisor_table[src & 3]);
+ int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
+ int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
+ if (src & TLV320AIC23_CLKIN_HALF) {
+ adc >>= 1;
+ dac >>= 1;
+ }
+ *sample_rate_adc = adc;
+ *sample_rate_dac = dac;
+}
+#endif
+
+static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
+ u32 sample_rate_adc, u32 sample_rate_dac)
+{
+ /* Search for the right sample rate */
+ int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
+ if (data < 0) {
+ printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
+ __func__, sample_rate_adc, sample_rate_dac);
+ return -EINVAL;
+ }
+ snd_soc_write(codec, TLV320AIC23_SRATE, data);
+#ifdef DEBUG
+ {
+ u32 adc, dac;
+ get_current_sample_rates(codec, mclk, &adc, &dac);
+ printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
+ adc, dac, data);
+ }
+#endif
+ return 0;
+}
+
+static int tlv320aic23_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;
+ u16 iface_reg;
+ int ret;
+ struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
+ u32 sample_rate_adc = aic23->requested_adc;
+ u32 sample_rate_dac = aic23->requested_dac;
+ u32 sample_rate = params_rate(params);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ aic23->requested_dac = sample_rate_dac = sample_rate;
+ if (!sample_rate_adc)
+ sample_rate_adc = sample_rate;
+ } else {
+ aic23->requested_adc = sample_rate_adc = sample_rate;
+ if (!sample_rate_dac)
+ sample_rate_dac = sample_rate;
+ }
+ ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
+ sample_rate_dac);
+ if (ret < 0)
+ return ret;
+
+ iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
+
+ switch (params_width(params)) {
+ case 16:
+ break;
+ case 20:
+ iface_reg |= (0x01 << 2);
+ break;
+ case 24:
+ iface_reg |= (0x02 << 2);
+ break;
+ case 32:
+ iface_reg |= (0x03 << 2);
+ break;
+ }
+ snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+ return 0;
+}
+
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+
+ /* set active */
+ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001);
+
+ return 0;
+}
+
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
+
+ /* deactivate */
+ if (!snd_soc_codec_is_active(codec)) {
+ udelay(50);
+ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
+ }
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aic23->requested_dac = 0;
+ else
+ aic23->requested_adc = 0;
+}
+
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 reg;
+
+ reg = snd_soc_read(codec, TLV320AIC23_DIGT);
+ if (mute)
+ reg |= TLV320AIC23_DACM_MUTE;
+
+ else
+ reg &= ~TLV320AIC23_DACM_MUTE;
+
+ snd_soc_write(codec, TLV320AIC23_DIGT, reg);
+
+ return 0;
+}
+
+static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface_reg;
+
+ iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface_reg |= TLV320AIC23_MS_MASTER;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ iface_reg &= ~TLV320AIC23_MS_MASTER;
+ break;
+ default:
+ return -EINVAL;
+
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface_reg |= TLV320AIC23_FOR_I2S;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface_reg |= TLV320AIC23_LRP_ON;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface_reg |= TLV320AIC23_FOR_DSP;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface_reg |= TLV320AIC23_FOR_LJUST;
+ break;
+ default:
+ return -EINVAL;
+
+ }
+
+ snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+ return 0;
+}
+
+static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct aic23 *aic23 = snd_soc_dai_get_drvdata(codec_dai);
+ aic23->mclk = freq;
+ return 0;
+}
+
+static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* vref/mid, osc on, dac unmute */
+ reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
+ TLV320AIC23_DAC_OFF);
+ snd_soc_write(codec, TLV320AIC23_PWR, reg);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* everything off except vref/vmid, */
+ snd_soc_write(codec, TLV320AIC23_PWR,
+ reg | TLV320AIC23_CLK_OFF);
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* everything off, dac mute, inactive */
+ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
+ snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff);
+ break;
+ }
+ codec->dapm.bias_level = level;
+ return 0;
+}
+
+#define AIC23_RATES SNDRV_PCM_RATE_8000_96000
+#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops tlv320aic23_dai_ops = {
+ .prepare = tlv320aic23_pcm_prepare,
+ .hw_params = tlv320aic23_hw_params,
+ .shutdown = tlv320aic23_shutdown,
+ .digital_mute = tlv320aic23_mute,
+ .set_fmt = tlv320aic23_set_dai_fmt,
+ .set_sysclk = tlv320aic23_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver tlv320aic23_dai = {
+ .name = "tlv320aic23-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC23_RATES,
+ .formats = AIC23_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AIC23_RATES,
+ .formats = AIC23_FORMATS,},
+ .ops = &tlv320aic23_dai_ops,
+};
+
+static int tlv320aic23_resume(struct snd_soc_codec *codec)
+{
+ struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
+ regcache_mark_dirty(aic23->regmap);
+ regcache_sync(aic23->regmap);
+
+ return 0;
+}
+
+static int tlv320aic23_codec_probe(struct snd_soc_codec *codec)
+{
+ /* Reset codec */
+ snd_soc_write(codec, TLV320AIC23_RESET, 0);
+
+ snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
+
+ /* Unmute input */
+ snd_soc_update_bits(codec, TLV320AIC23_LINVOL,
+ TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
+
+ snd_soc_update_bits(codec, TLV320AIC23_RINVOL,
+ TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
+
+ snd_soc_update_bits(codec, TLV320AIC23_ANLG,
+ TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED,
+ 0);
+
+ /* Default output volume */
+ snd_soc_write(codec, TLV320AIC23_LCHNVOL,
+ TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
+ snd_soc_write(codec, TLV320AIC23_RCHNVOL,
+ TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
+
+ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
+ .probe = tlv320aic23_codec_probe,
+ .resume = tlv320aic23_resume,
+ .set_bias_level = tlv320aic23_set_bias_level,
+ .suspend_bias_off = true,
+
+ .controls = tlv320aic23_snd_controls,
+ .num_controls = ARRAY_SIZE(tlv320aic23_snd_controls),
+ .dapm_widgets = tlv320aic23_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
+ .dapm_routes = tlv320aic23_intercon,
+ .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon),
+};
+
+int tlv320aic23_probe(struct device *dev, struct regmap *regmap)
+{
+ struct aic23 *aic23;
+
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ aic23 = devm_kzalloc(dev, sizeof(struct aic23), GFP_KERNEL);
+ if (aic23 == NULL)
+ return -ENOMEM;
+
+ aic23->regmap = regmap;
+
+ dev_set_drvdata(dev, aic23);
+
+ return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23,
+ &tlv320aic23_dai, 1);
+}
+EXPORT_SYMBOL(tlv320aic23_probe);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
Code Review / kvmfornfv.git / blobdiff