[PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Tomoya MORINAGA]

ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high
resistance to voltage noise. On chip regulator realizes power supply rejection
ratio (PSRR) be over 90dB so more than 50dB is improved than ever. ML26124-01HB/
ML26124-02GD can deliver stable audio performance without being affected by noise
from the power supply circuit and peripheral components. The chip also includes
a composite video signal output, which can be applied to various portable device
 requirements. The ML26124 is realized these functions into very small package
the size is only 2.56mm x 2.46mm therefore can be construct high quality sound
system easily.
ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package.

Signed-off-by: Tomoya MORINAGA <tomoya.rohm@gmail.com>
---
 sound/soc/codecs/Kconfig   |    4 +
 sound/soc/codecs/Makefile  |    2 +
 sound/soc/codecs/ml26124.c |  475 ++++++++++++++++++++++++++++++++++++++++++++
 sound/soc/codecs/ml26124.h |  122 +++++++++++
 4 files changed, 603 insertions(+), 0 deletions(-)
 create mode 100644 sound/soc/codecs/ml26124.c
 create mode 100644 sound/soc/codecs/ml26124.h

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 665d924..ce8e622 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -38,6 +38,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX98095 if I2C
 	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9877 if I2C
+	select SND_SOC_ML26124 if I2C
 	select SND_SOC_PCM3008
 	select SND_SOC_SGTL5000 if I2C
 	select SND_SOC_SN95031 if INTEL_SCU_IPC
@@ -211,6 +212,9 @@ config SND_SOC_MAX98095
 config SND_SOC_MAX9850
 	tristate
 
+config SND_SOC_ML26124
+	tristate
+
 config SND_SOC_PCM3008
        tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 5119a7e..c1353ff 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -24,6 +24,7 @@ snd-soc-l3-objs := l3.o
 snd-soc-max98088-objs := max98088.o
 snd-soc-max98095-objs := max98095.o
 snd-soc-max9850-objs := max9850.o
+snd-soc-ml26124-objs := ml26124.o
 snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-sgtl5000-objs := sgtl5000.o
 snd-soc-alc5623-objs := alc5623.o
@@ -122,6 +123,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(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
+obj-$(CONFIG_SND_SOC_ML26124)   += snd-soc-ml26124.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
 obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
new file mode 100644
index 0000000..ee323fb
--- /dev/null
+++ b/sound/soc/codecs/ml26124.c
@@ -0,0 +1,475 @@
+/*
+ * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd.
+ *
+ * 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.
+ *
+ * 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, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307, USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "ml26124.h"
+
+struct ml26124_priv {
+	struct snd_pcm_substream *substream;
+};
+
+/* ML26124 configuration */
+static const DECLARE_TLV_DB_SCALE(digital_vol, -7150, 50, 0);
+static const DECLARE_TLV_DB_SCALE(eq_band_gain, -7150, 50, 0);
+static const DECLARE_TLV_DB_SCALE(alclvl, -2250, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mingain, -1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(maxgain, -675, 600, 0);
+static const DECLARE_TLV_DB_SCALE(boost_vol, -1200, 75, 0);
+static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0);
+
+static const char * const ml26124_companding[] = {"16bit PCM", "u-law",
+						  "A-law"};
+static const struct soc_enum ml26124_adc_companding_enum
+	= SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding);
+static const struct soc_enum ml26124_dac_companding_enum
+	= SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding);
+
+static const struct snd_kcontrol_new ml26124_snd_controls[] = {
+	SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0,
+			0xff, 1, digital_vol),
+	SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0,
+			0xff, 1, digital_vol),
+	SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0,
+			0x3f, 0, boost_vol),
+	SOC_SINGLE_TLV("EQ Band0 Input Volume", ML26124_EQ_GAIN_BRAND0, 0,
+			0xff, 1, eq_band_gain),
+	SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0,
+			0xff, 1, eq_band_gain),
+	SOC_SINGLE_TLV("EQ Band2 Input Volume", ML26124_EQ_GAIN_BRAND2, 0,
+			0xff, 1, eq_band_gain),
+	SOC_SINGLE_TLV("EQ Band3 Input Volume", ML26124_EQ_GAIN_BRAND3, 0,
+			0xff, 1, eq_band_gain),
+	SOC_SINGLE_TLV("EQ Band4 Input Volume", ML26124_EQ_GAIN_BRAND4, 0,
+			0xff, 1, eq_band_gain),
+	SOC_SINGLE_TLV("ALC Target Level", ML26124_ALC_TARGET_LEV, 0,
+			0xf, 1, alclvl),
+	SOC_SINGLE_TLV("ALC Min Input Volume", ML26124_ALC_MAXMIN_GAIN, 0,
+			7, 0, mingain),
+	SOC_SINGLE_TLV("ALC MAX Input Volume", ML26124_ALC_MAXMIN_GAIN, 4,
+			7, 1, maxgain),
+	SOC_SINGLE_TLV("Playback Limitter Min Input Volume",
+			ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain),
+	SOC_SINGLE_TLV("Playback Limitter Max Input Volume",
+			ML26124_PL_MAXMIN_GAIN, 4, 7, 1, maxgain),
+	SOC_SINGLE_TLV("Playback Boost Volume", ML26124_PLYBAK_BOST_VOL, 0,
+			0x3f, 0, boost_vol),
+	SOC_SINGLE("DC High Pass Filter Switch", ML26124_FILTER_EN, 0, 1, 0),
+	SOC_SINGLE("NOISE High Pass Filter Switch", ML26124_FILTER_EN, 1, 1, 0),
+	SOC_SINGLE("EQ BAND0 Switch", ML26124_FILTER_EN, 2, 1, 0),
+	SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0),
+	SOC_SINGLE("EQ BAND2 Switch", ML26124_FILTER_EN, 4, 1, 0),
+	SOC_SINGLE("EQ BAND3 Switch", ML26124_FILTER_EN, 5, 1, 0),
+	SOC_SINGLE("EQ BAND4 Switch", ML26124_FILTER_EN, 6, 1, 0),
+	SOC_SINGLE("Play Limitter Switch", ML26124_DVOL_CTL, 0, 1, 0),
+	SOC_SINGLE("Capture Limitter Switch", ML26124_DVOL_CTL, 1, 1, 0),
+	SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0),
+	SOC_SINGLE("Digital Volume Switch", ML26124_DVOL_CTL, 4, 1, 0),
+	SOC_ENUM("DAC Companding", ml26124_dac_companding_enum),
+	SOC_ENUM("ADC Companding", ml26124_adc_companding_enum),
+};
+
+static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0),
+	SOC_DAPM_SINGLE("Line in Switch", ML26124_SPK_AMP_OUT, 3, 1, 0),
+	SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0),
+};
+
+/* Input mux */
+static const char * const ml26124_input_select[] = {"Analog MIC in",
+						    "Digital MIC in"};
+
+static const struct soc_enum ml26124_insel_enum =
+	SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 1, ml26124_input_select);
+
+static const struct snd_kcontrol_new ml26124_input_mux_controls =
+	SOC_DAPM_ENUM("Input Select", ml26124_insel_enum);
+
+static const struct snd_kcontrol_new ml26124_line_control =
+	SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0);
+
+static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = {
+	SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0,
+			   &ml26124_output_mixer_controls[0],
+			   ARRAY_SIZE(ml26124_output_mixer_controls)),
+	SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0),
+	SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0),
+	SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0),
+	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+			  &ml26124_input_mux_controls),
+	SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
+			     &ml26124_line_control),
+	SND_SOC_DAPM_INPUT("VIDEOIN"),
+	SND_SOC_DAPM_INPUT("MDIN"),
+	SND_SOC_DAPM_INPUT("MIN"),
+	SND_SOC_DAPM_INPUT("LIN"),
+	SND_SOC_DAPM_OUTPUT("VIDEOOUT"),
+	SND_SOC_DAPM_OUTPUT("SPOUT"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+};
+
+static const struct snd_soc_dapm_route ml26124_intercon[] = {
+	/* output mixer */
+	{"Output Mixer", "PGA Switch", "PGA"},
+	{"Output Mixer", "DAC Switch", "DAC"},
+	{"Output Mixer", NULL, "Line in Switch"},
+	/* outputs */
+	{"SPOUT", NULL, "Output Mixer"},
+	{"LOUT", NULL, "Output Mixer"},
+	{"SPOUT", NULL, "Output Mixer"},
+
+	/* input mux */
+	{"Input Mux", "Analog MIC in", "MICBIAS"},
+	{"Input Mux", "Digital MIC in", "MICBIAS"},
+	{"ADC", NULL, "Input Mux"},
+
+	/* inputs */
+	{"DAC", NULL, "MDIN"},
+	{"PGA", NULL, "MIN"},
+	{"SPOUT", NULL, "LIN"},
+};
+
+static int ml26124_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *hw_params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	switch (params_rate(hw_params)) {
+	case 16000:
+		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 3);
+		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c);
+		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0);
+		snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x20);
+		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00);
+		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04);
+		break;
+	case 32000:
+		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 6);
+		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c);
+		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0);
+		snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x20);
+		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00);
+		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04);
+		break;
+	case 48000:
+		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 0x08);
+		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c);
+		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0x00);
+		snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x30);
+		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00);
+		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04);
+		break;
+	default:
+		pr_err("%s:this rate is no support for ml26124\n", __func__);
+		return -EINVAL;
+	}
+
+	/* Record/Playback Running Control Register */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x2, 0x2);
+	else
+		snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x1, 0x1);
+
+	return 0;
+}
+
+static int snd_card_ml7213i2s_hw_free(struct snd_pcm_substream *substream,
+				      struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	/* soft reset assert */
+	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);
+
+	/* Stop  Record/Playback Running */
+	snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0);
+
+	return 0;
+}
+
+#define DVOL_CTL_DVMUTE_ON		BIT(4)	/* Digital volume MUTE On */
+#define DVOL_CTL_DVMUTE_OFF		0	/* Digital volume MUTE Off */
+
+static int ml26124_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	if (mute)
+		snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4),
+				    DVOL_CTL_DVMUTE_ON);
+	else
+		snd_soc_update_bits(codec, ML26124_DVOL_CTL, BIT(4),
+				    DVOL_CTL_DVMUTE_OFF);
+	return 0;
+}
+
+#define ML26124_SAI_NO_DELAY	BIT(1)
+#define ML26124_SAI_FRAME_SYNC	(BIT(5) | BIT(0))
+
+
+static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	unsigned char mode;
+	unsigned char mask = ML26124_SAI_NO_DELAY | ML26124_SAI_FRAME_SYNC;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		mode = 1;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		mode = 0;
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, ML26124_SAI_MODE_SEL, BIT(0), mode);
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* SAI configuration */
+	snd_soc_update_bits(codec, ML26124_SAI_TRANS_CTL, mask, mask);
+	snd_soc_update_bits(codec, ML26124_SAI_RCV_CTL, mask, mask);
+
+	return 0;
+}
+
+#define ML26134_CACHESIZE 79
+static const u16 ml26124_reg[ML26134_CACHESIZE] = {
+	0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
+	0x0, 0x0, 0x0,				/* System Control */
+	0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,	/* Power Management */
+	0x4,					/* Analog Reference Control */
+	0x10, 0x0, 0x33, 0x0, 0x0,	/* Input/Output Amplifier Control */
+	0x0, 0x0, 0x1,				/* Analog Path Contorl */
+	0x0, 0x0, 0x0,				/* Audio Interface Control */
+	0x1, 0x0, 0x0, 0xff, 0xff, 0x10,	/* DSP Control */
+	0x7, 0x7, 0x7, 0x7, 0x7,		/* DSP Control */
+	0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,	/* DSP Control */
+	0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,	/* DSP Control */
+	0x0, 0x0, 0x0,					/* DSP Control */
+	0x0, 0x2, 0x3, 0x0, 0xb, 0x70, 0x0, 0x0,	/* ALC Control */
+	0x4, 0x5, 0xd, 0x70, 0x10, 0x0,		/* Playback Limiter Control */
+	0x1, 0x1, 0x1				/* Video Amplifer Control */
+};
+
+#define ML26124_VMID	BIT(0)
+
+static int ml26124_set_bias_level(struct snd_soc_codec *codec,
+		enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* VMID ON */
+		snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
+				    ML26124_VMID, ML26124_VMID);
+	case SND_SOC_BIAS_PREPARE:
+	case SND_SOC_BIAS_STANDBY:
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* VMID OFF */
+		snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
+				    ML26124_VMID, 0);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define ML26124_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)
+#define ML26124_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+
+static const struct snd_soc_dai_ops ml26124_dai_ops = {
+	.hw_params	= ml26124_hw_params,
+	.hw_free	= snd_card_ml7213i2s_hw_free,
+	.digital_mute	= ml26124_mute,
+	.set_fmt	= ml26124_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver ml26124_dai = {
+	.name = "ml26124-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = ML26124_RATES,
+		.formats = ML26124_FORMATS,},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = ML26124_RATES,
+		.formats = ML26124_FORMATS,},
+	.ops = &ml26124_dai_ops,
+	.symmetric_rates = 1,
+};
+
+#ifdef CONFIG_PM
+static int ml26124_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	ml26124_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int ml26124_resume(struct snd_soc_codec *codec)
+{
+	ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+#else
+#define ml26124_suspend NULL
+#define ml26124_resume NULL
+#endif
+
+static int ml26124_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	/* Software Reset */
+	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);
+	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0);
+
+	ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	snd_soc_add_controls(codec, ml26124_snd_controls,
+			     ARRAY_SIZE(ml26124_snd_controls));
+
+	return 0;
+}
+
+/* power down chip */
+#define REG_CACHE_SIZE		0x79
+static struct snd_soc_codec_driver soc_codec_dev_ml26124 = {
+	.probe =	ml26124_probe,
+	.suspend =	ml26124_suspend,
+	.resume =	ml26124_resume,
+	.set_bias_level = ml26124_set_bias_level,
+	.reg_cache_size = REG_CACHE_SIZE,
+	.reg_word_size = sizeof(u8),
+	.reg_cache_default = ml26124_reg,
+	.dapm_widgets = ml26124_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets),
+	.dapm_routes = ml26124_intercon,
+	.num_dapm_routes = ARRAY_SIZE(ml26124_intercon),
+};
+
+static __devinit int ml26124_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct ml26124_priv *priv;
+	int ret;
+
+	priv = kzalloc(sizeof(*priv), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, priv);
+
+	ret = snd_soc_register_codec(&i2c->dev,
+			&soc_codec_dev_ml26124, &ml26124_dai, 1);
+	if (ret < 0)
+		kfree(priv);
+
+	return ret;
+}
+
+static __devexit int ml26124_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	kfree(i2c_get_clientdata(client));
+	return 0;
+}
+
+static const struct i2c_device_id ml26124_i2c_id[] = {
+	{ "ml26124", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ml26124_i2c_id);
+
+static struct i2c_driver ml26124_i2c_driver = {
+	.driver = {
+		.name = "ml26124",
+		.owner = THIS_MODULE,
+	},
+	.probe = ml26124_i2c_probe,
+	.remove = __devexit_p(ml26124_i2c_remove),
+	.id_table = ml26124_i2c_id,
+};
+
+static int __init ml26124_modinit(void)
+{
+	int ret = 0;
+
+	ret = i2c_add_driver(&ml26124_i2c_driver);
+	if (ret != 0)
+		pr_err("Failed to register ML26124 I2C driver: %d\n", ret);
+
+	return ret;
+}
+module_init(ml26124_modinit);
+
+static void __exit ml26124_exit(void)
+{
+	i2c_del_driver(&ml26124_i2c_driver);
+}
+module_exit(ml26124_exit);
+
+MODULE_AUTHOR("Tomoya MORINAGA <tomoya-linux@dsn.lapis-semi.com>");
+MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ml26124.h b/sound/soc/codecs/ml26124.h
new file mode 100644
index 0000000..d30b530
--- /dev/null
+++ b/sound/soc/codecs/ml26124.h
@@ -0,0 +1,122 @@
+/*
+ * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd.
+ *
+ * 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.
+ *
+ * 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, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307, USA.
+ */
+
+#ifndef ML26124_H
+#define ML26124_H
+
+/* Clock Control Register */
+#define ML26124_SMPLING_RATE		0x00
+#define ML26124_PLLNL			0x02
+#define ML26124_PLLNH			0x04
+#define ML26124_PLLML			0x06
+#define ML26124_PLLMH			0x08
+#define ML26124_PLLDIV			0x0a
+#define ML26124_CLK_EN			0x0c
+#define ML26124_CLK_CTL			0x0e
+
+/* System Control Register */
+#define ML26124_SW_RST			0x10
+#define ML26124_REC_PLYBAK_RUN		0x12
+#define ML26124_MIC_TIM			0x14
+
+/* Power Mnagement Register */
+#define ML26124_PW_REF_PW_MNG		0x20
+#define ML26124_PW_IN_PW_MNG		0x22
+#define ML26124_PW_DAC_PW_MNG		0x24
+#define ML26124_PW_SPAMP_PW_MNG		0x26
+#define ML26124_PW_LOUT_PW_MNG		0x28
+#define ML26124_PW_VOUT_PW_MNG		0x2a
+#define ML26124_PW_ZCCMP_PW_MNG		0x2e
+
+/* Analog Reference Control Register */
+#define ML26124_PW_MICBIAS_VOL		0x30
+
+/* Input/Output Amplifier Control Register */
+#define ML26124_PW_MIC_IN_VOL		0x32
+#define ML26124_PW_MIC_BOST_VOL		0x38
+#define ML26124_PW_SPK_AMP_VOL		0x3a
+#define ML26124_PW_AMP_VOL_FUNC		0x48
+#define ML26124_PW_AMP_VOL_FADE		0x4a
+
+/* Analog Path Control Register */
+#define ML26124_SPK_AMP_OUT		0x54
+#define ML26124_MIC_IF_CTL		0x5a
+#define ML26124_MIC_SELECT		0xe8
+
+/* Audio Interface Control Register */
+#define ML26124_SAI_TRANS_CTL		0x60
+#define ML26124_SAI_RCV_CTL		0x62
+#define ML26124_SAI_MODE_SEL		0x64
+
+/* DSP Control Register */
+#define ML26124_FILTER_EN		0x66
+#define ML26124_DVOL_CTL		0x68
+#define ML26124_MIXER_VOL_CTL		0x6a
+#define ML26124_RECORD_DIG_VOL		0x6c
+#define ML26124_PLBAK_DIG_VOL		0x70
+#define ML26124_DIGI_BOOST_VOL		0x72
+#define ML26124_EQ_GAIN_BRAND0		0x74
+#define ML26124_EQ_GAIN_BRAND1		0x76
+#define ML26124_EQ_GAIN_BRAND2		0x78
+#define ML26124_EQ_GAIN_BRAND3		0x7a
+#define ML26124_EQ_GAIN_BRAND4		0x7c
+#define ML26124_HPF2_CUTOFF		0x7e
+#define ML26124_EQBRAND0_F0L		0x80
+#define ML26124_EQBRAND0_F0H		0x82
+#define ML26124_EQBRAND0_F1L		0x84
+#define ML26124_EQBRAND0_F1H		0x86
+#define ML26124_EQBRAND1_F0L		0x88
+#define ML26124_EQBRAND1_F0H		0x8a
+#define ML26124_EQBRAND1_F1L		0x8c
+#define ML26124_EQBRAND1_F1H		0x8e
+#define ML26124_EQBRAND2_F0L		0x90
+#define ML26124_EQBRAND2_F0H		0x92
+#define ML26124_EQBRAND2_F1L		0x94
+#define ML26124_EQBRAND2_F1H		0x96
+#define ML26124_EQBRAND3_F0L		0x98
+#define ML26124_EQBRAND3_F0H		0x9a
+#define ML26124_EQBRAND3_F1L		0x9c
+#define ML26124_EQBRAND3_F1H		0x9e
+#define ML26124_EQBRAND4_F0L		0xa0
+#define ML26124_EQBRAND4_F0H		0xa2
+#define ML26124_EQBRAND4_F1L		0xa4
+#define ML26124_EQBRAND4_F1H		0xa6
+
+/* ALC Control Register */
+#define ML26124_ALC_MODE		0xb0
+#define ML26124_ALC_ATTACK_TIM		0xb2
+#define ML26124_ALC_DECAY_TIM		0xb4
+#define ML26124_ALC_HOLD_TIM		0xb6
+#define ML26124_ALC_TARGET_LEV		0xb8
+#define ML26124_ALC_MAXMIN_GAIN		0xba
+#define ML26124_NOIS_GATE_THRSH		0xbc
+#define ML26124_ALC_ZERO_TIMOUT		0xbe
+
+/* Playback Limiter Control Register */
+#define ML26124_PL_ATTACKTIME		0xc0
+#define ML26124_PL_DECAYTIME		0xc2
+#define ML26124_PL_TARGETTIME		0xc4
+#define ML26124_PL_MAXMIN_GAIN		0xc6
+#define ML26124_PLYBAK_BOST_VOL		0xc8
+#define ML26124_PL_0CROSS_TIMOUT	0xca
+
+/* Video Amplifer Control Register */
+#define ML26124_VIDEO_AMP_GAIN_CTL	0xd0
+#define ML26124_VIDEO_AMP_SETUP1	0xd2
+#define ML26124_VIDEO_AMP_CTL2		0xd4
+
+#endif
-- 
1.7.4.4

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Lars-Peter Clausen]

On 11/28/2011 11:05 AM, Tomoya MORINAGA wrote:
> ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high
> resistance to voltage noise. On chip regulator realizes power supply rejection
> ratio (PSRR) be over 90dB so more than 50dB is improved than ever. ML26124-01HB/
> ML26124-02GD can deliver stable audio performance without being affected by noise
> from the power supply circuit and peripheral components. The chip also includes
> a composite video signal output, which can be applied to various portable device
>  requirements. The ML26124 is realized these functions into very small package
> the size is only 2.56mm x 2.46mm therefore can be construct high quality sound
> system easily.
> ML26124-01HB is 25pin WCSP package; ML26124-02GD is 32pin WQFN package.
> 
> Signed-off-by: Tomoya MORINAGA <tomoya.rohm@gmail.com>
> ---
>  sound/soc/codecs/Kconfig   |    4 +
>  sound/soc/codecs/Makefile  |    2 +
>  sound/soc/codecs/ml26124.c |  475 ++++++++++++++++++++++++++++++++++++++++++++
>  sound/soc/codecs/ml26124.h |  122 +++++++++++
>  4 files changed, 603 insertions(+), 0 deletions(-)
>  create mode 100644 sound/soc/codecs/ml26124.c
>  create mode 100644 sound/soc/codecs/ml26124.h
> 
> diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
> index 665d924..ce8e622 100644
> --- a/sound/soc/codecs/Kconfig
> +++ b/sound/soc/codecs/Kconfig
> @@ -38,6 +38,7 @@ config SND_SOC_ALL_CODECS
>  	select SND_SOC_MAX98095 if I2C
>  	select SND_SOC_MAX9850 if I2C
>  	select SND_SOC_MAX9877 if I2C
> +	select SND_SOC_ML26124 if I2C
>  	select SND_SOC_PCM3008
>  	select SND_SOC_SGTL5000 if I2C
>  	select SND_SOC_SN95031 if INTEL_SCU_IPC
> @@ -211,6 +212,9 @@ config SND_SOC_MAX98095
>  config SND_SOC_MAX9850
>  	tristate
>  
> +config SND_SOC_ML26124
> +	tristate
> +
>  config SND_SOC_PCM3008
>         tristate
>  
> diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
> index 5119a7e..c1353ff 100644
> --- a/sound/soc/codecs/Makefile
> +++ b/sound/soc/codecs/Makefile
> @@ -24,6 +24,7 @@ snd-soc-l3-objs := l3.o
>  snd-soc-max98088-objs := max98088.o
>  snd-soc-max98095-objs := max98095.o
>  snd-soc-max9850-objs := max9850.o
> +snd-soc-ml26124-objs := ml26124.o
>  snd-soc-pcm3008-objs := pcm3008.o
>  snd-soc-sgtl5000-objs := sgtl5000.o
>  snd-soc-alc5623-objs := alc5623.o
> @@ -122,6 +123,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
>  obj-$(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
> +obj-$(CONFIG_SND_SOC_ML26124)   += snd-soc-ml26124.o
>  obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
>  obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
>  obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
> diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
> new file mode 100644
> index 0000000..ee323fb
> --- /dev/null
> +++ b/sound/soc/codecs/ml26124.c
> @@ -0,0 +1,475 @@
> +/*
> + * Copyright (C) 2011 LAPIS Semiconductor Co., Ltd.
> + *
> + * 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.
> + *
> + * 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, write to the Free Software
> + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307, USA.
> + */
> +
> +#include <linux/module.h>
> +#include <linux/moduleparam.h>
> +#include <linux/init.h>
> +#include <linux/delay.h>
> +#include <linux/pm.h>
> +#include <linux/i2c.h>
> +#include <linux/slab.h>
> +#include <linux/platform_device.h>
> +#include <sound/core.h>
> +#include <sound/pcm.h>
> +#include <sound/pcm_params.h>
> +#include <sound/soc.h>
> +#include <sound/tlv.h>
> +#include "ml26124.h"
> +
> +struct ml26124_priv {
> +	struct snd_pcm_substream *substream;

Is substream actually ever used?

> +};
> +
[...]
> +
> +static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = {
> +	SOC_DAPM_SINGLE("DAC Switch", ML26124_SPK_AMP_OUT, 1, 1, 0),
> +	SOC_DAPM_SINGLE("Line in Switch", ML26124_SPK_AMP_OUT, 3, 1, 0),
> +	SOC_DAPM_SINGLE("PGA Switch", ML26124_SPK_AMP_OUT, 5, 1, 0),
> +};
> +
> +/* Input mux */
> +static const char * const ml26124_input_select[] = {"Analog MIC in",
> +						    "Digital MIC in"};
> +
> +static const struct soc_enum ml26124_insel_enum =
> +	SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 1, ml26124_input_select);
> +
> +static const struct snd_kcontrol_new ml26124_input_mux_controls =
> +	SOC_DAPM_ENUM("Input Select", ml26124_insel_enum);
> +
> +static const struct snd_kcontrol_new ml26124_line_control =
> +	SOC_DAPM_SINGLE("Switch", ML26124_PW_LOUT_PW_MNG, 1, 1, 0);
> +
> +static const struct snd_soc_dapm_widget ml26124_dapm_widgets[] = {
> +	SND_SOC_DAPM_SUPPLY("MCLK", ML26124_CLK_EN, 0, 0, NULL, 0),
> +	SND_SOC_DAPM_SUPPLY("PLL", ML26124_CLK_EN, 1, 0, NULL, 0),
> +	SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 0, 0, NULL, 0),
> +	SND_SOC_DAPM_MIXER("Output Mixer", ML26124_PW_SPAMP_PW_MNG, 0, 0,
> +			   &ml26124_output_mixer_controls[0],
> +			   ARRAY_SIZE(ml26124_output_mixer_controls)),
> +	SND_SOC_DAPM_DAC("DAC", "Playback", ML26124_PW_DAC_PW_MNG, 1, 0),
> +	SND_SOC_DAPM_ADC("ADC", "Capture", ML26124_PW_IN_PW_MNG, 1, 0),
> +	SND_SOC_DAPM_PGA("PGA", ML26124_PW_IN_PW_MNG, 3, 0, NULL, 0),
> +	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
> +			  &ml26124_input_mux_controls),
> +	SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0,
> +			     &ml26124_line_control),
> +	SND_SOC_DAPM_INPUT("VIDEOIN"),
> +	SND_SOC_DAPM_INPUT("MDIN"),
> +	SND_SOC_DAPM_INPUT("MIN"),
> +	SND_SOC_DAPM_INPUT("LIN"),
> +	SND_SOC_DAPM_OUTPUT("VIDEOOUT"),
> +	SND_SOC_DAPM_OUTPUT("SPOUT"),
> +	SND_SOC_DAPM_OUTPUT("LOUT"),
> +};
> +
> +static const struct snd_soc_dapm_route ml26124_intercon[] = {
> +	/* output mixer */
> +	{"Output Mixer", "PGA Switch", "PGA"},
> +	{"Output Mixer", "DAC Switch", "DAC"},
> +	{"Output Mixer", NULL, "Line in Switch"},
> +	/* outputs */
> +	{"SPOUT", NULL, "Output Mixer"},
> +	{"LOUT", NULL, "Output Mixer"},
> +	{"SPOUT", NULL, "Output Mixer"},
> +
> +	/* input mux */
> +	{"Input Mux", "Analog MIC in", "MICBIAS"},
> +	{"Input Mux", "Digital MIC in", "MICBIAS"},
> +	{"ADC", NULL, "Input Mux"},
> +
> +	/* inputs */
> +	{"DAC", NULL, "MDIN"},
> +	{"PGA", NULL, "MIN"},
> +	{"SPOUT", NULL, "LIN"},

The routing looks a bit incomplete. There are quite a few dapm widgets which
are not used. E.g. MCLK, PLL, so they will never be enabled, which I would
assume would prevent the device from working. Also none of the inputs is
routed to the ADC, which seems a bit strange.

> +};
> +
[...]
> +static int snd_card_ml7213i2s_hw_free(struct snd_pcm_substream *substream,
> +				      struct snd_soc_dai *dai)

The name of the function doesn't really match the drivers name.

> +{
> +	struct snd_soc_codec *codec = dai->codec;
> +
> +	/* soft reset assert */
> +	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);
> +
> +	/* Stop  Record/Playback Running */
> +	snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0);

I would expect that this to cause problems with concurrent capture/playback
and one of them being stopped.

> +
> +	return 0;
> +}
> +
[...]
> +
> +static int ml26124_probe(struct snd_soc_codec *codec)
> +{
> +	int ret;
> +
> +	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
> +	if (ret < 0) {
> +		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
> +		return ret;
> +	}
> +
> +	/* Software Reset */
> +	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);
> +	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0);
> +
> +	ml26124_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
> +
> +	snd_soc_add_controls(codec, ml26124_snd_controls,
> +			     ARRAY_SIZE(ml26124_snd_controls));

Controls should use table based setup as well.

> +
> +	return 0;
> +}
> +
> +/* power down chip */
> +#define REG_CACHE_SIZE		0x79

You've defined ML26134_CACHESIZE with the same value above.

> +static struct snd_soc_codec_driver soc_codec_dev_ml26124 = {
> +	.probe =	ml26124_probe,
> +	.suspend =	ml26124_suspend,
> +	.resume =	ml26124_resume,
> +	.set_bias_level = ml26124_set_bias_level,
> +	.reg_cache_size = REG_CACHE_SIZE,
> +	.reg_word_size = sizeof(u8),
> +	.reg_cache_default = ml26124_reg,
> +	.dapm_widgets = ml26124_dapm_widgets,
> +	.num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets),
> +	.dapm_routes = ml26124_intercon,
> +	.num_dapm_routes = ARRAY_SIZE(ml26124_intercon),
> +};
> +
[...]

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Mark Brown]

On Mon, Nov 28, 2011 at 07:05:55PM +0900, Tomoya MORINAGA wrote:
> ML26124-01HB/ML26124-02GD is 16bit monaural audio CODEC which has high
> resistance to voltage noise. On chip regulator realizes power supply rejection
> ratio (PSRR) be over 90dB so more than 50dB is improved than ever. ML26124-01HB/

To repeat what I said last time please fix the word wrapping in your
commit message.

> +struct ml26124_priv {
> +	struct snd_pcm_substream *substream;
> +};

I can't see this being referenced anywhere in the code.

> +/* ML26124 configuration */
> +static const DECLARE_TLV_DB_SCALE(digital_vol, -7150, 50, 0);
> +static const DECLARE_TLV_DB_SCALE(eq_band_gain, -7150, 50, 0);

These are identical.

> +	SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0,
> +			0xff, 1, eq_band_gain),

> +	SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0),

These should mathc.

> +	SOC_SINGLE("Play Limitter Switch", ML26124_DVOL_CTL, 0, 1, 0),
> +	SOC_SINGLE("Capture Limitter Switch", ML26124_DVOL_CTL, 1, 1, 0),

Limiter.

> +	SOC_SINGLE("Digital Volume Switch", ML26124_DVOL_CTL, 4, 1, 0),

Volume Switch makes no sense.

> +	SOC_DAPM_SINGLE("Line in Switch", ML26124_SPK_AMP_OUT, 3, 1, 0),

Line In.

> +       case 16000:
> +               snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, 3);
> +               snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, 0x0c);
> +               snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, 0);
> +               snd_soc_update_bits(codec, ML26124_PLLML, 0xff, 0x20);
> +               snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, 0x00);
> +               snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, 0x04);
> +               break;

You're configuring the PLL here...  I assume there's some fixed clock
rate that your driver supports, you should at least say what it is.

> +static int snd_card_ml7213i2s_hw_free(struct snd_pcm_substream *substream,
> +				      struct snd_soc_dai *dai)
> +{
> +	struct snd_soc_codec *codec = dai->codec;
> +
> +	/* soft reset assert */
> +	snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1);

If you need to do this for some reason I'd expect it to be in the bias
configuration.  Apart from anything else this will run for both playback
and capture streams.  I'm also not seeing anything that restores the
register cache later on.

> +	/* Stop  Record/Playback Running */
> +	snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, 0x3, 0);

This seems a bit odd immediately after the device was reset...

> +#define ML26134_CACHESIZE 79
> +static const u16 ml26124_reg[ML26134_CACHESIZE] = {

As previously mentioned you should use regmap.

> +static int ml26124_set_bias_level(struct snd_soc_codec *codec,
> +		enum snd_soc_bias_level level)
> +{
> +	switch (level) {
> +	case SND_SOC_BIAS_ON:
> +		/* VMID ON */
> +		snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
> +				    ML26124_VMID, ML26124_VMID);
> +	case SND_SOC_BIAS_PREPARE:
> +	case SND_SOC_BIAS_STANDBY:
> +		break;
> +	case SND_SOC_BIAS_OFF:
> +		/* VMID OFF */
> +		snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
> +				    ML26124_VMID, 0);
> +		break;

I'd be very surprised if this works well - you're powering up VMID
after DAPM has finished and there's no delay at all for bringing up
VMID.

> +#define ML26124_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)

This doesn't match the set of rates for which you're configuring magic
values at all.

> +	priv = kzalloc(sizeof(*priv), GFP_KERNEL);
> +	if (priv == NULL)
> +		return -ENOMEM;

devm_kzalloc().

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Tomoya MORINAGA]

2011/11/28 Mark Brown <broonie@opensource.wolfsonmicro.com>:>> +
SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0),
> These should mathc.
I couldn't understand above.
Let me know this meaning.

Thanks

Tomoya
ROHM Co., Ltd.

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Tomoya MORINAGA]

2011/11/28 Lars-Peter Clausen <lars@metafoo.de>:

> Also none of the inputs is> routed to the ADC, which seems a bit strange.
This same as my referencing "wm8731.c".Could you give me more information?

Thanks,
Tomoya
ROHM Co., Ltd.

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Mark Brown]

On Tue, Nov 29, 2011 at 09:14:17PM +0900, Tomoya MORINAGA wrote:
> 2011/11/28 Mark Brown <broonie@opensource.wolfsonmicro.com>:>> +

> > These should mathc.
> I couldn't understand above.
> Let me know this meaning.

Match.

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Lars-Peter Clausen]

On 11/29/2011 01:17 PM, Tomoya MORINAGA wrote:
> 2011/11/28 Lars-Peter Clausen <lars@metafoo.de>:
> 
>> Also none of the inputs is> routed to the ADC, which seems a bit strange.
> This same as my referencing "wm8731.c".Could you give me more information?
> 

The DAPM routes are used to do power management based on active path and for
this the routes should more or less represent the routing of internal audio
signals in your CODEC. I would expect the analog signals to go to the ADC on
the CODEC. And if you don't connect them in your DAPM routings ASoC will
probably not power-up the ADC since there are no active source signals.

Also this doesn't make much sense:
+	{"Input Mux", "Analog MIC in", "MICBIAS"},
+	{"Input Mux", "Digital MIC in", "MICBIAS"},

You have mux, but both possible settings of the mux connect to the same
source, which pretty much makes the mux useless.

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Mark Brown]

On Tue, Nov 29, 2011 at 03:38:00PM +0100, Lars-Peter Clausen wrote:

> Also this doesn't make much sense:
> +	{"Input Mux", "Analog MIC in", "MICBIAS"},
> +	{"Input Mux", "Digital MIC in", "MICBIAS"},

> You have mux, but both possible settings of the mux connect to the same
> source, which pretty much makes the mux useless.

Plus the fact that the MICBIAS is unlikely to be directly connected to
either input within the CODEC, it will almost certainly be connected by
circuitry on the board and therefore mapped by the machine driver for
the system.

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Tomoya MORINAGA]

2011/11/29 Mark Brown <broonie@opensource.wolfsonmicro.com>:> > These
should mathc.
> Match.

Sorry, still I couldn't understand.
What does your "match" mean for anything?

Thanks

------
ROHM Co., Ltd.
tomoya

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Takashi Iwai]

At Thu, 1 Dec 2011 16:34:14 +0900,
Tomoya MORINAGA wrote:
> 
> 2011/11/29 Mark Brown <broonie@opensource.wolfsonmicro.com>:> > These
> should mathc.
> > Match.
> 
> Sorry, still I couldn't understand.
> What does your "match" mean for anything?

You need to have read Mark's comment carefully.  He quoted another
line before that.

================================================================
> +	SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0,
> +			0xff, 1, eq_band_gain),

> +	SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0),

These should mathc.
================================================================

So, you have two inconsistent names for EQ band1.


Takashi

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Tomoya MORINAGA]

2011/11/29 Lars-Peter Clausen <lars@metafoo.de>:
>
> Also this doesn't make much sense:
> +       {"Input Mux", "Analog MIC in", "MICBIAS"},
> +       {"Input Mux", "Digital MIC in", "MICBIAS"},
>
> You have mux, but both possible settings of the mux connect to the same
> source, which pretty much makes the mux useless.
>

Yes, you are right.
This setting was miss-settings.
This MICBIAS supplies to external MIC as reference voltage.
So, I think MICBIAS shouldn't appear at this setting.
I show modified settings.
       /* input */
       {"Input Mux", "Analog MIC SingleEnded in", "MIN"},
       {"Input Mux", "Analog MIC Differential in", "MIN"},
       {"Input Mux", "Digital MIC in", "MDIN"},
       {"ADC", "PGA", "Input Mux"},

Thanks,
------
ROHM Co., Ltd.
Tomoya MORINAGA

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Tomoya MORINAGA]

2011/12/1 Takashi Iwai <tiwai@suse.de>:
> You need to have read Mark's comment carefully.  He quoted another
> line before that.
>
> ================================================================
>> +     SOC_SINGLE_TLV("EQ Band1 Input Volume", ML26124_EQ_GAIN_BRAND1, 0,
>> +                     0xff, 1, eq_band_gain),
>
>> +     SOC_SINGLE("EQ BAND1 Switch", ML26124_FILTER_EN, 3, 1, 0),
>
> These should mathc.
> ================================================================
>
> So, you have two inconsistent names for EQ band1.
>

You mean "EQ Band1" and "EQ BAND1" are inconsistent names ?
If yes, can you accept the followings ?

EQ Band1 Input Volume
EQ Band1 Switch

Thanks,

------
tomoya
ROHM Co., Ltd.
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Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Mark Brown]

On Thu, Dec 01, 2011 at 04:57:52PM +0900, Tomoya MORINAGA wrote:

> You mean "EQ Band1" and "EQ BAND1" are inconsistent names ?
> If yes, can you accept the followings ?

> EQ Band1 Input Volume
> EQ Band1 Switch

"EQ Band1 Input" and "EQ Band1" are still different, they should be the
same.

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Lars-Peter Clausen]

On 12/01/2011 08:46 AM, Tomoya MORINAGA wrote:
> 2011/11/29 Lars-Peter Clausen <lars@metafoo.de>:
>>
>> Also this doesn't make much sense:
>> +       {"Input Mux", "Analog MIC in", "MICBIAS"},
>> +       {"Input Mux", "Digital MIC in", "MICBIAS"},
>>
>> You have mux, but both possible settings of the mux connect to the same
>> source, which pretty much makes the mux useless.
>>
> 
> Yes, you are right.
> This setting was miss-settings.
> This MICBIAS supplies to external MIC as reference voltage.
> So, I think MICBIAS shouldn't appear at this setting.
> I show modified settings.
>        /* input */
>        {"Input Mux", "Analog MIC SingleEnded in", "MIN"},
>        {"Input Mux", "Analog MIC Differential in", "MIN"},
>        {"Input Mux", "Digital MIC in", "MDIN"},
>        {"ADC", "PGA", "Input Mux"},
> 

The middle part(the control) of the path must either be a control attached to
the mixer on the left side(the sink) or a element of the mux's enum if a mux is
on the left side. If the left side is neither a mixer or a mux the middle part
is ignored.

The other problem probably is that you have one mux which lets you switch
between single-ended analog input, differential analog input and digital input.
The first two options are routed through PGA and ADC the later not.

- Lars

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Tomoya MORINAGA]

2011/12/1 Mark Brown <broonie@opensource.wolfsonmicro.com>:
> On Thu, Dec 01, 2011 at 04:57:52PM +0900, Tomoya MORINAGA wrote:
>
>> You mean "EQ Band1" and "EQ BAND1" are inconsistent names ?
>> If yes, can you accept the followings ?
>
>> EQ Band1 Input Volume
>> EQ Band1 Switch
>
> "EQ Band1 Input" and "EQ Band1" are still different, they should be the
> same.
>

Which description do you want ?

EQ Band1 Volume
EQ Band1 Switch

or

EQ Band1 Input Volume
EQ Band1 Input Switch

or

another

Thanks,
------
tomoya
ROHM Co., Ltd.

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Mark Brown]

On Fri, Dec 02, 2011 at 09:33:23AM +0900, Tomoya MORINAGA wrote:

> Which description do you want ?

> EQ Band1 Volume
> EQ Band1 Switch

> or

> EQ Band1 Input Volume
> EQ Band1 Input Switch

Either is fine, if I had to pick one I'd go with the first one.  The
important thing is that Switch and Volume use the same string.

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Tomoya MORINAGA]

2011/12/2 Mark Brown <broonie@opensource.wolfsonmicro.com>:
> On Fri, Dec 02, 2011 at 09:33:23AM +0900, Tomoya MORINAGA wrote:
>
>> Which description do you want ?
>
>> EQ Band1 Volume
>> EQ Band1 Switch
>
>> or
>
>> EQ Band1 Input Volume
>> EQ Band1 Input Switch
>
> Either is fine, if I had to pick one I'd go with the first one.  The
> important thing is that Switch and Volume use the same string.
>

OK, I would use the first one.

Thanks,

tomoya

Re: [PATCH v3] sound/soc/codecs: add LAPIS Semiconductor ML26124

[Tomoya MORINAGA]

2011/12/2 Lars-Peter Clausen <lars@metafoo.de>:
> The middle part(the control) of the path must either be a control attached to
> the mixer on the left side(the sink) or a element of the mux's enum if a mux is
> on the left side. If the left side is neither a mixer or a mux the middle part
> is ignored.
>
> The other problem probably is that you have one mux which lets you switch
> between single-ended analog input, differential analog input and digital input.
> The first two options are routed through PGA and ADC the later not.
I can understand.
The "digital input" description shouldn't be at snd_soc_dpram_route description.
So, I've modified like below. These satisfies your showed condition.

	/* Supply */
	{"DAC", NULL, "MCLK"},
	{"ADC", NULL, "MCLK"},
	{"DAC", NULL, "PLL"},
	{"ADC", NULL, "PLL"},

	/* output mixer */
	{"Output Mixer", "PGA Switch", "PGA"},
	{"Output Mixer", "DAC Switch", "DAC"},
	{"Output Mixer", "Line in Switch", "LIN"},

	/* outputs */
	{"LOUT", NULL, "Output Mixer"},
	{"SPOUT", NULL, "Output Mixer"},

	/* input */
	{"Input Mux", "Analog MIC SingleEnded in", "MIN"},
	{"Input Mux", "Analog MIC Differential in", "MIN"},
	{"ADC", NULL, "Input Mux"},


Thanks,
tomoya