Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This is the first big chunk for 3.5 merges of sound stuff.

  There are a few big changes in different areas.  First off, the
  streaming logic of USB-audio endpoints has been largely rewritten for
  the better support of "implicit feedback".  If anything about USB got
  broken, this change has to be checked.

  For HD-audio, the resume procedure was changed; instead of delaying
  the resume of the hardware until the first use, now waking up
  immediately at resume.  This is for buggy BIOS.

  For ASoC, dynamic PCM support and the improved support for digital
  links between off-SoC devices are major framework changes.

  Some highlights are below:

  * HD-audio
   - Avoid accesses of invalid pin-control bits that may stall the codec
   - V-ref setup cleanups
   - Fix the races in power-saving code
   - Fix the races in codec cache hashes and connection lists
   - Split some common codes for BIOS auto-parser to hda_auto_parser.c
   - Changed the PM resume code to wake up immediately for buggy BIOS
   - Creative SoundCore3D support
   - Add Conexant CX20751/2/3/4 codec support

  * ASoC
   - Dynamic PCM support, allowing support for SoCs with internal
     routing through components with tight sequencing and formatting
     constraints within their internal paths or where there are multiple
     components connected with CPU managed DMA controllers inside the
     SoC.
   - Greatly improved support for direct digital links between off-SoC
     devices, providing a much simpler way of connecting things like
     digital basebands to CODECs.
   - Much more fine grained and robust locking, cleaning up some of the
     confusion that crept in with multi-component.
   - CPU support for nVidia Tegra 30 I2S and audio hub controllers and
     ST-Ericsson MSP I2S controolers
   - New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
     Texas Instruments LM49453.
   - Some regmap changes needed by the Tegra I2S driver.
   - mc13783 audio support.

  * Misc
   - Rewrite with module_pci_driver()
   - Xonar DGX support for snd-oxygen
   - Improvement of packet handling in snd-firewire driver
   - New USB-endpoint streaming logic
   - Enhanced M-audio FTU quirks and relevant cleanups
   - Increment the support of OSS devices to 256
   - snd-aloop accuracy improvement

  There are a few more pending changes for 3.5, but they will be sent
  slightly later as partly depending on the changes of DRM."

Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree).  Also some
fairly trivial context conflicts in the imx and mcx soc drivers.

* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
  ALSA: snd-usb: fix stream info output in /proc
  ALSA: pcm - Add proper state checks to snd_pcm_drain()
  ALSA: sh: Fix up namespace collision in sh_dac_audio.
  ALSA: hda/realtek - Fix unused variable compile warning
  ASoC: sh: fsi: enable chip specific data transfer mode
  ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
  ASoC: sh: fsi: use same format for IN/OUT
  ASoC: sh: fsi: add fsi_version() and removed meaningless version check
  ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
  ASoC: tegra: Add machine driver for WM8753 codec
  ALSA: hda - Fix possible races of accesses to connection list array
  ASoC: OMAP: HDMI: Introduce codec
  ARM: mx31_3ds: Add sound support
  ASoC: imx-mc13783 cleanup
  mx31moboard: Add sound support
  ASoC: mc13783 codec cleanups
  ASoC: add imx-mc13783 sound support
  ASoC: Add mc13783 codec
  mfd: mc13xxx: add codec platform data
  ASoC: don't flip master of DT-instantiated DAI links
  ...

1 files changed, 681 insertions, 0 deletions

diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
new file mode 100644
index 00000000000..22cb5bf5927
--- /dev/null
+++ b/sound/soc/codecs/ml26124.c
@@ -0,0 +1,681 @@
+/*
+ * 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 <linux/regmap.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"
+
+#define DVOL_CTL_DVMUTE_ON		BIT(4)	/* Digital volume MUTE On */
+#define DVOL_CTL_DVMUTE_OFF		0	/* Digital volume MUTE Off */
+#define ML26124_SAI_NO_DELAY	BIT(1)
+#define ML26124_SAI_FRAME_SYNC	(BIT(5) | BIT(0)) /* For mono (Telecodec) */
+#define ML26134_CACHESIZE 212
+#define ML26124_VMID	BIT(1)
+#define ML26124_RATES (SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\
+		       SNDRV_PCM_RATE_48000)
+#define ML26124_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE |\
+			 SNDRV_PCM_FMTBIT_S32_LE)
+#define ML26124_NUM_REGISTER ML26134_CACHESIZE
+
+struct ml26124_priv {
+	u32 mclk;
+	u32 rate;
+	struct regmap *regmap;
+	int clk_in;
+	struct snd_pcm_substream *substream;
+};
+
+struct clk_coeff {
+	u32 mclk;
+	u32 rate;
+	u8 pllnl;
+	u8 pllnh;
+	u8 pllml;
+	u8 pllmh;
+	u8 plldiv;
+};
+
+/* ML26124 configuration */
+static const DECLARE_TLV_DB_SCALE(digital_tlv, -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_tlv),
+	SOC_SINGLE_TLV("Playback Digital Volume", ML26124_PLBAK_DIG_VOL, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("Digital Boost Volume", ML26124_DIGI_BOOST_VOL, 0,
+			0x3f, 0, boost_vol),
+	SOC_SINGLE_TLV("EQ Band0 Volume", ML26124_EQ_GAIN_BRAND0, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("EQ Band1 Volume", ML26124_EQ_GAIN_BRAND1, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("EQ Band2 Volume", ML26124_EQ_GAIN_BRAND2, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("EQ Band3 Volume", ML26124_EQ_GAIN_BRAND3, 0,
+			0xff, 1, digital_tlv),
+	SOC_SINGLE_TLV("EQ Band4 Volume", ML26124_EQ_GAIN_BRAND4, 0,
+			0xff, 1, digital_tlv),
+	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 Limiter Min Input Volume",
+			ML26124_PL_MAXMIN_GAIN, 0, 7, 0, mingain),
+	SOC_SINGLE_TLV("Playback Limiter 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("ZC Switch", ML26124_PW_ZCCMP_PW_MNG, 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 Limiter", ML26124_DVOL_CTL, 0, 1, 0),
+	SOC_SINGLE("Capture Limiter", ML26124_DVOL_CTL, 1, 1, 0),
+	SOC_SINGLE("Digital Volume Fade Switch", ML26124_DVOL_CTL, 3, 1, 0),
+	SOC_SINGLE("Digital 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 loopback 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 SingleEnded in",
+				"Digital MIC in", "Analog MIC Differential in"};
+
+static const struct soc_enum ml26124_insel_enum =
+	SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, 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("MCLKEN", ML26124_CLK_EN, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLLEN", ML26124_CLK_EN, 1, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLLOE", ML26124_CLK_EN, 2, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS", ML26124_PW_REF_PW_MNG, 2, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 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("MDIN"),
+	SND_SOC_DAPM_INPUT("MIN"),
+	SND_SOC_DAPM_INPUT("LIN"),
+	SND_SOC_DAPM_OUTPUT("SPOUT"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+};
+
+static const struct snd_soc_dapm_route ml26124_intercon[] = {
+	/* Supply */
+	{"DAC", NULL, "MCLKEN"},
+	{"ADC", NULL, "MCLKEN"},
+	{"DAC", NULL, "PLLEN"},
+	{"ADC", NULL, "PLLEN"},
+	{"DAC", NULL, "PLLOE"},
+	{"ADC", NULL, "PLLOE"},
+
+	/* output mixer */
+	{"Output Mixer", "DAC Switch", "DAC"},
+	{"Output Mixer", "Line in loopback Switch", "LIN"},
+
+	/* outputs */
+	{"LOUT", NULL, "Output Mixer"},
+	{"SPOUT", NULL, "Output Mixer"},
+	{"Line Out Enable", NULL, "LOUT"},
+
+	/* input */
+	{"ADC", NULL, "Input Mux"},
+	{"Input Mux", "Analog MIC SingleEnded in", "PGA"},
+	{"Input Mux", "Analog MIC Differential in", "PGA"},
+	{"PGA", NULL, "MIN"},
+};
+
+/* PLLOutputFreq(Hz) = InputMclkFreq(Hz) * PLLM / (PLLN * PLLDIV) */
+static const struct clk_coeff coeff_div[] = {
+	{12288000, 16000, 0xc, 0x0, 0x20, 0x0, 0x4},
+	{12288000, 32000, 0xc, 0x0, 0x20, 0x0, 0x4},
+	{12288000, 48000, 0xc, 0x0, 0x30, 0x0, 0x4},
+};
+
+static struct reg_default ml26124_reg[] = {
+	/* CLOCK control Register */
+	{0x00, 0x00 },	/* Sampling Rate */
+	{0x02, 0x00},	/* PLL NL */
+	{0x04, 0x00},	/* PLLNH */
+	{0x06, 0x00},	/* PLLML */
+	{0x08, 0x00},	/* MLLMH */
+	{0x0a, 0x00},	/* PLLDIV */
+	{0x0c, 0x00},	/* Clock Enable */
+	{0x0e, 0x00},	/* CLK Input/Output Control */
+
+	/* System Control Register */
+	{0x10, 0x00},	/* Software RESET */
+	{0x12, 0x00},	/* Record/Playback Run */
+	{0x14, 0x00},	/* Mic Input/Output control */
+
+	/* Power Management Register */
+	{0x20, 0x00},	/* Reference Power Management */
+	{0x22, 0x00},	/* Input Power Management */
+	{0x24, 0x00},	/* DAC Power Management */
+	{0x26, 0x00},	/* SP-AMP Power Management */
+	{0x28, 0x00},	/* LINEOUT Power Management */
+	{0x2a, 0x00},	/* VIDEO Power Management */
+	{0x2e, 0x00},	/* AC-CMP Power Management */
+
+	/* Analog reference Control Register */
+	{0x30, 0x04},	/* MICBIAS Voltage Control */
+
+	/* Input/Output Amplifier Control Register */
+	{0x32, 0x10},	/* MIC Input Volume */
+	{0x38, 0x00},	/* Mic Boost Volume */
+	{0x3a, 0x33},	/* Speaker AMP Volume */
+	{0x48, 0x00},	/* AMP Volume Control Function Enable */
+	{0x4a, 0x00},	/* Amplifier Volume Fader Control */
+
+	/* Analog Path Control Register */
+	{0x54, 0x00},	/* Speaker AMP Output Control */
+	{0x5a, 0x00},	/* Mic IF Control */
+	{0xe8, 0x01},	/* Mic Select Control */
+
+	/* Audio Interface Control Register */
+	{0x60, 0x00},	/* SAI-Trans Control */
+	{0x62, 0x00},	/* SAI-Receive Control */
+	{0x64, 0x00},	/* SAI Mode select */
+
+	/* DSP Control Register */
+	{0x66, 0x01},	/* Filter Func Enable */
+	{0x68, 0x00},	/* Volume Control Func Enable */
+	{0x6A, 0x00},	/* Mixer & Volume Control*/
+	{0x6C, 0xff},	/* Record Digital Volume */
+	{0x70, 0xff},	/* Playback Digital Volume */
+	{0x72, 0x10},	/* Digital Boost Volume */
+	{0x74, 0xe7},	/* EQ gain Band0 */
+	{0x76, 0xe7},	/* EQ gain Band1 */
+	{0x78, 0xe7},	/* EQ gain Band2 */
+	{0x7A, 0xe7},	/* EQ gain Band3 */
+	{0x7C, 0xe7},	/* EQ gain Band4 */
+	{0x7E, 0x00},	/* HPF2 CutOff*/
+	{0x80, 0x00},	/* EQ Band0 Coef0L */
+	{0x82, 0x00},	/* EQ Band0 Coef0H */
+	{0x84, 0x00},	/* EQ Band0 Coef0L */
+	{0x86, 0x00},	/* EQ Band0 Coef0H */
+	{0x88, 0x00},	/* EQ Band1 Coef0L */
+	{0x8A, 0x00},	/* EQ Band1 Coef0H */
+	{0x8C, 0x00},	/* EQ Band1 Coef0L */
+	{0x8E, 0x00},	/* EQ Band1 Coef0H */
+	{0x90, 0x00},	/* EQ Band2 Coef0L */
+	{0x92, 0x00},	/* EQ Band2 Coef0H */
+	{0x94, 0x00},	/* EQ Band2 Coef0L */
+	{0x96, 0x00},	/* EQ Band2 Coef0H */
+	{0x98, 0x00},	/* EQ Band3 Coef0L */
+	{0x9A, 0x00},	/* EQ Band3 Coef0H */
+	{0x9C, 0x00},	/* EQ Band3 Coef0L */
+	{0x9E, 0x00},	/* EQ Band3 Coef0H */
+	{0xA0, 0x00},	/* EQ Band4 Coef0L */
+	{0xA2, 0x00},	/* EQ Band4 Coef0H */
+	{0xA4, 0x00},	/* EQ Band4 Coef0L */
+	{0xA6, 0x00},	/* EQ Band4 Coef0H */
+
+	/* ALC Control Register */
+	{0xb0, 0x00},	/* ALC Mode */
+	{0xb2, 0x02},	/* ALC Attack Time */
+	{0xb4, 0x03},	/* ALC Decay Time */
+	{0xb6, 0x00},	/* ALC Hold Time */
+	{0xb8, 0x0b},	/* ALC Target Level */
+	{0xba, 0x70},	/* ALC Max/Min Gain */
+	{0xbc, 0x00},	/* Noise Gate Threshold */
+	{0xbe, 0x00},	/* ALC ZeroCross TimeOut */
+
+	/* Playback Limiter Control Register */
+	{0xc0, 0x04},	/* PL Attack Time */
+	{0xc2, 0x05},	/* PL Decay Time */
+	{0xc4, 0x0d},	/* PL Target Level */
+	{0xc6, 0x70},	/* PL Max/Min Gain */
+	{0xc8, 0x10},	/* Playback Boost Volume */
+	{0xca, 0x00},	/* PL ZeroCross TimeOut */
+
+	/* Video Amplifier Control Register */
+	{0xd0, 0x01},	/* VIDEO AMP Gain Control */
+	{0xd2, 0x01},	/* VIDEO AMP Setup 1 */
+	{0xd4, 0x01},	/* VIDEO AMP Control2 */
+};
+
+/* Get sampling rate value of sampling rate setting register (0x0) */
+static inline int get_srate(int rate)
+{
+	int srate;
+
+	switch (rate) {
+	case 16000:
+		srate = 3;
+		break;
+	case 32000:
+		srate = 6;
+		break;
+	case 48000:
+		srate = 8;
+		break;
+	default:
+		return -EINVAL;
+	}
+	return srate;
+}
+
+static inline int get_coeff(int mclk, int rate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+		if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+			return i;
+	}
+	return -EINVAL;
+}
+
+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;
+	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+	int i = get_coeff(priv->mclk, params_rate(hw_params));
+
+	priv->substream = substream;
+	priv->rate = params_rate(hw_params);
+
+	if (priv->clk_in) {
+		switch (priv->mclk / params_rate(hw_params)) {
+		case 256:
+			snd_soc_update_bits(codec, ML26124_CLK_CTL,
+					    BIT(0) | BIT(1), 1);
+			break;
+		case 512:
+			snd_soc_update_bits(codec, ML26124_CLK_CTL,
+					    BIT(0) | BIT(1), 2);
+			break;
+		case 1024:
+			snd_soc_update_bits(codec, ML26124_CLK_CTL,
+					    BIT(0) | BIT(1), 3);
+			break;
+		default:
+			dev_err(codec->dev, "Unsupported MCLKI\n");
+			break;
+		}
+	} else {
+		snd_soc_update_bits(codec, ML26124_CLK_CTL,
+				    BIT(0) | BIT(1), 0);
+	}
+
+	switch (params_rate(hw_params)) {
+	case 16000:
+		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+				    get_srate(params_rate(hw_params)));
+		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+				    coeff_div[i].pllnl);
+		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+				    coeff_div[i].pllnh);
+		snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+				    coeff_div[i].pllml);
+		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+				    coeff_div[i].pllmh);
+		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+				    coeff_div[i].plldiv);
+		break;
+	case 32000:
+		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+				    get_srate(params_rate(hw_params)));
+		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+				    coeff_div[i].pllnl);
+		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+				    coeff_div[i].pllnh);
+		snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+				    coeff_div[i].pllml);
+		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+				    coeff_div[i].pllmh);
+		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+				    coeff_div[i].plldiv);
+		break;
+	case 48000:
+		snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
+				    get_srate(params_rate(hw_params)));
+		snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
+				    coeff_div[i].pllnl);
+		snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
+				    coeff_div[i].pllnh);
+		snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
+				    coeff_div[i].pllml);
+		snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
+				    coeff_div[i].pllmh);
+		snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
+				    coeff_div[i].plldiv);
+		break;
+	default:
+		pr_err("%s:this rate is no support for ml26124\n", __func__);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int ml26124_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	switch (priv->substream->stream) {
+	case SNDRV_PCM_STREAM_CAPTURE:
+		snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(0), 1);
+		break;
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		snd_soc_update_bits(codec, ML26124_REC_PLYBAK_RUN, BIT(1), 2);
+		break;
+	}
+
+	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;
+}
+
+static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	unsigned char mode;
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	/* 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;
+	}
+
+	return 0;
+}
+
+static int ml26124_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 ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	switch (clk_id) {
+	case ML26124_USE_PLLOUT:
+		priv->clk_in = ML26124_USE_PLLOUT;
+		break;
+	case ML26124_USE_MCLKI:
+		priv->clk_in = ML26124_USE_MCLKI;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	priv->mclk = freq;
+
+	return 0;
+}
+
+static int ml26124_set_bias_level(struct snd_soc_codec *codec,
+		enum snd_soc_bias_level level)
+{
+	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG,
+				    ML26124_R26_MASK, ML26124_BLT_PREAMP_ON);
+		msleep(100);
+		snd_soc_update_bits(codec, ML26124_PW_SPAMP_PW_MNG,
+				    ML26124_R26_MASK,
+				    ML26124_MICBEN_ON | ML26124_BLT_ALL_ON);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* VMID ON */
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_update_bits(codec, ML26124_PW_REF_PW_MNG,
+					    ML26124_VMID, ML26124_VMID);
+			msleep(500);
+			regcache_sync(priv->regmap);
+		}
+		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;
+}
+
+static const struct snd_soc_dai_ops ml26124_dai_ops = {
+	.hw_params	= ml26124_hw_params,
+	.digital_mute	= ml26124_mute,
+	.set_fmt	= ml26124_set_dai_fmt,
+	.set_sysclk	= ml26124_set_dai_sysclk,
+};
+
+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)
+{
+	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;
+	struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
+	codec->control_data = priv->regmap;
+
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP);
+	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);
+
+	return 0;
+}
+
+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,
+	.dapm_widgets = ml26124_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ml26124_dapm_widgets),
+	.dapm_routes = ml26124_intercon,
+	.num_dapm_routes = ARRAY_SIZE(ml26124_intercon),
+	.controls = ml26124_snd_controls,
+	.num_controls = ARRAY_SIZE(ml26124_snd_controls),
+};
+
+static const struct regmap_config ml26124_i2c_regmap = {
+	.val_bits = 8,
+	.reg_bits = 8,
+	.max_register = ML26124_NUM_REGISTER,
+	.reg_defaults = ml26124_reg,
+	.num_reg_defaults = ARRAY_SIZE(ml26124_reg),
+	.cache_type = REGCACHE_RBTREE,
+	.write_flag_mask = 0x01,
+};
+
+static __devinit int ml26124_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct ml26124_priv *priv;
+	int ret;
+
+	priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, priv);
+
+	priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap);
+	if (IS_ERR(priv->regmap)) {
+		ret = PTR_ERR(priv->regmap);
+		dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret);
+		return ret;
+	}
+
+	return snd_soc_register_codec(&i2c->dev,
+			&soc_codec_dev_ml26124, &ml26124_dai, 1);
+}
+
+static __devexit int ml26124_i2c_remove(struct i2c_client *client)
+{
+	struct ml26124_priv *priv = i2c_get_clientdata(client);
+
+	snd_soc_unregister_codec(&client->dev);
+	regmap_exit(priv->regmap);
+	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,
+};
+
+module_i2c_driver(ml26124_i2c_driver);
+
+MODULE_AUTHOR("Tomoya MORINAGA <tomoya.rohm@gmail.com>");
+MODULE_DESCRIPTION("LAPIS Semiconductor ML26124 ALSA SoC codec driver");
+MODULE_LICENSE("GPL");