Add the rt linux 4.1.3-rt3 as base

Import the rt linux 4.1.3-rt3 as OPNFV kvm base.

It's from git://git.kernel.org/pub/scm/linux/kernel/git/rt/linux-rt-devel.git linux-4.1.y-rt and
the base is:

commit 0917f823c59692d751951bf5ea699a2d1e2f26a2
Author: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Date:   Sat Jul 25 12:13:34 2015 +0200

    Prepare v4.1.3-rt3

    Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>

We lose all the git history this way and it's not good. We
should apply another opnfv project repo in future.

Change-Id: I87543d81c9df70d99c5001fbdf646b202c19f423
Signed-off-by: Yunhong Jiang <yunhong.jiang@intel.com>

1 files changed, 581 insertions, 0 deletions

diff --git a/kernel/sound/soc/codecs/sirf-audio-codec.c b/kernel/sound/soc/codecs/sirf-audio-codec.c
new file mode 100644
index 000000000..0a8e43c98
--- /dev/null
+++ b/kernel/sound/soc/codecs/sirf-audio-codec.c
@@ -0,0 +1,581 @@
+/*
+ * SiRF audio codec driver
+ *
+ * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company.
+ *
+ * Licensed under GPLv2 or later.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/pm_runtime.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sirf-audio-codec.h"
+
+struct sirf_audio_codec {
+	struct clk *clk;
+	struct regmap *regmap;
+	u32 reg_ctrl0, reg_ctrl1;
+};
+
+static const char * const input_mode_mux[] = {"Single-ended",
+	"Differential"};
+
+static const struct soc_enum input_mode_mux_enum =
+	SOC_ENUM_SINGLE(AUDIO_IC_CODEC_CTRL1, 4, 2, input_mode_mux);
+
+static const struct snd_kcontrol_new sirf_audio_codec_input_mode_control =
+	SOC_DAPM_ENUM("Route", input_mode_mux_enum);
+
+static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -12400, 100, 0);
+static const DECLARE_TLV_DB_SCALE(capture_vol_tlv_prima2, 500, 100, 0);
+static const DECLARE_TLV_DB_RANGE(capture_vol_tlv_atlas6,
+	0, 7, TLV_DB_SCALE_ITEM(-100, 100, 0),
+	0x22, 0x3F, TLV_DB_SCALE_ITEM(700, 100, 0),
+);
+
+static struct snd_kcontrol_new volume_controls_atlas6[] = {
+	SOC_DOUBLE_TLV("Playback Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+			0x7F, 0, playback_vol_tlv),
+	SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 16, 10,
+			0x3F, 0, capture_vol_tlv_atlas6),
+};
+
+static struct snd_kcontrol_new volume_controls_prima2[] = {
+	SOC_DOUBLE_TLV("Speaker Volume", AUDIO_IC_CODEC_CTRL0, 21, 14,
+			0x7F, 0, playback_vol_tlv),
+	SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 15, 10,
+			0x1F, 0, capture_vol_tlv_prima2),
+};
+
+static struct snd_kcontrol_new left_input_path_controls[] = {
+	SOC_DAPM_SINGLE("Line Left Switch", AUDIO_IC_CODEC_CTRL1, 6, 1, 0),
+	SOC_DAPM_SINGLE("Mic Left Switch", AUDIO_IC_CODEC_CTRL1, 3, 1, 0),
+};
+
+static struct snd_kcontrol_new right_input_path_controls[] = {
+	SOC_DAPM_SINGLE("Line Right Switch", AUDIO_IC_CODEC_CTRL1, 5, 1, 0),
+	SOC_DAPM_SINGLE("Mic Right Switch", AUDIO_IC_CODEC_CTRL1, 2, 1, 0),
+};
+
+static struct snd_kcontrol_new left_dac_to_hp_left_amp_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 9, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_hp_right_amp_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 8, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_left_amp_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 7, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_hp_right_amp_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 6, 1, 0);
+
+static struct snd_kcontrol_new left_dac_to_speaker_lineout_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 11, 1, 0);
+
+static struct snd_kcontrol_new right_dac_to_speaker_lineout_switch_control =
+	SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 10, 1, 0);
+
+/* After enable adc, Delay 200ms to avoid pop noise */
+static int adc_enable_delay_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		msleep(200);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static void enable_and_reset_codec(struct regmap *regmap,
+		u32 codec_enable_bits, u32 codec_reset_bits)
+{
+	regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+			codec_enable_bits | codec_reset_bits,
+			codec_enable_bits);
+	msleep(20);
+	regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1,
+			codec_reset_bits, codec_reset_bits);
+}
+
+static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+#define ATLAS6_CODEC_ENABLE_BITS (1 << 29)
+#define ATLAS6_CODEC_RESET_BITS (1 << 28)
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec);
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		enable_and_reset_codec(sirf_audio_codec->regmap,
+			ATLAS6_CODEC_ENABLE_BITS, ATLAS6_CODEC_RESET_BITS);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		regmap_update_bits(sirf_audio_codec->regmap,
+			AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS, 0);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+#define PRIMA2_CODEC_ENABLE_BITS (1 << 27)
+#define PRIMA2_CODEC_RESET_BITS (1 << 26)
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec);
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		enable_and_reset_codec(sirf_audio_codec->regmap,
+			PRIMA2_CODEC_ENABLE_BITS, PRIMA2_CODEC_RESET_BITS);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		regmap_update_bits(sirf_audio_codec->regmap,
+			AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS, 0);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget atlas6_output_driver_dapm_widgets[] = {
+	SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+			25, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+			26, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+			27, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget prima2_output_driver_dapm_widgets[] = {
+	SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1,
+			23, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1,
+			24, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1,
+			25, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_widget atlas6_codec_clock_dapm_widget =
+	SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+			atlas6_codec_enable_and_reset_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget prima2_codec_clock_dapm_widget =
+	SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0,
+			prima2_codec_enable_and_reset_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD);
+
+static const struct snd_soc_dapm_widget sirf_audio_codec_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC left", NULL, AUDIO_IC_CODEC_CTRL0, 1, 0),
+	SND_SOC_DAPM_DAC("DAC right", NULL, AUDIO_IC_CODEC_CTRL0, 0, 0),
+	SND_SOC_DAPM_SWITCH("Left dac to hp left amp", SND_SOC_NOPM, 0, 0,
+			&left_dac_to_hp_left_amp_switch_control),
+	SND_SOC_DAPM_SWITCH("Left dac to hp right amp", SND_SOC_NOPM, 0, 0,
+			&left_dac_to_hp_right_amp_switch_control),
+	SND_SOC_DAPM_SWITCH("Right dac to hp left amp", SND_SOC_NOPM, 0, 0,
+			&right_dac_to_hp_left_amp_switch_control),
+	SND_SOC_DAPM_SWITCH("Right dac to hp right amp", SND_SOC_NOPM, 0, 0,
+			&right_dac_to_hp_right_amp_switch_control),
+	SND_SOC_DAPM_OUT_DRV("HP amp left driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+			NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HP amp right driver", AUDIO_IC_CODEC_CTRL0, 3, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_SWITCH("Left dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+			&left_dac_to_speaker_lineout_switch_control),
+	SND_SOC_DAPM_SWITCH("Right dac to speaker lineout", SND_SOC_NOPM, 0, 0,
+			&right_dac_to_speaker_lineout_switch_control),
+	SND_SOC_DAPM_OUT_DRV("Speaker amp driver", AUDIO_IC_CODEC_CTRL0, 4, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("HPOUTL"),
+	SND_SOC_DAPM_OUTPUT("HPOUTR"),
+	SND_SOC_DAPM_OUTPUT("SPKOUT"),
+
+	SND_SOC_DAPM_ADC_E("ADC left", NULL, AUDIO_IC_CODEC_CTRL1, 8, 0,
+			adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_ADC_E("ADC right", NULL, AUDIO_IC_CODEC_CTRL1, 7, 0,
+			adc_enable_delay_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_MIXER("Left PGA mixer", AUDIO_IC_CODEC_CTRL1, 1, 0,
+		&left_input_path_controls[0],
+		ARRAY_SIZE(left_input_path_controls)),
+	SND_SOC_DAPM_MIXER("Right PGA mixer", AUDIO_IC_CODEC_CTRL1, 0, 0,
+		&right_input_path_controls[0],
+		ARRAY_SIZE(right_input_path_controls)),
+
+	SND_SOC_DAPM_MUX("Mic input mode mux", SND_SOC_NOPM, 0, 0,
+			&sirf_audio_codec_input_mode_control),
+	SND_SOC_DAPM_MICBIAS("Mic Bias", AUDIO_IC_CODEC_PWR, 3, 0),
+	SND_SOC_DAPM_INPUT("MICIN1"),
+	SND_SOC_DAPM_INPUT("MICIN2"),
+	SND_SOC_DAPM_INPUT("LINEIN1"),
+	SND_SOC_DAPM_INPUT("LINEIN2"),
+
+	SND_SOC_DAPM_SUPPLY("HSL Phase Opposite", AUDIO_IC_CODEC_CTRL0,
+			30, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route sirf_audio_codec_map[] = {
+	{"SPKOUT", NULL, "Speaker Driver"},
+	{"Speaker Driver", NULL, "Speaker amp driver"},
+	{"Speaker amp driver", NULL, "Left dac to speaker lineout"},
+	{"Speaker amp driver", NULL, "Right dac to speaker lineout"},
+	{"Left dac to speaker lineout", "Switch", "DAC left"},
+	{"Right dac to speaker lineout", "Switch", "DAC right"},
+	{"HPOUTL", NULL, "HP Left Driver"},
+	{"HPOUTR", NULL, "HP Right Driver"},
+	{"HP Left Driver", NULL, "HP amp left driver"},
+	{"HP Right Driver", NULL, "HP amp right driver"},
+	{"HP amp left driver", NULL, "Right dac to hp left amp"},
+	{"HP amp right driver", NULL , "Right dac to hp right amp"},
+	{"HP amp left driver", NULL, "Left dac to hp left amp"},
+	{"HP amp right driver", NULL , "Right dac to hp right amp"},
+	{"Right dac to hp left amp", "Switch", "DAC left"},
+	{"Right dac to hp right amp", "Switch", "DAC right"},
+	{"Left dac to hp left amp", "Switch", "DAC left"},
+	{"Left dac to hp right amp", "Switch", "DAC right"},
+	{"DAC left", NULL, "codecclk"},
+	{"DAC right", NULL, "codecclk"},
+	{"DAC left", NULL, "Playback"},
+	{"DAC right", NULL, "Playback"},
+	{"DAC left", NULL, "HSL Phase Opposite"},
+	{"DAC right", NULL, "HSL Phase Opposite"},
+
+	{"Capture", NULL, "ADC left"},
+	{"Capture", NULL, "ADC right"},
+	{"ADC left", NULL, "codecclk"},
+	{"ADC right", NULL, "codecclk"},
+	{"ADC left", NULL, "Left PGA mixer"},
+	{"ADC right", NULL, "Right PGA mixer"},
+	{"Left PGA mixer", "Line Left Switch", "LINEIN2"},
+	{"Right PGA mixer", "Line Right Switch", "LINEIN1"},
+	{"Left PGA mixer", "Mic Left Switch", "MICIN2"},
+	{"Right PGA mixer", "Mic Right Switch", "Mic input mode mux"},
+	{"Mic input mode mux", "Single-ended", "MICIN1"},
+	{"Mic input mode mux", "Differential", "MICIN1"},
+};
+
+static void sirf_audio_codec_tx_enable(struct sirf_audio_codec *sirf_audio_codec)
+{
+	regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP,
+		AUDIO_FIFO_RESET, AUDIO_FIFO_RESET);
+	regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP,
+		AUDIO_FIFO_RESET, ~AUDIO_FIFO_RESET);
+	regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_INT_MSK, 0);
+	regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0);
+	regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP,
+		AUDIO_FIFO_START, AUDIO_FIFO_START);
+	regmap_update_bits(sirf_audio_codec->regmap,
+		AUDIO_PORT_IC_CODEC_TX_CTRL, IC_TX_ENABLE, IC_TX_ENABLE);
+}
+
+static void sirf_audio_codec_tx_disable(struct sirf_audio_codec *sirf_audio_codec)
+{
+	regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0);
+	regmap_update_bits(sirf_audio_codec->regmap,
+		AUDIO_PORT_IC_CODEC_TX_CTRL, IC_TX_ENABLE, ~IC_TX_ENABLE);
+}
+
+static void sirf_audio_codec_rx_enable(struct sirf_audio_codec *sirf_audio_codec,
+	int channels)
+{
+	regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP,
+		AUDIO_FIFO_RESET, AUDIO_FIFO_RESET);
+	regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP,
+		AUDIO_FIFO_RESET, ~AUDIO_FIFO_RESET);
+	regmap_write(sirf_audio_codec->regmap,
+		AUDIO_PORT_IC_RXFIFO_INT_MSK, 0);
+	regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP, 0);
+	regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP,
+		AUDIO_FIFO_START, AUDIO_FIFO_START);
+	if (channels == 1)
+		regmap_update_bits(sirf_audio_codec->regmap,
+			AUDIO_PORT_IC_CODEC_RX_CTRL,
+			IC_RX_ENABLE_MONO, IC_RX_ENABLE_MONO);
+	else
+		regmap_update_bits(sirf_audio_codec->regmap,
+			AUDIO_PORT_IC_CODEC_RX_CTRL,
+			IC_RX_ENABLE_STEREO, IC_RX_ENABLE_STEREO);
+}
+
+static void sirf_audio_codec_rx_disable(struct sirf_audio_codec *sirf_audio_codec)
+{
+	regmap_update_bits(sirf_audio_codec->regmap,
+			AUDIO_PORT_IC_CODEC_RX_CTRL,
+			IC_RX_ENABLE_STEREO, ~IC_RX_ENABLE_STEREO);
+}
+
+static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream,
+		int cmd,
+		struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec);
+	int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+
+	/*
+	 * This is a workaround, When stop playback,
+	 * need disable HP amp, avoid the current noise.
+	 */
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		if (playback) {
+			snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0,
+				IC_HSLEN | IC_HSREN, 0);
+			sirf_audio_codec_tx_disable(sirf_audio_codec);
+		} else
+			sirf_audio_codec_rx_disable(sirf_audio_codec);
+		break;
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (playback) {
+			sirf_audio_codec_tx_enable(sirf_audio_codec);
+			snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0,
+				IC_HSLEN | IC_HSREN, IC_HSLEN | IC_HSREN);
+		} else
+			sirf_audio_codec_rx_enable(sirf_audio_codec,
+				substream->runtime->channels);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+struct snd_soc_dai_ops sirf_audio_codec_dai_ops = {
+	.trigger = sirf_audio_codec_trigger,
+};
+
+struct snd_soc_dai_driver sirf_audio_codec_dai = {
+	.name = "sirf-audio-codec",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_48000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &sirf_audio_codec_dai_ops,
+};
+
+static int sirf_audio_codec_probe(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	pm_runtime_enable(codec->dev);
+
+	if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) {
+		snd_soc_dapm_new_controls(dapm,
+			prima2_output_driver_dapm_widgets,
+			ARRAY_SIZE(prima2_output_driver_dapm_widgets));
+		snd_soc_dapm_new_controls(dapm,
+			&prima2_codec_clock_dapm_widget, 1);
+		return snd_soc_add_codec_controls(codec,
+			volume_controls_prima2,
+			ARRAY_SIZE(volume_controls_prima2));
+	}
+	if (of_device_is_compatible(codec->dev->of_node, "sirf,atlas6-audio-codec")) {
+		snd_soc_dapm_new_controls(dapm,
+			atlas6_output_driver_dapm_widgets,
+			ARRAY_SIZE(atlas6_output_driver_dapm_widgets));
+		snd_soc_dapm_new_controls(dapm,
+			&atlas6_codec_clock_dapm_widget, 1);
+		return snd_soc_add_codec_controls(codec,
+			volume_controls_atlas6,
+			ARRAY_SIZE(volume_controls_atlas6));
+	}
+
+	return -EINVAL;
+}
+
+static int sirf_audio_codec_remove(struct snd_soc_codec *codec)
+{
+	pm_runtime_disable(codec->dev);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = {
+	.probe = sirf_audio_codec_probe,
+	.remove = sirf_audio_codec_remove,
+	.dapm_widgets = sirf_audio_codec_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets),
+	.dapm_routes = sirf_audio_codec_map,
+	.num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map),
+	.idle_bias_off = true,
+};
+
+static const struct of_device_id sirf_audio_codec_of_match[] = {
+	{ .compatible = "sirf,prima2-audio-codec" },
+	{ .compatible = "sirf,atlas6-audio-codec" },
+	{}
+};
+MODULE_DEVICE_TABLE(of, sirf_audio_codec_of_match);
+
+static const struct regmap_config sirf_audio_codec_regmap_config = {
+	.reg_bits = 32,
+	.reg_stride = 4,
+	.val_bits = 32,
+	.max_register = AUDIO_PORT_IC_RXFIFO_INT_MSK,
+	.cache_type = REGCACHE_NONE,
+};
+
+static int sirf_audio_codec_driver_probe(struct platform_device *pdev)
+{
+	int ret;
+	struct sirf_audio_codec *sirf_audio_codec;
+	void __iomem *base;
+	struct resource *mem_res;
+	const struct of_device_id *match;
+
+	match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node);
+
+	sirf_audio_codec = devm_kzalloc(&pdev->dev,
+		sizeof(struct sirf_audio_codec), GFP_KERNEL);
+	if (!sirf_audio_codec)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, sirf_audio_codec);
+
+	mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	base = devm_ioremap_resource(&pdev->dev, mem_res);
+	if (IS_ERR(base))
+		return PTR_ERR(base);
+
+	sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base,
+					    &sirf_audio_codec_regmap_config);
+	if (IS_ERR(sirf_audio_codec->regmap))
+		return PTR_ERR(sirf_audio_codec->regmap);
+
+	sirf_audio_codec->clk = devm_clk_get(&pdev->dev, NULL);
+	if (IS_ERR(sirf_audio_codec->clk)) {
+		dev_err(&pdev->dev, "Get clock failed.\n");
+		return PTR_ERR(sirf_audio_codec->clk);
+	}
+
+	ret = clk_prepare_enable(sirf_audio_codec->clk);
+	if (ret) {
+		dev_err(&pdev->dev, "Enable clock failed.\n");
+		return ret;
+	}
+
+	ret = snd_soc_register_codec(&(pdev->dev),
+			&soc_codec_device_sirf_audio_codec,
+			&sirf_audio_codec_dai, 1);
+	if (ret) {
+		dev_err(&pdev->dev, "Register Audio Codec dai failed.\n");
+		goto err_clk_put;
+	}
+
+	/*
+	 * Always open charge pump, if not, when the charge pump closed the
+	 * adc will not stable
+	 */
+	regmap_update_bits(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+		IC_CPFREQ, IC_CPFREQ);
+
+	if (of_device_is_compatible(pdev->dev.of_node, "sirf,atlas6-audio-codec"))
+		regmap_update_bits(sirf_audio_codec->regmap,
+				AUDIO_IC_CODEC_CTRL0, IC_CPEN, IC_CPEN);
+	return 0;
+
+err_clk_put:
+	clk_disable_unprepare(sirf_audio_codec->clk);
+	return ret;
+}
+
+static int sirf_audio_codec_driver_remove(struct platform_device *pdev)
+{
+	struct sirf_audio_codec *sirf_audio_codec = platform_get_drvdata(pdev);
+
+	clk_disable_unprepare(sirf_audio_codec->clk);
+	snd_soc_unregister_codec(&(pdev->dev));
+
+	return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int sirf_audio_codec_suspend(struct device *dev)
+{
+	struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+
+	regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+		&sirf_audio_codec->reg_ctrl0);
+	regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+		&sirf_audio_codec->reg_ctrl1);
+	clk_disable_unprepare(sirf_audio_codec->clk);
+
+	return 0;
+}
+
+static int sirf_audio_codec_resume(struct device *dev)
+{
+	struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev);
+	int ret;
+
+	ret = clk_prepare_enable(sirf_audio_codec->clk);
+	if (ret)
+		return ret;
+
+	regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0,
+		sirf_audio_codec->reg_ctrl0);
+	regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1,
+		sirf_audio_codec->reg_ctrl1);
+
+	return 0;
+}
+#endif
+
+static const struct dev_pm_ops sirf_audio_codec_pm_ops = {
+	SET_SYSTEM_SLEEP_PM_OPS(sirf_audio_codec_suspend, sirf_audio_codec_resume)
+};
+
+static struct platform_driver sirf_audio_codec_driver = {
+	.driver = {
+		.name = "sirf-audio-codec",
+		.of_match_table = sirf_audio_codec_of_match,
+		.pm = &sirf_audio_codec_pm_ops,
+	},
+	.probe = sirf_audio_codec_driver_probe,
+	.remove = sirf_audio_codec_driver_remove,
+};
+
+module_platform_driver(sirf_audio_codec_driver);
+
+MODULE_DESCRIPTION("SiRF audio codec driver");
+MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>");
+MODULE_LICENSE("GPL v2");