From 708b4351f08c08ea93f773fb9197bdd3f3b08273 Mon Sep 17 00:00:00 2001
From: Nicolin Chen <nicoleotsuka@gmail.com>
Date: Wed, 30 Jul 2014 19:27:38 +0800
Subject: ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support

The Freescale Generic ASoC Sound Card is a general ASoC DAI Link driver that
can be used, ideally, for all Freescale CPU DAI drivers and external CODECs.

The idea of this generic sound card is a bit like ASoC Simple Card. However,
for Freescale SoCs (especially those released in recent years), most of them
have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
this is a specific feature that might be painstakingly controlled and merged
into the Simple Card driver.

So having this driver will allow all Freescale SoC users to benefit from the
simplification to support a new card and the capability of wide sample rates
support through ASRC.

The driver is initially designed for sound card using I2S or PCM DAI formats.
However, it's also possible to merge those non-I2S/PCM type sound cards, such
as S/PDIF audio and HDMI audio, into this card as long as the merge will not
break the original function and as long as there is something redundant that
can be abstracted along with I2S type sound cards.

As an initial version, it only supports three cards that I can test:
imx-audio-cs42888, a new card that links ESAI with CS42888 CODEC
imx-audio-sgtl5000, just like the old imx-sgtl5000.c driver
imx-audio-wm8962, just like the old imx-wm8962.c driver

The driver is also compatible with the old Device Tree bindings of WM8962 and
SGTL5000. So we may consider to remove those two drivers after this driver is
totally enabled. (It needs to be added into defconfig)

Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/Kconfig         |  16 ++
 sound/soc/fsl/Makefile        |   2 +
 sound/soc/fsl/fsl-asoc-card.c | 573 ++++++++++++++++++++++++++++++++++++++++++
 3 files changed, 591 insertions(+)
 create mode 100644 sound/soc/fsl/fsl-asoc-card.c

(limited to 'sound')

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f54a8fc99291..2b99a9e86899 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -59,6 +59,22 @@ config SND_SOC_FSL_ESAI
 config SND_SOC_FSL_UTILS
 	tristate
 
+config SND_SOC_FSL_ASOC_CARD
+	tristate "Generic ASoC Sound Card with ASRC support"
+	depends on OF && I2C
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_FSL_ESAI
+	select SND_SOC_FSL_SAI
+	select SND_SOC_FSL_SSI
+	select SND_SOC_CS42XX8_I2C
+	select SND_SOC_SGTL5000
+	select SND_SOC_WM8962
+	help
+	 ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
+	 ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
+	 and SGTL5000.
+	 Say Y if you want to add support for Freescale Generic ASoC Sound Card.
+
 config SND_SOC_IMX_PCM_DMA
 	tristate
 	select SND_SOC_GENERIC_DMAENGINE_PCM
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 9ff59267eac9..8f6d84efa973 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -11,6 +11,7 @@ snd-soc-p1022-rdk-objs := p1022_rdk.o
 obj-$(CONFIG_SND_SOC_P1022_RDK) += snd-soc-p1022-rdk.o
 
 # Freescale SSI/DMA/SAI/SPDIF Support
+snd-soc-fsl-asoc-card-objs := fsl-asoc-card.o
 snd-soc-fsl-asrc-objs := fsl_asrc.o fsl_asrc_dma.o
 snd-soc-fsl-sai-objs := fsl_sai.o
 snd-soc-fsl-ssi-y := fsl_ssi.o
@@ -19,6 +20,7 @@ snd-soc-fsl-spdif-objs := fsl_spdif.o
 snd-soc-fsl-esai-objs := fsl_esai.o
 snd-soc-fsl-utils-objs := fsl_utils.o
 snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o
 obj-$(CONFIG_SND_SOC_FSL_ASRC) += snd-soc-fsl-asrc.o
 obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
 obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
new file mode 100644
index 000000000000..cf3f1f47f1e8
--- /dev/null
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -0,0 +1,573 @@
+/*
+ * Freescale Generic ASoC Sound Card driver with ASRC
+ *
+ * Copyright (C) 2014 Freescale Semiconductor, Inc.
+ *
+ * Author: Nicolin Chen <nicoleotsuka@gmail.com>
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2. This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "fsl_esai.h"
+#include "fsl_sai.h"
+#include "imx-audmux.h"
+
+#include "../codecs/sgtl5000.h"
+#include "../codecs/wm8962.h"
+
+#define RX 0
+#define TX 1
+
+/* Default DAI format without Master and Slave flag */
+#define DAI_FMT_BASE (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF)
+
+/**
+ * CODEC private data
+ *
+ * @mclk_freq: Clock rate of MCLK
+ * @mclk_id: MCLK (or main clock) id for set_sysclk()
+ * @fll_id: FLL (or secordary clock) id for set_sysclk()
+ * @pll_id: PLL id for set_pll()
+ */
+struct codec_priv {
+	unsigned long mclk_freq;
+	u32 mclk_id;
+	u32 fll_id;
+	u32 pll_id;
+};
+
+/**
+ * CPU private data
+ *
+ * @sysclk_freq[2]: SYSCLK rates for set_sysclk()
+ * @sysclk_dir[2]: SYSCLK directions for set_sysclk()
+ * @sysclk_id[2]: SYSCLK ids for set_sysclk()
+ *
+ * Note: [1] for tx and [0] for rx
+ */
+struct cpu_priv {
+	unsigned long sysclk_freq[2];
+	u32 sysclk_dir[2];
+	u32 sysclk_id[2];
+};
+
+/**
+ * Freescale Generic ASOC card private data
+ *
+ * @dai_link[3]: DAI link structure including normal one and DPCM link
+ * @pdev: platform device pointer
+ * @codec_priv: CODEC private data
+ * @cpu_priv: CPU private data
+ * @card: ASoC card structure
+ * @sample_rate: Current sample rate
+ * @sample_format: Current sample format
+ * @asrc_rate: ASRC sample rate used by Back-Ends
+ * @asrc_format: ASRC sample format used by Back-Ends
+ * @dai_fmt: DAI format between CPU and CODEC
+ * @name: Card name
+ */
+
+struct fsl_asoc_card_priv {
+	struct snd_soc_dai_link dai_link[3];
+	struct platform_device *pdev;
+	struct codec_priv codec_priv;
+	struct cpu_priv cpu_priv;
+	struct snd_soc_card card;
+	u32 sample_rate;
+	u32 sample_format;
+	u32 asrc_rate;
+	u32 asrc_format;
+	u32 dai_fmt;
+	char name[32];
+};
+
+/**
+ * This dapm route map exsits for DPCM link only.
+ * The other routes shall go through Device Tree.
+ */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"CPU-Playback",  NULL, "ASRC-Playback"},
+	{"Playback",  NULL, "CPU-Playback"},
+	{"ASRC-Capture",  NULL, "CPU-Capture"},
+	{"CPU-Capture",  NULL, "Capture"},
+};
+
+/* Add all possible widgets into here without being redundant */
+static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Line Out Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_MIC("AMIC", NULL),
+	SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream,
+				   struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	struct cpu_priv *cpu_priv = &priv->cpu_priv;
+	struct device *dev = rtd->card->dev;
+	int ret;
+
+	priv->sample_rate = params_rate(params);
+	priv->sample_format = params_format(params);
+
+	if (priv->card.set_bias_level)
+		return 0;
+
+	/* Specific configurations of DAIs starts from here */
+	ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, cpu_priv->sysclk_id[tx],
+				     cpu_priv->sysclk_freq[tx],
+				     cpu_priv->sysclk_dir[tx]);
+	if (ret) {
+		dev_err(dev, "failed to set sysclk for cpu dai\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops fsl_asoc_card_ops = {
+	.hw_params = fsl_asoc_card_hw_params,
+};
+
+static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+			      struct snd_pcm_hw_params *params)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	struct snd_interval *rate;
+	struct snd_mask *mask;
+
+	rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	rate->max = rate->min = priv->asrc_rate;
+
+	mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	snd_mask_none(mask);
+	snd_mask_set(mask, priv->asrc_format);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link fsl_asoc_card_dai[] = {
+	/* Default ASoC DAI Link*/
+	{
+		.name = "HiFi",
+		.stream_name = "HiFi",
+		.ops = &fsl_asoc_card_ops,
+	},
+	/* DPCM Link between Front-End and Back-End (Optional) */
+	{
+		.name = "HiFi-ASRC-FE",
+		.stream_name = "HiFi-ASRC-FE",
+		.codec_name = "snd-soc-dummy",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.dynamic = 1,
+	},
+	{
+		.name = "HiFi-ASRC-BE",
+		.stream_name = "HiFi-ASRC-BE",
+		.platform_name = "snd-soc-dummy",
+		.be_hw_params_fixup = be_hw_params_fixup,
+		.ops = &fsl_asoc_card_ops,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.no_pcm = 1,
+	},
+};
+
+static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card,
+					struct snd_soc_dapm_context *dapm,
+					enum snd_soc_bias_level level)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct codec_priv *codec_priv = &priv->codec_priv;
+	struct device *dev = card->dev;
+	unsigned int pll_out;
+	int ret;
+
+	if (dapm->dev != codec_dai->dev)
+		return 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_PREPARE:
+		if (dapm->bias_level != SND_SOC_BIAS_STANDBY)
+			break;
+
+		if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE)
+			pll_out = priv->sample_rate * 384;
+		else
+			pll_out = priv->sample_rate * 256;
+
+		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id,
+					  codec_priv->mclk_id,
+					  codec_priv->mclk_freq, pll_out);
+		if (ret) {
+			dev_err(dev, "failed to start FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id,
+					     pll_out, SND_SOC_CLOCK_IN);
+		if (ret) {
+			dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (dapm->bias_level != SND_SOC_BIAS_PREPARE)
+			break;
+
+		ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+					     codec_priv->mclk_freq,
+					     SND_SOC_CLOCK_IN);
+		if (ret) {
+			dev_err(dev, "failed to switch away from FLL: %d\n", ret);
+			return ret;
+		}
+
+		ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0);
+		if (ret) {
+			dev_err(dev, "failed to stop FLL: %d\n", ret);
+			return ret;
+		}
+		break;
+
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_audmux_init(struct device_node *np,
+				     struct fsl_asoc_card_priv *priv)
+{
+	struct device *dev = &priv->pdev->dev;
+	u32 int_ptcr = 0, ext_ptcr = 0;
+	int int_port, ext_port;
+	int ret;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(dev, "mux-int-port missing or invalid\n");
+		return ret;
+	}
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(dev, "mux-ext-port missing or invalid\n");
+		return ret;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the AUDMUX API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+
+	/*
+	 * Use asynchronous mode (6 wires) for all cases.
+	 * If only 4 wires are needed, just set SSI into
+	 * synchronous mode and enable 4 PADs in IOMUX.
+	 */
+	switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		int_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | ext_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR;
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		ext_ptcr = IMX_AUDMUX_V2_PTCR_RFSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_RCSEL(8 | int_port) |
+			   IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+			   IMX_AUDMUX_V2_PTCR_RFSDIR |
+			   IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			   IMX_AUDMUX_V2_PTCR_TFSDIR |
+			   IMX_AUDMUX_V2_PTCR_TCLKDIR;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Asynchronous mode can not be set along with RCLKDIR */
+	ret = imx_audmux_v2_configure_port(int_port, 0,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(int_port, int_ptcr,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(ext_port, 0,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr,
+					   IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_late_probe(struct snd_soc_card *card)
+{
+	struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card);
+	struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+	struct codec_priv *codec_priv = &priv->codec_priv;
+	struct device *dev = card->dev;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id,
+				     codec_priv->mclk_freq, SND_SOC_CLOCK_IN);
+	if (ret) {
+		dev_err(dev, "failed to set sysclk in %s\n", __func__);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int fsl_asoc_card_probe(struct platform_device *pdev)
+{
+	struct device_node *cpu_np, *codec_np, *asrc_np;
+	struct device_node *np = pdev->dev.of_node;
+	struct platform_device *asrc_pdev = NULL;
+	struct platform_device *cpu_pdev;
+	struct fsl_asoc_card_priv *priv;
+	struct i2c_client *codec_dev;
+	struct clk *codec_clk;
+	u32 width;
+	int ret;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
+	/* Give a chance to old DT binding */
+	if (!cpu_np)
+		cpu_np = of_parse_phandle(np, "ssi-controller", 0);
+	codec_np = of_parse_phandle(np, "audio-codec", 0);
+	if (!cpu_np || !codec_np) {
+		dev_err(&pdev->dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	cpu_pdev = of_find_device_by_node(cpu_np);
+	if (!cpu_pdev) {
+		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	codec_dev = of_find_i2c_device_by_node(codec_np);
+	if (!codec_dev) {
+		dev_err(&pdev->dev, "failed to find codec platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	asrc_np = of_parse_phandle(np, "audio-asrc", 0);
+	if (asrc_np)
+		asrc_pdev = of_find_device_by_node(asrc_np);
+
+	/* Get the MCLK rate only, and leave it controlled by CODEC drivers */
+	codec_clk = clk_get(&codec_dev->dev, NULL);
+	if (!IS_ERR(codec_clk)) {
+		priv->codec_priv.mclk_freq = clk_get_rate(codec_clk);
+		clk_put(codec_clk);
+	}
+
+	/* Default sample rate and format, will be updated in hw_params() */
+	priv->sample_rate = 44100;
+	priv->sample_format = SNDRV_PCM_FORMAT_S16_LE;
+
+	/* Assign a default DAI format, and allow each card to overwrite it */
+	priv->dai_fmt = DAI_FMT_BASE;
+
+	/* Diversify the card configurations */
+	if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) {
+		priv->card.set_bias_level = NULL;
+		priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq;
+		priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq;
+		priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT;
+		priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) {
+		priv->codec_priv.mclk_id = SGTL5000_SYSCLK;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+	} else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) {
+		priv->card.set_bias_level = fsl_asoc_card_set_bias_level;
+		priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK;
+		priv->codec_priv.fll_id = WM8962_SYSCLK_FLL;
+		priv->codec_priv.pll_id = WM8962_FLL;
+		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+	} else {
+		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
+		return -EINVAL;
+	}
+
+	/* Common settings for corresponding Freescale CPU DAI driver */
+	if (strstr(cpu_np->name, "ssi")) {
+		/* Only SSI needs to configure AUDMUX */
+		ret = fsl_asoc_card_audmux_init(np, priv);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to init audmux\n");
+			goto fail;
+		}
+	} else if (strstr(cpu_np->name, "esai")) {
+		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
+		priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL;
+	} else if (strstr(cpu_np->name, "sai")) {
+		priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1;
+		priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1;
+	}
+
+	sprintf(priv->name, "%s-audio", codec_dev->name);
+
+	/* Initialize sound card */
+	priv->pdev = pdev;
+	priv->card.dev = &pdev->dev;
+	priv->card.name = priv->name;
+	priv->card.dai_link = priv->dai_link;
+	priv->card.dapm_routes = audio_map;
+	priv->card.late_probe = fsl_asoc_card_late_probe;
+	priv->card.num_dapm_routes = ARRAY_SIZE(audio_map);
+	priv->card.dapm_widgets = fsl_asoc_card_dapm_widgets;
+	priv->card.num_dapm_widgets = ARRAY_SIZE(fsl_asoc_card_dapm_widgets);
+
+	memcpy(priv->dai_link, fsl_asoc_card_dai,
+	       sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link));
+
+	/* Normal DAI Link */
+	priv->dai_link[0].cpu_of_node = cpu_np;
+	priv->dai_link[0].codec_of_node = codec_np;
+	priv->dai_link[0].codec_dai_name = codec_dev->name;
+	priv->dai_link[0].platform_of_node = cpu_np;
+	priv->dai_link[0].dai_fmt = priv->dai_fmt;
+	priv->card.num_links = 1;
+
+	if (asrc_pdev) {
+		/* DPCM DAI Links only if ASRC exsits */
+		priv->dai_link[1].cpu_of_node = asrc_np;
+		priv->dai_link[1].platform_of_node = asrc_np;
+		priv->dai_link[2].codec_dai_name = codec_dev->name;
+		priv->dai_link[2].codec_of_node = codec_np;
+		priv->dai_link[2].cpu_of_node = cpu_np;
+		priv->dai_link[2].dai_fmt = priv->dai_fmt;
+		priv->card.num_links = 3;
+
+		ret = of_property_read_u32(asrc_np, "fsl,asrc-rate",
+					   &priv->asrc_rate);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to get output rate\n");
+			ret = -EINVAL;
+			goto fail;
+		}
+
+		ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
+		if (ret) {
+			dev_err(&pdev->dev, "failed to get output rate\n");
+			ret = -EINVAL;
+			goto fail;
+		}
+
+		if (width == 24)
+			priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
+		else
+			priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
+	}
+
+	/* Finish card registering */
+	platform_set_drvdata(pdev, priv);
+	snd_soc_card_set_drvdata(&priv->card, priv);
+
+	ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+
+fail:
+	of_node_put(codec_np);
+	of_node_put(asrc_np);
+	of_node_put(cpu_np);
+
+	return ret;
+}
+
+static const struct of_device_id fsl_asoc_card_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-cs42888", },
+	{ .compatible = "fsl,imx-audio-sgtl5000", },
+	{ .compatible = "fsl,imx-audio-wm8962", },
+	{}
+};
+
+static struct platform_driver fsl_asoc_card_driver = {
+	.probe = fsl_asoc_card_probe,
+	.driver = {
+		.name = "fsl-asoc-card",
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = fsl_asoc_card_dt_ids,
+	},
+};
+module_platform_driver(fsl_asoc_card_driver);
+
+MODULE_DESCRIPTION("Freescale Generic ASoC Sound Card driver with ASRC");
+MODULE_AUTHOR("Nicolin Chen <nicoleotsuka@gmail.com>");
+MODULE_ALIAS("platform:fsl-asoc-card");
+MODULE_LICENSE("GPL");
-- 
cgit v1.2.3


From de0d712a6dd1eed097dc6aa4f97ee461949414fe Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@freescale.com>
Date: Fri, 8 Aug 2014 14:47:21 +0800
Subject: ASoC: fsl_esai: refine esai for TDM support

Original driver didn't store the number of slots, just fix the slot number
to 2, use this default number to calculate bclk and pins for TX/RX.
In this patch, add one parameter for slots, and update the calculation of
bclk and pins of TX/RX. Then driver will be compatible with slots > 2 in
TDM mode.

Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_esai.c | 14 +++++++++++---
 sound/soc/fsl/fsl_esai.h |  8 ++++----
 2 files changed, 15 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 72d154e7dd03..f252370073e5 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -38,6 +38,7 @@
  * @fsysclk: system clock source to derive HCK, SCK and FS
  * @fifo_depth: depth of tx/rx FIFO
  * @slot_width: width of each DAI slot
+ * @slots: number of slots
  * @hck_rate: clock rate of desired HCKx clock
  * @sck_rate: clock rate of desired SCKx clock
  * @hck_dir: the direction of HCKx pads
@@ -56,6 +57,7 @@ struct fsl_esai {
 	struct clk *fsysclk;
 	u32 fifo_depth;
 	u32 slot_width;
+	u32 slots;
 	u32 hck_rate[2];
 	u32 sck_rate[2];
 	bool hck_dir[2];
@@ -363,6 +365,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask,
 			   ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask));
 
 	esai_priv->slot_width = slot_width;
+	esai_priv->slots = slots;
 
 	return 0;
 }
@@ -510,10 +513,11 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u32 width = snd_pcm_format_width(params_format(params));
 	u32 channels = params_channels(params);
+	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
 	u32 bclk, mask, val;
 	int ret;
 
-	bclk = params_rate(params) * esai_priv->slot_width * 2;
+	bclk = params_rate(params) * esai_priv->slot_width * esai_priv->slots;
 
 	ret = fsl_esai_set_bclk(dai, tx, bclk);
 	if (ret)
@@ -530,7 +534,7 @@ static int fsl_esai_hw_params(struct snd_pcm_substream *substream,
 	mask = ESAI_xFCR_xFR_MASK | ESAI_xFCR_xWA_MASK | ESAI_xFCR_xFWM_MASK |
 	      (tx ? ESAI_xFCR_TE_MASK | ESAI_xFCR_TIEN : ESAI_xFCR_RE_MASK);
 	val = ESAI_xFCR_xWA(width) | ESAI_xFCR_xFWM(esai_priv->fifo_depth) |
-	     (tx ? ESAI_xFCR_TE(channels) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(channels));
+	     (tx ? ESAI_xFCR_TE(pins) | ESAI_xFCR_TIEN : ESAI_xFCR_RE(pins));
 
 	regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), mask, val);
 
@@ -565,6 +569,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai);
 	bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
 	u8 i, channels = substream->runtime->channels;
+	u32 pins = DIV_ROUND_UP(channels, esai_priv->slots);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -579,7 +584,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd,
 
 		regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx),
 				   tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK,
-				   tx ? ESAI_xCR_TE(channels) : ESAI_xCR_RE(channels));
+				   tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins));
 		break;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
@@ -783,6 +788,9 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	/* Set a default slot size */
 	esai_priv->slot_width = 32;
 
+	/* Set a default slot number */
+	esai_priv->slots = 2;
+
 	/* Set a default master/slave state */
 	esai_priv->slave_mode = true;
 
diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h
index 75e14033e8d8..91a550f4a10d 100644
--- a/sound/soc/fsl/fsl_esai.h
+++ b/sound/soc/fsl/fsl_esai.h
@@ -130,8 +130,8 @@
 #define ESAI_xFCR_RE_WIDTH	4
 #define ESAI_xFCR_TE_MASK	(((1 << ESAI_xFCR_TE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
 #define ESAI_xFCR_RE_MASK	(((1 << ESAI_xFCR_RE_WIDTH) - 1) << ESAI_xFCR_xE_SHIFT)
-#define ESAI_xFCR_TE(x) 	((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_TE_MASK)
-#define ESAI_xFCR_RE(x) 	((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xFCR_RE_MASK)
+#define ESAI_xFCR_TE(x) 	((ESAI_xFCR_TE_MASK >> (ESAI_xFCR_TE_WIDTH - x)) & ESAI_xFCR_TE_MASK)
+#define ESAI_xFCR_RE(x) 	((ESAI_xFCR_RE_MASK >> (ESAI_xFCR_RE_WIDTH - x)) & ESAI_xFCR_RE_MASK)
 #define ESAI_xFCR_xFR_SHIFT	1
 #define ESAI_xFCR_xFR_MASK	(1 << ESAI_xFCR_xFR_SHIFT)
 #define ESAI_xFCR_xFR		(1 << ESAI_xFCR_xFR_SHIFT)
@@ -272,8 +272,8 @@
 #define ESAI_xCR_RE_WIDTH	4
 #define ESAI_xCR_TE_MASK	(((1 << ESAI_xCR_TE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
 #define ESAI_xCR_RE_MASK	(((1 << ESAI_xCR_RE_WIDTH) - 1) << ESAI_xCR_xE_SHIFT)
-#define ESAI_xCR_TE(x) 		((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_TE_MASK)
-#define ESAI_xCR_RE(x) 		((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - ((x + 1) >> 1))) & ESAI_xCR_RE_MASK)
+#define ESAI_xCR_TE(x) 		((ESAI_xCR_TE_MASK >> (ESAI_xCR_TE_WIDTH - x)) & ESAI_xCR_TE_MASK)
+#define ESAI_xCR_RE(x) 		((ESAI_xCR_RE_MASK >> (ESAI_xCR_RE_WIDTH - x)) & ESAI_xCR_RE_MASK)
 
 /*
  * Transmit Clock Control Register -- REG_ESAI_TCCR 0xD8
-- 
cgit v1.2.3


From 567e4f98922ce5542f8c2aa469a0c6ddf182b6ea Mon Sep 17 00:00:00 2001
From: Sean Cross <xobs@kosagi.com>
Date: Thu, 31 Jul 2014 10:43:36 +0800
Subject: ASoC: add es8328 codec driver

Add a codec driver for the Everest ES8328.  It supports two separate audio
outputs and two separate audio inputs.

Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/es8328.txt |  38 ++
 sound/soc/codecs/Kconfig                           |  13 +
 sound/soc/codecs/Makefile                          |   6 +
 sound/soc/codecs/es8328-i2c.c                      |  60 ++
 sound/soc/codecs/es8328-spi.c                      |  49 ++
 sound/soc/codecs/es8328.c                          | 756 +++++++++++++++++++++
 sound/soc/codecs/es8328.h                          | 314 +++++++++
 7 files changed, 1236 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/es8328.txt
 create mode 100644 sound/soc/codecs/es8328-i2c.c
 create mode 100644 sound/soc/codecs/es8328-spi.c
 create mode 100644 sound/soc/codecs/es8328.c
 create mode 100644 sound/soc/codecs/es8328.h

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/es8328.txt b/Documentation/devicetree/bindings/sound/es8328.txt
new file mode 100644
index 000000000000..30ea8a318ae9
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/es8328.txt
@@ -0,0 +1,38 @@
+Everest ES8328 audio CODEC
+
+This device supports both I2C and SPI.
+
+Required properties:
+
+  - compatible : "everest,es8328"
+  - DVDD-supply : Regulator providing digital core supply voltage 1.8 - 3.6V
+  - AVDD-supply : Regulator providing analog supply voltage 3.3V
+  - PVDD-supply : Regulator providing digital IO supply voltage 1.8 - 3.6V
+  - IPVDD-supply : Regulator providing analog output voltage 3.3V
+  - clocks : A 22.5792 or 11.2896 MHz clock
+  - reg : the I2C address of the device for I2C, the chip select number for SPI
+
+Pins on the device (for linking into audio routes):
+
+  * LOUT1
+  * LOUT2
+  * ROUT1
+  * ROUT2
+  * LINPUT1
+  * RINPUT1
+  * LINPUT2
+  * RINPUT2
+  * Mic Bias
+
+
+Example:
+
+codec: es8328@11 {
+	compatible = "everest,es8328";
+	DVDD-supply = <&reg_3p3v>;
+	AVDD-supply = <&reg_3p3v>;
+	PVDD-supply = <&reg_3p3v>;
+	HPVDD-supply = <&reg_3p3v>;
+	clocks = <&clks 169>;
+	reg = <0x11>;
+};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 8838838e25ed..8bca6343d8a3 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -57,6 +57,8 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_DA732X if I2C
 	select SND_SOC_DA9055 if I2C
 	select SND_SOC_BT_SCO
+	select SND_SOC_ES8328_SPI if SPI_MASTER
+	select SND_SOC_ES8328_I2C if I2C
 	select SND_SOC_ISABELLE if I2C
 	select SND_SOC_JZ4740_CODEC
 	select SND_SOC_LM4857 if I2C
@@ -405,6 +407,17 @@ config SND_SOC_DMIC
 config SND_SOC_HDMI_CODEC
        tristate "HDMI stub CODEC"
 
+config SND_SOC_ES8328
+	tristate "Everest Semi ES8328 CODEC"
+
+config SND_SOC_ES8328_I2C
+	tristate
+	select SND_SOC_ES8328
+
+config SND_SOC_ES8328_SPI
+	tristate
+	select SND_SOC_ES8328
+
 config SND_SOC_ISABELLE
         tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 20afe0f0c5be..31a8283006d1 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -49,6 +49,9 @@ snd-soc-da732x-objs := da732x.o
 snd-soc-da9055-objs := da9055.o
 snd-soc-bt-sco-objs := bt-sco.o
 snd-soc-dmic-objs := dmic.o
+snd-soc-es8328-objs := es8328.o
+snd-soc-es8328-i2c-objs := es8328-i2c.o
+snd-soc-es8328-spi-objs := es8328-spi.o
 snd-soc-isabelle-objs := isabelle.o
 snd-soc-jz4740-codec-objs := jz4740.o
 snd-soc-l3-objs := l3.o
@@ -220,6 +223,9 @@ obj-$(CONFIG_SND_SOC_DA732X)	+= snd-soc-da732x.o
 obj-$(CONFIG_SND_SOC_DA9055)	+= snd-soc-da9055.o
 obj-$(CONFIG_SND_SOC_BT_SCO)	+= snd-soc-bt-sco.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
+obj-$(CONFIG_SND_SOC_ES8328)	+= snd-soc-es8328.o
+obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o
+obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o
 obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
diff --git a/sound/soc/codecs/es8328-i2c.c b/sound/soc/codecs/es8328-i2c.c
new file mode 100644
index 000000000000..aae410d122ee
--- /dev/null
+++ b/sound/soc/codecs/es8328-i2c.c
@@ -0,0 +1,60 @@
+/*
+ * es8328-i2c.c  --  ES8328 ALSA SoC I2C Audio driver
+ *
+ * Copyright 2014 Sutajio Ko-Usagi PTE LTD
+ *
+ * Author: Sean Cross <xobs@kosagi.com>
+ *
+ * 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.
+ */
+
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+
+#include <sound/soc.h>
+
+#include "es8328.h"
+
+static const struct i2c_device_id es8328_id[] = {
+	{ "everest,es8328", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, es8328_id);
+
+static const struct of_device_id es8328_of_match[] = {
+	{ .compatible = "everest,es8328", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, es8328_of_match);
+
+static int es8328_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	return es8328_probe(&i2c->dev,
+			devm_regmap_init_i2c(i2c, &es8328_regmap_config));
+}
+
+static int es8328_i2c_remove(struct i2c_client *i2c)
+{
+	snd_soc_unregister_codec(&i2c->dev);
+	return 0;
+}
+
+static struct i2c_driver es8328_i2c_driver = {
+	.driver = {
+		.name		= "es8328",
+		.of_match_table = es8328_of_match,
+	},
+	.probe    = es8328_i2c_probe,
+	.remove   = es8328_i2c_remove,
+	.id_table = es8328_id,
+};
+
+module_i2c_driver(es8328_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC ES8328 audio CODEC I2C driver");
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/es8328-spi.c b/sound/soc/codecs/es8328-spi.c
new file mode 100644
index 000000000000..8fbd935e1c76
--- /dev/null
+++ b/sound/soc/codecs/es8328-spi.c
@@ -0,0 +1,49 @@
+/*
+ * es8328.c  --  ES8328 ALSA SoC SPI Audio driver
+ *
+ * Copyright 2014 Sutajio Ko-Usagi PTE LTD
+ *
+ * Author: Sean Cross <xobs@kosagi.com>
+ *
+ * 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.
+ */
+
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <linux/spi/spi.h>
+#include <sound/soc.h>
+#include "es8328.h"
+
+static const struct of_device_id es8328_of_match[] = {
+	{ .compatible = "everest,es8328", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, es8328_of_match);
+
+static int es8328_spi_probe(struct spi_device *spi)
+{
+	return es8328_probe(&spi->dev,
+			devm_regmap_init_spi(spi, &es8328_regmap_config));
+}
+
+static int es8328_spi_remove(struct spi_device *spi)
+{
+	snd_soc_unregister_codec(&spi->dev);
+	return 0;
+}
+
+static struct spi_driver es8328_spi_driver = {
+	.driver = {
+		.name		= "es8328",
+		.of_match_table	= es8328_of_match,
+	},
+	.probe	= es8328_spi_probe,
+	.remove	= es8328_spi_remove,
+};
+
+module_spi_driver(es8328_spi_driver);
+MODULE_DESCRIPTION("ASoC ES8328 audio CODEC SPI driver");
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c
new file mode 100644
index 000000000000..7a9f65ad183d
--- /dev/null
+++ b/sound/soc/codecs/es8328.c
@@ -0,0 +1,756 @@
+/*
+ * es8328.c  --  ES8328 ALSA SoC Audio driver
+ *
+ * Copyright 2014 Sutajio Ko-Usagi PTE LTD
+ *
+ * Author: Sean Cross <xobs@kosagi.com>
+ *
+ * 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.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/of_device.h>
+#include <linux/module.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include "es8328.h"
+
+#define ES8328_SYSCLK_RATE_1X 11289600
+#define ES8328_SYSCLK_RATE_2X 22579200
+
+/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */
+static struct {
+	int rate;
+	u8 ratio;
+} mclk_ratios[] = {
+	{ 8000, 9 },
+	{11025, 7 },
+	{22050, 4 },
+	{44100, 2 },
+};
+
+/* regulator supplies for sgtl5000, VDDD is an optional external supply */
+enum sgtl5000_regulator_supplies {
+	DVDD,
+	AVDD,
+	PVDD,
+	HPVDD,
+	ES8328_SUPPLY_NUM
+};
+
+/* vddd is optional supply */
+static const char * const supply_names[ES8328_SUPPLY_NUM] = {
+	"DVDD",
+	"AVDD",
+	"PVDD",
+	"HPVDD",
+};
+
+#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \
+		SNDRV_PCM_RATE_22050 | \
+		SNDRV_PCM_RATE_11025)
+#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE)
+
+struct es8328_priv {
+	struct regmap *regmap;
+	struct clk *clk;
+	int playback_fs;
+	bool deemph;
+	struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM];
+};
+
+/*
+ * ES8328 Controls
+ */
+
+static const char * const adcpol_txt[] = {"Normal", "L Invert", "R Invert",
+					  "L + R Invert"};
+static SOC_ENUM_SINGLE_DECL(adcpol,
+			    ES8328_ADCCONTROL6, 6, adcpol_txt);
+
+static const DECLARE_TLV_DB_SCALE(play_tlv, -3000, 100, 0);
+static const DECLARE_TLV_DB_SCALE(dac_adc_tlv, -9600, 50, 0);
+static const DECLARE_TLV_DB_SCALE(pga_tlv, 0, 300, 0);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 300, 0);
+
+static const int deemph_settings[] = { 0, 32000, 44100, 48000 };
+
+static int es8328_set_deemph(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int val, i, best;
+
+	/*
+	 * If we're using deemphasis select the nearest available sample
+	 * rate.
+	 */
+	if (es8328->deemph) {
+		best = 1;
+		for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
+			if (abs(deemph_settings[i] - es8328->playback_fs) <
+			    abs(deemph_settings[best] - es8328->playback_fs))
+				best = i;
+		}
+
+		val = best << 1;
+	} else {
+		val = 0;
+	}
+
+	dev_dbg(codec->dev, "Set deemphasis %d\n", val);
+
+	return snd_soc_update_bits(codec, ES8328_DACCONTROL6, 0x6, val);
+}
+
+static int es8328_get_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.enumerated.item[0] = es8328->deemph;
+	return 0;
+}
+
+static int es8328_put_deemph(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int deemph = ucontrol->value.enumerated.item[0];
+	int ret;
+
+	if (deemph > 1)
+		return -EINVAL;
+
+	ret = es8328_set_deemph(codec);
+	if (ret < 0)
+		return ret;
+
+	es8328->deemph = deemph;
+
+	return 0;
+}
+
+
+
+static const struct snd_kcontrol_new es8328_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Capture Digital Volume",
+		ES8328_ADCCONTROL8, ES8328_ADCCONTROL9,
+		 0, 0xc0, 1, dac_adc_tlv),
+	SOC_SINGLE("Capture ZC Switch", ES8328_ADCCONTROL7, 6, 1, 0),
+
+	SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0,
+		    es8328_get_deemph, es8328_put_deemph),
+
+	SOC_ENUM("Capture Polarity", adcpol),
+
+	SOC_SINGLE_TLV("Left Mixer Left Bypass Volume",
+			ES8328_DACCONTROL17, 3, 7, 1, bypass_tlv),
+	SOC_SINGLE_TLV("Left Mixer Right Bypass Volume",
+			ES8328_DACCONTROL19, 3, 7, 1, bypass_tlv),
+	SOC_SINGLE_TLV("Right Mixer Left Bypass Volume",
+			ES8328_DACCONTROL18, 3, 7, 1, bypass_tlv),
+	SOC_SINGLE_TLV("Right Mixer Right Bypass Volume",
+			ES8328_DACCONTROL20, 3, 7, 1, bypass_tlv),
+
+	SOC_DOUBLE_R_TLV("PCM Volume",
+			ES8328_LDACVOL, ES8328_RDACVOL,
+			0, ES8328_DACVOL_MAX, 1, dac_adc_tlv),
+
+	SOC_DOUBLE_R_TLV("Output 1 Playback Volume",
+			ES8328_LOUT1VOL, ES8328_ROUT1VOL,
+			0, ES8328_OUT1VOL_MAX, 0, play_tlv),
+
+	SOC_DOUBLE_R_TLV("Output 2 Playback Volume",
+			ES8328_LOUT2VOL, ES8328_ROUT2VOL,
+			0, ES8328_OUT2VOL_MAX, 0, play_tlv),
+
+	SOC_DOUBLE_TLV("Mic PGA Volume", ES8328_ADCCONTROL1,
+			4, 0, 8, 0, mic_tlv),
+};
+
+/*
+ * DAPM Controls
+ */
+
+static const char * const es8328_line_texts[] = {
+	"Line 1", "Line 2", "PGA", "Differential"};
+
+static const struct soc_enum es8328_lline_enum =
+	SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 3,
+			      ARRAY_SIZE(es8328_line_texts),
+			      es8328_line_texts);
+static const struct snd_kcontrol_new es8328_left_line_controls =
+	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+
+static const struct soc_enum es8328_rline_enum =
+	SOC_ENUM_SINGLE(ES8328_DACCONTROL16, 0,
+			      ARRAY_SIZE(es8328_line_texts),
+			      es8328_line_texts);
+static const struct snd_kcontrol_new es8328_right_line_controls =
+	SOC_DAPM_ENUM("Route", es8328_lline_enum);
+
+/* Left Mixer */
+static const struct snd_kcontrol_new es8328_left_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL17, 8, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL17, 7, 1, 0),
+	SOC_DAPM_SINGLE("Right Playback Switch", ES8328_DACCONTROL18, 8, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL18, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new es8328_right_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Left Playback Switch", ES8328_DACCONTROL19, 8, 1, 0),
+	SOC_DAPM_SINGLE("Left Bypass Switch", ES8328_DACCONTROL19, 7, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", ES8328_DACCONTROL20, 8, 1, 0),
+	SOC_DAPM_SINGLE("Right Bypass Switch", ES8328_DACCONTROL20, 7, 1, 0),
+};
+
+static const char * const es8328_pga_sel[] = {
+	"Line 1", "Line 2", "Line 3", "Differential"};
+
+/* Left PGA Mux */
+static const struct soc_enum es8328_lpga_enum =
+	SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 6,
+			      ARRAY_SIZE(es8328_pga_sel),
+			      es8328_pga_sel);
+static const struct snd_kcontrol_new es8328_left_pga_controls =
+	SOC_DAPM_ENUM("Route", es8328_lpga_enum);
+
+/* Right PGA Mux */
+static const struct soc_enum es8328_rpga_enum =
+	SOC_ENUM_SINGLE(ES8328_ADCCONTROL2, 4,
+			      ARRAY_SIZE(es8328_pga_sel),
+			      es8328_pga_sel);
+static const struct snd_kcontrol_new es8328_right_pga_controls =
+	SOC_DAPM_ENUM("Route", es8328_rpga_enum);
+
+/* Differential Mux */
+static const char * const es8328_diff_sel[] = {"Line 1", "Line 2"};
+static SOC_ENUM_SINGLE_DECL(diffmux,
+			    ES8328_ADCCONTROL3, 7, es8328_diff_sel);
+static const struct snd_kcontrol_new es8328_diffmux_controls =
+	SOC_DAPM_ENUM("Route", diffmux);
+
+/* Mono ADC Mux */
+static const char * const es8328_mono_mux[] = {"Stereo", "Mono (Left)",
+	"Mono (Right)", "Digital Mono"};
+static SOC_ENUM_SINGLE_DECL(monomux,
+			    ES8328_ADCCONTROL3, 3, es8328_mono_mux);
+static const struct snd_kcontrol_new es8328_monomux_controls =
+	SOC_DAPM_ENUM("Route", monomux);
+
+static const struct snd_soc_dapm_widget es8328_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_diffmux_controls),
+	SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_monomux_controls),
+	SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_monomux_controls),
+
+	SND_SOC_DAPM_MUX("Left PGA Mux", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_AINL_OFF, 1,
+			&es8328_left_pga_controls),
+	SND_SOC_DAPM_MUX("Right PGA Mux", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_AINR_OFF, 1,
+			&es8328_right_pga_controls),
+
+	SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_left_line_controls),
+	SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+		&es8328_right_line_controls),
+
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_ADCR_OFF, 1),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_ADCL_OFF, 1),
+
+	SND_SOC_DAPM_SUPPLY("Mic Bias", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_MIC_BIAS_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Mic Bias Gen", ES8328_ADCPOWER,
+			ES8328_ADCPOWER_ADC_BIAS_GEN_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DAC STM", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACSTM_RESET, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC STM", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCSTM_RESET, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DAC DIG", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACDIG_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC DIG", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCDIG_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("DAC DLL", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACDLL_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("ADC DLL", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCDLL_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_SUPPLY("ADC Vref", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_ADCVREF_OFF, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DAC Vref", ES8328_CHIPPOWER,
+			ES8328_CHIPPOWER_DACVREF_OFF, 1, NULL, 0),
+
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback", ES8328_DACPOWER,
+			ES8328_DACPOWER_RDAC_OFF, 1),
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", ES8328_DACPOWER,
+			ES8328_DACPOWER_LDAC_OFF, 1),
+
+	SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+		&es8328_left_mixer_controls[0],
+		ARRAY_SIZE(es8328_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+		&es8328_right_mixer_controls[0],
+		ARRAY_SIZE(es8328_right_mixer_controls)),
+
+	SND_SOC_DAPM_PGA("Right Out 2", ES8328_DACPOWER,
+			ES8328_DACPOWER_ROUT2_ON, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 2", ES8328_DACPOWER,
+			ES8328_DACPOWER_LOUT2_ON, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Out 1", ES8328_DACPOWER,
+			ES8328_DACPOWER_ROUT1_ON, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Left Out 1", ES8328_DACPOWER,
+			ES8328_DACPOWER_LOUT1_ON, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LOUT1"),
+	SND_SOC_DAPM_OUTPUT("ROUT1"),
+	SND_SOC_DAPM_OUTPUT("LOUT2"),
+	SND_SOC_DAPM_OUTPUT("ROUT2"),
+
+	SND_SOC_DAPM_INPUT("LINPUT1"),
+	SND_SOC_DAPM_INPUT("LINPUT2"),
+	SND_SOC_DAPM_INPUT("RINPUT1"),
+	SND_SOC_DAPM_INPUT("RINPUT2"),
+};
+
+static const struct snd_soc_dapm_route es8328_dapm_routes[] = {
+
+	{ "Left Line Mux", "Line 1", "LINPUT1" },
+	{ "Left Line Mux", "Line 2", "LINPUT2" },
+	{ "Left Line Mux", "PGA", "Left PGA Mux" },
+	{ "Left Line Mux", "Differential", "Differential Mux" },
+
+	{ "Right Line Mux", "Line 1", "RINPUT1" },
+	{ "Right Line Mux", "Line 2", "RINPUT2" },
+	{ "Right Line Mux", "PGA", "Right PGA Mux" },
+	{ "Right Line Mux", "Differential", "Differential Mux" },
+
+	{ "Left PGA Mux", "Line 1", "LINPUT1" },
+	{ "Left PGA Mux", "Line 2", "LINPUT2" },
+	{ "Left PGA Mux", "Differential", "Differential Mux" },
+
+	{ "Right PGA Mux", "Line 1", "RINPUT1" },
+	{ "Right PGA Mux", "Line 2", "RINPUT2" },
+	{ "Right PGA Mux", "Differential", "Differential Mux" },
+
+	{ "Differential Mux", "Line 1", "LINPUT1" },
+	{ "Differential Mux", "Line 1", "RINPUT1" },
+	{ "Differential Mux", "Line 2", "LINPUT2" },
+	{ "Differential Mux", "Line 2", "RINPUT2" },
+
+	{ "Left ADC Mux", "Stereo", "Left PGA Mux" },
+	{ "Left ADC Mux", "Mono (Left)", "Left PGA Mux" },
+	{ "Left ADC Mux", "Digital Mono", "Left PGA Mux" },
+
+	{ "Right ADC Mux", "Stereo", "Right PGA Mux" },
+	{ "Right ADC Mux", "Mono (Right)", "Right PGA Mux" },
+	{ "Right ADC Mux", "Digital Mono", "Right PGA Mux" },
+
+	{ "Left ADC", NULL, "Left ADC Mux" },
+	{ "Right ADC", NULL, "Right ADC Mux" },
+
+	{ "ADC DIG", NULL, "ADC STM" },
+	{ "ADC DIG", NULL, "ADC Vref" },
+	{ "ADC DIG", NULL, "ADC DLL" },
+
+	{ "Left ADC", NULL, "ADC DIG" },
+	{ "Right ADC", NULL, "ADC DIG" },
+
+	{ "Mic Bias", NULL, "Mic Bias Gen" },
+
+	{ "Left Line Mux", "Line 1", "LINPUT1" },
+	{ "Left Line Mux", "Line 2", "LINPUT2" },
+	{ "Left Line Mux", "PGA", "Left PGA Mux" },
+	{ "Left Line Mux", "Differential", "Differential Mux" },
+
+	{ "Right Line Mux", "Line 1", "RINPUT1" },
+	{ "Right Line Mux", "Line 2", "RINPUT2" },
+	{ "Right Line Mux", "PGA", "Right PGA Mux" },
+	{ "Right Line Mux", "Differential", "Differential Mux" },
+
+	{ "Left Out 1", NULL, "Left DAC" },
+	{ "Right Out 1", NULL, "Right DAC" },
+	{ "Left Out 2", NULL, "Left DAC" },
+	{ "Right Out 2", NULL, "Right DAC" },
+
+	{ "Left Mixer", "Playback Switch", "Left DAC" },
+	{ "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
+	{ "Left Mixer", "Right Playback Switch", "Right DAC" },
+	{ "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+	{ "Right Mixer", "Left Playback Switch", "Left DAC" },
+	{ "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
+	{ "Right Mixer", "Playback Switch", "Right DAC" },
+	{ "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+	{ "DAC DIG", NULL, "DAC STM" },
+	{ "DAC DIG", NULL, "DAC Vref" },
+	{ "DAC DIG", NULL, "DAC DLL" },
+
+	{ "Left DAC", NULL, "DAC DIG" },
+	{ "Right DAC", NULL, "DAC DIG" },
+
+	{ "Left Out 1", NULL, "Left Mixer" },
+	{ "LOUT1", NULL, "Left Out 1" },
+	{ "Right Out 1", NULL, "Right Mixer" },
+	{ "ROUT1", NULL, "Right Out 1" },
+
+	{ "Left Out 2", NULL, "Left Mixer" },
+	{ "LOUT2", NULL, "Left Out 2" },
+	{ "Right Out 2", NULL, "Right Mixer" },
+	{ "ROUT2", NULL, "Right Out 2" },
+};
+
+static int es8328_mute(struct snd_soc_dai *dai, int mute)
+{
+	return snd_soc_update_bits(dai->codec, ES8328_DACCONTROL3,
+			ES8328_DACCONTROL3_DACMUTE,
+			mute ? ES8328_DACCONTROL3_DACMUTE : 0);
+}
+
+static int es8328_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int clk_rate;
+	int i;
+	int reg;
+	u8 ratio;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = ES8328_DACCONTROL2;
+	else
+		reg = ES8328_ADCCONTROL5;
+
+	clk_rate = clk_get_rate(es8328->clk);
+
+	if ((clk_rate != ES8328_SYSCLK_RATE_1X) &&
+		(clk_rate != ES8328_SYSCLK_RATE_2X)) {
+		dev_err(codec->dev,
+			"%s: clock is running at %d Hz, not %d or %d Hz\n",
+			 __func__, clk_rate,
+			 ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X);
+		return -EINVAL;
+	}
+
+	/* find master mode MCLK to sampling frequency ratio */
+	ratio = mclk_ratios[0].rate;
+	for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++)
+		if (params_rate(params) <= mclk_ratios[i].rate)
+			ratio = mclk_ratios[i].ratio;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		es8328->playback_fs = params_rate(params);
+		es8328_set_deemph(codec);
+	}
+
+	return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio);
+}
+
+static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec);
+	int clk_rate;
+	u8 mode = ES8328_DACCONTROL1_DACWL_16;
+
+	/* set master/slave audio interface */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM)
+		return -EINVAL;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		mode |= ES8328_DACCONTROL1_DACFORMAT_I2S;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF)
+		return -EINVAL;
+
+	snd_soc_write(codec, ES8328_DACCONTROL1, mode);
+	snd_soc_write(codec, ES8328_ADCCONTROL4, mode);
+
+	/* Master serial port mode, with BCLK generated automatically */
+	clk_rate = clk_get_rate(es8328->clk);
+	if (clk_rate == ES8328_SYSCLK_RATE_1X)
+		snd_soc_write(codec, ES8328_MASTERMODE,
+				ES8328_MASTERMODE_MSC);
+	else
+		snd_soc_write(codec, ES8328_MASTERMODE,
+				ES8328_MASTERMODE_MCLKDIV2 |
+				ES8328_MASTERMODE_MSC);
+
+	return 0;
+}
+
+static int es8328_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* VREF, VMID=2x50k, digital enabled */
+		snd_soc_write(codec, ES8328_CHIPPOWER, 0);
+		snd_soc_update_bits(codec, ES8328_CONTROL1,
+				ES8328_CONTROL1_VMIDSEL_MASK |
+				ES8328_CONTROL1_ENREF,
+				ES8328_CONTROL1_VMIDSEL_50k |
+				ES8328_CONTROL1_ENREF);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_update_bits(codec, ES8328_CONTROL1,
+					ES8328_CONTROL1_VMIDSEL_MASK |
+					ES8328_CONTROL1_ENREF,
+					ES8328_CONTROL1_VMIDSEL_5k |
+					ES8328_CONTROL1_ENREF);
+
+			/* Charge caps */
+			msleep(100);
+		}
+
+		snd_soc_write(codec, ES8328_CONTROL2,
+				ES8328_CONTROL2_OVERCURRENT_ON |
+				ES8328_CONTROL2_THERMAL_SHUTDOWN_ON);
+
+		/* VREF, VMID=2*500k, digital stopped */
+		snd_soc_update_bits(codec, ES8328_CONTROL1,
+				ES8328_CONTROL1_VMIDSEL_MASK |
+				ES8328_CONTROL1_ENREF,
+				ES8328_CONTROL1_VMIDSEL_500k |
+				ES8328_CONTROL1_ENREF);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, ES8328_CONTROL1,
+				ES8328_CONTROL1_VMIDSEL_MASK |
+				ES8328_CONTROL1_ENREF,
+				0);
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+static const struct snd_soc_dai_ops es8328_dai_ops = {
+	.hw_params	= es8328_hw_params,
+	.digital_mute	= es8328_mute,
+	.set_fmt	= es8328_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver es8328_dai = {
+	.name = "es8328-hifi-analog",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = ES8328_RATES,
+		.formats = ES8328_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = ES8328_RATES,
+		.formats = ES8328_FORMATS,
+	},
+	.ops = &es8328_dai_ops,
+};
+
+static int es8328_suspend(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328;
+	int ret;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	es8328_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	clk_disable_unprepare(es8328->clk);
+
+	ret = regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
+			es8328->supplies);
+	if (ret) {
+		dev_err(codec->dev, "unable to disable regulators\n");
+		return ret;
+	}
+	return 0;
+}
+
+static int es8328_resume(struct snd_soc_codec *codec)
+{
+	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+	struct es8328_priv *es8328;
+	int ret;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	ret = clk_prepare_enable(es8328->clk);
+	if (ret) {
+		dev_err(codec->dev, "unable to enable clock\n");
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies),
+					es8328->supplies);
+	if (ret) {
+		dev_err(codec->dev, "unable to enable regulators\n");
+		return ret;
+	}
+
+	regcache_mark_dirty(regmap);
+	ret = regcache_sync(regmap);
+	if (ret) {
+		dev_err(codec->dev, "unable to sync regcache\n");
+		return ret;
+	}
+
+	es8328_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+static int es8328_codec_probe(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328;
+	int ret;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(es8328->supplies),
+					es8328->supplies);
+	if (ret) {
+		dev_err(codec->dev, "unable to enable regulators\n");
+		return ret;
+	}
+
+	/* Setup clocks */
+	es8328->clk = devm_clk_get(codec->dev, NULL);
+	if (IS_ERR(es8328->clk)) {
+		dev_err(codec->dev, "codec clock missing or invalid\n");
+		goto clk_fail;
+	}
+
+	ret = clk_prepare_enable(es8328->clk);
+	if (ret) {
+		dev_err(codec->dev, "unable to prepare codec clk\n");
+		goto clk_fail;
+	}
+
+	return 0;
+
+clk_fail:
+	regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
+			       es8328->supplies);
+	return ret;
+}
+
+static int es8328_remove(struct snd_soc_codec *codec)
+{
+	struct es8328_priv *es8328;
+
+	es8328 = snd_soc_codec_get_drvdata(codec);
+
+	if (es8328->clk)
+		clk_disable_unprepare(es8328->clk);
+
+	regulator_bulk_disable(ARRAY_SIZE(es8328->supplies),
+			       es8328->supplies);
+
+	return 0;
+}
+
+const struct regmap_config es8328_regmap_config = {
+	.reg_bits	= 8,
+	.val_bits	= 8,
+	.max_register	= ES8328_REG_MAX,
+	.cache_type	= REGCACHE_RBTREE,
+};
+EXPORT_SYMBOL_GPL(es8328_regmap_config);
+
+static struct snd_soc_codec_driver es8328_codec_driver = {
+	.probe		  = es8328_codec_probe,
+	.suspend	  = es8328_suspend,
+	.resume		  = es8328_resume,
+	.remove		  = es8328_remove,
+	.set_bias_level	  = es8328_set_bias_level,
+	.controls	  = es8328_snd_controls,
+	.num_controls	  = ARRAY_SIZE(es8328_snd_controls),
+	.dapm_widgets	  = es8328_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(es8328_dapm_widgets),
+	.dapm_routes	  = es8328_dapm_routes,
+	.num_dapm_routes  = ARRAY_SIZE(es8328_dapm_routes),
+};
+
+int es8328_probe(struct device *dev, struct regmap *regmap)
+{
+	struct es8328_priv *es8328;
+	int ret;
+	int i;
+
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	es8328 = devm_kzalloc(dev, sizeof(*es8328), GFP_KERNEL);
+	if (es8328 == NULL)
+		return -ENOMEM;
+
+	es8328->regmap = regmap;
+
+	for (i = 0; i < ARRAY_SIZE(es8328->supplies); i++)
+		es8328->supplies[i].supply = supply_names[i];
+
+	ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(es8328->supplies),
+				es8328->supplies);
+	if (ret) {
+		dev_err(dev, "unable to get regulators\n");
+		return ret;
+	}
+
+	dev_set_drvdata(dev, es8328);
+
+	return snd_soc_register_codec(dev,
+			&es8328_codec_driver, &es8328_dai, 1);
+}
+EXPORT_SYMBOL_GPL(es8328_probe);
+
+MODULE_DESCRIPTION("ASoC ES8328 driver");
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h
new file mode 100644
index 000000000000..cb36afe10c0e
--- /dev/null
+++ b/sound/soc/codecs/es8328.h
@@ -0,0 +1,314 @@
+/*
+ * es8328.h  --  ES8328 ALSA SoC Audio driver
+ */
+
+#ifndef _ES8328_H
+#define _ES8328_H
+
+#include <linux/regmap.h>
+
+struct device;
+
+extern const struct regmap_config es8328_regmap_config;
+int es8328_probe(struct device *dev, struct regmap *regmap);
+
+#define ES8328_DACLVOL 46
+#define ES8328_DACRVOL 47
+#define ES8328_DACCTL 28
+#define ES8328_RATEMASK (0x1f << 0)
+
+#define ES8328_CONTROL1		0x00
+#define ES8328_CONTROL1_VMIDSEL_OFF (0 << 0)
+#define ES8328_CONTROL1_VMIDSEL_50k (1 << 0)
+#define ES8328_CONTROL1_VMIDSEL_500k (2 << 0)
+#define ES8328_CONTROL1_VMIDSEL_5k (3 << 0)
+#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0)
+#define ES8328_CONTROL1_ENREF (1 << 2)
+#define ES8328_CONTROL1_SEQEN (1 << 3)
+#define ES8328_CONTROL1_SAMEFS (1 << 4)
+#define ES8328_CONTROL1_DACMCLK_ADC (0 << 5)
+#define ES8328_CONTROL1_DACMCLK_DAC (1 << 5)
+#define ES8328_CONTROL1_LRCM (1 << 6)
+#define ES8328_CONTROL1_SCP_RESET (1 << 7)
+
+#define ES8328_CONTROL2		0x01
+#define ES8328_CONTROL2_VREF_BUF_OFF (1 << 0)
+#define ES8328_CONTROL2_VREF_LOWPOWER (1 << 1)
+#define ES8328_CONTROL2_IBIASGEN_OFF (1 << 2)
+#define ES8328_CONTROL2_ANALOG_OFF (1 << 3)
+#define ES8328_CONTROL2_VREF_BUF_LOWPOWER (1 << 4)
+#define ES8328_CONTROL2_VCM_MOD_LOWPOWER (1 << 5)
+#define ES8328_CONTROL2_OVERCURRENT_ON (1 << 6)
+#define ES8328_CONTROL2_THERMAL_SHUTDOWN_ON (1 << 7)
+
+#define ES8328_CHIPPOWER	0x02
+#define ES8328_CHIPPOWER_DACVREF_OFF 0
+#define ES8328_CHIPPOWER_ADCVREF_OFF 1
+#define ES8328_CHIPPOWER_DACDLL_OFF 2
+#define ES8328_CHIPPOWER_ADCDLL_OFF 3
+#define ES8328_CHIPPOWER_DACSTM_RESET 4
+#define ES8328_CHIPPOWER_ADCSTM_RESET 5
+#define ES8328_CHIPPOWER_DACDIG_OFF 6
+#define ES8328_CHIPPOWER_ADCDIG_OFF 7
+
+#define ES8328_ADCPOWER		0x03
+#define ES8328_ADCPOWER_INT1_LOWPOWER 0
+#define ES8328_ADCPOWER_FLASH_ADC_LOWPOWER 1
+#define ES8328_ADCPOWER_ADC_BIAS_GEN_OFF 2
+#define ES8328_ADCPOWER_MIC_BIAS_OFF 3
+#define ES8328_ADCPOWER_ADCR_OFF 4
+#define ES8328_ADCPOWER_ADCL_OFF 5
+#define ES8328_ADCPOWER_AINR_OFF 6
+#define ES8328_ADCPOWER_AINL_OFF 7
+
+#define ES8328_DACPOWER		0x04
+#define ES8328_DACPOWER_OUT3_ON 0
+#define ES8328_DACPOWER_MONO_ON 1
+#define ES8328_DACPOWER_ROUT2_ON 2
+#define ES8328_DACPOWER_LOUT2_ON 3
+#define ES8328_DACPOWER_ROUT1_ON 4
+#define ES8328_DACPOWER_LOUT1_ON 5
+#define ES8328_DACPOWER_RDAC_OFF 6
+#define ES8328_DACPOWER_LDAC_OFF 7
+
+#define ES8328_CHIPLOPOW1	0x05
+#define ES8328_CHIPLOPOW2	0x06
+#define ES8328_ANAVOLMANAG	0x07
+
+#define ES8328_MASTERMODE	0x08
+#define ES8328_MASTERMODE_BCLKDIV (0 << 0)
+#define ES8328_MASTERMODE_BCLK_INV (1 << 5)
+#define ES8328_MASTERMODE_MCLKDIV2 (1 << 6)
+#define ES8328_MASTERMODE_MSC (1 << 7)
+
+#define ES8328_ADCCONTROL1	0x09
+#define ES8328_ADCCONTROL2	0x0a
+#define ES8328_ADCCONTROL3	0x0b
+#define ES8328_ADCCONTROL4	0x0c
+#define ES8328_ADCCONTROL5	0x0d
+#define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0)
+
+#define ES8328_ADCCONTROL6	0x0e
+
+#define ES8328_ADCCONTROL7	0x0f
+#define ES8328_ADCCONTROL7_ADC_MUTE (1 << 2)
+#define ES8328_ADCCONTROL7_ADC_LER (1 << 3)
+#define ES8328_ADCCONTROL7_ADC_ZERO_CROSS (1 << 4)
+#define ES8328_ADCCONTROL7_ADC_SOFT_RAMP (1 << 5)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_4 (0 << 6)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_8 (1 << 6)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_16 (2 << 6)
+#define ES8328_ADCCONTROL7_ADC_RAMP_RATE_32 (3 << 6)
+
+#define ES8328_ADCCONTROL8	0x10
+#define ES8328_ADCCONTROL9	0x11
+#define ES8328_ADCCONTROL10	0x12
+#define ES8328_ADCCONTROL11	0x13
+#define ES8328_ADCCONTROL12	0x14
+#define ES8328_ADCCONTROL13	0x15
+#define ES8328_ADCCONTROL14	0x16
+
+#define ES8328_DACCONTROL1	0x17
+#define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1)
+#define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1)
+#define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1)
+#define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1)
+#define ES8328_DACCONTROL1_DACWL_24 (0 << 3)
+#define ES8328_DACCONTROL1_DACWL_20 (1 << 3)
+#define ES8328_DACCONTROL1_DACWL_18 (2 << 3)
+#define ES8328_DACCONTROL1_DACWL_16 (3 << 3)
+#define ES8328_DACCONTROL1_DACWL_32 (4 << 3)
+#define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6)
+#define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6)
+#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6)
+#define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK1 (1 << 6)
+#define ES8328_DACCONTROL1_LRSWAP (1 << 7)
+
+#define ES8328_DACCONTROL2	0x18
+#define ES8328_DACCONTROL2_RATEMASK (0x1f << 0)
+#define ES8328_DACCONTROL2_DOUBLESPEED (1 << 5)
+
+#define ES8328_DACCONTROL3	0x19
+#define ES8328_DACCONTROL3_AUTOMUTE (1 << 2)
+#define ES8328_DACCONTROL3_DACMUTE (1 << 2)
+#define ES8328_DACCONTROL3_LEFTGAINVOL (1 << 3)
+#define ES8328_DACCONTROL3_DACZEROCROSS (1 << 4)
+#define ES8328_DACCONTROL3_DACSOFTRAMP (1 << 5)
+#define ES8328_DACCONTROL3_DACRAMPRATE (3 << 6)
+
+#define ES8328_LDACVOL 0x1a
+#define ES8328_LDACVOL_MASK (0 << 0)
+#define ES8328_LDACVOL_MAX (0xc0)
+
+#define ES8328_RDACVOL 0x1b
+#define ES8328_RDACVOL_MASK (0 << 0)
+#define ES8328_RDACVOL_MAX (0xc0)
+
+#define ES8328_DACVOL_MAX (0xc0)
+
+#define ES8328_DACCONTROL4	0x1a
+#define ES8328_DACCONTROL5	0x1b
+
+#define ES8328_DACCONTROL6	0x1c
+#define ES8328_DACCONTROL6_CLICKFREE (1 << 3)
+#define ES8328_DACCONTROL6_DAC_INVR (1 << 4)
+#define ES8328_DACCONTROL6_DAC_INVL (1 << 5)
+#define ES8328_DACCONTROL6_DEEMPH_OFF (0 << 6)
+#define ES8328_DACCONTROL6_DEEMPH_32k (1 << 6)
+#define ES8328_DACCONTROL6_DEEMPH_44_1k (2 << 6)
+#define ES8328_DACCONTROL6_DEEMPH_48k (3 << 6)
+
+#define ES8328_DACCONTROL7	0x1d
+#define ES8328_DACCONTROL7_VPP_SCALE_3p5	(0 << 0)
+#define ES8328_DACCONTROL7_VPP_SCALE_4p0	(1 << 0)
+#define ES8328_DACCONTROL7_VPP_SCALE_3p0	(2 << 0)
+#define ES8328_DACCONTROL7_VPP_SCALE_2p5	(3 << 0)
+#define ES8328_DACCONTROL7_SHELVING_STRENGTH (1 << 2) /* In eights */
+#define ES8328_DACCONTROL7_MONO		(1 << 5)
+#define ES8328_DACCONTROL7_ZEROR	(1 << 6)
+#define ES8328_DACCONTROL7_ZEROL	(1 << 7)
+
+/* Shelving filter */
+#define ES8328_DACCONTROL8	0x1e
+#define ES8328_DACCONTROL9	0x1f
+#define ES8328_DACCONTROL10	0x20
+#define ES8328_DACCONTROL11	0x21
+#define ES8328_DACCONTROL12	0x22
+#define ES8328_DACCONTROL13	0x23
+#define ES8328_DACCONTROL14	0x24
+#define ES8328_DACCONTROL15	0x25
+
+#define ES8328_DACCONTROL16	0x26
+#define ES8328_DACCONTROL16_RMIXSEL_RIN1 (0 << 0)
+#define ES8328_DACCONTROL16_RMIXSEL_RIN2 (1 << 0)
+#define ES8328_DACCONTROL16_RMIXSEL_RIN3 (2 << 0)
+#define ES8328_DACCONTROL16_RMIXSEL_RADC (3 << 0)
+#define ES8328_DACCONTROL16_LMIXSEL_LIN1 (0 << 3)
+#define ES8328_DACCONTROL16_LMIXSEL_LIN2 (1 << 3)
+#define ES8328_DACCONTROL16_LMIXSEL_LIN3 (2 << 3)
+#define ES8328_DACCONTROL16_LMIXSEL_LADC (3 << 3)
+
+#define ES8328_DACCONTROL17	0x27
+#define ES8328_DACCONTROL17_LI2LOVOL (7 << 3)
+#define ES8328_DACCONTROL17_LI2LO (1 << 6)
+#define ES8328_DACCONTROL17_LD2LO (1 << 7)
+
+#define ES8328_DACCONTROL18	0x28
+#define ES8328_DACCONTROL18_RI2LOVOL (7 << 3)
+#define ES8328_DACCONTROL18_RI2LO (1 << 6)
+#define ES8328_DACCONTROL18_RD2LO (1 << 7)
+
+#define ES8328_DACCONTROL19	0x29
+#define ES8328_DACCONTROL19_LI2ROVOL (7 << 3)
+#define ES8328_DACCONTROL19_LI2RO (1 << 6)
+#define ES8328_DACCONTROL19_LD2RO (1 << 7)
+
+#define ES8328_DACCONTROL20	0x2a
+#define ES8328_DACCONTROL20_RI2ROVOL (7 << 3)
+#define ES8328_DACCONTROL20_RI2RO (1 << 6)
+#define ES8328_DACCONTROL20_RD2RO (1 << 7)
+
+#define ES8328_DACCONTROL21	0x2b
+#define ES8328_DACCONTROL21_LI2MOVOL (7 << 3)
+#define ES8328_DACCONTROL21_LI2MO (1 << 6)
+#define ES8328_DACCONTROL21_LD2MO (1 << 7)
+
+#define ES8328_DACCONTROL22	0x2c
+#define ES8328_DACCONTROL22_RI2MOVOL (7 << 3)
+#define ES8328_DACCONTROL22_RI2MO (1 << 6)
+#define ES8328_DACCONTROL22_RD2MO (1 << 7)
+
+#define ES8328_DACCONTROL23	0x2d
+#define ES8328_DACCONTROL23_MOUTINV		(1 << 1)
+#define ES8328_DACCONTROL23_HPSWPOL		(1 << 2)
+#define ES8328_DACCONTROL23_HPSWEN		(1 << 3)
+#define ES8328_DACCONTROL23_VROI_1p5k		(0 << 4)
+#define ES8328_DACCONTROL23_VROI_40k		(1 << 4)
+#define ES8328_DACCONTROL23_OUT3_VREF		(0 << 5)
+#define ES8328_DACCONTROL23_OUT3_ROUT1		(1 << 5)
+#define ES8328_DACCONTROL23_OUT3_MONOOUT	(2 << 5)
+#define ES8328_DACCONTROL23_OUT3_RIGHT_MIXER	(3 << 5)
+#define ES8328_DACCONTROL23_ROUT2INV		(1 << 7)
+
+/* LOUT1 Amplifier */
+#define ES8328_LOUT1VOL 0x2e
+#define ES8328_LOUT1VOL_MASK (0 << 5)
+#define ES8328_LOUT1VOL_MAX (0x24)
+
+/* ROUT1 Amplifier */
+#define ES8328_ROUT1VOL 0x2f
+#define ES8328_ROUT1VOL_MASK (0 << 5)
+#define ES8328_ROUT1VOL_MAX (0x24)
+
+#define ES8328_OUT1VOL_MAX (0x24)
+
+/* LOUT2 Amplifier */
+#define ES8328_LOUT2VOL 0x30
+#define ES8328_LOUT2VOL_MASK (0 << 5)
+#define ES8328_LOUT2VOL_MAX (0x24)
+
+/* ROUT2 Amplifier */
+#define ES8328_ROUT2VOL 0x31
+#define ES8328_ROUT2VOL_MASK (0 << 5)
+#define ES8328_ROUT2VOL_MAX (0x24)
+
+#define ES8328_OUT2VOL_MAX (0x24)
+
+/* Mono Out Amplifier */
+#define ES8328_MONOOUTVOL 0x32
+#define ES8328_MONOOUTVOL_MASK (0 << 5)
+#define ES8328_MONOOUTVOL_MAX (0x24)
+
+#define ES8328_DACCONTROL29	0x33
+#define ES8328_DACCONTROL30	0x34
+
+#define ES8328_SYSCLK		0
+
+#define ES8328_REG_MAX		0x35
+
+#define ES8328_PLL1		0
+#define ES8328_PLL2		1
+
+/* clock inputs */
+#define ES8328_MCLK		0
+#define ES8328_PCMCLK		1
+
+/* clock divider id's */
+#define ES8328_PCMDIV		0
+#define ES8328_BCLKDIV		1
+#define ES8328_VXCLKDIV		2
+
+/* PCM clock dividers */
+#define ES8328_PCM_DIV_1	(0 << 6)
+#define ES8328_PCM_DIV_3	(2 << 6)
+#define ES8328_PCM_DIV_5_5	(3 << 6)
+#define ES8328_PCM_DIV_2	(4 << 6)
+#define ES8328_PCM_DIV_4	(5 << 6)
+#define ES8328_PCM_DIV_6	(6 << 6)
+#define ES8328_PCM_DIV_8	(7 << 6)
+
+/* BCLK clock dividers */
+#define ES8328_BCLK_DIV_1	(0 << 7)
+#define ES8328_BCLK_DIV_2	(1 << 7)
+#define ES8328_BCLK_DIV_4	(2 << 7)
+#define ES8328_BCLK_DIV_8	(3 << 7)
+
+/* VXCLK clock dividers */
+#define ES8328_VXCLK_DIV_1	(0 << 6)
+#define ES8328_VXCLK_DIV_2	(1 << 6)
+#define ES8328_VXCLK_DIV_4	(2 << 6)
+#define ES8328_VXCLK_DIV_8	(3 << 6)
+#define ES8328_VXCLK_DIV_16	(4 << 6)
+
+#define ES8328_DAI_HIFI		0
+#define ES8328_DAI_VOICE	1
+
+#define ES8328_1536FS		1536
+#define ES8328_1024FS		1024
+#define ES8328_768FS		768
+#define ES8328_512FS		512
+#define ES8328_384FS		384
+#define ES8328_256FS		256
+#define ES8328_128FS		128
+
+#endif
-- 
cgit v1.2.3


From 7e7292dba2155c1433ce9f9a819f1acb9090747b Mon Sep 17 00:00:00 2001
From: Sean Cross <xobs@kosagi.com>
Date: Thu, 31 Jul 2014 10:43:37 +0800
Subject: ASoC: fsl: add imx-es8328 machine driver

This adds an initial machine driver for the ES8328 audio codec on Freescale
boards.  The driver supports headphones and an audio regulator for an onboard
speaker amp.

Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 .../devicetree/bindings/sound/imx-audio-es8328.txt |  60 ++++++
 sound/soc/fsl/Kconfig                              |  14 ++
 sound/soc/fsl/Makefile                             |   2 +
 sound/soc/fsl/imx-es8328.c                         | 232 +++++++++++++++++++++
 4 files changed, 308 insertions(+)
 create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
 create mode 100644 sound/soc/fsl/imx-es8328.c

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
new file mode 100644
index 000000000000..07b68ab206fb
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt
@@ -0,0 +1,60 @@
+Freescale i.MX audio complex with ES8328 codec
+
+Required properties:
+- compatible       : "fsl,imx-audio-es8328"
+- model            : The user-visible name of this sound complex
+- ssi-controller   : The phandle of the i.MX SSI controller
+- jack-gpio        : Optional GPIO for headphone jack
+- audio-amp-supply : Power regulator for speaker amps
+- audio-codec      : The phandle of the ES8328 audio codec
+- audio-routing    : A list of the connections between audio components.
+                     Each entry is a pair of strings, the first being the
+		     connection's sink, the second being the connection's
+		     source. Valid names could be power supplies, ES8328
+		     pins, and the jacks on the board:
+
+			Power supplies:
+			   * audio-amp
+
+			ES8328 pins:
+			   * LOUT1
+			   * LOUT2
+			   * ROUT1
+			   * ROUT2
+			   * LINPUT1
+			   * LINPUT2
+			   * RINPUT1
+			   * RINPUT2
+			   * Mic PGA
+
+			Board connectors:
+			   * Headphone
+			   * Speaker
+			   * Mic Jack
+- mux-int-port     : The internal port of the i.MX audio muxer (AUDMUX)
+- mux-ext-port     : The external port of the i.MX audio muxer (AUDMIX)
+
+Note: The AUDMUX port numbering should start at 1, which is consistent with
+hardware manual.
+
+Example:
+
+sound {
+	compatible = "fsl,imx-audio-es8328";
+	model = "imx-audio-es8328";
+	ssi-controller = <&ssi1>;
+	audio-codec = <&codec>;
+	jack-gpio = <&gpio5 15 0>;
+	audio-amp-supply = <&reg_audio_amp>;
+	audio-routing =
+		"Speaker", "LOUT2",
+		"Speaker", "ROUT2",
+		"Speaker", "audio-amp",
+		"Headphone", "ROUT1",
+		"Headphone", "LOUT1",
+		"LINPUT1", "Mic Jack",
+		"RINPUT1", "Mic Jack",
+		"Mic Jack", "Mic Bias";
+	mux-int-port = <1>;
+	mux-ext-port = <3>;
+};
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 2b99a9e86899..fa90340dc13a 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -257,6 +257,20 @@ config SND_SOC_IMX_WM8962
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a wm8962 codec.
 
+config SND_SOC_IMX_ES8328
+	tristate "SoC Audio support for i.MX boards with the ES8328 codec"
+	depends on OF && (I2C || SPI)
+	select SND_SOC_ES8328_I2C if I2C
+	select SND_SOC_ES8328_SPI if SPI_MASTER
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_FSL_SSI
+	select SND_SOC_FSL_UTILS
+	select SND_SOC_IMX_PCM_FIQ
+	help
+	  Say Y if you want to add support for the ES8328 audio codec connected
+	  via SSI/I2S over either SPI or I2C.
+
 config SND_SOC_IMX_SGTL5000
 	tristate "SoC Audio support for i.MX boards with sgtl5000"
 	depends on OF && I2C
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 8f6d84efa973..d28dc25c9375 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -52,6 +52,7 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
 snd-soc-phycore-ac97-objs := phycore-ac97.o
 snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
+snd-soc-imx-es8328-objs := imx-es8328.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-wm8962-objs := imx-wm8962.o
 snd-soc-imx-spdif-objs := imx-spdif.o
@@ -61,6 +62,7 @@ obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
 obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
+obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
 obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c
new file mode 100644
index 000000000000..653e66d150c8
--- /dev/null
+++ b/sound/soc/fsl/imx-es8328.c
@@ -0,0 +1,232 @@
+/*
+ * Copyright 2012 Freescale Semiconductor, Inc.
+ * Copyright 2012 Linaro Ltd.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/gpio.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/of_gpio.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE	32
+#define MUX_PORT_MAX	7
+
+struct imx_es8328_data {
+	struct device *dev;
+	struct snd_soc_dai_link dai;
+	struct snd_soc_card card;
+	char codec_dai_name[DAI_NAME_SIZE];
+	char platform_name[DAI_NAME_SIZE];
+	int jack_gpio;
+};
+
+static struct snd_soc_jack_gpio headset_jack_gpios[] = {
+	{
+		.gpio = -1,
+		.name = "headset-gpio",
+		.report = SND_JACK_HEADSET,
+		.invert = 0,
+		.debounce_time = 200,
+	},
+};
+
+static struct snd_soc_jack headset_jack;
+
+static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct imx_es8328_data *data = container_of(rtd->card,
+					struct imx_es8328_data, card);
+	int ret = 0;
+
+	/* Headphone jack detection */
+	if (gpio_is_valid(data->jack_gpio)) {
+		ret = snd_soc_jack_new(rtd->codec, "Headphone",
+				       SND_JACK_HEADPHONE | SND_JACK_BTN_0,
+				       &headset_jack);
+		if (ret)
+			return ret;
+
+		headset_jack_gpios[0].gpio = data->jack_gpio;
+		ret = snd_soc_jack_add_gpios(&headset_jack,
+					     ARRAY_SIZE(headset_jack_gpios),
+					     headset_jack_gpios);
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_dapm_widget imx_es8328_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("audio-amp", 1, 0),
+};
+
+static int imx_es8328_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct device_node *ssi_np, *codec_np;
+	struct platform_device *ssi_pdev;
+	struct imx_es8328_data *data;
+	u32 int_port, ext_port;
+	int ret;
+	struct device *dev = &pdev->dev;
+
+	ret = of_property_read_u32(np, "mux-int-port", &int_port);
+	if (ret) {
+		dev_err(dev, "mux-int-port missing or invalid\n");
+		goto fail;
+	}
+	if (int_port > MUX_PORT_MAX || int_port == 0) {
+		dev_err(dev, "mux-int-port: hardware only has %d mux ports\n",
+			MUX_PORT_MAX);
+		goto fail;
+	}
+
+	ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+	if (ret) {
+		dev_err(dev, "mux-ext-port missing or invalid\n");
+		goto fail;
+	}
+	if (ext_port > MUX_PORT_MAX || ext_port == 0) {
+		dev_err(dev, "mux-ext-port: hardware only has %d mux ports\n",
+			MUX_PORT_MAX);
+		goto fail;
+	}
+
+	/*
+	 * The port numbering in the hardware manual starts at 1, while
+	 * the audmux API expects it starts at 0.
+	 */
+	int_port--;
+	ext_port--;
+	ret = imx_audmux_v2_configure_port(int_port,
+			IMX_AUDMUX_V2_PTCR_SYN |
+			IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) |
+			IMX_AUDMUX_V2_PTCR_TFSDIR |
+			IMX_AUDMUX_V2_PTCR_TCLKDIR,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+	if (ret) {
+		dev_err(dev, "audmux internal port setup failed\n");
+		return ret;
+	}
+	ret = imx_audmux_v2_configure_port(ext_port,
+			IMX_AUDMUX_V2_PTCR_SYN,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+	if (ret) {
+		dev_err(dev, "audmux external port setup failed\n");
+		return ret;
+	}
+
+	ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0);
+	codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+	if (!ssi_np || !codec_np) {
+		dev_err(dev, "phandle missing or invalid\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	ssi_pdev = of_find_device_by_node(ssi_np);
+	if (!ssi_pdev) {
+		dev_err(dev, "failed to find SSI platform device\n");
+		ret = -EINVAL;
+		goto fail;
+	}
+
+	data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		ret = -ENOMEM;
+		goto fail;
+	}
+
+	data->dev = dev;
+
+	data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0);
+
+	data->dai.name = "hifi";
+	data->dai.stream_name = "hifi";
+	data->dai.codec_dai_name = "es8328-hifi-analog";
+	data->dai.codec_of_node = codec_np;
+	data->dai.cpu_of_node = ssi_np;
+	data->dai.platform_of_node = ssi_np;
+	data->dai.init = &imx_es8328_dai_init;
+	data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			    SND_SOC_DAIFMT_CBM_CFM;
+
+	data->card.dev = dev;
+	data->card.dapm_widgets = imx_es8328_dapm_widgets;
+	data->card.num_dapm_widgets = ARRAY_SIZE(imx_es8328_dapm_widgets);
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret) {
+		dev_err(dev, "Unable to parse card name\n");
+		goto fail;
+	}
+	ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+	if (ret) {
+		dev_err(dev, "Unable to parse routing: %d\n", ret);
+		goto fail;
+	}
+	data->card.num_links = 1;
+	data->card.owner = THIS_MODULE;
+	data->card.dai_link = &data->dai;
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(dev, "Unable to register: %d\n", ret);
+		goto fail;
+	}
+
+	platform_set_drvdata(pdev, data);
+fail:
+	of_node_put(ssi_np);
+	of_node_put(codec_np);
+
+	return ret;
+}
+
+static int imx_es8328_remove(struct platform_device *pdev)
+{
+	struct imx_es8328_data *data = platform_get_drvdata(pdev);
+
+	snd_soc_jack_free_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios),
+				headset_jack_gpios);
+
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_es8328_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-es8328", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_es8328_dt_ids);
+
+static struct platform_driver imx_es8328_driver = {
+	.driver = {
+		.name = "imx-es8328",
+		.of_match_table = imx_es8328_dt_ids,
+	},
+	.probe = imx_es8328_probe,
+	.remove = imx_es8328_remove,
+};
+module_platform_driver(imx_es8328_driver);
+
+MODULE_AUTHOR("Sean Cross <xobs@kosagi.com>");
+MODULE_DESCRIPTION("Kosagi i.MX6 ES8328 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-audio-es8328");
-- 
cgit v1.2.3


From 5f37671e004eeca017b93f6b26f2425acbb8d411 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@freescale.com>
Date: Mon, 18 Aug 2014 16:38:39 +0800
Subject: ASoC: fsl-asoc-card: Fix build warning for maybe-uninitialized

When build fsl-asoc-card as module, there is following error:

sound/soc/fsl/fsl-asoc-card.c: In function 'fsl_asoc_card_probe':
>> sound/soc/fsl/fsl-asoc-card.c:547:13: warning: 'asrc_np' may be used uninitialized in this function [-Wmaybe-uninitialized]
     of_node_put(asrc_np);
                ^

vim +/asrc_np +547 sound/soc/fsl/fsl-asoc-card.c

   531                  if (width == 24)
   532                          priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
   533                  else
   534                          priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
   535          }
   536
   537          /* Finish card registering */
   538          platform_set_drvdata(pdev, priv);
   539          snd_soc_card_set_drvdata(&priv->card, priv);
   540
   541          ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
   542          if (ret)
   543                  dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
   544
   545  fail:
   546          of_node_put(codec_np);
 > 547          of_node_put(asrc_np);
   548          of_node_put(cpu_np);
   549
   550          return ret;
   551  }
   552
   553  static const struct of_device_id fsl_asoc_card_dt_ids[] = {
   554          { .compatible = "fsl,imx-audio-cs42888", },
   555          { .compatible = "fsl,imx-audio-sgtl5000", },

Add 'asrc_fail' branch for error jump after asrc_np initialized.

Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl-asoc-card.c | 9 +++++----
 1 file changed, 5 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index cf3f1f47f1e8..007c772f3cef 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -469,7 +469,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		ret = fsl_asoc_card_audmux_init(np, priv);
 		if (ret) {
 			dev_err(&pdev->dev, "failed to init audmux\n");
-			goto fail;
+			goto asrc_fail;
 		}
 	} else if (strstr(cpu_np->name, "esai")) {
 		priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL;
@@ -518,14 +518,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		if (ret) {
 			dev_err(&pdev->dev, "failed to get output rate\n");
 			ret = -EINVAL;
-			goto fail;
+			goto asrc_fail;
 		}
 
 		ret = of_property_read_u32(asrc_np, "fsl,asrc-width", &width);
 		if (ret) {
 			dev_err(&pdev->dev, "failed to get output rate\n");
 			ret = -EINVAL;
-			goto fail;
+			goto asrc_fail;
 		}
 
 		if (width == 24)
@@ -542,9 +542,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	if (ret)
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
 
+asrc_fail:
+	of_node_put(asrc_np);
 fail:
 	of_node_put(codec_np);
-	of_node_put(asrc_np);
 	of_node_put(cpu_np);
 
 	return ret;
-- 
cgit v1.2.3


From 499898d66d88cc626a2e01b02c3b819536bdf169 Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@freescale.com>
Date: Mon, 18 Aug 2014 16:38:40 +0800
Subject: ASoC: fsl: fsl-asoc-card: Select SND_SOC_IMX_AUDMUX

Building kernel with SND_SOC_IMX_AUDMUX=n leads to the following error:

   sound/built-in.o: In function `fsl_asoc_card_probe':
>> fsl-asoc-card.c:(.text+0x1467b5): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x1467d0): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x1467ed): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x146807): undefined reference to `imx_audmux_v2_configure_port'

Update Kconfig to select SND_SOC_IMX_AUDMUX when SND_SOC_FSL_ASOC_CARD=y.

Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/Kconfig | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index fa90340dc13a..4698c01af684 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -62,6 +62,7 @@ config SND_SOC_FSL_UTILS
 config SND_SOC_FSL_ASOC_CARD
 	tristate "Generic ASoC Sound Card with ASRC support"
 	depends on OF && I2C
+	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_IMX_PCM_DMA
 	select SND_SOC_FSL_ESAI
 	select SND_SOC_FSL_SAI
-- 
cgit v1.2.3


From cdec729765659adafba983d6b6760ad52c71d5d8 Mon Sep 17 00:00:00 2001
From: Sean Cross <xobs@kosagi.com>
Date: Tue, 19 Aug 2014 12:49:34 +0800
Subject: ASoC: fsl: Fix building of imx-es8328 on PPC

The imx-es8328 driver fails to build on PPC because it explicitly depends on
SND_SOC_IMX_PCM_FIQ, which itself doesn't build on PPC.  Instead, rely on
the SND_SOC_FSL_SSI config option to pull in the necessary libraries.

While we're at it, remove SND_SOC_FSL_UTILS, which also is not needed.

Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/Kconfig | 2 --
 1 file changed, 2 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 4698c01af684..3154f43b11ab 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -266,8 +266,6 @@ config SND_SOC_IMX_ES8328
 	select SND_SOC_IMX_PCM_DMA
 	select SND_SOC_IMX_AUDMUX
 	select SND_SOC_FSL_SSI
-	select SND_SOC_FSL_UTILS
-	select SND_SOC_IMX_PCM_FIQ
 	help
 	  Say Y if you want to add support for the ES8328 audio codec connected
 	  via SSI/I2S over either SPI or I2C.
-- 
cgit v1.2.3


From 38c6e4bb67760db1392b9c5ee0082af07c0db20d Mon Sep 17 00:00:00 2001
From: Shengjiu Wang <shengjiu.wang@freescale.com>
Date: Tue, 19 Aug 2014 17:36:41 +0800
Subject: ASoC: fsl-asoc-card: move 'config SND_SOC_FSL_ASOC_CARD' to 'if
 SND_IMX_SOC'

Build kernel with SND_SOC_FSL_ASOC_CARD=m && SND_SOC_FSL_{SSI,SAI,ESAI}=y
leads the following error:

   sound/built-in.o: In function `fsl_sai_probe':
>> fsl_sai.c:(.text+0x5f662): undefined reference to `imx_pcm_dma_init'
   sound/built-in.o: In function `fsl_esai_probe':
>> fsl_esai.c:(.text+0x6044b): undefined reference to `imx_pcm_dma_init'

The config SND_SOC_FSL_ASOC_CARD is for IMX SOC, So move it under condition
of 'if SND_IMX_SOC'.

Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/Kconfig | 34 +++++++++++++++++-----------------
 1 file changed, 17 insertions(+), 17 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 3154f43b11ab..7c1da8ede975 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -59,23 +59,6 @@ config SND_SOC_FSL_ESAI
 config SND_SOC_FSL_UTILS
 	tristate
 
-config SND_SOC_FSL_ASOC_CARD
-	tristate "Generic ASoC Sound Card with ASRC support"
-	depends on OF && I2C
-	select SND_SOC_IMX_AUDMUX
-	select SND_SOC_IMX_PCM_DMA
-	select SND_SOC_FSL_ESAI
-	select SND_SOC_FSL_SAI
-	select SND_SOC_FSL_SSI
-	select SND_SOC_CS42XX8_I2C
-	select SND_SOC_SGTL5000
-	select SND_SOC_WM8962
-	help
-	 ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
-	 ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
-	 and SGTL5000.
-	 Say Y if you want to add support for Freescale Generic ASoC Sound Card.
-
 config SND_SOC_IMX_PCM_DMA
 	tristate
 	select SND_SOC_GENERIC_DMAENGINE_PCM
@@ -298,6 +281,23 @@ config SND_SOC_IMX_MC13783
 	select SND_SOC_MC13783
 	select SND_SOC_IMX_PCM_DMA
 
+config SND_SOC_FSL_ASOC_CARD
+	tristate "Generic ASoC Sound Card with ASRC support"
+	depends on OF && I2C
+	select SND_SOC_IMX_AUDMUX
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_FSL_ESAI
+	select SND_SOC_FSL_SAI
+	select SND_SOC_FSL_SSI
+	select SND_SOC_CS42XX8_I2C
+	select SND_SOC_SGTL5000
+	select SND_SOC_WM8962
+	help
+	 ALSA SoC Audio support with ASRC feature for Freescale SoCs that have
+	 ESAI/SAI/SSI and connect with external CODECs such as WM8962, CS42888
+	 and SGTL5000.
+	 Say Y if you want to add support for Freescale Generic ASoC Sound Card.
+
 endif # SND_IMX_SOC
 
 endmenu
-- 
cgit v1.2.3


From bf16d883263dedefb6149916e41b3e2779bb1573 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Mon, 25 Aug 2014 11:30:59 +0800
Subject: ASoC: fsl-asrc: Convert to use regmap framework's endianness method.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_asrc.c | 6 +-----
 1 file changed, 1 insertion(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c
index 822110420b71..3b145313f93e 100644
--- a/sound/soc/fsl/fsl_asrc.c
+++ b/sound/soc/fsl/fsl_asrc.c
@@ -684,7 +684,7 @@ static bool fsl_asrc_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_asrc_regmap_config = {
+static const struct regmap_config fsl_asrc_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -802,10 +802,6 @@ static int fsl_asrc_probe(struct platform_device *pdev)
 
 	asrc_priv->paddr = res->start;
 
-	/* Register regmap and let it prepare core clock */
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_asrc_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	asrc_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "mem", regs,
 						      &fsl_asrc_regmap_config);
 	if (IS_ERR(asrc_priv->regmap)) {
-- 
cgit v1.2.3


From 92bd0334b27845f250f1fadb091242140391c99b Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Mon, 25 Aug 2014 11:31:00 +0800
Subject: ASoC: fsl-esai: Convert to use regmap framework's endianness method.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_esai.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index 72d154e7dd03..2882fc66a10d 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -707,7 +707,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_esai_regmap_config = {
+static const struct regmap_config fsl_esai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -733,9 +733,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	esai_priv->pdev = pdev;
 	strcpy(esai_priv->name, np->name);
 
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);
-- 
cgit v1.2.3


From 664915074e750614c5d140093d5098a165a24e3d Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Mon, 25 Aug 2014 11:31:01 +0800
Subject: ASoC: fsl-spdif: Convert to use regmap framework's endianness method.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 sound/soc/fsl/fsl_spdif.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 70acfe4a9bd5..ae4e408810ec 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1040,7 +1040,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_spdif_regmap_config = {
+static const struct regmap_config fsl_spdif_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -1184,9 +1184,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai));
 	spdif_priv->cpu_dai_drv.name = spdif_priv->name;
 
-	if (of_property_read_bool(np, "big-endian"))
-		fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	/* Get the addresses and IRQ */
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
 	regs = devm_ioremap_resource(&pdev->dev, res);
-- 
cgit v1.2.3


From 014fd22ef9c6a7e9536b7e16635714a1a34810a8 Mon Sep 17 00:00:00 2001
From: Xiubo Li <Li.Xiubo@freescale.com>
Date: Mon, 25 Aug 2014 11:31:02 +0800
Subject: ASoC: fsl-sai: Convert to use regmap framework's endianness method.

Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
---
 Documentation/devicetree/bindings/sound/fsl-sai.txt | 7 +++----
 sound/soc/fsl/fsl_sai.c                             | 6 +-----
 sound/soc/fsl/fsl_sai.h                             | 1 -
 3 files changed, 4 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt
index 0f4e23828190..5f239b8bcddd 100644
--- a/Documentation/devicetree/bindings/sound/fsl-sai.txt
+++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt
@@ -18,9 +18,8 @@ Required properties:
 - pinctrl-names: Must contain a "default" entry.
 - pinctrl-NNN: One property must exist for each entry in pinctrl-names.
   See ../pinctrl/pinctrl-bindings.txt for details of the property values.
-- big-endian-regs: If this property is absent, the little endian mode will
-  be in use as default, or the big endian mode will be in use for all the
-  device registers.
+- big-endian: Boolean property, required if all the FTM_PWM registers
+  are big-endian rather than little-endian.
 - big-endian-data: If this property is absent, the little endian mode will
   be in use as default, or the big endian mode will be in use for all the
   fifo data.
@@ -38,6 +37,6 @@ sai2: sai@40031000 {
 	      dma-names = "tx", "rx";
 	      dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>,
 		   <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>;
-	      big-endian-regs;
+	      big-endian;
 	      big-endian-data;
 };
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index faa049797897..52d1e9982639 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -539,7 +539,7 @@ static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg)
 	}
 }
 
-static struct regmap_config fsl_sai_regmap_config = {
+static const struct regmap_config fsl_sai_regmap_config = {
 	.reg_bits = 32,
 	.reg_stride = 4,
 	.val_bits = 32,
@@ -568,10 +568,6 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai"))
 		sai->sai_on_imx = true;
 
-	sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs");
-	if (sai->big_endian_regs)
-		fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG;
-
 	sai->big_endian_data = of_property_read_bool(np, "big-endian-data");
 
 	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h
index 0e6c9f595d75..20e3e53ce6ea 100644
--- a/sound/soc/fsl/fsl_sai.h
+++ b/sound/soc/fsl/fsl_sai.h
@@ -131,7 +131,6 @@ struct fsl_sai {
 	struct clk *bus_clk;
 	struct clk *mclk_clk[FSL_SAI_MCLK_MAX];
 
-	bool big_endian_regs;
 	bool big_endian_data;
 	bool is_dsp_mode;
 	bool sai_on_imx;
-- 
cgit v1.2.3