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Diffstat (limited to 'sound/soc/fsl/imx-wm8962.c')
-rw-r--r--sound/soc/fsl/imx-wm8962.c860
1 files changed, 860 insertions, 0 deletions
diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c
new file mode 100644
index 000000000000..cb50a6fa48a9
--- /dev/null
+++ b/sound/soc/fsl/imx-wm8962.c
@@ -0,0 +1,860 @@
+/*
+ * Copyright (C) 2013-2016 Freescale Semiconductor, Inc.
+ *
+ * Based on imx-sgtl5000.c
+ * Copyright (C) 2012 Freescale Semiconductor, Inc.
+ * Copyright (C) 2012 Linaro Ltd.
+ * Copyright 2017 NXP
+ *
+ * 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/module.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/of_gpio.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/control.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include <linux/pinctrl/consumer.h>
+
+#include "../codecs/wm8962.h"
+#include "imx-audmux.h"
+
+#define DAI_NAME_SIZE 32
+
+struct imx_wm8962_data {
+ struct snd_soc_dai_link dai[3];
+ struct snd_soc_card card;
+ char codec_dai_name[DAI_NAME_SIZE];
+ char platform_name[DAI_NAME_SIZE];
+ struct clk *codec_clk;
+ unsigned int clk_frequency;
+ bool is_codec_master;
+};
+
+struct imx_priv {
+ int hp_gpio;
+ int hp_active_low;
+ int mic_gpio;
+ int mic_active_low;
+ bool amic_mono;
+ bool dmic_mono;
+ struct snd_soc_component *component;
+ struct platform_device *pdev;
+ struct snd_pcm_substream *first_stream;
+ struct snd_pcm_substream *second_stream;
+ struct platform_device *asrc_pdev;
+ u32 asrc_rate;
+ u32 asrc_format;
+};
+static struct imx_priv card_priv;
+
+#ifdef CONFIG_SND_SOC_IMX_WM8962_ANDROID
+static int sample_rate = 44100;
+static snd_pcm_format_t sample_format = SNDRV_PCM_FORMAT_S16_LE;
+#endif
+
+static struct snd_soc_jack imx_hp_jack;
+static struct snd_soc_jack_pin imx_hp_jack_pins[] = {
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+static struct snd_soc_jack_gpio imx_hp_jack_gpio = {
+ .name = "headphone detect",
+ .report = SND_JACK_HEADPHONE,
+ .debounce_time = 250,
+ .invert = 0,
+};
+
+static struct snd_soc_jack imx_mic_jack;
+static struct snd_soc_jack_pin imx_mic_jack_pins[] = {
+ {
+ .pin = "AMIC",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+static struct snd_soc_jack_gpio imx_mic_jack_gpio = {
+ .name = "microphone detect",
+ .report = SND_JACK_MICROPHONE,
+ .debounce_time = 250,
+ .invert = 0,
+};
+
+static int hpjack_status_check(void *data)
+{
+ struct imx_priv *priv = &card_priv;
+ struct platform_device *pdev = priv->pdev;
+ char *envp[3], *buf;
+ int hp_status, ret;
+
+ if (!gpio_is_valid(priv->hp_gpio))
+ return 0;
+
+ hp_status = gpio_get_value(priv->hp_gpio) ? 1 : 0;
+
+ buf = kmalloc(32, GFP_ATOMIC);
+ if (!buf) {
+ dev_err(&pdev->dev, "%s kmalloc failed\n", __func__);
+ return -ENOMEM;
+ }
+
+ if (hp_status != priv->hp_active_low) {
+ snprintf(buf, 32, "STATE=%d", 2);
+ snd_soc_dapm_disable_pin(snd_soc_component_get_dapm(priv->component), "Ext Spk");
+ ret = imx_hp_jack_gpio.report;
+ } else {
+ snprintf(buf, 32, "STATE=%d", 0);
+ snd_soc_dapm_enable_pin(snd_soc_component_get_dapm(priv->component), "Ext Spk");
+ ret = 0;
+ }
+
+ envp[0] = "NAME=headphone";
+ envp[1] = buf;
+ envp[2] = NULL;
+ kobject_uevent_env(&pdev->dev.kobj, KOBJ_CHANGE, envp);
+ kfree(buf);
+
+ return ret;
+}
+
+static int micjack_status_check(void *data)
+{
+ struct imx_priv *priv = &card_priv;
+ struct platform_device *pdev = priv->pdev;
+ char *envp[3], *buf;
+ int mic_status, ret;
+
+ if (!gpio_is_valid(priv->mic_gpio))
+ return 0;
+
+ mic_status = gpio_get_value(priv->mic_gpio) ? 1 : 0;
+
+ if ((mic_status != priv->mic_active_low && priv->amic_mono)
+ || (mic_status == priv->mic_active_low && priv->dmic_mono))
+ snd_soc_component_update_bits(priv->component, WM8962_THREED1,
+ WM8962_ADC_MONOMIX_MASK, WM8962_ADC_MONOMIX);
+
+ else
+ snd_soc_component_update_bits(priv->component, WM8962_THREED1,
+ WM8962_ADC_MONOMIX_MASK, 0);
+
+ buf = kmalloc(32, GFP_ATOMIC);
+ if (!buf) {
+ dev_err(&pdev->dev, "%s kmalloc failed\n", __func__);
+ return -ENOMEM;
+ }
+
+ if (mic_status != priv->mic_active_low) {
+ snprintf(buf, 32, "STATE=%d", 2);
+ snd_soc_dapm_disable_pin(snd_soc_component_get_dapm(priv->component), "DMIC");
+ ret = imx_mic_jack_gpio.report;
+ } else {
+ snprintf(buf, 32, "STATE=%d", 0);
+ snd_soc_dapm_enable_pin(snd_soc_component_get_dapm(priv->component), "DMIC");
+ ret = 0;
+ }
+
+ envp[0] = "NAME=microphone";
+ envp[1] = buf;
+ envp[2] = NULL;
+ kobject_uevent_env(&pdev->dev.kobj, KOBJ_CHANGE, envp);
+ kfree(buf);
+
+ return ret;
+}
+
+
+static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_MIC("AMIC", NULL),
+ SND_SOC_DAPM_MIC("DMIC", NULL),
+};
+
+static u32 imx_wm8962_rates[] = {32000, 48000, 96000};
+static struct snd_pcm_hw_constraint_list imx_wm8962_rate_constraints = {
+ .count = ARRAY_SIZE(imx_wm8962_rates),
+ .list = imx_wm8962_rates,
+};
+
+static int imx_hifi_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_card *card = rtd->card;
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
+ int ret;
+
+ if (!data->is_codec_master) {
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &imx_wm8962_rate_constraints);
+ if (ret)
+ return ret;
+ }
+
+ return 0;
+}
+
+#ifdef CONFIG_SND_SOC_IMX_WM8962_ANDROID
+static int imx_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct imx_priv *priv = &card_priv;
+ struct device *dev = &priv->pdev->dev;
+ u32 dai_format;
+ int ret = 0;
+
+ sample_rate = params_rate(params);
+ sample_format = params_format(params);
+
+ if (data->is_codec_master)
+ dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+ else
+ dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
+ if (ret) {
+ dev_err(dev, "failed to set codec dai fmt: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops imx_hifi_ops = {
+ .startup = imx_hifi_startup,
+ .hw_params = imx_hifi_hw_params,
+};
+
+static int imx_wm8962_set_bias_level(struct snd_soc_card *card,
+ struct snd_soc_dapm_context *dapm,
+ enum snd_soc_bias_level level)
+{
+ struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai;
+ struct imx_priv *priv = &card_priv;
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
+ struct device *dev = &priv->pdev->dev;
+ unsigned int pll_out;
+ int ret;
+
+ if (dapm->dev != codec_dai->dev)
+ return 0;
+
+ data->clk_frequency = clk_get_rate(data->codec_clk);
+
+ switch (level) {
+ case SND_SOC_BIAS_PREPARE:
+ if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
+ if (sample_format == SNDRV_PCM_FORMAT_S24_LE
+ || sample_format == SNDRV_PCM_FORMAT_S20_3LE)
+ pll_out = sample_rate * 384;
+ else
+ pll_out = sample_rate * 256;
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ WM8962_FLL_MCLK, data->clk_frequency,
+ pll_out);
+ if (ret < 0) {
+ dev_err(dev, "failed to start FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8962_SYSCLK_FLL, pll_out,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ 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) {
+ ret = snd_soc_dai_set_sysclk(codec_dai,
+ WM8962_SYSCLK_MCLK, data->clk_frequency,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(dev,
+ "failed to switch away from FLL: %d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ 0, 0, 0);
+ if (ret < 0) {
+ dev_err(dev, "failed to stop FLL: %d\n", ret);
+ return ret;
+ }
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+#else
+
+static int imx_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct imx_priv *priv = &card_priv;
+ struct device *dev = &priv->pdev->dev;
+ struct snd_soc_card *card = platform_get_drvdata(priv->pdev);
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
+ unsigned int sample_rate = params_rate(params);
+ snd_pcm_format_t sample_format = params_format(params);
+ u32 dai_format, pll_out;
+ int ret = 0;
+
+ if (!priv->first_stream) {
+ priv->first_stream = substream;
+ } else {
+ priv->second_stream = substream;
+
+ /* We suppose the two substream are using same params */
+ return 0;
+ }
+
+ data->clk_frequency = clk_get_rate(data->codec_clk);
+
+ if (data->is_codec_master)
+ dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+ else
+ dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS;
+
+ /* set codec DAI configuration */
+ ret = snd_soc_dai_set_fmt(codec_dai, dai_format);
+ if (ret) {
+ dev_err(dev, "failed to set codec dai fmt: %d\n", ret);
+ return ret;
+ }
+
+ if (sample_format == SNDRV_PCM_FORMAT_S24_LE
+ || sample_format == SNDRV_PCM_FORMAT_S20_3LE)
+ pll_out = sample_rate * 384;
+ else
+ pll_out = sample_rate * 256;
+
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK,
+ data->clk_frequency, pll_out);
+ if (ret) {
+ dev_err(dev, "failed to start FLL: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL,
+ pll_out, SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(dev, "failed to set SYSCLK: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int imx_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct imx_priv *priv = &card_priv;
+ struct device *dev = &priv->pdev->dev;
+ int ret;
+
+ /* We don't need to handle anything if there's no substream running */
+ if (!priv->first_stream)
+ return 0;
+
+ if (priv->first_stream == substream)
+ priv->first_stream = priv->second_stream;
+ priv->second_stream = NULL;
+
+ if (!priv->first_stream) {
+ /*
+ * Continuously setting FLL would cause playback distortion.
+ * We can fix it just by mute codec after playback.
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dai_digital_mute(codec_dai, 1, substream->stream);
+
+ /*
+ * WM8962 doesn't allow us to continuously setting FLL,
+ * So we set MCLK as sysclk once, which'd remove the limitation.
+ */
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+ 0, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(dev, "failed to switch away from FLL: %d\n", ret);
+ return ret;
+ }
+
+ /* Disable FLL and let codec do pm_runtime_put() */
+ ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
+ WM8962_FLL_MCLK, 0, 0);
+ if (ret < 0) {
+ dev_err(dev, "failed to stop FLL: %d\n", ret);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops imx_hifi_ops = {
+ .startup = imx_hifi_startup,
+ .hw_params = imx_hifi_hw_params,
+ .hw_free = imx_hifi_hw_free,
+};
+#endif /* CONFIG_SND_SOC_IMX_WM8962_ANDROID */
+
+static int imx_wm8962_gpio_init(struct snd_soc_card *card)
+{
+ struct snd_soc_pcm_runtime *rtd = list_first_entry(
+ &card->rtd_list, struct snd_soc_pcm_runtime, list);
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct imx_priv *priv = &card_priv;
+
+ priv->component = codec_dai->component;
+
+ if (gpio_is_valid(priv->hp_gpio)) {
+ imx_hp_jack_gpio.gpio = priv->hp_gpio;
+ imx_hp_jack_gpio.jack_status_check = hpjack_status_check;
+
+ snd_soc_card_jack_new(card, "Headphone Jack",
+ SND_JACK_HEADPHONE, &imx_hp_jack,
+ imx_hp_jack_pins, ARRAY_SIZE(imx_hp_jack_pins));
+
+ snd_soc_jack_add_gpios(&imx_hp_jack, 1, &imx_hp_jack_gpio);
+ }
+
+ if (gpio_is_valid(priv->mic_gpio)) {
+ imx_mic_jack_gpio.gpio = priv->mic_gpio;
+ imx_mic_jack_gpio.jack_status_check = micjack_status_check;
+
+ snd_soc_card_jack_new(card, "AMIC",
+ SND_JACK_MICROPHONE, &imx_mic_jack,
+ imx_mic_jack_pins, ARRAY_SIZE(imx_mic_jack_pins));
+
+ snd_soc_jack_add_gpios(&imx_mic_jack, 1, &imx_mic_jack_gpio);
+ } else if (priv->amic_mono || priv->dmic_mono) {
+ /*
+ * Permanent set monomix bit if only one microphone
+ * is present on the board while it needs monomix.
+ */
+ snd_soc_component_update_bits(priv->component, WM8962_THREED1,
+ WM8962_ADC_MONOMIX_MASK, WM8962_ADC_MONOMIX);
+ }
+
+ return 0;
+}
+
+static ssize_t headphone_show(struct device_driver *dev, char *buf)
+{
+ struct imx_priv *priv = &card_priv;
+ int hp_status;
+
+ if (!gpio_is_valid(priv->hp_gpio)) {
+ strcpy(buf, "no detect gpio connected\n");
+ return strlen(buf);
+ }
+
+ /* Check if headphone is plugged in */
+ hp_status = gpio_get_value(priv->hp_gpio) ? 1 : 0;
+
+ if (hp_status != priv->hp_active_low)
+ strcpy(buf, "headphone\n");
+ else
+ strcpy(buf, "speaker\n");
+
+ return strlen(buf);
+}
+
+static DRIVER_ATTR_RO(headphone);
+
+static ssize_t microphone_show(struct device_driver *dev, char *buf)
+{
+ struct imx_priv *priv = &card_priv;
+ int mic_status;
+
+ if (!gpio_is_valid(priv->mic_gpio)) {
+ strcpy(buf, "no detect gpio connected\n");
+ return strlen(buf);
+ }
+
+ /* Check if analog microphone is plugged in */
+ mic_status = gpio_get_value(priv->mic_gpio) ? 1 : 0;
+
+ if (mic_status != priv->mic_active_low)
+ strcpy(buf, "amic\n");
+ else
+ strcpy(buf, "dmic\n");
+
+ return strlen(buf);
+}
+
+static DRIVER_ATTR_RO(microphone);
+
+static int imx_wm8962_late_probe(struct snd_soc_card *card)
+{
+ struct snd_soc_pcm_runtime *rtd;
+ struct snd_soc_dai *codec_dai;
+ struct imx_priv *priv = &card_priv;
+ struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card);
+ struct device *dev = &priv->pdev->dev;
+ int ret;
+
+ data->clk_frequency = clk_get_rate(data->codec_clk);
+ rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
+ codec_dai = rtd->codec_dai;
+ ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
+ data->clk_frequency, SND_SOC_CLOCK_IN);
+ if (ret < 0)
+ dev_err(dev, "failed to set sysclk in %s\n", __func__);
+
+ return ret;
+}
+
+static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params) {
+ struct imx_priv *priv = &card_priv;
+ struct snd_interval *rate;
+ struct snd_mask *mask;
+
+ if (!priv->asrc_pdev)
+ return -EINVAL;
+
+ 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 int imx_wm8962_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct device_node *cpu_np = NULL, *codec_np = NULL;
+ struct platform_device *cpu_pdev;
+ struct imx_priv *priv = &card_priv;
+ struct i2c_client *codec_dev;
+ struct imx_wm8962_data *data;
+ int int_port, ext_port, tmp_port;
+ int ret;
+ struct platform_device *asrc_pdev = NULL;
+ struct device_node *asrc_np;
+ struct snd_soc_dai_link_component *dlc;
+ u32 width;
+
+ priv->pdev = pdev;
+ priv->asrc_pdev = NULL;
+
+ dlc = devm_kzalloc(&pdev->dev, 9 * sizeof(*dlc), GFP_KERNEL);
+ if (!dlc)
+ return -ENOMEM;
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
+ if (of_property_read_bool(pdev->dev.of_node, "codec-master"))
+ data->is_codec_master = true;
+
+ cpu_np = of_parse_phandle(pdev->dev.of_node, "audio-cpu", 0);
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ if (!strstr(cpu_np->name, "ssi"))
+ goto audmux_bypass;
+
+ ret = of_property_read_u32(np, "mux-int-port", &int_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-int-port missing or invalid\n");
+ goto fail;
+ }
+ ret = of_property_read_u32(np, "mux-ext-port", &ext_port);
+ if (ret) {
+ dev_err(&pdev->dev, "mux-ext-port missing or invalid\n");
+ 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--;
+ if (data->is_codec_master) {
+ tmp_port = int_port;
+ int_port = ext_port;
+ ext_port = tmp_port;
+ }
+
+ ret = imx_audmux_v2_configure_port(ext_port,
+ IMX_AUDMUX_V2_PTCR_SYN |
+ IMX_AUDMUX_V2_PTCR_TFSEL(int_port) |
+ IMX_AUDMUX_V2_PTCR_TCSEL(int_port) |
+ IMX_AUDMUX_V2_PTCR_TFSDIR |
+ IMX_AUDMUX_V2_PTCR_TCLKDIR,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux internal port setup failed\n");
+ goto fail;
+ }
+ ret = imx_audmux_v2_configure_port(int_port,
+ IMX_AUDMUX_V2_PTCR_SYN,
+ IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port));
+ if (ret) {
+ dev_err(&pdev->dev, "audmux external port setup failed\n");
+ goto fail;
+ }
+
+audmux_bypass:
+ codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+ if (!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 SSI platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+ codec_dev = of_find_i2c_device_by_node(codec_np);
+ if (!codec_dev || !codec_dev->dev.driver) {
+ dev_err(&pdev->dev, "failed to find codec platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ asrc_np = of_parse_phandle(pdev->dev.of_node, "asrc-controller", 0);
+ if (asrc_np) {
+ asrc_pdev = of_find_device_by_node(asrc_np);
+ priv->asrc_pdev = asrc_pdev;
+ }
+
+ priv->first_stream = NULL;
+ priv->second_stream = NULL;
+
+ data->codec_clk = clk_get(&codec_dev->dev, NULL);
+ if (IS_ERR(data->codec_clk)) {
+ ret = PTR_ERR(data->codec_clk);
+ dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
+ goto fail;
+ }
+
+ priv->amic_mono = of_property_read_bool(codec_np, "amic-mono");
+ priv->dmic_mono = of_property_read_bool(codec_np, "dmic-mono");
+
+ priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0,
+ (enum of_gpio_flags *)&priv->hp_active_low);
+ priv->mic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0,
+ (enum of_gpio_flags *)&priv->mic_active_low);
+
+ data->dai[0].cpus = &dlc[0];
+ data->dai[0].num_cpus = 1;
+ data->dai[0].platforms = &dlc[1];
+ data->dai[0].num_platforms = 1;
+ data->dai[0].codecs = &dlc[2];
+ data->dai[0].num_codecs = 1;
+
+ data->dai[0].name = "HiFi";
+ data->dai[0].stream_name = "HiFi";
+ data->dai[0].codecs->dai_name = "wm8962";
+ data->dai[0].codecs->of_node = codec_np;
+ data->dai[0].cpus->dai_name = dev_name(&cpu_pdev->dev);
+ data->dai[0].platforms->of_node = cpu_np;
+ data->dai[0].ops = &imx_hifi_ops;
+ data->dai[0].dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF;
+ if (data->is_codec_master)
+ data->dai[0].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
+ else
+ data->dai[0].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;
+
+ data->card.num_links = 1;
+
+ if (asrc_pdev) {
+ data->dai[1].cpus = &dlc[3];
+ data->dai[1].num_cpus = 1;
+ data->dai[1].platforms = &dlc[4];
+ data->dai[1].num_platforms = 1;
+ data->dai[1].codecs = &dlc[5];
+ data->dai[1].num_codecs = 1;
+
+ data->dai[2].cpus = &dlc[6];
+ data->dai[2].num_cpus = 1;
+ data->dai[2].platforms = &dlc[7];
+ data->dai[2].num_platforms = 1;
+ data->dai[2].codecs = &dlc[8];
+ data->dai[2].num_codecs = 1;
+
+ data->dai[0].ignore_pmdown_time = 1;
+ data->dai[1].name = "HiFi-ASRC-FE";
+ data->dai[1].stream_name = "HiFi-ASRC-FE";
+ data->dai[1].codecs->name = "snd-soc-dummy";
+ data->dai[1].codecs->dai_name = "snd-soc-dummy-dai";
+ data->dai[1].cpus->of_node = asrc_np;
+ data->dai[1].platforms->of_node = asrc_np;
+ data->dai[1].dynamic = 1;
+ data->dai[1].ignore_pmdown_time = 1;
+ data->dai[1].dpcm_playback = 1;
+ data->dai[1].dpcm_capture = 1;
+ data->dai[1].dpcm_merged_chan = 1;
+
+ data->dai[2].name = "HiFi-ASRC-BE";
+ data->dai[2].stream_name = "HiFi-ASRC-BE";
+ data->dai[2].codecs->dai_name = "wm8962";
+ data->dai[2].codecs->of_node = codec_np;
+ data->dai[2].cpus->dai_name = dev_name(&cpu_pdev->dev);
+ data->dai[2].platforms->name = "snd-soc-dummy";
+ data->dai[2].ops = &imx_hifi_ops;
+ data->dai[2].be_hw_params_fixup = be_hw_params_fixup;
+ data->dai[2].no_pcm = 1;
+ data->dai[2].ignore_pmdown_time = 1;
+ data->dai[2].dpcm_playback = 1;
+ data->dai[2].dpcm_capture = 1;
+ data->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;
+ }
+
+ data->card.dev = &pdev->dev;
+ ret = snd_soc_of_parse_card_name(&data->card, "model");
+ if (ret)
+ goto fail;
+ ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing");
+ if (ret)
+ goto fail;
+ data->card.owner = THIS_MODULE;
+ data->card.dai_link = data->dai;
+ data->card.dapm_widgets = imx_wm8962_dapm_widgets;
+ data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets);
+
+ data->card.late_probe = imx_wm8962_late_probe;
+
+#ifdef CONFIG_SND_SOC_IMX_WM8962_ANDROID
+ data->card.set_bias_level = imx_wm8962_set_bias_level;
+#endif
+ platform_set_drvdata(pdev, &data->card);
+ snd_soc_card_set_drvdata(&data->card, data);
+
+ ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ goto fail;
+ }
+
+ imx_wm8962_gpio_init(&data->card);
+
+ if (gpio_is_valid(priv->hp_gpio)) {
+ ret = driver_create_file(pdev->dev.driver, &driver_attr_headphone);
+ if (ret) {
+ dev_err(&pdev->dev, "create hp attr failed (%d)\n", ret);
+ goto fail_hp;
+ }
+ }
+
+ if (gpio_is_valid(priv->mic_gpio)) {
+ ret = driver_create_file(pdev->dev.driver, &driver_attr_microphone);
+ if (ret) {
+ dev_err(&pdev->dev, "create mic attr failed (%d)\n", ret);
+ goto fail_mic;
+ }
+ }
+
+ goto fail;
+
+fail_mic:
+ driver_remove_file(pdev->dev.driver, &driver_attr_headphone);
+fail_hp:
+fail:
+ of_node_put(cpu_np);
+ of_node_put(codec_np);
+
+ return ret;
+}
+
+static int imx_wm8962_remove(struct platform_device *pdev)
+{
+ driver_remove_file(pdev->dev.driver, &driver_attr_microphone);
+ driver_remove_file(pdev->dev.driver, &driver_attr_headphone);
+
+ return 0;
+}
+
+static const struct of_device_id imx_wm8962_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-wm8962", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids);
+
+static struct platform_driver imx_wm8962_driver = {
+ .driver = {
+ .name = "imx-wm8962",
+ .pm = &snd_soc_pm_ops,
+ .of_match_table = imx_wm8962_dt_ids,
+ },
+ .probe = imx_wm8962_probe,
+ .remove = imx_wm8962_remove,
+};
+module_platform_driver(imx_wm8962_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-wm8962");