--- /dev/null
+/*
+ * Copyright (C) 2010-2014 Freescale Semiconductor, Inc. All Rights Reserved.
+ */
+
+/*
+ * 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.h>
+#include <linux/of_platform.h>
+#include <linux/slab.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_esai.h"
+
+#define CODEC_CLK_EXTER_OSC 1
+#define CODEC_CLK_ESAI_HCKT 2
+#define SUPPORT_RATE_NUM 10
+
+struct imx_priv {
+ unsigned int mclk_freq;
+ struct platform_device *pdev;
+ struct platform_device *asrc_pdev;
+ u32 asrc_rate;
+ u32 asrc_format;
+};
+
+static struct imx_priv card_priv;
+
+static int imx_cs42888_surround_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 *cpu_dai = rtd->cpu_dai;
+ struct snd_soc_dai *codec_dai = rtd->codec_dai;
+ struct imx_priv *priv = &card_priv;
+ u32 dai_format = 0;
+
+ dai_format = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKT_EXTAL,
+ priv->mclk_freq, SND_SOC_CLOCK_OUT);
+ else
+ snd_soc_dai_set_sysclk(cpu_dai, ESAI_HCKR_EXTAL,
+ priv->mclk_freq, SND_SOC_CLOCK_OUT);
+ snd_soc_dai_set_sysclk(codec_dai, 0, priv->mclk_freq, SND_SOC_CLOCK_IN);
+
+ /* set cpu DAI configuration */
+ snd_soc_dai_set_fmt(cpu_dai, dai_format);
+ /* set i.MX active slot mask */
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
+
+ /* set codec DAI configuration */
+ snd_soc_dai_set_fmt(codec_dai, dai_format);
+ return 0;
+}
+
+static int imx_cs42888_surround_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ static struct snd_pcm_hw_constraint_list constraint_rates;
+ struct imx_priv *priv = &card_priv;
+ struct device *dev = &priv->pdev->dev;
+ static u32 support_rates[SUPPORT_RATE_NUM];
+ int ret;
+
+ if (priv->mclk_freq == 24576000) {
+ support_rates[0] = 48000;
+ support_rates[1] = 96000;
+ support_rates[2] = 192000;
+ constraint_rates.list = support_rates;
+ constraint_rates.count = 3;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE,
+ &constraint_rates);
+ if (ret)
+ return ret;
+ } else
+ dev_warn(dev, "mclk may be not supported %d\n", priv->mclk_freq);
+
+ return 0;
+}
+
+static struct snd_soc_ops imx_cs42888_surround_ops = {
+ .startup = imx_cs42888_surround_startup,
+ .hw_params = imx_cs42888_surround_hw_params,
+};
+
+/**
+ * imx_cs42888_surround_startup() is to set constrain for hw parameter, but
+ * backend use same runtime as frontend, for p2p backend need to use different
+ * parameter, so backend can't use the startup.
+ */
+static struct snd_soc_ops imx_cs42888_surround_ops_be = {
+ .hw_params = imx_cs42888_surround_hw_params,
+};
+
+
+static const struct snd_soc_dapm_widget imx_cs42888_dapm_widgets[] = {
+ SND_SOC_DAPM_LINE("Line Out Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Line out jack */
+ {"Line Out Jack", NULL, "AOUT1L"},
+ {"Line Out Jack", NULL, "AOUT1R"},
+ {"Line Out Jack", NULL, "AOUT2L"},
+ {"Line Out Jack", NULL, "AOUT2R"},
+ {"Line Out Jack", NULL, "AOUT3L"},
+ {"Line Out Jack", NULL, "AOUT3R"},
+ {"Line Out Jack", NULL, "AOUT4L"},
+ {"Line Out Jack", NULL, "AOUT4R"},
+ {"AIN1L", NULL, "Line In Jack"},
+ {"AIN1R", NULL, "Line In Jack"},
+ {"AIN2L", NULL, "Line In Jack"},
+ {"AIN2R", NULL, "Line In Jack"},
+ {"CPU-Playback", NULL, "ASRC-Playback"},
+ {"Playback", NULL, "CPU-Playback"},/* dai route for be and fe */
+ {"ASRC-Capture", NULL, "CPU-Capture"},
+ {"CPU-Capture", NULL, "Capture"},
+};
+
+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 struct snd_soc_dai_link imx_cs42888_dai[] = {
+ {
+ .name = "HiFi",
+ .stream_name = "HiFi",
+ .codec_dai_name = "cs42888",
+ .ops = &imx_cs42888_surround_ops,
+ .ignore_pmdown_time = 1,
+ },
+ {
+ .name = "HiFi-ASRC-FE",
+ .stream_name = "HiFi-ASRC-FE",
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .dynamic = 1,
+ .ignore_pmdown_time = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "HiFi-ASRC-BE",
+ .stream_name = "HiFi-ASRC-BE",
+ .codec_dai_name = "cs42888",
+ .platform_name = "snd-soc-dummy",
+ .no_pcm = 1,
+ .ignore_pmdown_time = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &imx_cs42888_surround_ops_be,
+ .be_hw_params_fixup = be_hw_params_fixup,
+ },
+};
+
+static struct snd_soc_card snd_soc_card_imx_cs42888 = {
+ .name = "cs42888-audio",
+ .dai_link = imx_cs42888_dai,
+ .dapm_widgets = imx_cs42888_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(imx_cs42888_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+/*
+ * This function will register the snd_soc_pcm_link drivers.
+ */
+static int imx_cs42888_probe(struct platform_device *pdev)
+{
+ struct device_node *esai_np, *codec_np;
+ struct device_node *asrc_np;
+ struct platform_device *esai_pdev;
+ struct platform_device *asrc_pdev = NULL;
+ struct i2c_client *codec_dev;
+ struct imx_priv *priv = &card_priv;
+ struct clk *codec_clk = NULL;
+ int ret;
+ u32 width;
+
+ priv->pdev = pdev;
+ priv->asrc_pdev = NULL;
+
+ esai_np = of_parse_phandle(pdev->dev.of_node, "esai-controller", 0);
+ codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0);
+ if (!esai_np || !codec_np) {
+ dev_err(&pdev->dev, "phandle missing or invalid\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;
+ }
+
+ esai_pdev = of_find_device_by_node(esai_np);
+ if (!esai_pdev) {
+ dev_err(&pdev->dev, "failed to find ESAI platform 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;
+ }
+
+ /*if there is no asrc controller, we only enable one device*/
+ if (!asrc_pdev) {
+ imx_cs42888_dai[0].codec_of_node = codec_np;
+ imx_cs42888_dai[0].cpu_dai_name = dev_name(&esai_pdev->dev);
+ imx_cs42888_dai[0].platform_of_node = esai_np;
+ snd_soc_card_imx_cs42888.num_links = 1;
+ } else {
+ imx_cs42888_dai[0].codec_of_node = codec_np;
+ imx_cs42888_dai[0].cpu_dai_name = dev_name(&esai_pdev->dev);
+ imx_cs42888_dai[0].platform_of_node = esai_np;
+ imx_cs42888_dai[1].cpu_of_node = asrc_np;
+ imx_cs42888_dai[1].platform_of_node = asrc_np;
+ imx_cs42888_dai[2].codec_of_node = codec_np;
+ imx_cs42888_dai[2].cpu_dai_name = dev_name(&esai_pdev->dev);
+ snd_soc_card_imx_cs42888.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;
+ }
+
+ codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+ if (IS_ERR(codec_clk)) {
+ ret = PTR_ERR(codec_clk);
+ dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret);
+ goto fail;
+ }
+ priv->mclk_freq = clk_get_rate(codec_clk);
+
+ snd_soc_card_imx_cs42888.dev = &pdev->dev;
+
+ platform_set_drvdata(pdev, &snd_soc_card_imx_cs42888);
+
+ ret = snd_soc_register_card(&snd_soc_card_imx_cs42888);
+ if (ret)
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+fail:
+ if (asrc_np)
+ of_node_put(asrc_np);
+ if (esai_np)
+ of_node_put(esai_np);
+ if (codec_np)
+ of_node_put(codec_np);
+ return ret;
+}
+
+static int imx_cs42888_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&snd_soc_card_imx_cs42888);
+ return 0;
+}
+
+static const struct of_device_id imx_cs42888_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-cs42888", },
+ { /* sentinel */ }
+};
+
+static struct platform_driver imx_cs42888_driver = {
+ .probe = imx_cs42888_probe,
+ .remove = imx_cs42888_remove,
+ .driver = {
+ .name = "imx-cs42888",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ .of_match_table = imx_cs42888_dt_ids,
+ },
+};
+module_platform_driver(imx_cs42888_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("ALSA SoC cs42888 Machine Layer Driver");
+MODULE_ALIAS("platform:imx-cs42888");
+MODULE_LICENSE("GPL");
projects / karo-tx-linux.git / commitdiff