@@ -220,4 +220,14 @@ config SND_SOC_FSL_PCM
tristate
select SND_SOC_GENERIC_DMAENGINE_PCM
+config SND_SOC_FSL_SGTL5000
+ tristate "SoC Audio support for FSL boards with sgtl5000"
+ depends on OF && I2C
+ select SND_SOC_FSL_SAI
+ select SND_SOC_FSL_PCM
+ select SND_SOC_SGTL5000
+ help
+ Say Y if you want to add support for SoC audio on an FSL board with
+ a sgtl5000 codec.
+
endif # SND_FSL_SOC
@@ -58,6 +58,8 @@ obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
# FSL ARM SAI/SGT15000 Platform Support
snd-soc-fsl-sai-objs := fsl-sai.o
snd-soc-fsl-pcm-objs := fsl-pcm-dma.o
+snd-soc-fsl-sgtl5000-objs := fsl-sgtl5000.o
obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o
obj-$(CONFIG_SND_SOC_FSL_PCM) += snd-soc-fsl-pcm.o
+obj-$(CONFIG_SND_SOC_FSL_SGTL5000) += snd-soc-fsl-sgtl5000.o
new file mode 100644
@@ -0,0 +1,208 @@
+/*
+ * Freeacale ALSA SoC Audio using SGT1500 as codec.
+ *
+ * Copyright 2012-2013 Freescale Semiconductor, Inc.
+ *
+ * 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:
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/i2c.h>
+#include <linux/clk.h>
+
+#include "../codecs/sgtl5000.h"
+#include "fsl-sai.h"
+
+static unsigned int sysclk_rate;
+
+static int fsl_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret;
+ struct device *dev = rtd->card->dev;
+
+ ret = snd_soc_dai_set_sysclk(rtd->codec_dai, SGTL5000_SYSCLK,
+ sysclk_rate, SND_SOC_CLOCK_IN);
+ if (ret) {
+ dev_err(dev, "could not set codec driver clock params :%d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, FSL_SAI_CLK_BUS,
+ sysclk_rate, SND_SOC_CLOCK_OUT);
+ if (ret) {
+ dev_err(dev, "could not set cpu dai driver clock params :%d\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int sgtl5000_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;
+ unsigned int channels = params_channels(params);
+
+ /* TODO: The SAI driver should figure this out for us */
+ switch (channels) {
+ case 2:
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffc, 0xfffffffc, 2, 0);
+ break;
+ case 1:
+ snd_soc_dai_set_tdm_slot(cpu_dai, 0xfffffffe, 0xfffffffe, 1, 0);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_ops fsl_sgtl5000_hifi_ops = {
+ .hw_params = sgtl5000_params,
+};
+
+static struct snd_soc_dai_link fsl_sgtl5000_dai = {
+ .name = "HiFi",
+ .stream_name = "HiFi",
+ .codec_dai_name = "sgtl5000",
+ .init = &fsl_sgtl5000_dai_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM,
+ .ops = &fsl_sgtl5000_hifi_ops,
+};
+
+static const struct snd_soc_dapm_widget fsl_sgtl5000_dapm_widgets[] = {
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In Jack", NULL),
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Line Out Jack", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static struct snd_soc_card fsl_sgt1500_card = {
+ .owner = THIS_MODULE,
+ .num_links = 1,
+ .dai_link = &fsl_sgtl5000_dai,
+ .dapm_widgets = fsl_sgtl5000_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(fsl_sgtl5000_dapm_widgets),
+};
+
+static int fsl_sgtl5000_parse_dt(struct platform_device *pdev)
+{
+ int ret;
+ struct device_node *sai_np, *codec_np;
+ struct clk *codec_clk;
+ struct i2c_client *codec_dev;
+ struct device_node *np = pdev->dev.of_node;
+
+ ret = snd_soc_of_parse_card_name(&fsl_sgt1500_card, "model");
+ if (ret)
+ return ret;
+
+ ret = snd_soc_of_parse_audio_routing(&fsl_sgt1500_card,
+ "audio-routing");
+ if (ret)
+ return ret;
+
+ sai_np = of_parse_phandle(np, "saif-controller", 0);
+ if (!sai_np) {
+ dev_err(&pdev->dev, "\"saif-controller\" phandle missing or "
+ "invalid\n");
+ return -EINVAL;
+ }
+ fsl_sgtl5000_dai.cpu_of_node = sai_np;
+ fsl_sgtl5000_dai.platform_of_node = sai_np;
+
+ codec_np = of_parse_phandle(np, "audio-codec", 0);
+ if (!codec_np) {
+ dev_err(&pdev->dev, "\"audio-codec\" phandle missing or "
+ "invalid\n");
+ ret = -EINVAL;
+ goto sai_np_fail;
+ }
+ fsl_sgtl5000_dai.codec_of_node = codec_np;
+
+ 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 = PTR_ERR(codec_dev);
+ goto codec_np_fail;
+ }
+
+ codec_clk = devm_clk_get(&codec_dev->dev, NULL);
+ if (IS_ERR(codec_clk)) {
+ dev_err(&pdev->dev, "failed to get codec clock\n");
+ ret = PTR_ERR(codec_clk);
+ goto codec_np_fail;
+ }
+
+ sysclk_rate = clk_get_rate(codec_clk);
+
+codec_np_fail:
+ of_node_put(codec_np);
+sai_np_fail:
+ of_node_put(sai_np);
+
+ return ret;
+}
+
+static int fsl_sgtl5000_probe(struct platform_device *pdev)
+{
+ int ret;
+
+ fsl_sgt1500_card.dev = &pdev->dev;
+
+ ret = fsl_sgtl5000_parse_dt(pdev);
+ if (ret) {
+ dev_err(&pdev->dev,
+ "parse sgtl5000 device tree failed :%d\n",
+ ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_card(&fsl_sgt1500_card);
+ if (ret) {
+ dev_err(&pdev->dev, "register soc sound card failed :%d\n",
+ ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int fsl_sgtl5000_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&fsl_sgt1500_card);
+
+ return 0;
+}
+
+static const struct of_device_id fsl_sgtl5000_dt_ids[] = {
+ { .compatible = "fsl,vf610-sgtl5000", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, fsl_sgtl5000_dt_ids);
+
+static struct platform_driver fsl_sgtl5000_driver = {
+ .driver = {
+ .name = "fsl-sgtl5000",
+ .owner = THIS_MODULE,
+ .of_match_table = fsl_sgtl5000_dt_ids,
+ },
+ .probe = fsl_sgtl5000_probe,
+ .remove = fsl_sgtl5000_remove,
+};
+module_platform_driver(fsl_sgtl5000_driver);
+
+MODULE_AUTHOR("Xiubo Li <Li.Xiubo@freescale.com>");
+MODULE_DESCRIPTION("Freescale SGTL5000 ASoC driver");
+MODULE_LICENSE("GPL");