1 files changed, 274 insertions, 0 deletions
diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c
new file mode 100644
index 000000000..2179dedb2
--- /dev/null
+++ b/sound/soc/intel/boards/bytcht_da7213.c
@@ -0,0 +1,274 @@
+/*
+ *  bytcht-da7213.c - ASoc Machine driver for Intel Baytrail and
+ *             Cherrytrail-based platforms, with Dialog DA7213 codec
+ *
+ *  Copyright (C) 2017 Intel Corporation
+ *  Author: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
+ *
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful, but
+ *  WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/module.h>
+#include <linux/acpi.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <asm/platform_sst_audio.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-acpi.h>
+#include "../../codecs/da7213.h"
+#include "../atom/sst-atom-controls.h"
+
+static const struct snd_kcontrol_new controls[] = {
+	SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Mic"),
+	SOC_DAPM_PIN_SWITCH("Aux In"),
+};
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+	SND_SOC_DAPM_LINE("Aux In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "HPL"},
+	{"Headphone Jack", NULL, "HPR"},
+
+	{"AUXL", NULL, "Aux In"},
+	{"AUXR", NULL, "Aux In"},
+
+	/* Assume Mic1 is linked to Headset and Mic2 to on-board mic */
+	{"MIC1", NULL, "Headset Mic"},
+	{"MIC2", NULL, "Mic"},
+
+	/* SOC-codec link */
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx"},
+	{"codec_in1", NULL, "ssp2 Rx"},
+
+	{"Playback", NULL, "ssp2 Tx"},
+	{"ssp2 Rx", NULL, "Capture"},
+};
+
+static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
+		       struct snd_pcm_hw_params *params)
+{
+	int ret;
+	struct snd_interval *rate = hw_param_interval(params,
+			SNDRV_PCM_HW_PARAM_RATE);
+	struct snd_interval *channels = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_CHANNELS);
+
+	/* The DSP will convert the FE rate to 48k, stereo, 24bits */
+	rate->min = rate->max = 48000;
+	channels->min = channels->max = 2;
+
+	/* set SSP2 to 24-bit */
+	params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+
+	/*
+	 * Default mode for SSP configuration is TDM 4 slot, override config
+	 * with explicit setting to I2S 2ch 24-bit. The word length is set with
+	 * dai_set_tdm_slot() since there is no other API exposed
+	 */
+	ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
+				  SND_SOC_DAIFMT_I2S     |
+				  SND_SOC_DAIFMT_NB_NF   |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
+	if (ret < 0) {
+		dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_single(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, 48000);
+}
+
+static int aif1_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;
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK,
+				     19200000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		dev_err(codec_dai->dev, "can't set codec sysclk configuration\n");
+
+	ret = snd_soc_dai_set_pll(codec_dai, 0,
+			DA7213_SYSCLK_PLL_SRM, 0, DA7213_PLL_FREQ_OUT_98304000);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret);
+		return -EIO;
+	}
+
+	return ret;
+}
+
+static int aif1_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;
+	int ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, 0,
+				  DA7213_SYSCLK_MCLK, 0, 0);
+	if (ret < 0) {
+		dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret);
+		return -EIO;
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_ops aif1_ops = {
+	.startup = aif1_startup,
+};
+
+static const struct snd_soc_ops ssp2_ops = {
+	.hw_params = aif1_hw_params,
+	.hw_free = aif1_hw_free,
+
+};
+
+static struct snd_soc_dai_link dailink[] = {
+	[MERR_DPCM_AUDIO] = {
+		.name = "Audio Port",
+		.stream_name = "Audio",
+		.cpu_dai_name = "media-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &aif1_ops,
+	},
+	[MERR_DPCM_DEEP_BUFFER] = {
+		.name = "Deep-Buffer Audio Port",
+		.stream_name = "Deep-Buffer Audio",
+		.cpu_dai_name = "deepbuffer-cpu-dai",
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.platform_name = "sst-mfld-platform",
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.ops = &aif1_ops,
+	},
+	/* CODEC<->CODEC link */
+	/* back ends */
+	{
+		.name = "SSP2-Codec",
+		.id = 0,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "da7213-hifi",
+		.codec_name = "i2c-DLGS7213:00",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+						| SND_SOC_DAIFMT_CBS_CFS,
+		.be_hw_params_fixup = codec_fixup,
+		.nonatomic = true,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+		.ops = &ssp2_ops,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card bytcht_da7213_card = {
+	.name = "bytcht-da7213",
+	.owner = THIS_MODULE,
+	.dai_link = dailink,
+	.num_links = ARRAY_SIZE(dailink),
+	.controls = controls,
+	.num_controls = ARRAY_SIZE(controls),
+	.dapm_widgets = dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static char codec_name[SND_ACPI_I2C_ID_LEN];
+
+static int bytcht_da7213_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card;
+	struct snd_soc_acpi_mach *mach;
+	const char *i2c_name = NULL;
+	int dai_index = 0;
+	int ret_val = 0;
+	int i;
+
+	mach = (&pdev->dev)->platform_data;
+	card = &bytcht_da7213_card;
+	card->dev = &pdev->dev;
+
+	/* fix index of codec dai */
+	for (i = 0; i < ARRAY_SIZE(dailink); i++) {
+		if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) {
+			dai_index = i;
+			break;
+		}
+	}
+
+	/* fixup codec name based on HID */
+	i2c_name = acpi_dev_get_first_match_name(mach->id, NULL, -1);
+	if (i2c_name) {
+		snprintf(codec_name, sizeof(codec_name),
+			"%s%s", "i2c-", i2c_name);
+		dailink[dai_index].codec_name = codec_name;
+	}
+
+	ret_val = devm_snd_soc_register_card(&pdev->dev, card);
+	if (ret_val) {
+		dev_err(&pdev->dev,
+			"snd_soc_register_card failed %d\n", ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, card);
+	return ret_val;
+}
+
+static struct platform_driver bytcht_da7213_driver = {
+	.driver = {
+		.name = "bytcht_da7213",
+	},
+	.probe = bytcht_da7213_probe,
+};
+module_platform_driver(bytcht_da7213_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail+DA7213 Machine driver");
+MODULE_AUTHOR("Pierre-Louis Bossart");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytcht_da7213");