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Diffstat (limited to 'sound/soc/intel/boards/bytcht_da7213.c')
-rw-r--r--sound/soc/intel/boards/bytcht_da7213.c283
1 files changed, 283 insertions, 0 deletions
diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c
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+++ b/sound/soc/intel/boards/bytcht_da7213.c
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+/*
+ * 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 "../../codecs/da7213.h"
+#include "../atom/sst-atom-controls.h"
+#include "../common/sst-acpi.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,
+ },
+ [MERR_DPCM_COMPR] = {
+ .name = "Compressed Port",
+ .stream_name = "Compress",
+ .cpu_dai_name = "compress-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ },
+ /* CODEC<->CODEC link */
+ /* back ends */
+ {
+ .name = "SSP2-Codec",
+ .id = 1,
+ .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[16]; /* i2c-<HID>:00 with HID being 8 chars */
+
+static int bytcht_da7213_probe(struct platform_device *pdev)
+{
+ int ret_val = 0;
+ int i;
+ struct snd_soc_card *card;
+ struct sst_acpi_mach *mach;
+ const char *i2c_name = NULL;
+ int dai_index = 0;
+
+ mach = (&pdev->dev)->platform_data;
+ card = &bytcht_da7213_card;
+ card->dev = &pdev->dev;
+
+ /* fix index of codec dai */
+ dai_index = MERR_DPCM_COMPR + 1;
+ 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 = sst_acpi_find_name_from_hid(mach->id);
+ if (i2c_name != NULL) {
+ 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");