Adding upstream version 4.19.249.upstream/4.19.249upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 200 insertions, 0 deletions

diff --git a/sound/soc/intel/boards/bytcht_nocodec.c b/sound/soc/intel/boards/bytcht_nocodec.c
new file mode 100644
index 000000000..b80ec027a
--- /dev/null
+++ b/sound/soc/intel/boards/bytcht_nocodec.c
@@ -0,0 +1,200 @@
+/*
+ *  bytcht_nocodec.c - ASoc Machine driver for MinnowBoard Max and Up
+ *  to make I2S signals observable on the Low-Speed connector. Audio codec
+ *  is not managed by ASoC/DAPM
+ *
+ *  Copyright (C) 2015-2017 Intel Corp
+ *
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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 <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../atom/sst-atom-controls.h"
+
+static const struct snd_soc_dapm_widget widgets[] = {
+	SND_SOC_DAPM_MIC("Mic", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_kcontrol_new controls[] = {
+	SOC_DAPM_PIN_SWITCH("Mic"),
+	SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"ssp2 Tx", NULL, "codec_out0"},
+	{"ssp2 Tx", NULL, "codec_out1"},
+	{"codec_in0", NULL, "ssp2 Rx"},
+	{"codec_in1", NULL, "ssp2 Rx"},
+
+	{"ssp2 Rx", NULL, "Mic"},
+	{"Speaker", NULL, "ssp2 Tx"},
+};
+
+static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
+			    struct snd_pcm_hw_params *params)
+{
+	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);
+	int ret;
+
+	/* 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 const unsigned int rates_48000[] = {
+	48000,
+};
+
+static const struct snd_pcm_hw_constraint_list constraints_48000 = {
+	.count = ARRAY_SIZE(rates_48000),
+	.list  = rates_48000,
+};
+
+static int aif1_startup(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE,
+			&constraints_48000);
+}
+
+static struct snd_soc_ops aif1_ops = {
+	.startup = aif1_startup,
+};
+
+static struct snd_soc_dai_link dais[] = {
+	[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",
+		.ignore_suspend = 1,
+		.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",
+		.ignore_suspend = 1,
+		.nonatomic = true,
+		.dynamic = 1,
+		.dpcm_playback = 1,
+		.ops = &aif1_ops,
+	},
+	/* CODEC<->CODEC link */
+	/* back ends */
+	{
+		.name = "SSP2-LowSpeed Connector",
+		.id = 0,
+		.cpu_dai_name = "ssp2-port",
+		.platform_name = "sst-mfld-platform",
+		.no_pcm = 1,
+		.codec_dai_name = "snd-soc-dummy-dai",
+		.codec_name = "snd-soc-dummy",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+						| SND_SOC_DAIFMT_CBS_CFS,
+		.be_hw_params_fixup = codec_fixup,
+		.ignore_suspend = 1,
+		.nonatomic = true,
+		.dpcm_playback = 1,
+		.dpcm_capture = 1,
+	},
+};
+
+/* SoC card */
+static struct snd_soc_card bytcht_nocodec_card = {
+	.name = "bytcht-nocodec",
+	.owner = THIS_MODULE,
+	.dai_link = dais,
+	.num_links = ARRAY_SIZE(dais),
+	.dapm_widgets = widgets,
+	.num_dapm_widgets = ARRAY_SIZE(widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+	.controls = controls,
+	.num_controls = ARRAY_SIZE(controls),
+	.fully_routed = true,
+};
+
+static int snd_bytcht_nocodec_mc_probe(struct platform_device *pdev)
+{
+	int ret_val = 0;
+
+	/* register the soc card */
+	bytcht_nocodec_card.dev = &pdev->dev;
+
+	ret_val = devm_snd_soc_register_card(&pdev->dev, &bytcht_nocodec_card);
+
+	if (ret_val) {
+		dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n",
+			ret_val);
+		return ret_val;
+	}
+	platform_set_drvdata(pdev, &bytcht_nocodec_card);
+	return ret_val;
+}
+
+static struct platform_driver snd_bytcht_nocodec_mc_driver = {
+	.driver = {
+		.name = "bytcht_nocodec",
+	},
+	.probe = snd_bytcht_nocodec_mc_probe,
+};
+module_platform_driver(snd_bytcht_nocodec_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail Nocodec Machine driver");
+MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytcht_nocodec");