1 files changed, 349 insertions, 0 deletions

diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c
new file mode 100644
index 0000000..93bb8ee
--- /dev/null
+++ b/sound/soc/omap/omap-abe-twl6040.c
@@ -0,0 +1,349 @@
+/*
+ * omap-abe-twl6040.c  --  SoC audio for TI OMAP based boards with ABE and
+ *			   twl6040 codec
+ *
+ * Author: Misael Lopez Cruz <misael.lopez@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/twl6040.h>
+#include <linux/platform_data/omap-abe-twl6040.h>
+#include <linux/module.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+#include <plat/hardware.h>
+#include <plat/mux.h>
+
+#include "omap-dmic.h"
+#include "omap-mcpdm.h"
+#include "omap-pcm.h"
+#include "../codecs/twl6040.h"
+
+static int omap_abe_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;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = codec->card;
+	struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+	int clk_id, freq;
+	int ret;
+
+	clk_id = twl6040_get_clk_id(rtd->codec);
+	if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
+		freq = pdata->mclk_freq;
+	else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
+		freq = 32768;
+	else
+		return -EINVAL;
+
+	/* set the codec mclk */
+	ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
+				SND_SOC_CLOCK_IN);
+	if (ret) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+	return ret;
+}
+
+static struct snd_soc_ops omap_abe_ops = {
+	.hw_params = omap_abe_hw_params,
+};
+
+static int omap_abe_dmic_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;
+	int ret = 0;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
+				     19200000, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set DMIC cpu system clock\n");
+		return ret;
+	}
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
+				     SND_SOC_CLOCK_OUT);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set DMIC output clock\n");
+		return ret;
+	}
+	return 0;
+}
+
+static struct snd_soc_ops omap_abe_dmic_ops = {
+	.hw_params = omap_abe_dmic_hw_params,
+};
+
+/* Headset jack */
+static struct snd_soc_jack hs_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headset Stereophone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+/* SDP4430 machine DAPM */
+static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
+	/* Outputs */
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_SPK("Earphone Spk", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_SPK("Vibrator", NULL),
+
+	/* Inputs */
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
+	SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Routings for outputs */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+
+	{"Earphone Spk", NULL, "EP"},
+
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	{"Line Out", NULL, "AUXL"},
+	{"Line Out", NULL, "AUXR"},
+
+	{"Vibrator", NULL, "VIBRAL"},
+	{"Vibrator", NULL, "VIBRAR"},
+
+	/* Routings for inputs */
+	{"HSMIC", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "Headset Mic Bias"},
+
+	{"MAINMIC", NULL, "Main Handset Mic"},
+	{"Main Handset Mic", NULL, "Main Mic Bias"},
+
+	{"SUBMIC", NULL, "Sub Handset Mic"},
+	{"Sub Handset Mic", NULL, "Main Mic Bias"},
+
+	{"AFML", NULL, "Line In"},
+	{"AFMR", NULL, "Line In"},
+};
+
+static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm,
+					  int connected, char *pin)
+{
+	if (!connected)
+		snd_soc_dapm_disable_pin(dapm, pin);
+}
+
+static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = codec->card;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev);
+	int hs_trim;
+	int ret = 0;
+
+	/* Disable not connected paths if not used */
+	twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
+	twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk");
+	twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk");
+	twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out");
+	twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vinrator");
+	twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
+	twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic");
+	twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic");
+	twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In");
+
+	/*
+	 * Configure McPDM offset cancellation based on the HSOTRIM value from
+	 * twl6040.
+	 */
+	hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
+	omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
+					TWL6040_HSF_TRIM_RIGHT(hs_trim));
+
+	/* Headset jack detection only if it is supported */
+	if (pdata->jack_detection) {
+		ret = snd_soc_jack_new(codec, "Headset Jack",
+					SND_JACK_HEADSET, &hs_jack);
+		if (ret)
+			return ret;
+
+		ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+					hs_jack_pins);
+		twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Digital Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route dmic_audio_map[] = {
+	{"DMic", NULL, "Digital Mic"},
+	{"Digital Mic", NULL, "Digital Mic1 Bias"},
+};
+
+static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
+				ARRAY_SIZE(dmic_dapm_widgets));
+	if (ret)
+		return ret;
+
+	return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
+				ARRAY_SIZE(dmic_audio_map));
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link twl6040_dmic_dai[] = {
+	{
+		.name = "TWL6040",
+		.stream_name = "TWL6040",
+		.cpu_dai_name = "omap-mcpdm",
+		.codec_dai_name = "twl6040-legacy",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "twl6040-codec",
+		.init = omap_abe_twl6040_init,
+		.ops = &omap_abe_ops,
+	},
+	{
+		.name = "DMIC",
+		.stream_name = "DMIC Capture",
+		.cpu_dai_name = "omap-dmic",
+		.codec_dai_name = "dmic-hifi",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "dmic-codec",
+		.init = omap_abe_dmic_init,
+		.ops = &omap_abe_dmic_ops,
+	},
+};
+
+static struct snd_soc_dai_link twl6040_only_dai[] = {
+	{
+		.name = "TWL6040",
+		.stream_name = "TWL6040",
+		.cpu_dai_name = "omap-mcpdm",
+		.codec_dai_name = "twl6040-legacy",
+		.platform_name = "omap-pcm-audio",
+		.codec_name = "twl6040-codec",
+		.init = omap_abe_twl6040_init,
+		.ops = &omap_abe_ops,
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card omap_abe_card = {
+	.owner = THIS_MODULE,
+
+	.dapm_widgets = twl6040_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static __devinit int omap_abe_probe(struct platform_device *pdev)
+{
+	struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev);
+	struct snd_soc_card *card = &omap_abe_card;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	if (!pdata) {
+		dev_err(&pdev->dev, "Missing pdata\n");
+		return -ENODEV;
+	}
+
+	if (pdata->card_name) {
+		card->name = pdata->card_name;
+	} else {
+		dev_err(&pdev->dev, "Card name is not provided\n");
+		return -ENODEV;
+	}
+
+	if (!pdata->mclk_freq) {
+		dev_err(&pdev->dev, "MCLK frequency missing\n");
+		return -ENODEV;
+	}
+
+	if (pdata->has_dmic) {
+		card->dai_link = twl6040_dmic_dai;
+		card->num_links = ARRAY_SIZE(twl6040_dmic_dai);
+	} else {
+		card->dai_link = twl6040_only_dai;
+		card->num_links = ARRAY_SIZE(twl6040_only_dai);
+	}
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+static int __devexit omap_abe_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver omap_abe_driver = {
+	.driver = {
+		.name = "omap-abe-twl6040",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = omap_abe_probe,
+	.remove = __devexit_p(omap_abe_remove),
+};
+
+module_platform_driver(omap_abe_driver);
+
+MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-abe-twl6040");