ASoC: ti: Merge davinci and omap directories

Create new directory to contain all Texas Instruments specific DAI,
platform and machine drivers instead of scattering them under davinci and
omap directories.

There is already inter dependency between the two directories becasue of
McASP (on dra7x it is serviced by sDMA, not EDMA).

With the upcoming AM654 we will need to introduce new platform driver for
UDMA and it does not fit under davinci, nor under omap.

With the move I have restructured the Kconfig to be more usable in the era
of simple-sound-card:
CPU DAIs can be selected individually and they will select the platform
driver they can be served with.

To avoid breakage, I have moved over deprecated Kconfig options so
defconfig builds will work without regression.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
For sound/soc/{omap => ti}:
Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

1 files changed, 594 insertions, 0 deletions

diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
new file mode 100644
index 000000000000..4dce494dfbd3
--- /dev/null
+++ b/sound/soc/ti/ams-delta.c
@@ -0,0 +1,594 @@
+/*
+ * ams-delta.c  --  SoC audio for Amstrad E3 (Delta) videophone
+ *
+ * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
+ *
+ * Initially based on sound/soc/omap/osk5912.x
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * 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/gpio/consumer.h>
+#include <linux/spinlock.h>
+#include <linux/tty.h>
+#include <linux/module.h>
+
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include <linux/platform_data/asoc-ti-mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "../codecs/cx20442.h"
+
+/* Board specific DAPM widgets */
+static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
+	/* Handset */
+	SND_SOC_DAPM_MIC("Mouthpiece", NULL),
+	SND_SOC_DAPM_HP("Earpiece", NULL),
+	/* Handsfree/Speakerphone */
+	SND_SOC_DAPM_MIC("Microphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* How they are connected to codec pins */
+static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
+	{"TELIN", NULL, "Mouthpiece"},
+	{"Earpiece", NULL, "TELOUT"},
+
+	{"MIC", NULL, "Microphone"},
+	{"Speaker", NULL, "SPKOUT"},
+};
+
+/*
+ * Controls, functional after the modem line discipline is activated.
+ */
+
+/* Virtual switch: audio input/output constellations */
+static const char *ams_delta_audio_mode[] =
+	{"Mixed", "Handset", "Handsfree", "Speakerphone"};
+
+/* Selection <-> pin translation */
+#define AMS_DELTA_MOUTHPIECE	0
+#define AMS_DELTA_EARPIECE	1
+#define AMS_DELTA_MICROPHONE	2
+#define AMS_DELTA_SPEAKER	3
+#define AMS_DELTA_AGC		4
+
+#define AMS_DELTA_MIXED		((1 << AMS_DELTA_EARPIECE) | \
+						(1 << AMS_DELTA_MICROPHONE))
+#define AMS_DELTA_HANDSET	((1 << AMS_DELTA_MOUTHPIECE) | \
+						(1 << AMS_DELTA_EARPIECE))
+#define AMS_DELTA_HANDSFREE	((1 << AMS_DELTA_MICROPHONE) | \
+						(1 << AMS_DELTA_SPEAKER))
+#define AMS_DELTA_SPEAKERPHONE	(AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
+
+static const unsigned short ams_delta_audio_mode_pins[] = {
+	AMS_DELTA_MIXED,
+	AMS_DELTA_HANDSET,
+	AMS_DELTA_HANDSFREE,
+	AMS_DELTA_SPEAKERPHONE,
+};
+
+static unsigned short ams_delta_audio_agc;
+
+/*
+ * Used for passing a codec structure pointer
+ * from the board initialization code to the tty line discipline.
+ */
+static struct snd_soc_component *cx20442_codec;
+
+static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+	struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
+	unsigned short pins;
+	int pin, changed = 0;
+
+	/* Refuse any mode changes if we are not able to control the codec. */
+	if (!cx20442_codec->card->pop_time)
+		return -EUNATCH;
+
+	if (ucontrol->value.enumerated.item[0] >= control->items)
+		return -EINVAL;
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	/* Translate selection to bitmap */
+	pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
+
+	/* Setup pins after corresponding bits if changed */
+	pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
+
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin_unlocked(dapm, "Mouthpiece");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "Earpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "Microphone");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_AGC));
+	if (pin != ams_delta_audio_agc) {
+		ams_delta_audio_agc = pin;
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin_unlocked(dapm, "AGCIN");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
+	}
+
+	if (changed)
+		snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	return changed;
+}
+
+static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+	unsigned short pins, mode;
+
+	pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
+							AMS_DELTA_MOUTHPIECE) |
+			(snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
+							AMS_DELTA_EARPIECE));
+	if (pins)
+		pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
+							AMS_DELTA_MICROPHONE);
+	else
+		pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
+							AMS_DELTA_MICROPHONE) |
+			(snd_soc_dapm_get_pin_status(dapm, "Speaker") <<
+							AMS_DELTA_SPEAKER) |
+			(ams_delta_audio_agc << AMS_DELTA_AGC));
+
+	for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
+		if (pins == ams_delta_audio_mode_pins[mode])
+			break;
+
+	if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
+		return -EINVAL;
+
+	ucontrol->value.enumerated.item[0] = mode;
+
+	return 0;
+}
+
+static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum,
+				      ams_delta_audio_mode);
+
+static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
+	SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum,
+			ams_delta_get_audio_mode, ams_delta_set_audio_mode),
+};
+
+/* Hook switch */
+static struct snd_soc_jack ams_delta_hook_switch;
+static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
+	{
+		.name = "hook_switch",
+		.report = SND_JACK_HEADSET,
+		.invert = 1,
+		.debounce_time = 150,
+	}
+};
+
+/* After we are able to control the codec over the modem,
+ * the hook switch can be used for dynamic DAPM reconfiguration. */
+static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
+	/* Handset */
+	{
+		.pin = "Mouthpiece",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Earpiece",
+		.mask = SND_JACK_HEADPHONE,
+	},
+	/* Handsfree */
+	{
+		.pin = "Microphone",
+		.mask = SND_JACK_MICROPHONE,
+		.invert = 1,
+	},
+	{
+		.pin = "Speaker",
+		.mask = SND_JACK_HEADPHONE,
+		.invert = 1,
+	},
+};
+
+
+/*
+ * Modem line discipline, required for making above controls functional.
+ * Activated from userspace with ldattach, possibly invoked from udev rule.
+ */
+
+/* To actually apply any modem controlled configuration changes to the codec,
+ * we must connect codec DAI pins to the modem for a moment.  Be careful not
+ * to interfere with our digital mute function that shares the same hardware. */
+static struct timer_list cx81801_timer;
+static bool cx81801_cmd_pending;
+static bool ams_delta_muted;
+static DEFINE_SPINLOCK(ams_delta_lock);
+static struct gpio_desc *gpiod_modem_codec;
+
+static void cx81801_timeout(struct timer_list *unused)
+{
+	int muted;
+
+	spin_lock(&ams_delta_lock);
+	cx81801_cmd_pending = 0;
+	muted = ams_delta_muted;
+	spin_unlock(&ams_delta_lock);
+
+	/* Reconnect the codec DAI back from the modem to the CPU DAI
+	 * only if digital mute still off */
+	if (!muted)
+		gpiod_set_value(gpiod_modem_codec, 0);
+}
+
+/* Line discipline .open() */
+static int cx81801_open(struct tty_struct *tty)
+{
+	int ret;
+
+	if (!cx20442_codec)
+		return -ENODEV;
+
+	/*
+	 * Pass the codec structure pointer for use by other ldisc callbacks,
+	 * both the card and the codec specific parts.
+	 */
+	tty->disc_data = cx20442_codec;
+
+	ret = v253_ops.open(tty);
+
+	if (ret < 0)
+		tty->disc_data = NULL;
+
+	return ret;
+}
+
+/* Line discipline .close() */
+static void cx81801_close(struct tty_struct *tty)
+{
+	struct snd_soc_component *component = tty->disc_data;
+	struct snd_soc_dapm_context *dapm = &component->card->dapm;
+
+	del_timer_sync(&cx81801_timer);
+
+	/* Prevent the hook switch from further changing the DAPM pins */
+	INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
+
+	if (!component)
+		return;
+
+	v253_ops.close(tty);
+
+	/* Revert back to default audio input/output constellation */
+	snd_soc_dapm_mutex_lock(dapm);
+
+	snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+/* Line discipline .hangup() */
+static int cx81801_hangup(struct tty_struct *tty)
+{
+	cx81801_close(tty);
+	return 0;
+}
+
+/* Line discipline .receive_buf() */
+static void cx81801_receive(struct tty_struct *tty,
+				const unsigned char *cp, char *fp, int count)
+{
+	struct snd_soc_component *component = tty->disc_data;
+	const unsigned char *c;
+	int apply, ret;
+
+	if (!component)
+		return;
+
+	if (!component->card->pop_time) {
+		/* First modem response, complete setup procedure */
+
+		/* Initialize timer used for config pulse generation */
+		timer_setup(&cx81801_timer, cx81801_timeout, 0);
+
+		v253_ops.receive_buf(tty, cp, fp, count);
+
+		/* Link hook switch to DAPM pins */
+		ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
+					ARRAY_SIZE(ams_delta_hook_switch_pins),
+					ams_delta_hook_switch_pins);
+		if (ret)
+			dev_warn(component->dev,
+				"Failed to link hook switch to DAPM pins, "
+				"will continue with hook switch unlinked.\n");
+
+		return;
+	}
+
+	v253_ops.receive_buf(tty, cp, fp, count);
+
+	for (c = &cp[count - 1]; c >= cp; c--) {
+		if (*c != '\r')
+			continue;
+		/* Complete modem response received, apply config to codec */
+
+		spin_lock_bh(&ams_delta_lock);
+		mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
+		apply = !ams_delta_muted && !cx81801_cmd_pending;
+		cx81801_cmd_pending = 1;
+		spin_unlock_bh(&ams_delta_lock);
+
+		/* Apply config pulse by connecting the codec to the modem
+		 * if not already done */
+		if (apply)
+			gpiod_set_value(gpiod_modem_codec, 1);
+		break;
+	}
+}
+
+/* Line discipline .write_wakeup() */
+static void cx81801_wakeup(struct tty_struct *tty)
+{
+	v253_ops.write_wakeup(tty);
+}
+
+static struct tty_ldisc_ops cx81801_ops = {
+	.magic = TTY_LDISC_MAGIC,
+	.name = "cx81801",
+	.owner = THIS_MODULE,
+	.open = cx81801_open,
+	.close = cx81801_close,
+	.hangup = cx81801_hangup,
+	.receive_buf = cx81801_receive,
+	.write_wakeup = cx81801_wakeup,
+};
+
+
+/*
+ * Even if not very useful, the sound card can still work without any of the
+ * above functonality activated.  You can still control its audio input/output
+ * constellation and speakerphone gain from userspace by issuing AT commands
+ * over the modem port.
+ */
+
+static struct snd_soc_ops ams_delta_ops;
+
+
+/* Digital mute implemented using modem/CPU multiplexer.
+ * Shares hardware with codec config pulse generation */
+static bool ams_delta_muted = 1;
+
+static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	int apply;
+
+	if (ams_delta_muted == mute)
+		return 0;
+
+	spin_lock_bh(&ams_delta_lock);
+	ams_delta_muted = mute;
+	apply = !cx81801_cmd_pending;
+	spin_unlock_bh(&ams_delta_lock);
+
+	if (apply)
+		gpiod_set_value(gpiod_modem_codec, !!mute);
+	return 0;
+}
+
+/* Our codec DAI probably doesn't have its own .ops structure */
+static const struct snd_soc_dai_ops ams_delta_dai_ops = {
+	.digital_mute = ams_delta_digital_mute,
+};
+
+/* Will be used if the codec ever has its own digital_mute function */
+static int ams_delta_startup(struct snd_pcm_substream *substream)
+{
+	return ams_delta_digital_mute(NULL, 0);
+}
+
+static void ams_delta_shutdown(struct snd_pcm_substream *substream)
+{
+	ams_delta_digital_mute(NULL, 1);
+}
+
+
+/*
+ * Card initialization
+ */
+
+static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+	int ret;
+	/* Codec is ready, now add/activate board specific controls */
+
+	/* Store a pointer to the codec structure for tty ldisc use */
+	cx20442_codec = rtd->codec_dai->component;
+
+	/* Add hook switch - can be used to control the codec from userspace
+	 * even if line discipline fails */
+	ret = snd_soc_card_jack_new(card, "hook_switch", SND_JACK_HEADSET,
+				    &ams_delta_hook_switch, NULL, 0);
+	if (ret)
+		dev_warn(card->dev,
+				"Failed to allocate resources for hook switch, "
+				"will continue without one.\n");
+	else {
+		ret = snd_soc_jack_add_gpiods(card->dev, &ams_delta_hook_switch,
+					ARRAY_SIZE(ams_delta_hook_switch_gpios),
+					ams_delta_hook_switch_gpios);
+		if (ret)
+			dev_warn(card->dev,
+				"Failed to set up hook switch GPIO line, "
+				"will continue with hook switch inactive.\n");
+	}
+
+	gpiod_modem_codec = devm_gpiod_get(card->dev, "modem_codec",
+					   GPIOD_OUT_HIGH);
+	if (IS_ERR(gpiod_modem_codec)) {
+		dev_warn(card->dev, "Failed to obtain modem_codec GPIO\n");
+		return 0;
+	}
+
+	/* Set up digital mute if not provided by the codec */
+	if (!codec_dai->driver->ops) {
+		codec_dai->driver->ops = &ams_delta_dai_ops;
+	} else {
+		ams_delta_ops.startup = ams_delta_startup;
+		ams_delta_ops.shutdown = ams_delta_shutdown;
+	}
+
+	/* Register optional line discipline for over the modem control */
+	ret = tty_register_ldisc(N_V253, &cx81801_ops);
+	if (ret) {
+		dev_warn(card->dev,
+				"Failed to register line discipline, "
+				"will continue without any controls.\n");
+		return 0;
+	}
+
+	/* Set up initial pin constellation */
+	snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+	snd_soc_dapm_disable_pin(dapm, "Speaker");
+	snd_soc_dapm_disable_pin(dapm, "AGCIN");
+	snd_soc_dapm_disable_pin(dapm, "AGCOUT");
+
+	return 0;
+}
+
+/* DAI glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ams_delta_dai_link = {
+	.name = "CX20442",
+	.stream_name = "CX20442",
+	.cpu_dai_name = "omap-mcbsp.1",
+	.codec_dai_name = "cx20442-voice",
+	.init = ams_delta_cx20442_init,
+	.platform_name = "omap-mcbsp.1",
+	.codec_name = "cx20442-codec",
+	.ops = &ams_delta_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBM_CFM,
+};
+
+/* Audio card driver */
+static struct snd_soc_card ams_delta_audio_card = {
+	.name = "AMS_DELTA",
+	.owner = THIS_MODULE,
+	.dai_link = &ams_delta_dai_link,
+	.num_links = 1,
+
+	.controls = ams_delta_audio_controls,
+	.num_controls = ARRAY_SIZE(ams_delta_audio_controls),
+	.dapm_widgets = ams_delta_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ams_delta_dapm_widgets),
+	.dapm_routes = ams_delta_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(ams_delta_audio_map),
+};
+
+/* Module init/exit */
+static int ams_delta_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &ams_delta_audio_card;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		card->dev = NULL;
+		return ret;
+	}
+	return 0;
+}
+
+static int ams_delta_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	if (tty_unregister_ldisc(N_V253) != 0)
+		dev_warn(&pdev->dev,
+			"failed to unregister V253 line discipline\n");
+
+	snd_soc_unregister_card(card);
+	card->dev = NULL;
+	return 0;
+}
+
+#define DRV_NAME "ams-delta-audio"
+
+static struct platform_driver ams_delta_driver = {
+	.driver = {
+		.name = DRV_NAME,
+	},
+	.probe = ams_delta_probe,
+	.remove = ams_delta_remove,
+};
+
+module_platform_driver(ams_delta_driver);
+
+MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
+MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);