1 files changed, 646 insertions, 0 deletions

diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
new file mode 100644
index 000000000000..5a5166ac7279
--- /dev/null
+++ b/sound/soc/omap/ams-delta.c
@@ -0,0 +1,646 @@
+/*
+ * 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.h>
+#include <linux/spinlock.h>
+#include <linux/tty.h>
+
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include <mach/board-ams-delta.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/cx20442.h"
+
+
+/* Board specific DAPM widgets */
+ 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))
+
+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;
+
+static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	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 (!codec->control_data)
+		return -EUNATCH;
+
+	if (ucontrol->value.enumerated.item[0] >= control->max)
+		return -EINVAL;
+
+	mutex_lock(&codec->mutex);
+
+	/* 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(codec, "Mouthpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(codec, "Mouthpiece");
+		else
+			snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
+	if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(codec, "Earpiece");
+		else
+			snd_soc_dapm_disable_pin(codec, "Earpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
+	if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(codec, "Microphone");
+		else
+			snd_soc_dapm_disable_pin(codec, "Microphone");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
+	if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(codec, "Speaker");
+		else
+			snd_soc_dapm_disable_pin(codec, "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(codec, "AGCIN");
+		else
+			snd_soc_dapm_disable_pin(codec, "AGCIN");
+	}
+	if (changed)
+		snd_soc_dapm_sync(codec);
+
+	mutex_unlock(&codec->mutex);
+
+	return changed;
+}
+
+static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	unsigned short pins, mode;
+
+	pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") <<
+							AMS_DELTA_MOUTHPIECE) |
+			(snd_soc_dapm_get_pin_status(codec, "Earpiece") <<
+							AMS_DELTA_EARPIECE));
+	if (pins)
+		pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+							AMS_DELTA_MICROPHONE);
+	else
+		pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+							AMS_DELTA_MICROPHONE) |
+			(snd_soc_dapm_get_pin_status(codec, "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 struct soc_enum ams_delta_audio_enum[] = {
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
+						ams_delta_audio_mode),
+};
+
+static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
+	SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
+			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[] = {
+	{
+		.gpio = 4,
+		.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 carefull 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 void cx81801_timeout(unsigned long data)
+{
+	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)
+		ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
+}
+
+/* Line discipline .open() */
+static int cx81801_open(struct tty_struct *tty)
+{
+	return v253_ops.open(tty);
+}
+
+/* Line discipline .close() */
+static void cx81801_close(struct tty_struct *tty)
+{
+	struct snd_soc_codec *codec = tty->disc_data;
+
+	del_timer_sync(&cx81801_timer);
+
+	v253_ops.close(tty);
+
+	/* Prevent the hook switch from further changing the DAPM pins */
+	INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
+
+	/* Revert back to default audio input/output constellation */
+	snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+	snd_soc_dapm_enable_pin(codec, "Earpiece");
+	snd_soc_dapm_enable_pin(codec, "Microphone");
+	snd_soc_dapm_disable_pin(codec, "Speaker");
+	snd_soc_dapm_disable_pin(codec, "AGCIN");
+	snd_soc_dapm_sync(codec);
+}
+
+/* Line discipline .hangup() */
+static int cx81801_hangup(struct tty_struct *tty)
+{
+	cx81801_close(tty);
+	return 0;
+}
+
+/* Line discipline .recieve_buf() */
+static void cx81801_receive(struct tty_struct *tty,
+				const unsigned char *cp, char *fp, int count)
+{
+	struct snd_soc_codec *codec = tty->disc_data;
+	const unsigned char *c;
+	int apply, ret;
+
+	if (!codec->control_data) {
+		/* First modem response, complete setup procedure */
+
+		/* Initialize timer used for config pulse generation */
+		setup_timer(&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(codec->socdev->card->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)
+			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+						AMS_DELTA_LATCH2_MODEM_CODEC);
+		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 usefull, 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 issueing AT commands
+ * over the modem port.
+ */
+
+static int ams_delta_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* Set cpu DAI configuration */
+	return snd_soc_dai_set_fmt(rtd->dai->cpu_dai,
+				   SND_SOC_DAIFMT_DSP_A |
+				   SND_SOC_DAIFMT_NB_NF |
+				   SND_SOC_DAIFMT_CBM_CFM);
+}
+
+static struct snd_soc_ops ams_delta_ops = {
+	.hw_params = ams_delta_hw_params,
+};
+
+
+/* Board specific codec bias level control */
+static int ams_delta_set_bias_level(struct snd_soc_card *card,
+					enum snd_soc_bias_level level)
+{
+	struct snd_soc_codec *codec = card->codec;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF)
+			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+						AMS_DELTA_LATCH2_MODEM_NRESET);
+		break;
+	case SND_SOC_BIAS_OFF:
+		if (codec->bias_level != SND_SOC_BIAS_OFF)
+			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+						0);
+	}
+	codec->bias_level = level;
+
+	return 0;
+}
+
+/* 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)
+		ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+				mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
+	return 0;
+}
+
+/* Our codec DAI probably doesn't have its own .ops structure */
+static 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_codec *codec)
+{
+	struct snd_soc_dai *codec_dai = codec->dai;
+	struct snd_soc_card *card = codec->socdev->card;
+	int ret;
+	/* Codec is ready, now add/activate board specific controls */
+
+	/* Set up digital mute if not provided by the codec */
+	if (!codec_dai->ops) {
+		codec_dai->ops = &ams_delta_dai_ops;
+	} else if (!codec_dai->ops->digital_mute) {
+		codec_dai->ops->digital_mute = ams_delta_digital_mute;
+	} else {
+		ams_delta_ops.startup = ams_delta_startup;
+		ams_delta_ops.shutdown = ams_delta_shutdown;
+	}
+
+	/* Set codec bias level */
+	ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+
+	/* Add hook switch - can be used to control the codec from userspace
+	 * even if line discipline fails */
+	ret = snd_soc_jack_new(card, "hook_switch",
+				SND_JACK_HEADSET, &ams_delta_hook_switch);
+	if (ret)
+		dev_warn(card->dev,
+				"Failed to allocate resources for hook switch, "
+				"will continue without one.\n");
+	else {
+		ret = snd_soc_jack_add_gpios(&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");
+	}
+
+	/* 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;
+	}
+
+	/* Add board specific DAPM widgets and routes */
+	ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets,
+					ARRAY_SIZE(ams_delta_dapm_widgets));
+	if (ret) {
+		dev_warn(card->dev,
+				"Failed to register DAPM controls, "
+				"will continue without any.\n");
+		return 0;
+	}
+
+	ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map,
+					ARRAY_SIZE(ams_delta_audio_map));
+	if (ret) {
+		dev_warn(card->dev,
+				"Failed to set up DAPM routes, "
+				"will continue with codec default map.\n");
+		return 0;
+	}
+
+	/* Set up initial pin constellation */
+	snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+	snd_soc_dapm_enable_pin(codec, "Earpiece");
+	snd_soc_dapm_enable_pin(codec, "Microphone");
+	snd_soc_dapm_disable_pin(codec, "Speaker");
+	snd_soc_dapm_disable_pin(codec, "AGCIN");
+	snd_soc_dapm_disable_pin(codec, "AGCOUT");
+	snd_soc_dapm_sync(codec);
+
+	/* Add virtual switch */
+	ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
+					ARRAY_SIZE(ams_delta_audio_controls));
+	if (ret)
+		dev_warn(card->dev,
+				"Failed to register audio mode control, "
+				"will continue without it.\n");
+
+	return 0;
+}
+
+/* DAI glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ams_delta_dai_link = {
+	.name = "CX20442",
+	.stream_name = "CX20442",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &cx20442_dai,
+	.init = ams_delta_cx20442_init,
+	.ops = &ams_delta_ops,
+};
+
+/* Audio card driver */
+static struct snd_soc_card ams_delta_audio_card = {
+	.name = "AMS_DELTA",
+	.platform = &omap_soc_platform,
+	.dai_link = &ams_delta_dai_link,
+	.num_links = 1,
+	.set_bias_level = ams_delta_set_bias_level,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device ams_delta_snd_soc_device = {
+	.card = &ams_delta_audio_card,
+	.codec_dev = &cx20442_codec_dev,
+};
+
+/* Module init/exit */
+static struct platform_device *ams_delta_audio_platform_device;
+static struct platform_device *cx20442_platform_device;
+
+static int __init ams_delta_module_init(void)
+{
+	int ret;
+
+	if (!(machine_is_ams_delta()))
+		return -ENODEV;
+
+	ams_delta_audio_platform_device =
+			platform_device_alloc("soc-audio", -1);
+	if (!ams_delta_audio_platform_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(ams_delta_audio_platform_device,
+				&ams_delta_snd_soc_device);
+	ams_delta_snd_soc_device.dev = &ams_delta_audio_platform_device->dev;
+	*(unsigned int *)ams_delta_dai_link.cpu_dai->private_data = OMAP_MCBSP1;
+
+	ret = platform_device_add(ams_delta_audio_platform_device);
+	if (ret)
+		goto err;
+
+	/*
+	 * Codec platform device could be registered from elsewhere (board?),
+	 * but I do it here as it makes sense only if used with the card.
+	 */
+	cx20442_platform_device = platform_device_register_simple("cx20442",
+								-1, NULL, 0);
+	return 0;
+err:
+	platform_device_put(ams_delta_audio_platform_device);
+	return ret;
+}
+module_init(ams_delta_module_init);
+
+static void __exit ams_delta_module_exit(void)
+{
+	struct snd_soc_codec *codec;
+	struct tty_struct *tty;
+
+	if (ams_delta_audio_card.codec) {
+		codec = ams_delta_audio_card.codec;
+
+		if (codec->control_data) {
+			tty = codec->control_data;
+
+			tty_hangup(tty);
+		}
+	}
+
+	if (tty_unregister_ldisc(N_V253) != 0)
+		dev_warn(&ams_delta_audio_platform_device->dev,
+			"failed to unregister V253 line discipline\n");
+
+	snd_soc_jack_free_gpios(&ams_delta_hook_switch,
+			ARRAY_SIZE(ams_delta_hook_switch_gpios),
+			ams_delta_hook_switch_gpios);
+
+	/* Keep modem power on */
+	ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+
+	platform_device_unregister(cx20442_platform_device);
+	platform_device_unregister(ams_delta_audio_platform_device);
+}
+module_exit(ams_delta_module_exit);
+
+MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
+MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
+MODULE_LICENSE("GPL");