--- /dev/null
+/*
+ * 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 <linux/module.h>
+
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include <mach/board-ams-delta.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_codec *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->hw_write)
+ 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[] = {
+ {
+ .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 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 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)
+{
+ 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_codec *codec = tty->disc_data;
+ struct snd_soc_dapm_context *dapm = &codec->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 (!codec)
+ 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_codec *codec = tty->disc_data;
+ const unsigned char *c;
+ int apply, ret;
+
+ if (!codec)
+ return;
+
+ if (!codec->hw_write) {
+ /* 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->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 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)
+ 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 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;
+
+ /* 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;
+ }
+
+ /* 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_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;
+ }
+
+ /* 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;
+}
+
+static int ams_delta_card_remove(struct snd_soc_card *card)
+{
+ snd_soc_jack_free_gpios(&ams_delta_hook_switch,
+ ARRAY_SIZE(ams_delta_hook_switch_gpios),
+ ams_delta_hook_switch_gpios);
+
+ 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,
+ .remove = ams_delta_card_remove,
+ .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);
Code Review / kvmfornfv.git / blobdiff