OMAP: DSS2: DSI: Increase HS TX timeout

[linux-2.6/linux-acpi-2.6/ibm-acpi-2.6.git] / sound / soc / omap / ams-delta.c

/*
 * 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 <plat/board-ams-delta.h>
#include <plat/mcbsp.h>

#include "omap-mcbsp.h"
#include "omap-pcm.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;

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");