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Release 4.7 sound/soc/omap/ams-delta.c

Directory: sound/soc/omap
/*
 * 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; }

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lars-peter clausenlars-peter clausen41.04%116.67%
takashi iwaitakashi iwai10.26%116.67%
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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; }

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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); }

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/* 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; }

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/* 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); }

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/* Line discipline .hangup() */
static int cx81801_hangup(struct tty_struct *tty) { cx81801_close(tty); return 0; }

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/* 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; } }

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/* Line discipline .write_wakeup() */
static void cx81801_wakeup(struct tty_struct *tty) { v253_ops.write_wakeup(tty); }

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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; }

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/* 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); }

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static void ams_delta_shutdown(struct snd_pcm_substream *substream) { ams_delta_digital_mute(NULL, 1); }

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/* * 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; }

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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; }

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/* 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; }

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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; }

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#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);

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stephen warrenstephen warren301.42%13.85%
charles keepaxcharles keepax301.42%13.85%
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Directory: sound/soc/omap
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