| Author | Tokens | Token Proportion | Commits | Commit Proportion |
|---|---|---|---|---|
| Janusz Krzysztofik | 1830 | 86.48% | 6 | 22.22% |
| Liam Girdwood | 129 | 6.10% | 2 | 7.41% |
| Lars-Peter Clausen | 84 | 3.97% | 6 | 22.22% |
| Charles Keepax | 30 | 1.42% | 1 | 3.70% |
| Kuninori Morimoto | 20 | 0.95% | 2 | 7.41% |
| Kees Cook | 5 | 0.24% | 1 | 3.70% |
| Axel Lin | 5 | 0.24% | 1 | 3.70% |
| Takashi Iwai | 4 | 0.19% | 2 | 7.41% |
| Paul Gortmaker | 3 | 0.14% | 1 | 3.70% |
| Lucas De Marchi | 2 | 0.09% | 1 | 3.70% |
| Peter Ujfalusi | 2 | 0.09% | 2 | 7.41% |
| Joe Perches | 1 | 0.05% | 1 | 3.70% |
| Arnd Bergmann | 1 | 0.05% | 1 | 3.70% |
| Total | 2116 | 27 |
/* * 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);
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