/* * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone * * Copyright (C) 2009 Janusz Krzysztofik * * 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 #include #include #include #include #include #include #include #include #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 snd_soc_dapm_context *dapm = &codec->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 (!codec->hw_write) 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(dapm, "Mouthpiece")) { changed = 1; if (pin) snd_soc_dapm_enable_pin(dapm, "Mouthpiece"); else snd_soc_dapm_disable_pin(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(dapm, "Earpiece"); else snd_soc_dapm_disable_pin(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(dapm, "Microphone"); else snd_soc_dapm_disable_pin(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(dapm, "Speaker"); else snd_soc_dapm_disable_pin(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(dapm, "AGCIN"); else snd_soc_dapm_disable_pin(dapm, "AGCIN"); } if (changed) snd_soc_dapm_sync(dapm); 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); struct snd_soc_dapm_context *dapm = &codec->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 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 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); } /* * Used for passing a codec structure pointer * from the board initialization code to the tty line discipline. */ static struct snd_soc_codec *cx20442_codec; /* 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->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_disable_pin(dapm, "Mouthpiece"); snd_soc_dapm_enable_pin(dapm, "Earpiece"); snd_soc_dapm_enable_pin(dapm, "Microphone"); snd_soc_dapm_disable_pin(dapm, "Speaker"); snd_soc_dapm_disable_pin(dapm, "AGCIN"); snd_soc_dapm_sync(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 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->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, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: if (card->dapm.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 (card->dapm.bias_level != SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, 0); } card->dapm.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 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_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_card *card = rtd->card; int ret; /* Codec is ready, now add/activate board specific controls */ /* Store a pointer to the codec structure for tty ldisc use */ cx20442_codec = 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; } /* Set codec bias level */ ams_delta_set_bias_level(card, dapm, 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(rtd->codec, "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(dapm, 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(dapm, 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(dapm, "Mouthpiece"); snd_soc_dapm_enable_pin(dapm, "Earpiece"); snd_soc_dapm_enable_pin(dapm, "Microphone"); snd_soc_dapm_disable_pin(dapm, "Speaker"); snd_soc_dapm_disable_pin(dapm, "AGCIN"); snd_soc_dapm_disable_pin(dapm, "AGCOUT"); /* 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_name ="omap-mcbsp-dai.0", .codec_dai_name = "cx20442-voice", .init = ams_delta_cx20442_init, .platform_name = "omap-pcm-audio", .codec_name = "cx20442-codec", .ops = &ams_delta_ops, }; /* 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, .set_bias_level = ams_delta_set_bias_level, }; /* 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_audio_card); 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-codec", -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) { 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, &ams_delta_audio_card.rtd[0].codec->dapm, 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 "); MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone"); MODULE_LICENSE("GPL");