| /* |
| * 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 <mach/board-ams-delta.h> |
| #include <mach/mcbsp.h> |
| |
| #include "omap-mcbsp.h" |
| #include "omap-pcm.h" |
| #include "../codecs/cx20442.h" |
| |
| |
| /* Board specific DAPM widgets */ |
| 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)) |
| |
| 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_AMSDELTA, &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_AMSDELTA) != 0) |
| dev_warn(&ams_delta_audio_platform_device->dev, |
| "failed to unregister AMSDELTA 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"); |