[RFC,3/3] ASoC: add support for Amstrad E3 (Delta) machine

Message ID	200907220523.01107.jkrzyszt@tis.icnet.pl (mailing list archive)
State	Awaiting Upstream, archived
Headers	show Received: from vger.kernel.org (vger.kernel.org [209.132.176.167]) by demeter.kernel.org (8.14.2/8.14.2) with ESMTP id n6M3PUm2006568 for <patchwork-omap@patchwork.kernel.org>; Wed, 22 Jul 2009 03:25:32 GMT Received: (majordomo@vger.kernel.org) by vger.kernel.org via listexpand id S1756467AbZGVDYa (ORCPT <rfc822;patchwork-omap@patchwork.kernel.org>); Tue, 21 Jul 2009 23:24:30 -0400 Received: (majordomo@vger.kernel.org) by vger.kernel.org id S1756409AbZGVDYa (ORCPT <rfc822;linux-omap-outgoing>); Tue, 21 Jul 2009 23:24:30 -0400 Received: from d1.icnet.pl ([212.160.220.21]:39242 "EHLO d1.icnet.pl" rhost-flags-OK-OK-OK-OK) by vger.kernel.org with ESMTP id S1755927AbZGVDY1 (ORCPT <rfc822;linux-omap@vger.kernel.org>); Tue, 21 Jul 2009 23:24:27 -0400 Received: from 87-205-12-81.ip.netia.com.pl ([87.205.12.81] helo=acer.intranet) by d1.icnet.pl with asmtp (TLS-1.0:DHE_RSA_AES_128_CBC_SHA:16) (Exim 4.34) id 1MTSRH-0000g2-76; Wed, 22 Jul 2009 05:24:19 +0200 From: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Organization: Tele-Info-System, Poznan, PL To: Mark Brown <broonie@opensource.wolfsonmicro.com> Subject: [RFC] [PATCH 3/3] ASoC: add support for Amstrad E3 (Delta) machine Date: Wed, 22 Jul 2009 05:22:59 +0200 User-Agent: KMail/1.9.10 Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>, Jarkko Nikula <jhnikula@gmail.com>, Jonathan McDowell <noodles@earth.li>, Peter Ujfalusi <peter.ujfalusi@nokia.com>, Tony Lindgren <tony@atomide.com>, alsa-devel@alsa-project.org, "linux-serial@vger.kernel.org" <linux-serial@vger.kernel.org>, "linux-omap@vger.kernel.org" <linux-omap@vger.kernel.org>, "linux-kernel@vger.kernel.org" <linux-kernel@vger.kernel.org>, e3-hacking@earth.li, Arun KS <arunks@mistralsolutions.com> MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Content-Disposition: inline Message-Id: <200907220523.01107.jkrzyszt@tis.icnet.pl> X-SA-Exim-Scanned: No (on d1.icnet); SAEximRunCond expanded to false Sender: linux-omap-owner@vger.kernel.org Precedence: bulk List-ID: <linux-omap.vger.kernel.org> X-Mailing-List: linux-omap@vger.kernel.org

--- linux-2.6.31-rc1/sound/soc/omap/Kconfig.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/Kconfig 2009-07-22 01:06:51.000000000 +0200 @@ -15,6 +15,14 @@ config SND_OMAP_SOC_N810 help Say Y if you want to add support for SoC audio on Nokia N810. +config SND_OMAP_SOC_AMS_DELTA + tristate "SoC Audio support for Amstrad E3 (Delta) videophone" + depends on SND_OMAP_SOC && MACH_AMS_DELTA + select SND_OMAP_SOC_MCBSP + select SND_SOC_CX20442 + help + Say Y if you want to add support for SoC audio on Amstrad Delta. + config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C --- linux-2.6.31-rc1/sound/soc/omap/Makefile.orig 2009-07-04 16:40:00.000000000 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/Makefile 2009-07-22 01:06:51.000000000 +0200 @@ -7,6 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd- # OMAP Machine Support snd-soc-n810-objs := n810.o +snd-soc-ams-delta-objs := ams-delta.o snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o @@ -16,6 +17,7 @@ snd-soc-omap3pandora-objs := omap3pandor snd-soc-omap3beagle-objs := omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o +obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o --- /dev/null 2009-06-24 10:33:03.437003511 +0200 +++ linux-2.6.31-rc1/sound/soc/omap/ams-delta.c 2009-07-22 01:06:51.000000000 +0200 @@ -0,0 +1,618 @@ +/* + * ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone + * + * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> + * + * 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/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" + + +/* + * Controls available after the modem line discipline has been 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_EARPHONE 1 +#define AMS_DELTA_MICROPHONE 2 +#define AMS_DELTA_SPEAKER 3 +#define AMS_DELTA_AGC 4 + +#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPHONE) | \ + (1 << AMS_DELTA_MICROPHONE)) +#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \ + (1 << AMS_DELTA_EARPHONE)) +#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 bool ams_delta_audio_agc = 0; + +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 function; + bool changed = 0; + + if (ucontrol->value.enumerated.item[0] >= control->max) + return -EINVAL; + + mutex_lock(&codec->mutex); + + /* Translate selection to bitmap */ + function = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]]; + + /* Setup pins after corresponding bits if changed */ + if ((bool)snd_soc_dapm_get_pin_status(codec, "Speaker") != + (bool)(function & (1 << AMS_DELTA_SPEAKER))) { + changed = 1; + if (function & (1 << AMS_DELTA_SPEAKER)) + snd_soc_dapm_enable_pin(codec, "Speaker"); + else + snd_soc_dapm_disable_pin(codec, "Speaker"); + } + if ((bool)snd_soc_dapm_get_pin_status(codec, "Microphone") != + (bool)(function & (1 << AMS_DELTA_MICROPHONE))) { + changed = 1; + if (function & (1 << AMS_DELTA_MICROPHONE)) + snd_soc_dapm_enable_pin(codec, "Microphone"); + else + snd_soc_dapm_disable_pin(codec, "Microphone"); + } + if ((bool)snd_soc_dapm_get_pin_status(codec, "Earphone") != + (bool)(function & (1 << AMS_DELTA_EARPHONE))) { + changed = 1; + if (function & (1 << AMS_DELTA_EARPHONE)) + snd_soc_dapm_enable_pin(codec, "Earphone"); + else + snd_soc_dapm_disable_pin(codec, "Earphone"); + } + if ((bool)snd_soc_dapm_get_pin_status(codec, "Mouthpiece") != + (bool)(function & (1 << AMS_DELTA_MOUTHPIECE))) { + changed = 1; + if (function & (1 << AMS_DELTA_MOUTHPIECE)) + snd_soc_dapm_enable_pin(codec, "Mouthpiece"); + else + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + } + if (ams_delta_audio_agc != (bool)(function & (1 << AMS_DELTA_AGC))) { + ams_delta_audio_agc = (bool)(function & (1 << AMS_DELTA_AGC)); + changed = 1; + if (ams_delta_audio_agc) + 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 value, mux; + + value = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") << + AMS_DELTA_MOUTHPIECE) | + (snd_soc_dapm_get_pin_status(codec, "Earphone") << + AMS_DELTA_EARPHONE)); + if (value) + value |= (snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE); + else + value = ((snd_soc_dapm_get_pin_status(codec, "Microphone") << + AMS_DELTA_MICROPHONE) | + (snd_soc_dapm_get_pin_status(codec, "Speaker") << + AMS_DELTA_SPEAKER) | + ((unsigned short)ams_delta_audio_agc << AMS_DELTA_AGC)); + + for (mux = 0; mux < ARRAY_SIZE(ams_delta_audio_mode); mux++) + if (value == ams_delta_audio_mode_pins[mux]) + break; + + if (mux >= ARRAY_SIZE(ams_delta_audio_mode)) + return -EINVAL; + + ucontrol->value.enumerated.item[0] = mux; + + 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 Function", ams_delta_audio_enum[0], + ams_delta_get_audio_mode, ams_delta_set_audio_mode), +}; + +/* Now that we are able to control the codec over the modem, + * our hook switch can be used for dynamic DAPM reconfiguration */ +static struct snd_soc_jack ams_delta_hook_switch; +static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[]; +static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = { + { + .pin = "Mouthpiece", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Earphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Speaker", + .mask = SND_JACK_HEADPHONE, + .invert = 1, + }, + { + .pin = "Microphone", + .mask = SND_JACK_MICROPHONE, + .invert = 1, + }, +}; + +/* 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 digital mute function that shares the same hardware. */ +static struct timer_list cx81801_timer; +static bool cx81801_cmd_pending = 0; +static bool ams_delta_muted; + +static void cx81801_timeout(unsigned long data) +{ + /* REVISIT - locking? */ + if(!ams_delta_muted) + /* Reconnect the codec DAI back from the modem to the CPU DAI + * only if digital mute still off */ + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0); + cx81801_cmd_pending = 0; +} + +/* Modem init: echo off, digital speaker off, quiet off, voice mode */ +static const char *cx81801_init = "ate0m0q0+fclass=8\r"; + + +/* + * Modem line discipline required for making above controls functional. + * Activated from userspace with ldattach, possibly invoked from udev rule. + */ + +static struct snd_soc_card ams_delta_audio_card; + +/* Line discipline setup */ +static int cx81801_open(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = ams_delta_audio_card.codec; + int ret, len = strlen(cx81801_init); + + /* Doesn't make sense without write callback */ + if (!tty->ops->write) + return -EINVAL; + + /* Pass codec structure address to other ldisc callbacks */ + tty->disc_data = codec; + + if (tty->ops->write(tty, cx81801_init, len) != len) { + ret = -EIO; + goto err; + } + /* Actual setup will be performed after the modem responds. */ + return 0; +err: + tty->disc_data = NULL; + return ret; +} + +/* Can this empty callback be omitted? */ +static void cx81801_wakeup(struct tty_struct *tty) +{ +} + +/* Modem response callback */ +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; + + if (!codec->control_data) { + /* First modem response, complete setup procedure */ + + /* Initialize timer used for config pulse generation */ + setup_timer(&cx81801_timer, cx81801_timeout, 0); + + /* Set up codec access to modem controls */ + codec->control_data = tty; + codec->hw_write = (hw_write_t)tty->ops->write; + codec->pop_time = 1; + + /* Link hook switch to DAPM pins */ + snd_soc_jack_add_pins(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_pins), + ams_delta_hook_switch_pins); + + /* Add virtual switches */ + snd_soc_add_controls(codec, ams_delta_audio_controls, + ARRAY_SIZE(ams_delta_audio_controls)); + + /* Set DAPM pins after hook switch present state */ +#if 0 + /* Fails for switch state matching initial gpio->state = 0 */ + snd_soc_jack_report(&ams_delta_ams_delta_hook_switch, + gpio_get_value(ams_delta_hook_switch_gpios[0].gpio) ? + 0 : SND_JACK_HEADSET, SND_JACK_HEADSET); +#else + if (gpio_get_value(ams_delta_hook_switch_gpios[0].gpio)) { + snd_soc_dapm_enable_pin(codec, "Speaker"); + snd_soc_dapm_enable_pin(codec, "Microphone"); + snd_soc_dapm_disable_pin(codec, "Earphone"); + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + } else { + snd_soc_dapm_disable_pin(codec, "Speaker"); + snd_soc_dapm_disable_pin(codec, "Microphone"); + snd_soc_dapm_enable_pin(codec, "Earphone"); + snd_soc_dapm_enable_pin(codec, "Mouthpiece"); + } + snd_soc_dapm_sync(codec); +#endif + return; + } + for (c = &cp[count - 1]; c >= cp; c--) { + if (*c != '\r') + continue; + /* Complete modem response received, apply config to codec */ + + /* REVISIT - locking? */ + /* Reconnect the codec to the modem if not already done */ + if (!ams_delta_muted && !cx81801_cmd_pending) + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + AMS_DELTA_LATCH2_MODEM_CODEC); + + /* Keep config pulse long enough for proper config activation */ + cx81801_cmd_pending = 1; + mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(200)); + break; + } +} + +static void cx81801_close(struct tty_struct *tty) +{ + struct snd_soc_codec *codec = tty->disc_data; + + del_timer_sync(&cx81801_timer); + + /* Revert back to default audio input/output constellation */ + snd_soc_dapm_disable_pin(codec, "AGCIN"); + snd_soc_dapm_disable_pin(codec, "AGCOUT"); + snd_soc_jack_report(&ams_delta_hook_switch, + SND_JACK_HEADPHONE, SND_JACK_HEADSET); + + /* REVISIT - Don't know how to do that */ + /* + * Remove controls that we expose when over the modem control available: + * - virtual audio mode switch, + * - hook switch to DAPM pins links + */ + + /* Prevent the codec driver from further accessing the modem */ + codec->hw_write = NULL; + codec->control_data = NULL; + codec->pop_time = 0; + + tty->disc_data = NULL; +} + +static struct tty_ldisc_ops cx81801_ops = { + .magic = TTY_LDISC_MAGIC, + .name = "cx81801", + .owner = THIS_MODULE, + .open = cx81801_open, + .close = cx81801_close, + .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 DAPM widgets + */ + +static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Microphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_HP("Earphone", NULL), + SND_SOC_DAPM_MIC("Mouthpiece", NULL), +}; + +static const struct snd_soc_dapm_route ams_delta_audio_map[] = { + {"TELIN", NULL, "Mouthpiece"}, + {"Earphone", NULL, "TELOUT"}, + + {"MIC", NULL, "Microphone"}, + {"Speaker", NULL, "SPKOUT"}, +}; + +/* 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 = 100, + } +}; + +/* 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) +{ + /* REVISIT - locking? */ + if (ams_delta_muted == mute) + return 0; + ams_delta_muted = mute; + if (cx81801_cmd_pending) + return 0; + + 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; + /* Codec is ready, now add/activate board specific controls */ + + /* Set up digital mute if not provided by 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(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY); + + /* Add board specific DAPM controls */ + if (!snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets, + ARRAY_SIZE(ams_delta_dapm_widgets))) { + if (!snd_soc_dapm_add_routes(codec, ams_delta_audio_map, + ARRAY_SIZE(ams_delta_audio_map))) { + /* Actual pin constellation if no modem control */ + snd_soc_dapm_disable_pin(codec, "Mouthpiece"); + snd_soc_dapm_enable_pin(codec, "Earphone"); + 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 hook switch */ + if (!snd_soc_jack_new(&ams_delta_audio_card, "hook_switch", + SND_JACK_HEADSET, &ams_delta_hook_switch)) { + if (!snd_soc_jack_add_gpios(&ams_delta_hook_switch, + ARRAY_SIZE(ams_delta_hook_switch_gpios), + ams_delta_hook_switch_gpios)) { +#ifdef CONFIG_GPIO_SYSFS + /* Expose hook switch over sysfs if configured */ + gpio_export(ams_delta_hook_switch_gpios[0].gpio, false); +#endif + } + } + /* Register optional line discipline for over the modem control */ + tty_register_ldisc(N_AMSDELTA, &cx81801_ops); + + 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) +{ + tty_unregister_ldisc(N_AMSDELTA); + +#ifdef CONFIG_GPIO_SYSFS + gpio_unexport(ams_delta_hook_switch_gpios[0].gpio); +#endif + 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");

[RFC,3/3] ASoC: add support for Amstrad E3 (Delta) machine

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