[RFC] [PATCH 3/3] ASoC: add support for Amstrad E3 (Delta) machine
This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC. Created and tested against linux-2.6.31-rc3. Applies and works with linux-omap-2.6 commit 7c5cb7862d32cb344be7831d466535d5255e35ac as well. Depends on patch 2 form this series: TTY: Add definition of a new line discipline required by Amstrad E3 (Delta) ASoC driver. Credits to Mark Underwood for his initial, omap-alsa based sound driver for this machine. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> --- CPU DAI parameters best matching the codec DAI has been selected out empirically for best user experience. Board specific audio function control (with related DAPM widgets) has been modeled after empirically discovered codec capabilities. Unlike other ASoC machine drivers, this one contains a line discipline. This is required for providig the codec driver with support for talking to the modem chip that can control the codec behavoiur. As the line discipline code contains board specific bits, I decided to make it a part of the ASoC machine driver, not the codec, and not a separate source file. Otherwise, some kind of a glue, like a bus over a tty, could help in linking the board specific part (bus device) to a more generic line discipline (bus adapter)[1]. In order to work at all, this driver requires a working McBSP1. On OMAP1510 based machines (not sure if other OMAP1 variants as well), where McBSP1 is a DSP public peripheral, that means the kernel must provide basic DSP support, ie. omap_dsp_init(), in order to power up the DSP. This used to be included in linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c. Without that, the driver would not work, ie. not shift in/out any bits over the CPU DAI[2]. This limitation is not board, but CPU specific, and may apply to other code that makes use of McBSP1/McBSP3 on affected machines. I provide an extra patch [4/3] as a temporary solution. To work correctly in playback mode, this driver requires my prevoiusly submitted patch that corrects pcm pointer calculation for OMAP1510 based machines[3] (already included in linux-2.6.31-rc3). To support codec controls, this driver requires my previously submitted patch that adds support for modem found on Amstrad Delta[4]. [1] http://www.spinics.net/lists/linux-serial/msg01856.html [2] http://www.spinics.net/lists/linux-omap/msg15114.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.html [4] http://www.spinics.net/lists/linux-omap/msg15432.html --- 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");
On Wed, Jul 22, 2009 at 05:22:59AM +0200, Janusz Krzysztofik wrote:
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> --- CPU DAI parameters best matching the codec DAI has been selected out empirically for best user experience.
Again, all the documentation you've got here could quite happily go in the commit message and there's a bunch of checkpatch issues.
+#include <sound/soc-dapm.h> +#include <sound/jack.h>
ASoC will pull that one in for you, not that it really matters.
+ /* Setup pins after corresponding bits if changed */ + if ((bool)snd_soc_dapm_get_pin_status(codec, "Speaker") != + (bool)(function & (1 << AMS_DELTA_SPEAKER))) {
Don't like these casts... why are they needed?
+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), +};
Is it possible to control all the functions of the audio mode independantly or are only certain combinations possible?
+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, + }, +};
I guess microphone and speaker are for speakerphone mode while mouthpiece and earpiece are a headset? Might be nice to come up with names that make the paring a bit clearer, or possibly just put a comment in there.
+/* 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? */
Yeah, locking is probably a good idea :)
+ /* 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);
Hrm. We should just fix that so that adding a pin forces a sync rather than just doing a report. The GPIO jack wrapper ought to just do everything you need without any code in the machine driver.
+ + /* 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 + */
You can't do that. Rather than adding and removing the controls dynamically I'd suggest hiding the controls from applications while they're inactive - this will also avoid any renumbering which might confuse them. However, that API isn't in mainline yet. Takashi, there is a patch adding a snd_ctl_activate_id() call in your unstable tree which hasn't made it into mainline and should help here - would there be any problem merging it? I've got another use case for it in ASoC that I'm hoping to find time to look at soon.
+ /* 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))) {
The error handling here is a bit odd...
+ /* 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
The gpio_export() should be in the ASoC GPIO code rather than here, I'd expect - care to cook up a patch?
At Wed, 22 Jul 2009 12:03:28 +0100, Mark Brown wrote:
+ /* Setup pins after corresponding bits if changed */ + if ((bool)snd_soc_dapm_get_pin_status(codec, "Speaker") != + (bool)(function & (1 << AMS_DELTA_SPEAKER))) {
Don't like these casts... why are they needed?
Because the right side is the bit operation? The cast doesn't look nice, though... Takashi
On Wed, Jul 22, 2009 at 01:39:22PM +0200, Takashi Iwai wrote:
Mark Brown wrote:
+ /* Setup pins after corresponding bits if changed */ + if ((bool)snd_soc_dapm_get_pin_status(codec, "Speaker") != + (bool)(function & (1 << AMS_DELTA_SPEAKER))) {
Don't like these casts... why are they needed?
Because the right side is the bit operation?
Ick, yes.
The cast doesn't look nice, though...
Indeed. I'd suggest rewriting to try to do less in the if statement - a helper function seems to be in order here since the same code is repeated several times with different pins and bitmasks.
Mark Brown wrote:
On Wed, Jul 22, 2009 at 01:39:22PM +0200, Takashi Iwai wrote:
Mark Brown wrote:
+ /* Setup pins after corresponding bits if changed */ + if ((bool)snd_soc_dapm_get_pin_status(codec, "Speaker") != + (bool)(function & (1 << AMS_DELTA_SPEAKER))) {
Don't like these casts... why are they needed?
Because the right side is the bit operation?
Ick, yes.
The cast doesn't look nice, though...
Indeed. I'd suggest rewriting to try to do less in the if statement - a helper function seems to be in order here since the same code is repeated several times with different pins and bitmasks.
Thanks, I'll rewrite as you suggest. Janusz
Hi Mark, Mark Brown wrote:
On Wed, Jul 22, 2009 at 05:22:59AM +0200, Janusz Krzysztofik wrote:
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> --- CPU DAI parameters best matching the codec DAI has been selected out empirically for best user experience.
Again, all the documentation you've got here could quite happily go in the commit message and there's a bunch of checkpatch issues.
OK, I promiss to remember about this for the future.
+#include <sound/soc-dapm.h> +#include <sound/jack.h>
ASoC will pull that one in for you, not that it really matters.
Maybe it should, but without <sound/jack.h> I get: sound/soc/omap/ams-delta.c:184: error: 'SND_JACK_MICROPHONE' undeclared here (not in a function) sound/soc/omap/ams-delta.c:188: error: 'SND_JACK_HEADPHONE' undeclared here (not in a function) sound/soc/omap/ams-delta.c: In function 'cx81801_close': sound/soc/omap/ams-delta.c:336: error: 'SND_JACK_HEADSET' undeclared (first use in this function) ...
+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), +};
Is it possible to control all the functions of the audio mode independantly or are only certain combinations possible?
Certain combinations only. For example, no way of turning on the mouthpiece without turning on the earphone as well, turning on AGC automatically selects the speakerphone and turns off the handset.
+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, + }, +};
I guess microphone and speaker are for speakerphone mode while mouthpiece and earpiece are a headset? Might be nice to come up with
Exactly.
names that make the paring a bit clearer, or possibly just put a comment in there.
With my limited English skills, I can only replace Earphone with Earpiece and add a comment. Please someone suggest better namings if not enough.
+/* 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? */
Yeah, locking is probably a good idea :)
I'll have to learn about locking first. Could somebody point me to an example code?
+ /* 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))) {
The error handling here is a bit odd...
Do you mean those negations? Would be better if replaced with "== 0"? I am not sure if this is acceptable, but I just tried to avoid giving up with a partialy working driver in case of errors on optional parts.
+ /* 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
The gpio_export() should be in the ASoC GPIO code rather than here, I'd expect - care to cook up a patch?
When I tried to push a similiar code into the gpio-keys dirver, Dmitry said I was wrong because my application would be limited to gpio based devices only. However, it looks like there are no jacks other than gpio based in ASoC, so maybe that objection does not apply here. Or maybe the ASoC framework could just provide its own sysfs file with actual jack state, whether gpio based or not? BTW, I decided to reuse already existing jack/input event types instead of inventing new. Anyway, should I CC: linux-input? Thanks, Janusz
On Wed, Jul 22, 2009 at 04:53:12PM +0200, Janusz Krzysztofik wrote:
Mark Brown wrote:
+#include <sound/jack.h>
ASoC will pull that one in for you, not that it really matters.
Maybe it should, but without <sound/jack.h> I get: sound/soc/omap/ams-delta.c:184: error: 'SND_JACK_MICROPHONE' undeclared here (not in a function)
Hrmpf, that's no use. I'll add it - it'd be good if you could report stuff like this when you find it, you've got several workarounds for core code in here.
Is it possible to control all the functions of the audio mode independantly or are only certain combinations possible?
Certain combinations only. For example, no way of turning on the mouthpiece without turning on the earphone as well, turning on AGC automatically selects the speakerphone and turns off the handset.
OK, then this mode selection is fine.
names that make the paring a bit clearer, or possibly just put a comment in there.
With my limited English skills, I can only replace Earphone with Earpiece and add a comment. Please someone suggest better namings if not enough.
Prefixes of "Headset" on the ones that apply to that? Or the comment, this is only really visible internally.
+static void cx81801_timeout(unsigned long data) +{ + /* REVISIT - locking? */
Yeah, locking is probably a good idea :)
I'll have to learn about locking first. Could somebody point me to an example code?
In what sense? For an overview of the various APIs Linux Device Drivers is probably still good: http://lwn.net/Kernel/LDD3
+ /* 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))) {
The error handling here is a bit odd...
Do you mean those negations? Would be better if replaced with "== 0"? I am not sure if this is acceptable, but I just tried to avoid giving up with a partialy working driver in case of errors on optional parts.
The negations, the nesting and the lack of any reporting of failures. Probably just calling all the functions one after another and logging any errors would be OK.
+#ifdef CONFIG_GPIO_SYSFS + /* Expose hook switch over sysfs if configured */ + gpio_export(ams_delta_hook_switch_gpios[0].gpio, false); +#endif
The gpio_export() should be in the ASoC GPIO code rather than here, I'd expect - care to cook up a patch?
When I tried to push a similiar code into the gpio-keys dirver, Dmitry said I was wrong because my application would be limited to gpio based devices only. However, it looks like there are no jacks
Userspace applications are a separate issue - it looked like this was just for diagnostic purposes? Obviously any user space applications using the GPIO interface will be limited to your device but that applies no matter where you put this code.
other than gpio based in ASoC, so maybe that objection does not apply here. Or maybe the ASoC framework could just provide its own sysfs file with actual jack state, whether gpio based or not?
There's at least a driver for the WM8350 headphone detection (possibly others, I'd need to grep) and an awful lot of hardware which could use this in-codec. TLV320AIC3x has some code that ought to be moved over to the standard framework too.
BTW, I decided to reuse already existing jack/input event types instead of inventing new. Anyway, should I CC: linux-input?
No need if you're not changing it.
Hi, Wednesday 22 July 2009 17:07:09 Mark Brown wrote:
On Wed, Jul 22, 2009 at 04:53:12PM +0200, Janusz Krzysztofik wrote:
Mark Brown wrote:
+#include <sound/jack.h>
ASoC will pull that one in for you, not that it really matters.
Maybe it should, but without <sound/jack.h> I get: sound/soc/omap/ams-delta.c:184: error: 'SND_JACK_MICROPHONE' undeclared here (not in a function)
Hrmpf, that's no use. I'll add it - it'd be good if you could report stuff like this when you find it, you've got several workarounds for core code in here.
But first I would have to learn how to distinguish such upper layer misses from my own bugs ;). I'll try to do my best.
Is it possible to control all the functions of the audio mode independantly or are only certain combinations possible?
Certain combinations only. For example, no way of turning on the mouthpiece without turning on the earphone as well, turning on AGC automatically selects the speakerphone and turns off the handset.
OK, then this mode selection is fine.
Actually I do not support two more possible I/O constellations: speaker only and microphone only. But I don't think anybody would find them really usefull. Even that Mixed configuration (Microphone+Earpiece) could be probably removed, I keep it only because it matches initail state, before the line discipline comes into action.
+static void cx81801_timeout(unsigned long data) +{ + /* REVISIT - locking? */
Yeah, locking is probably a good idea :)
I'll have to learn about locking first. Could somebody point me to an example code?
In what sense? For an overview of the various APIs Linux Device Drivers is probably still good: http://lwn.net/Kernel/LDD3
Thanks, I've already started with reading. There are several options AFAICS, I'll try to select the most suitable one. As the code that requires locking belongs to the line discipline, I'll wait a little more for possible commemts from Alan or other serial guys, if any, before I try to implement any locking.
+ /* 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))) {
The error handling here is a bit odd...
Do you mean those negations? Would be better if replaced with "== 0"? I am not sure if this is acceptable, but I just tried to avoid giving up with a partialy working driver in case of errors on optional parts.
The negations, the nesting and the lack of any reporting of failures. Probably just calling all the functions one after another and logging any errors would be OK.
Thanks, will be rewriten this way.
+#ifdef CONFIG_GPIO_SYSFS + /* Expose hook switch over sysfs if configured */ + gpio_export(ams_delta_hook_switch_gpios[0].gpio, false); +#endif
The gpio_export() should be in the ASoC GPIO code rather than here, I'd expect - care to cook up a patch?
When I tried to push a similiar code into the gpio-keys dirver, Dmitry said I was wrong because my application would be limited to gpio based devices only. However, it looks like there are no jacks
Userspace applications are a separate issue - it looked like this was just for diagnostic purposes? Obviously any user space applications using the GPIO interface will be limited to your device but that applies no matter where you put this code.
other than gpio based in ASoC, so maybe that objection does not apply here. Or maybe the ASoC framework could just provide its own sysfs file with actual jack state, whether gpio based or not?
There's at least a driver for the WM8350 headphone detection (possibly others, I'd need to grep) and an awful lot of hardware which could use this in-codec. TLV320AIC3x has some code that ought to be moved over to the standard framework too.
Mark, After all, I am not sure what your preferences finally are. If you still think that exporting gpio based jack state from ASoC framework over gpiolib sysfs doesn't hurt and can be usefull, I'll remove the code from my driver and provide a patch against sound/soc/soc-jack.c. But if you would rather prefere exposing jack state in a more general, not gpio limited way instead, or depend on input layer provided EVIOCGSW ioctl, as Dmitry suggested (unfortunatelly, unlike generic ldattach, there is no utility ready for use!), I'd rather keep this gpio_export() in my driver, let's say for diagnostic purposes, at least until that different option is ready for use in a convenient way. Thanks, Janusz
On Wed, Jul 22, 2009 at 09:18:05PM +0200, Janusz Krzysztofik wrote:
After all, I am not sure what your preferences finally are. If you still think that exporting gpio based jack state from ASoC framework over gpiolib sysfs doesn't hurt and can be usefull, I'll remove the code from my driver and provide a patch against sound/soc/soc-jack.c. But if you would rather prefere exposing jack state in a more general, not gpio limited way instead, or depend on input layer provided EVIOCGSW ioctl, as Dmitry suggested (unfortunatelly, unlike generic ldattach, there is no utility ready for use!), I'd rather keep this gpio_export() in my driver, let's say for diagnostic purposes, at least until that different option is ready for use in a convenient way.
Jack state is already exposed via the input layer by the ALSA jack core. Reporting of GPIO state via gpio_export() would be additional to that and would be purely for debug purposes, it would not be recommended for use by userspace applications.
participants (3)
-
Janusz Krzysztofik -
Mark Brown -
Takashi Iwai