- 根目录:
- sound
- soc
- omap
- ams-delta.c
/*
* 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 <linux/module.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <asm/mach-types.h>
#include <plat/board-ams-delta.h>
#include <plat/mcbsp.h>
#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 <jkrzyszt@tis.icnet.pl>");
MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
MODULE_LICENSE("GPL");