25985edced
Fixes generated by 'codespell' and manually reviewed. Signed-off-by: Lucas De Marchi <lucas.demarchi@profusion.mobi>
660 lines
18 KiB
C
660 lines
18 KiB
C
/*
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* ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone
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*
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* Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
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*
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* Initially based on sound/soc/omap/osk5912.x
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* Copyright (C) 2008 Mistral Solutions
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* version 2 as published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
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* 02110-1301 USA
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*
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*/
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#include <linux/gpio.h>
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#include <linux/spinlock.h>
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#include <linux/tty.h>
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#include <sound/soc.h>
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#include <sound/jack.h>
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#include <asm/mach-types.h>
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#include <plat/board-ams-delta.h>
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#include <plat/mcbsp.h>
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#include "omap-mcbsp.h"
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#include "omap-pcm.h"
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#include "../codecs/cx20442.h"
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/* Board specific DAPM widgets */
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static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
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/* Handset */
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SND_SOC_DAPM_MIC("Mouthpiece", NULL),
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SND_SOC_DAPM_HP("Earpiece", NULL),
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/* Handsfree/Speakerphone */
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SND_SOC_DAPM_MIC("Microphone", NULL),
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SND_SOC_DAPM_SPK("Speaker", NULL),
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};
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/* How they are connected to codec pins */
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static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
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{"TELIN", NULL, "Mouthpiece"},
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{"Earpiece", NULL, "TELOUT"},
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{"MIC", NULL, "Microphone"},
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{"Speaker", NULL, "SPKOUT"},
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};
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/*
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* Controls, functional after the modem line discipline is activated.
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*/
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/* Virtual switch: audio input/output constellations */
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static const char *ams_delta_audio_mode[] =
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{"Mixed", "Handset", "Handsfree", "Speakerphone"};
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/* Selection <-> pin translation */
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#define AMS_DELTA_MOUTHPIECE 0
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#define AMS_DELTA_EARPIECE 1
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#define AMS_DELTA_MICROPHONE 2
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#define AMS_DELTA_SPEAKER 3
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#define AMS_DELTA_AGC 4
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#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPIECE) | \
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(1 << AMS_DELTA_MICROPHONE))
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#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \
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(1 << AMS_DELTA_EARPIECE))
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#define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \
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(1 << AMS_DELTA_SPEAKER))
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#define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
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static const unsigned short ams_delta_audio_mode_pins[] = {
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AMS_DELTA_MIXED,
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AMS_DELTA_HANDSET,
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AMS_DELTA_HANDSFREE,
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AMS_DELTA_SPEAKERPHONE,
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};
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static unsigned short ams_delta_audio_agc;
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static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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struct snd_soc_dapm_context *dapm = &codec->dapm;
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struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
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unsigned short pins;
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int pin, changed = 0;
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/* Refuse any mode changes if we are not able to control the codec. */
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if (!codec->hw_write)
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return -EUNATCH;
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if (ucontrol->value.enumerated.item[0] >= control->max)
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return -EINVAL;
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mutex_lock(&codec->mutex);
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/* Translate selection to bitmap */
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pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
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/* Setup pins after corresponding bits if changed */
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pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
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if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin(dapm, "Mouthpiece");
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else
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snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
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}
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pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
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if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin(dapm, "Earpiece");
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else
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snd_soc_dapm_disable_pin(dapm, "Earpiece");
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}
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pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
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if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin(dapm, "Microphone");
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else
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snd_soc_dapm_disable_pin(dapm, "Microphone");
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}
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pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
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if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin(dapm, "Speaker");
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else
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snd_soc_dapm_disable_pin(dapm, "Speaker");
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}
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pin = !!(pins & (1 << AMS_DELTA_AGC));
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if (pin != ams_delta_audio_agc) {
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ams_delta_audio_agc = pin;
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin(dapm, "AGCIN");
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else
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snd_soc_dapm_disable_pin(dapm, "AGCIN");
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}
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if (changed)
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snd_soc_dapm_sync(dapm);
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mutex_unlock(&codec->mutex);
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return changed;
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}
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static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
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struct snd_soc_dapm_context *dapm = &codec->dapm;
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unsigned short pins, mode;
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pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
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AMS_DELTA_MOUTHPIECE) |
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(snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
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AMS_DELTA_EARPIECE));
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if (pins)
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pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
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AMS_DELTA_MICROPHONE);
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else
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pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
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AMS_DELTA_MICROPHONE) |
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(snd_soc_dapm_get_pin_status(dapm, "Speaker") <<
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AMS_DELTA_SPEAKER) |
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(ams_delta_audio_agc << AMS_DELTA_AGC));
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for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
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if (pins == ams_delta_audio_mode_pins[mode])
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break;
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if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
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return -EINVAL;
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ucontrol->value.enumerated.item[0] = mode;
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return 0;
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}
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static const struct soc_enum ams_delta_audio_enum[] = {
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SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
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ams_delta_audio_mode),
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};
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static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
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SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
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ams_delta_get_audio_mode, ams_delta_set_audio_mode),
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};
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/* Hook switch */
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static struct snd_soc_jack ams_delta_hook_switch;
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static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
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{
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.gpio = 4,
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.name = "hook_switch",
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.report = SND_JACK_HEADSET,
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.invert = 1,
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.debounce_time = 150,
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}
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};
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/* After we are able to control the codec over the modem,
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* the hook switch can be used for dynamic DAPM reconfiguration. */
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static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
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/* Handset */
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{
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.pin = "Mouthpiece",
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.mask = SND_JACK_MICROPHONE,
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},
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{
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.pin = "Earpiece",
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.mask = SND_JACK_HEADPHONE,
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},
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/* Handsfree */
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{
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.pin = "Microphone",
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.mask = SND_JACK_MICROPHONE,
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.invert = 1,
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},
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{
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.pin = "Speaker",
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.mask = SND_JACK_HEADPHONE,
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.invert = 1,
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},
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};
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/*
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* Modem line discipline, required for making above controls functional.
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* Activated from userspace with ldattach, possibly invoked from udev rule.
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*/
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/* To actually apply any modem controlled configuration changes to the codec,
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* we must connect codec DAI pins to the modem for a moment. Be careful not
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* to interfere with our digital mute function that shares the same hardware. */
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static struct timer_list cx81801_timer;
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static bool cx81801_cmd_pending;
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static bool ams_delta_muted;
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static DEFINE_SPINLOCK(ams_delta_lock);
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static void cx81801_timeout(unsigned long data)
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{
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int muted;
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spin_lock(&ams_delta_lock);
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cx81801_cmd_pending = 0;
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muted = ams_delta_muted;
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spin_unlock(&ams_delta_lock);
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/* Reconnect the codec DAI back from the modem to the CPU DAI
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* only if digital mute still off */
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if (!muted)
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ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
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}
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/*
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* Used for passing a codec structure pointer
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* from the board initialization code to the tty line discipline.
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*/
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static struct snd_soc_codec *cx20442_codec;
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/* Line discipline .open() */
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static int cx81801_open(struct tty_struct *tty)
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{
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int ret;
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if (!cx20442_codec)
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return -ENODEV;
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/*
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* Pass the codec structure pointer for use by other ldisc callbacks,
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* both the card and the codec specific parts.
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*/
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tty->disc_data = cx20442_codec;
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ret = v253_ops.open(tty);
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if (ret < 0)
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tty->disc_data = NULL;
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return ret;
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}
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/* Line discipline .close() */
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static void cx81801_close(struct tty_struct *tty)
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{
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struct snd_soc_codec *codec = tty->disc_data;
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struct snd_soc_dapm_context *dapm = &codec->dapm;
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del_timer_sync(&cx81801_timer);
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/* Prevent the hook switch from further changing the DAPM pins */
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INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
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if (!codec)
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return;
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v253_ops.close(tty);
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/* Revert back to default audio input/output constellation */
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snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
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snd_soc_dapm_enable_pin(dapm, "Earpiece");
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snd_soc_dapm_enable_pin(dapm, "Microphone");
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snd_soc_dapm_disable_pin(dapm, "Speaker");
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snd_soc_dapm_disable_pin(dapm, "AGCIN");
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snd_soc_dapm_sync(dapm);
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}
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/* Line discipline .hangup() */
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static int cx81801_hangup(struct tty_struct *tty)
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{
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cx81801_close(tty);
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return 0;
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}
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/* Line discipline .recieve_buf() */
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static void cx81801_receive(struct tty_struct *tty,
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const unsigned char *cp, char *fp, int count)
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{
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struct snd_soc_codec *codec = tty->disc_data;
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const unsigned char *c;
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int apply, ret;
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if (!codec)
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return;
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if (!codec->hw_write) {
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/* First modem response, complete setup procedure */
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/* Initialize timer used for config pulse generation */
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setup_timer(&cx81801_timer, cx81801_timeout, 0);
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v253_ops.receive_buf(tty, cp, fp, count);
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/* Link hook switch to DAPM pins */
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ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
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ARRAY_SIZE(ams_delta_hook_switch_pins),
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ams_delta_hook_switch_pins);
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if (ret)
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dev_warn(codec->dev,
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"Failed to link hook switch to DAPM pins, "
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"will continue with hook switch unlinked.\n");
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return;
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}
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v253_ops.receive_buf(tty, cp, fp, count);
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for (c = &cp[count - 1]; c >= cp; c--) {
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if (*c != '\r')
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continue;
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/* Complete modem response received, apply config to codec */
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spin_lock_bh(&ams_delta_lock);
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mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
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apply = !ams_delta_muted && !cx81801_cmd_pending;
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cx81801_cmd_pending = 1;
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spin_unlock_bh(&ams_delta_lock);
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/* Apply config pulse by connecting the codec to the modem
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* if not already done */
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if (apply)
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ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
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AMS_DELTA_LATCH2_MODEM_CODEC);
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break;
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}
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}
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/* Line discipline .write_wakeup() */
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static void cx81801_wakeup(struct tty_struct *tty)
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{
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v253_ops.write_wakeup(tty);
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}
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static struct tty_ldisc_ops cx81801_ops = {
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.magic = TTY_LDISC_MAGIC,
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.name = "cx81801",
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.owner = THIS_MODULE,
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.open = cx81801_open,
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.close = cx81801_close,
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.hangup = cx81801_hangup,
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.receive_buf = cx81801_receive,
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.write_wakeup = cx81801_wakeup,
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};
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/*
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* Even if not very useful, the sound card can still work without any of the
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* above functonality activated. You can still control its audio input/output
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* constellation and speakerphone gain from userspace by issuing AT commands
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* over the modem port.
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*/
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static int ams_delta_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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/* Set cpu DAI configuration */
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return snd_soc_dai_set_fmt(rtd->cpu_dai,
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SND_SOC_DAIFMT_DSP_A |
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SND_SOC_DAIFMT_NB_NF |
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SND_SOC_DAIFMT_CBM_CFM);
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}
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static struct snd_soc_ops ams_delta_ops = {
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.hw_params = ams_delta_hw_params,
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};
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/* Board specific codec bias level control */
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static int ams_delta_set_bias_level(struct snd_soc_card *card,
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enum snd_soc_bias_level level)
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{
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struct snd_soc_codec *codec = card->rtd->codec;
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switch (level) {
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case SND_SOC_BIAS_ON:
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case SND_SOC_BIAS_PREPARE:
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case SND_SOC_BIAS_STANDBY:
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if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
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ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
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AMS_DELTA_LATCH2_MODEM_NRESET);
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break;
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case SND_SOC_BIAS_OFF:
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if (codec->dapm.bias_level != SND_SOC_BIAS_OFF)
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ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
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0);
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}
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codec->dapm.bias_level = level;
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return 0;
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}
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/* Digital mute implemented using modem/CPU multiplexer.
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* Shares hardware with codec config pulse generation */
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static bool ams_delta_muted = 1;
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static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
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{
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int apply;
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if (ams_delta_muted == mute)
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return 0;
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spin_lock_bh(&ams_delta_lock);
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ams_delta_muted = mute;
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apply = !cx81801_cmd_pending;
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spin_unlock_bh(&ams_delta_lock);
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if (apply)
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ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
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mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
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return 0;
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}
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/* Our codec DAI probably doesn't have its own .ops structure */
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static struct snd_soc_dai_ops ams_delta_dai_ops = {
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.digital_mute = ams_delta_digital_mute,
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};
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/* Will be used if the codec ever has its own digital_mute function */
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static int ams_delta_startup(struct snd_pcm_substream *substream)
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{
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return ams_delta_digital_mute(NULL, 0);
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}
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static void ams_delta_shutdown(struct snd_pcm_substream *substream)
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{
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ams_delta_digital_mute(NULL, 1);
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}
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/*
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* Card initialization
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*/
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static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
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{
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struct snd_soc_codec *codec = rtd->codec;
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struct snd_soc_dapm_context *dapm = &codec->dapm;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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struct snd_soc_card *card = rtd->card;
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int ret;
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/* Codec is ready, now add/activate board specific controls */
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/* Store a pointer to the codec structure for tty ldisc use */
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cx20442_codec = codec;
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/* Set up digital mute if not provided by the codec */
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if (!codec_dai->driver->ops) {
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codec_dai->driver->ops = &ams_delta_dai_ops;
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} else {
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ams_delta_ops.startup = ams_delta_startup;
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ams_delta_ops.shutdown = ams_delta_shutdown;
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}
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/* Set codec bias level */
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ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
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/* Add hook switch - can be used to control the codec from userspace
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* even if line discipline fails */
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ret = snd_soc_jack_new(rtd->codec, "hook_switch",
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SND_JACK_HEADSET, &ams_delta_hook_switch);
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if (ret)
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dev_warn(card->dev,
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"Failed to allocate resources for hook switch, "
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"will continue without one.\n");
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else {
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ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch,
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ARRAY_SIZE(ams_delta_hook_switch_gpios),
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ams_delta_hook_switch_gpios);
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if (ret)
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dev_warn(card->dev,
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"Failed to set up hook switch GPIO line, "
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"will continue with hook switch inactive.\n");
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}
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/* Register optional line discipline for over the modem control */
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ret = tty_register_ldisc(N_V253, &cx81801_ops);
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if (ret) {
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dev_warn(card->dev,
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"Failed to register line discipline, "
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"will continue without any controls.\n");
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return 0;
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}
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/* Add board specific DAPM widgets and routes */
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ret = snd_soc_dapm_new_controls(dapm, ams_delta_dapm_widgets,
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ARRAY_SIZE(ams_delta_dapm_widgets));
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if (ret) {
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dev_warn(card->dev,
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"Failed to register DAPM controls, "
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|
"will continue without any.\n");
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return 0;
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}
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|
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ret = snd_soc_dapm_add_routes(dapm, ams_delta_audio_map,
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ARRAY_SIZE(ams_delta_audio_map));
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if (ret) {
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dev_warn(card->dev,
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"Failed to set up DAPM routes, "
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|
"will continue with codec default map.\n");
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return 0;
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}
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/* Set up initial pin constellation */
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snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
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snd_soc_dapm_enable_pin(dapm, "Earpiece");
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snd_soc_dapm_enable_pin(dapm, "Microphone");
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snd_soc_dapm_disable_pin(dapm, "Speaker");
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snd_soc_dapm_disable_pin(dapm, "AGCIN");
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snd_soc_dapm_disable_pin(dapm, "AGCOUT");
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snd_soc_dapm_sync(dapm);
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/* Add virtual switch */
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ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
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ARRAY_SIZE(ams_delta_audio_controls));
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if (ret)
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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,
|
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.platform_name = "omap-pcm-audio",
|
|
.codec_name = "cx20442-codec",
|
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.ops = &ams_delta_ops,
|
|
};
|
|
|
|
/* Audio card driver */
|
|
static struct snd_soc_card ams_delta_audio_card = {
|
|
.name = "AMS_DELTA",
|
|
.dai_link = &ams_delta_dai_link,
|
|
.num_links = 1,
|
|
.set_bias_level = ams_delta_set_bias_level,
|
|
};
|
|
|
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/* 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, 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");
|