linux/sound/soc/ti/n810.c
Mark Brown a9b696c851
GPIO descriptors for TI ASoC codecs
Merge series from Linus Walleij <linus.walleij@linaro.org>:

This cleans up and rewrites the GPIO usage in the TI
ASoC components to use GPIO descriptors exclusively.

Signed-off-by: Linus Walleij <linus.walleij@linaro.org>
---
Linus Walleij (5):
      ASoC: ti: Convert N810 ASoC to GPIO descriptors
      ASoC: ti: Convert RX51 to use exclusively GPIO descriptors
      ASoC: ti: Convert TWL4030 to use GPIO descriptors
      ASoC: ti: Convert Pandora ASoC to GPIO descriptors
      ASoC: ti: osk5912: Drop unused include

 arch/arm/mach-omap2/board-n8x0.c           | 10 +++++
 arch/arm/mach-omap2/pdata-quirks.c         | 10 +++++
 include/linux/platform_data/omap-twl4030.h |  3 --
 sound/soc/ti/n810.c                        | 31 ++++++++-------
 sound/soc/ti/omap-twl4030.c                | 20 ++++------
 sound/soc/ti/omap3pandora.c                | 63 +++++++++++-------------------
 sound/soc/ti/osk5912.c                     |  1 -
 sound/soc/ti/rx51.c                        | 19 ++-------
 8 files changed, 72 insertions(+), 85 deletions(-)
---
base-commit: 0bb80ecc33
change-id: 20230922-descriptors-asoc-ti-a852eff479ed

Best regards,
--
Linus Walleij <linus.walleij@linaro.org>
2023-10-02 16:17:47 +01:00

372 lines
9.0 KiB
C

// SPDX-License-Identifier: GPL-2.0-only
/*
* n810.c -- SoC audio for Nokia N810
*
* Copyright (C) 2008 Nokia Corporation
*
* Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
*/
#include <linux/clk.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <asm/mach-types.h>
#include <linux/gpio/consumer.h>
#include <linux/module.h>
#include <linux/platform_data/asoc-ti-mcbsp.h>
#include "omap-mcbsp.h"
static struct gpio_desc *n810_headset_amp;
static struct gpio_desc *n810_speaker_amp;
enum {
N810_JACK_DISABLED,
N810_JACK_HP,
N810_JACK_HS,
N810_JACK_MIC,
};
static struct clk *sys_clkout2;
static struct clk *sys_clkout2_src;
static struct clk *func96m_clk;
static int n810_spk_func;
static int n810_jack_func;
static int n810_dmic_func;
static void n810_ext_control(struct snd_soc_dapm_context *dapm)
{
int hp = 0, line1l = 0;
switch (n810_jack_func) {
case N810_JACK_HS:
line1l = 1;
fallthrough;
case N810_JACK_HP:
hp = 1;
break;
case N810_JACK_MIC:
line1l = 1;
break;
}
snd_soc_dapm_mutex_lock(dapm);
if (n810_spk_func)
snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
if (hp)
snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
if (line1l)
snd_soc_dapm_enable_pin_unlocked(dapm, "HS Mic");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
if (n810_dmic_func)
snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
else
snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
snd_soc_dapm_sync_unlocked(dapm);
snd_soc_dapm_mutex_unlock(dapm);
}
static int n810_startup(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
n810_ext_control(&rtd->card->dapm);
return clk_prepare_enable(sys_clkout2);
}
static void n810_shutdown(struct snd_pcm_substream *substream)
{
clk_disable_unprepare(sys_clkout2);
}
static int n810_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0);
int err;
/* Set the codec system clock for DAC and ADC */
err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000,
SND_SOC_CLOCK_IN);
return err;
}
static const struct snd_soc_ops n810_ops = {
.startup = n810_startup,
.hw_params = n810_hw_params,
.shutdown = n810_shutdown,
};
static int n810_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = n810_spk_func;
return 0;
}
static int n810_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (n810_spk_func == ucontrol->value.enumerated.item[0])
return 0;
n810_spk_func = ucontrol->value.enumerated.item[0];
n810_ext_control(&card->dapm);
return 1;
}
static int n810_get_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = n810_jack_func;
return 0;
}
static int n810_set_jack(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (n810_jack_func == ucontrol->value.enumerated.item[0])
return 0;
n810_jack_func = ucontrol->value.enumerated.item[0];
n810_ext_control(&card->dapm);
return 1;
}
static int n810_get_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.enumerated.item[0] = n810_dmic_func;
return 0;
}
static int n810_set_input(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
if (n810_dmic_func == ucontrol->value.enumerated.item[0])
return 0;
n810_dmic_func = ucontrol->value.enumerated.item[0];
n810_ext_control(&card->dapm);
return 1;
}
static int n810_spk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
if (SND_SOC_DAPM_EVENT_ON(event))
gpiod_set_value(n810_speaker_amp, 1);
else
gpiod_set_value(n810_speaker_amp, 0);
return 0;
}
static int n810_jack_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
if (SND_SOC_DAPM_EVENT_ON(event))
gpiod_set_value(n810_headset_amp, 1);
else
gpiod_set_value(n810_headset_amp, 0);
return 0;
}
static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
SND_SOC_DAPM_MIC("DMic", NULL),
SND_SOC_DAPM_MIC("HS Mic", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
{"Headphone Jack", NULL, "HPLOUT"},
{"Headphone Jack", NULL, "HPROUT"},
{"Ext Spk", NULL, "LLOUT"},
{"Ext Spk", NULL, "RLOUT"},
{"DMic Rate 64", NULL, "DMic"},
{"DMic", NULL, "Mic Bias"},
/*
* Note that the mic bias is coming from Retu/Vilma and we don't have
* control over it atm. The analog HS mic is not working. <- TODO
*/
{"LINE1L", NULL, "HS Mic"},
};
static const char *spk_function[] = {"Off", "On"};
static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"};
static const char *input_function[] = {"ADC", "Digital Mic"};
static const struct soc_enum n810_enum[] = {
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
};
static const struct snd_kcontrol_new aic33_n810_controls[] = {
SOC_ENUM_EXT("Speaker Function", n810_enum[0],
n810_get_spk, n810_set_spk),
SOC_ENUM_EXT("Jack Function", n810_enum[1],
n810_get_jack, n810_set_jack),
SOC_ENUM_EXT("Input Select", n810_enum[2],
n810_get_input, n810_set_input),
};
/* Digital audio interface glue - connects codec <--> CPU */
SND_SOC_DAILINK_DEFS(aic33,
DAILINK_COMP_ARRAY(COMP_CPU("48076000.mcbsp")),
DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
"tlv320aic3x-hifi")),
DAILINK_COMP_ARRAY(COMP_PLATFORM("48076000.mcbsp")));
static struct snd_soc_dai_link n810_dai = {
.name = "TLV320AIC33",
.stream_name = "AIC33",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
.ops = &n810_ops,
SND_SOC_DAILINK_REG(aic33),
};
/* Audio machine driver */
static struct snd_soc_card snd_soc_n810 = {
.name = "N810",
.owner = THIS_MODULE,
.dai_link = &n810_dai,
.num_links = 1,
.controls = aic33_n810_controls,
.num_controls = ARRAY_SIZE(aic33_n810_controls),
.dapm_widgets = aic33_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
.fully_routed = true,
};
static struct platform_device *n810_snd_device;
static int __init n810_soc_init(void)
{
int err;
struct device *dev;
if (!of_have_populated_dt() ||
(!of_machine_is_compatible("nokia,n810") &&
!of_machine_is_compatible("nokia,n810-wimax")))
return -ENODEV;
n810_snd_device = platform_device_alloc("soc-audio", -1);
if (!n810_snd_device)
return -ENOMEM;
platform_set_drvdata(n810_snd_device, &snd_soc_n810);
err = platform_device_add(n810_snd_device);
if (err)
goto err1;
dev = &n810_snd_device->dev;
sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
if (IS_ERR(sys_clkout2_src)) {
dev_err(dev, "Could not get sys_clkout2_src clock\n");
err = PTR_ERR(sys_clkout2_src);
goto err2;
}
sys_clkout2 = clk_get(dev, "sys_clkout2");
if (IS_ERR(sys_clkout2)) {
dev_err(dev, "Could not get sys_clkout2\n");
err = PTR_ERR(sys_clkout2);
goto err3;
}
/*
* Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
* 96 MHz as its parent in order to get 12 MHz
*/
func96m_clk = clk_get(dev, "func_96m_ck");
if (IS_ERR(func96m_clk)) {
dev_err(dev, "Could not get func 96M clock\n");
err = PTR_ERR(func96m_clk);
goto err4;
}
clk_set_parent(sys_clkout2_src, func96m_clk);
clk_set_rate(sys_clkout2, 12000000);
n810_headset_amp = devm_gpiod_get(&n810_snd_device->dev,
"headphone", GPIOD_OUT_LOW);
if (IS_ERR(n810_headset_amp)) {
err = PTR_ERR(n810_headset_amp);
goto err4;
}
n810_speaker_amp = devm_gpiod_get(&n810_snd_device->dev,
"speaker", GPIOD_OUT_LOW);
if (IS_ERR(n810_speaker_amp)) {
err = PTR_ERR(n810_speaker_amp);
goto err4;
}
return 0;
err4:
clk_put(sys_clkout2);
err3:
clk_put(sys_clkout2_src);
err2:
platform_device_del(n810_snd_device);
err1:
platform_device_put(n810_snd_device);
return err;
}
static void __exit n810_soc_exit(void)
{
clk_put(sys_clkout2_src);
clk_put(sys_clkout2);
clk_put(func96m_clk);
platform_device_unregister(n810_snd_device);
}
module_init(n810_soc_init);
module_exit(n810_soc_exit);
MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
MODULE_DESCRIPTION("ALSA SoC Nokia N810");
MODULE_LICENSE("GPL");