linux/sound/soc/samsung/tobermory.c
Kuninori Morimoto 21b6cd54c9
ASoC: samsung: convert not to use asoc_xxx()
ASoC is now unified asoc_xxx() into snd_soc_xxx().
This patch convert asoc_xxx() to snd_soc_xxx().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87fs3kqnhi.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
2023-09-25 14:16:34 +02:00

252 lines
5.7 KiB
C

// SPDX-License-Identifier: GPL-2.0+
//
// Tobermory audio support
//
// Copyright 2011 Wolfson Microelectronics
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/jack.h>
#include <linux/gpio.h>
#include <linux/module.h>
#include "../codecs/wm8962.h"
static int sample_rate = 44100;
static int tobermory_set_bias_level(struct snd_soc_card *card,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_dai *codec_dai;
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
codec_dai = snd_soc_rtd_to_codec(rtd, 0);
if (dapm->dev != codec_dai->dev)
return 0;
switch (level) {
case SND_SOC_BIAS_PREPARE:
if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
WM8962_FLL_MCLK, 32768,
sample_rate * 512);
if (ret < 0)
pr_err("Failed to start FLL: %d\n", ret);
ret = snd_soc_dai_set_sysclk(codec_dai,
WM8962_SYSCLK_FLL,
sample_rate * 512,
SND_SOC_CLOCK_IN);
if (ret < 0) {
pr_err("Failed to set SYSCLK: %d\n", ret);
snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
0, 0, 0);
return ret;
}
}
break;
default:
break;
}
return 0;
}
static int tobermory_set_bias_level_post(struct snd_soc_card *card,
struct snd_soc_dapm_context *dapm,
enum snd_soc_bias_level level)
{
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_dai *codec_dai;
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
codec_dai = snd_soc_rtd_to_codec(rtd, 0);
if (dapm->dev != codec_dai->dev)
return 0;
switch (level) {
case SND_SOC_BIAS_STANDBY:
ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
32768, SND_SOC_CLOCK_IN);
if (ret < 0) {
pr_err("Failed to switch away from FLL: %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
0, 0, 0);
if (ret < 0) {
pr_err("Failed to stop FLL: %d\n", ret);
return ret;
}
break;
default:
break;
}
dapm->bias_level = level;
return 0;
}
static int tobermory_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
sample_rate = params_rate(params);
return 0;
}
static const struct snd_soc_ops tobermory_ops = {
.hw_params = tobermory_hw_params,
};
SND_SOC_DAILINK_DEFS(cpu,
DAILINK_COMP_ARRAY(COMP_CPU("samsung-i2s.0")),
DAILINK_COMP_ARRAY(COMP_CODEC("wm8962.1-001a", "wm8962")),
DAILINK_COMP_ARRAY(COMP_PLATFORM("samsung-i2s.0")));
static struct snd_soc_dai_link tobermory_dai[] = {
{
.name = "CPU",
.stream_name = "CPU",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBM_CFM,
.ops = &tobermory_ops,
SND_SOC_DAILINK_REG(cpu),
},
};
static const struct snd_kcontrol_new controls[] = {
SOC_DAPM_PIN_SWITCH("Main Speaker"),
SOC_DAPM_PIN_SWITCH("DMIC"),
};
static const struct snd_soc_dapm_widget widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("DMIC", NULL),
SND_SOC_DAPM_MIC("AMIC", NULL),
SND_SOC_DAPM_SPK("Main Speaker", NULL),
};
static const struct snd_soc_dapm_route audio_paths[] = {
{ "Headphone", NULL, "HPOUTL" },
{ "Headphone", NULL, "HPOUTR" },
{ "Main Speaker", NULL, "SPKOUTL" },
{ "Main Speaker", NULL, "SPKOUTR" },
{ "Headset Mic", NULL, "MICBIAS" },
{ "IN4L", NULL, "Headset Mic" },
{ "IN4R", NULL, "Headset Mic" },
{ "AMIC", NULL, "MICBIAS" },
{ "IN1L", NULL, "AMIC" },
{ "IN1R", NULL, "AMIC" },
{ "DMIC", NULL, "MICBIAS" },
{ "DMICDAT", NULL, "DMIC" },
};
static struct snd_soc_jack tobermory_headset;
/* Headset jack detection DAPM pins */
static struct snd_soc_jack_pin tobermory_headset_pins[] = {
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
{
.pin = "Headphone",
.mask = SND_JACK_MICROPHONE,
},
};
static int tobermory_late_probe(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd;
struct snd_soc_component *component;
struct snd_soc_dai *codec_dai;
int ret;
rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
component = snd_soc_rtd_to_codec(rtd, 0)->component;
codec_dai = snd_soc_rtd_to_codec(rtd, 0);
ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
32768, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET |
SND_JACK_BTN_0, &tobermory_headset,
tobermory_headset_pins,
ARRAY_SIZE(tobermory_headset_pins));
if (ret)
return ret;
wm8962_mic_detect(component, &tobermory_headset);
return 0;
}
static struct snd_soc_card tobermory = {
.name = "Tobermory",
.owner = THIS_MODULE,
.dai_link = tobermory_dai,
.num_links = ARRAY_SIZE(tobermory_dai),
.set_bias_level = tobermory_set_bias_level,
.set_bias_level_post = tobermory_set_bias_level_post,
.controls = controls,
.num_controls = ARRAY_SIZE(controls),
.dapm_widgets = widgets,
.num_dapm_widgets = ARRAY_SIZE(widgets),
.dapm_routes = audio_paths,
.num_dapm_routes = ARRAY_SIZE(audio_paths),
.fully_routed = true,
.late_probe = tobermory_late_probe,
};
static int tobermory_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &tobermory;
int ret;
card->dev = &pdev->dev;
ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err_probe(&pdev->dev, ret, "snd_soc_register_card() failed\n");
return ret;
}
static struct platform_driver tobermory_driver = {
.driver = {
.name = "tobermory",
.pm = &snd_soc_pm_ops,
},
.probe = tobermory_probe,
};
module_platform_driver(tobermory_driver);
MODULE_DESCRIPTION("Tobermory audio support");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:tobermory");